diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h index 5d02b0e2ae..bc4981bbb1 100644 --- a/webrtc/audio/audio_receive_stream.h +++ b/webrtc/audio/audio_receive_stream.h @@ -14,7 +14,7 @@ #include "webrtc/audio_receive_stream.h" #include "webrtc/audio/scoped_voe_interface.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/voice_engine/include/voe_base.h" namespace webrtc { diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc index 594ddf5c97..f83d32050d 100644 --- a/webrtc/call/call.cc +++ b/webrtc/call/call.cc @@ -25,9 +25,9 @@ #include "webrtc/call/rtc_event_log.h" #include "webrtc/common.h" #include "webrtc/config.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/system_wrappers/include/cpu_info.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index c37b83bab4..95fd32e258 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -19,7 +19,7 @@ #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/rtp_to_ntp.h" diff --git a/webrtc/call/congestion_controller.cc b/webrtc/call/congestion_controller.cc index 1ec361e898..52704154bb 100644 --- a/webrtc/call/congestion_controller.cc +++ b/webrtc/call/congestion_controller.cc @@ -20,8 +20,8 @@ #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h" #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/video_engine/call_stats.h" diff --git a/webrtc/common_video/video_render_frames.cc b/webrtc/common_video/video_render_frames.cc index f4ece5e8e9..a7db15c7ea 100644 --- a/webrtc/common_video/video_render_frames.cc +++ b/webrtc/common_video/video_render_frames.cc @@ -12,7 +12,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc index 79d6cbc4b7..7d5b99b9da 100644 --- a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc +++ b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc @@ -18,7 +18,7 @@ #include "webrtc/base/arraysize.h" #include "webrtc/examples/android/media_demo/jni/jni_helpers.h" -#include "webrtc/modules/utility/interface/helpers_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" #include "webrtc/test/channel_transport/include/channel_transport.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 839a1439e6..13f9a5d306 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -45,7 +45,7 @@ source_set("rent_a_codec") { config("audio_coding_config") { include_dirs = [ "main/include", - "../interface", + "../include", ] } diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h index 9e812b20b6..4b080ba93a 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h +++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h @@ -24,7 +24,7 @@ #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc index dbbf8f8d7d..3aee3445d5 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc @@ -33,7 +33,7 @@ #include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/event_wrapper.h" diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/webrtc/modules/audio_coding/main/acm2/call_statistics.h index 2aece0ff40..e2df9210ff 100644 --- a/webrtc/modules/audio_coding/main/acm2/call_statistics.h +++ b/webrtc/modules/audio_coding/main/acm2/call_statistics.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_ #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" // // This class is for book keeping of calls to ACM. It is not useful to log API diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h index c6942ec285..6b50dd07f8 100644 --- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h +++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/audio_coding_module.gypi index 6fb37d25fa..088a3197fd 100644 --- a/webrtc/modules/audio_coding/main/audio_coding_module.gypi +++ b/webrtc/modules/audio_coding/main/audio_coding_module.gypi @@ -81,13 +81,13 @@ ], 'include_dirs': [ 'include', - '../../interface', + '../../include', '<(webrtc_root)', ], 'direct_dependent_settings': { 'include_dirs': [ 'include', - '../../interface', + '../../include', '<(webrtc_root)', ], }, diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module.h b/webrtc/modules/audio_coding/main/include/audio_coding_module.h index 1792a25c1b..00f8d5477f 100644 --- a/webrtc/modules/audio_coding/main/include/audio_coding_module.h +++ b/webrtc/modules/audio_coding/main/include/audio_coding_module.h @@ -18,7 +18,7 @@ #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h b/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h index 489df406f4..1ca6f9d13c 100644 --- a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h +++ b/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h @@ -13,7 +13,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h index 39d4dabd98..ff6937ec08 100644 --- a/webrtc/modules/audio_coding/main/test/Channel.h +++ b/webrtc/modules/audio_coding/main/test/Channel.h @@ -14,7 +14,7 @@ #include #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/main/test/PCMFile.cc index d0ae7830de..0466e0222c 100644 --- a/webrtc/modules/audio_coding/main/test/PCMFile.cc +++ b/webrtc/modules/audio_coding/main/test/PCMFile.cc @@ -15,7 +15,7 @@ #include #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/main/test/PCMFile.h index 8353898f03..785ed667fe 100644 --- a/webrtc/modules/audio_coding/main/test/PCMFile.h +++ b/webrtc/modules/audio_coding/main/test/PCMFile.h @@ -16,7 +16,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h index c79b63e164..6bad755af9 100644 --- a/webrtc/modules/audio_coding/main/test/RTPFile.h +++ b/webrtc/modules/audio_coding/main/test/RTPFile.h @@ -15,7 +15,7 @@ #include #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc index 5b1e07e478..857381d250 100644 --- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc +++ b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc @@ -17,7 +17,7 @@ #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/main/test/Channel.h" #include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc index 20b10a376e..9d0986a013 100644 --- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc +++ b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc @@ -14,7 +14,7 @@ #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/main/test/utility.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/sleep.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc index d3f6fa6dd4..0252d1cdfa 100644 --- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc +++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc @@ -20,7 +20,7 @@ #include "webrtc/modules/audio_coding/neteq/expand.h" #include "webrtc/modules/audio_coding/neteq/packet_buffer.h" #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc index 5140c0620f..806d02b8de 100644 --- a/webrtc/modules/audio_coding/neteq/delay_manager.cc +++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc @@ -17,7 +17,7 @@ #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/logging.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/nack.cc b/webrtc/modules/audio_coding/neteq/nack.cc index fd3d762605..011914b3d9 100644 --- a/webrtc/modules/audio_coding/neteq/nack.cc +++ b/webrtc/modules/audio_coding/neteq/nack.cc @@ -15,7 +15,7 @@ #include // For std::max. #include "webrtc/base/checks.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/logging.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc index 9ed0fc8a3c..d9ae15e7fa 100644 --- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc +++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc @@ -42,7 +42,7 @@ #include "webrtc/modules/audio_coding/neteq/preemptive_expand.h" #include "webrtc/modules/audio_coding/neteq/sync_buffer.h" #include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" // Modify the code to obtain backwards bit-exactness. Once bit-exactness is no diff --git a/webrtc/modules/audio_coding/neteq/packet.h b/webrtc/modules/audio_coding/neteq/packet.h index 723ed8b0a3..64b325e027 100644 --- a/webrtc/modules/audio_coding/neteq/packet.h +++ b/webrtc/modules/audio_coding/neteq/packet.h @@ -13,7 +13,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc index cf8e0280bb..7ef40bc814 100644 --- a/webrtc/modules/audio_coding/neteq/rtcp.cc +++ b/webrtc/modules/audio_coding/neteq/rtcp.cc @@ -15,7 +15,7 @@ #include #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h index 3fbce8be5c..56ed72fcee 100644 --- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h +++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h @@ -14,7 +14,7 @@ #include #include #include "webrtc/typedefs.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" enum stereoModes { stereoModeMono, diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h index 3bd2df5ca8..489a8b2ad8 100644 --- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h +++ b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_ #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h index 0a41c6ec20..c9fe11f656 100644 --- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h +++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h @@ -14,7 +14,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { namespace test { diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc index adce1cfd79..7a52408705 100644 --- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc +++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc @@ -34,7 +34,7 @@ #include "webrtc/modules/audio_coding/neteq/tools/packet.h" #include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h" #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/rtp_file_reader.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.cc b/webrtc/modules/audio_coding/neteq/tools/packet.cc index b8b27afdec..2b2fcc286e 100644 --- a/webrtc/modules/audio_coding/neteq/tools/packet.cc +++ b/webrtc/modules/audio_coding/neteq/tools/packet.cc @@ -12,8 +12,8 @@ #include -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" namespace webrtc { namespace test { diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc index 9b17ba8f64..dad72eaecd 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc +++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc @@ -18,7 +18,7 @@ #include "webrtc/base/checks.h" #include "webrtc/call/rtc_event_log.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" // Files generated at build-time by the protobuf compiler. #ifdef WEBRTC_ANDROID_PLATFORM_BUILD diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h index 7150bcfe89..90d5931224 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h +++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h @@ -16,7 +16,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc index e3a829bd24..b7a3109c01 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc +++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc @@ -20,7 +20,7 @@ #include "webrtc/base/checks.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/test/rtp_file_reader.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h index cd7d7e874c..2febf68b91 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h +++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h @@ -18,7 +18,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h index 6c16192daa..53371be8f6 100644 --- a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h +++ b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_ #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn index 3b9e2769ac..36391c7abc 100644 --- a/webrtc/modules/audio_conference_mixer/BUILD.gn +++ b/webrtc/modules/audio_conference_mixer/BUILD.gn @@ -9,15 +9,15 @@ config("audio_conference_mixer_config") { visibility = [ ":*" ] # Only targets in this file can depend on this. include_dirs = [ - "interface", - "../interface", + "include", + "../include", ] } source_set("audio_conference_mixer") { sources = [ - "interface/audio_conference_mixer.h", - "interface/audio_conference_mixer_defines.h", + "include/audio_conference_mixer.h", + "include/audio_conference_mixer_defines.h", "source/audio_conference_mixer_impl.cc", "source/audio_conference_mixer_impl.h", "source/audio_frame_manipulator.cc", diff --git a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi index 5aa3cc449b..9d7179504c 100644 --- a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi +++ b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi @@ -17,8 +17,8 @@ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', ], 'sources': [ - 'interface/audio_conference_mixer.h', - 'interface/audio_conference_mixer_defines.h', + 'include/audio_conference_mixer.h', + 'include/audio_conference_mixer_defines.h', 'source/audio_frame_manipulator.cc', 'source/audio_frame_manipulator.h', 'source/memory_pool.h', diff --git a/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h new file mode 100644 index 0000000000..7370442704 --- /dev/null +++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h @@ -0,0 +1,77 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_ +#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_ + +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" + +namespace webrtc { +class AudioMixerOutputReceiver; +class MixerParticipant; +class Trace; + +class AudioConferenceMixer : public Module +{ +public: + enum {kMaximumAmountOfMixedParticipants = 3}; + enum Frequency + { + kNbInHz = 8000, + kWbInHz = 16000, + kSwbInHz = 32000, + kFbInHz = 48000, + kLowestPossible = -1, + kDefaultFrequency = kWbInHz + }; + + // Factory method. Constructor disabled. + static AudioConferenceMixer* Create(int id); + virtual ~AudioConferenceMixer() {} + + // Module functions + int64_t TimeUntilNextProcess() override = 0; + int32_t Process() override = 0; + + // Register/unregister a callback class for receiving the mixed audio. + virtual int32_t RegisterMixedStreamCallback( + AudioMixerOutputReceiver* receiver) = 0; + virtual int32_t UnRegisterMixedStreamCallback() = 0; + + // Add/remove participants as candidates for mixing. + virtual int32_t SetMixabilityStatus(MixerParticipant* participant, + bool mixable) = 0; + // Returns true if a participant is a candidate for mixing. + virtual bool MixabilityStatus( + const MixerParticipant& participant) const = 0; + + // Inform the mixer that the participant should always be mixed and not + // count toward the number of mixed participants. Note that a participant + // must have been added to the mixer (by calling SetMixabilityStatus()) + // before this function can be successfully called. + virtual int32_t SetAnonymousMixabilityStatus( + MixerParticipant* participant, bool mixable) = 0; + // Returns true if the participant is mixed anonymously. + virtual bool AnonymousMixabilityStatus( + const MixerParticipant& participant) const = 0; + + // Set the minimum sampling frequency at which to mix. The mixing algorithm + // may still choose to mix at a higher samling frequency to avoid + // downsampling of audio contributing to the mixed audio. + virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0; + +protected: + AudioConferenceMixer() {} +}; +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_ diff --git a/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h new file mode 100644 index 0000000000..5d58f42435 --- /dev/null +++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h @@ -0,0 +1,60 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ +#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ + +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +class MixHistory; + +// A callback class that all mixer participants must inherit from/implement. +class MixerParticipant +{ +public: + // The implementation of this function should update audioFrame with new + // audio every time it's called. + // + // If it returns -1, the frame will not be added to the mix. + virtual int32_t GetAudioFrame(int32_t id, + AudioFrame* audioFrame) = 0; + + // Returns true if the participant was mixed this mix iteration. + bool IsMixed() const; + + // This function specifies the sampling frequency needed for the AudioFrame + // for future GetAudioFrame(..) calls. + virtual int32_t NeededFrequency(int32_t id) const = 0; + + MixHistory* _mixHistory; +protected: + MixerParticipant(); + virtual ~MixerParticipant(); +}; + +class AudioMixerOutputReceiver +{ +public: + // This callback function provides the mixed audio for this mix iteration. + // Note that uniqueAudioFrames is an array of AudioFrame pointers with the + // size according to the size parameter. + virtual void NewMixedAudio(const int32_t id, + const AudioFrame& generalAudioFrame, + const AudioFrame** uniqueAudioFrames, + const uint32_t size) = 0; +protected: + AudioMixerOutputReceiver() {} + virtual ~AudioMixerOutputReceiver() {} +}; +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h index 7ff39579ee..7536ce9c28 100644 --- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h +++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h @@ -8,12 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_ -#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_ +#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_ -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/interface/module_common_types.h" +#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include") + +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { class AudioMixerOutputReceiver; @@ -74,4 +76,4 @@ protected: }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_ +#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_ diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h index d15b7fca02..057c37f267 100644 --- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h +++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h @@ -8,10 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ -#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ +#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ +#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ -#include "webrtc/modules/interface/module_common_types.h" +#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include") + +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -57,4 +59,4 @@ protected: }; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ +#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_ diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc index 2d2cf9dbb8..0ac9eae691 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc +++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" #include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h" #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h index bc9a27e9f0..2466112769 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h +++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h @@ -16,10 +16,10 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h" #include "webrtc/modules/audio_conference_mixer/source/memory_pool.h" #include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { class AudioProcessing; diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc index 636698e9c1..9c5d3b939d 100644 --- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc +++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc @@ -9,7 +9,7 @@ */ #include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace { diff --git a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc index d4fbd205f1..293bfa0db9 100644 --- a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc +++ b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc @@ -10,8 +10,8 @@ #include "testing/gmock/include/gmock/gmock.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h" -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn index 8875178c15..b5e503a9e3 100644 --- a/webrtc/modules/audio_device/BUILD.gn +++ b/webrtc/modules/audio_device/BUILD.gn @@ -10,7 +10,7 @@ import("../../build/webrtc.gni") config("audio_device_config") { include_dirs = [ - "../interface", + "../include", "include", "dummy", # Contains dummy audio device implementations. ] diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc index 260e793d60..522010e052 100644 --- a/webrtc/modules/audio_device/android/audio_manager.cc +++ b/webrtc/modules/audio_device/android/audio_manager.cc @@ -16,7 +16,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_device/android/audio_common.h" -#include "webrtc/modules/utility/interface/helpers_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" #define TAG "AudioManager" #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__) diff --git a/webrtc/modules/audio_device/android/audio_manager.h b/webrtc/modules/audio_device/android/audio_manager.h index 9cceaacfca..26caf61afe 100644 --- a/webrtc/modules/audio_device/android/audio_manager.h +++ b/webrtc/modules/audio_device/android/audio_manager.h @@ -19,8 +19,8 @@ #include "webrtc/modules/audio_device/audio_device_config.h" #include "webrtc/modules/audio_device/include/audio_device_defines.h" #include "webrtc/modules/audio_device/audio_device_generic.h" -#include "webrtc/modules/utility/interface/helpers_android.h" -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h index efd516425a..a84718078d 100644 --- a/webrtc/modules/audio_device/android/audio_record_jni.h +++ b/webrtc/modules/audio_device/android/audio_record_jni.h @@ -17,8 +17,8 @@ #include "webrtc/modules/audio_device/android/audio_manager.h" #include "webrtc/modules/audio_device/include/audio_device_defines.h" #include "webrtc/modules/audio_device/audio_device_generic.h" -#include "webrtc/modules/utility/interface/helpers_android.h" -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/android/audio_track_jni.h b/webrtc/modules/audio_device/android/audio_track_jni.h index 43bfcad657..067dc6c651 100644 --- a/webrtc/modules/audio_device/android/audio_track_jni.h +++ b/webrtc/modules/audio_device/android/audio_track_jni.h @@ -18,8 +18,8 @@ #include "webrtc/modules/audio_device/android/audio_manager.h" #include "webrtc/modules/audio_device/include/audio_device_defines.h" #include "webrtc/modules/audio_device/audio_device_generic.h" -#include "webrtc/modules/utility/interface/helpers_android.h" -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/android/build_info.cc b/webrtc/modules/audio_device/android/build_info.cc index cb5dc293d7..6289697073 100644 --- a/webrtc/modules/audio_device/android/build_info.cc +++ b/webrtc/modules/audio_device/android/build_info.cc @@ -10,7 +10,7 @@ #include "webrtc/modules/audio_device/android/build_info.h" -#include "webrtc/modules/utility/interface/helpers_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/android/build_info.h b/webrtc/modules/audio_device/android/build_info.h index d9b2871841..1490fa0772 100644 --- a/webrtc/modules/audio_device/android/build_info.h +++ b/webrtc/modules/audio_device/android/build_info.h @@ -14,7 +14,7 @@ #include #include -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/android/ensure_initialized.cc b/webrtc/modules/audio_device/android/ensure_initialized.cc index e8197b7ca0..11b2b25e13 100644 --- a/webrtc/modules/audio_device/android/ensure_initialized.cc +++ b/webrtc/modules/audio_device/android/ensure_initialized.cc @@ -18,7 +18,7 @@ #include "webrtc/base/checks.h" #include "webrtc/modules/audio_device/android/audio_record_jni.h" #include "webrtc/modules/audio_device/android/audio_track_jni.h" -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" namespace webrtc { namespace audiodevicemodule { diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h index d96388b6b5..4c4d72434d 100644 --- a/webrtc/modules/audio_device/android/opensles_player.h +++ b/webrtc/modules/audio_device/android/opensles_player.h @@ -22,7 +22,7 @@ #include "webrtc/modules/audio_device/android/opensles_common.h" #include "webrtc/modules/audio_device/include/audio_device_defines.h" #include "webrtc/modules/audio_device/audio_device_generic.h" -#include "webrtc/modules/utility/interface/helpers_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/audio_device.gypi b/webrtc/modules/audio_device/audio_device.gypi index 0678d33802..c3b6fbaa46 100644 --- a/webrtc/modules/audio_device/audio_device.gypi +++ b/webrtc/modules/audio_device/audio_device.gypi @@ -20,13 +20,13 @@ ], 'include_dirs': [ '.', - '../interface', + '../include', 'include', 'dummy', # Contains dummy audio device implementations. ], 'direct_dependent_settings': { 'include_dirs': [ - '../interface', + '../include', 'include', ], }, diff --git a/webrtc/modules/audio_device/include/audio_device.h b/webrtc/modules/audio_device/include/audio_device.h index c2c2b88103..15e08730c7 100644 --- a/webrtc/modules/audio_device/include/audio_device.h +++ b/webrtc/modules/audio_device/include/audio_device.h @@ -12,7 +12,7 @@ #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_ #include "webrtc/modules/audio_device/include/audio_device_defines.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm index f26e9f1cc7..5bf6b3b780 100644 --- a/webrtc/modules/audio_device/ios/audio_device_ios.mm +++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm @@ -21,7 +21,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/modules/audio_device/fine_audio_buffer.h" -#include "webrtc/modules/utility/interface/helpers_ios.h" +#include "webrtc/modules/utility/include/helpers_ios.h" namespace webrtc { diff --git a/webrtc/modules/audio_device/test/audio_device_test_defines.h b/webrtc/modules/audio_device/test/audio_device_test_defines.h index 5068646bdd..cc8e3e3aef 100644 --- a/webrtc/modules/audio_device/test/audio_device_test_defines.h +++ b/webrtc/modules/audio_device/test/audio_device_test_defines.h @@ -13,7 +13,7 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_device/include/audio_device.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/system_wrappers/include/trace.h" #ifdef _WIN32 diff --git a/webrtc/modules/audio_processing/agc/agc.cc b/webrtc/modules/audio_processing/agc/agc.cc index 706b963aa1..fc78f07ebb 100644 --- a/webrtc/modules/audio_processing/agc/agc.cc +++ b/webrtc/modules/audio_processing/agc/agc.cc @@ -19,7 +19,7 @@ #include "webrtc/base/checks.h" #include "webrtc/modules/audio_processing/agc/histogram.h" #include "webrtc/modules/audio_processing/agc/utility.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { namespace { diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc index 867022dcbf..e4b753bc3d 100644 --- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc +++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc @@ -19,7 +19,7 @@ #include "webrtc/modules/audio_processing/agc/gain_map_internal.h" #include "webrtc/modules/audio_processing/gain_control_impl.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/logging.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/agc/agc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_unittest.cc index 66a8a2b1b3..25b99d8773 100644 --- a/webrtc/modules/audio_processing/agc/agc_unittest.cc +++ b/webrtc/modules/audio_processing/agc/agc_unittest.cc @@ -13,7 +13,7 @@ #include "gmock/gmock.h" #include "gtest/gtest.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/tools/agc/test_utils.h" diff --git a/webrtc/modules/audio_processing/agc/histogram.cc b/webrtc/modules/audio_processing/agc/histogram.cc index 1d3035fe12..5c66727a9f 100644 --- a/webrtc/modules/audio_processing/agc/histogram.cc +++ b/webrtc/modules/audio_processing/agc/histogram.cc @@ -13,7 +13,7 @@ #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/agc/mock_agc.h b/webrtc/modules/audio_processing/agc/mock_agc.h index 13dbd2edd5..e362200d86 100644 --- a/webrtc/modules/audio_processing/agc/mock_agc.h +++ b/webrtc/modules/audio_processing/agc/mock_agc.h @@ -14,7 +14,7 @@ #include "webrtc/modules/audio_processing/agc/agc.h" #include "gmock/gmock.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h index 864633f267..48c9488eb7 100644 --- a/webrtc/modules/audio_processing/audio_buffer.h +++ b/webrtc/modules/audio_processing/audio_buffer.h @@ -15,7 +15,7 @@ #include "webrtc/common_audio/channel_buffer.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/audio_processing/splitting_filter.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/scoped_vector.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index c6574151d0..668ec11e91 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -36,7 +36,7 @@ extern "C" { #include "webrtc/modules/audio_processing/processing_component.h" #include "webrtc/modules/audio_processing/transient/transient_suppressor.h" #include "webrtc/modules/audio_processing/voice_detection_impl.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/file_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc index f4c36d0009..ed20daaa61 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc @@ -14,7 +14,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/config.h" #include "webrtc/modules/audio_processing/test/test_utils.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" using ::testing::Invoke; using ::testing::Return; diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc index eac165c542..eff791d129 100644 --- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc +++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc @@ -24,7 +24,7 @@ #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/audio_processing/test/protobuf_utils.h" #include "webrtc/modules/audio_processing/test/test_utils.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc index 555d9d6133..1383bbe719 100644 --- a/webrtc/modules/audio_processing/test/process_test.cc +++ b/webrtc/modules/audio_processing/test/process_test.cc @@ -22,7 +22,7 @@ #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/audio_processing/test/protobuf_utils.h" #include "webrtc/modules/audio_processing/test/test_utils.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/cpu_features_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h index f53d8ac81c..93a0138c16 100644 --- a/webrtc/modules/audio_processing/test/test_utils.h +++ b/webrtc/modules/audio_processing/test/test_utils.h @@ -22,7 +22,7 @@ #include "webrtc/common_audio/channel_buffer.h" #include "webrtc/common_audio/wav_file.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc index 506abaf203..b7b7595abf 100644 --- a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc +++ b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc @@ -19,7 +19,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_audio/include/audio_util.h" #include "webrtc/modules/audio_processing/agc/agc.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/audio_processing/typing_detection.h b/webrtc/modules/audio_processing/typing_detection.h index 5fa6456e9e..40608f885d 100644 --- a/webrtc/modules/audio_processing/typing_detection.h +++ b/webrtc/modules/audio_processing/typing_detection.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_ #define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc index 39ec37e6ec..fce144de6b 100644 --- a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc +++ b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc @@ -18,7 +18,7 @@ #include "webrtc/modules/audio_processing/vad/common.h" #include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h" #include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/vad/standalone_vad.cc b/webrtc/modules/audio_processing/vad/standalone_vad.cc index 468b8ff3f0..1209526a92 100644 --- a/webrtc/modules/audio_processing/vad/standalone_vad.cc +++ b/webrtc/modules/audio_processing/vad/standalone_vad.cc @@ -12,8 +12,8 @@ #include -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc index 942008e733..5462d05d37 100644 --- a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc +++ b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc @@ -14,7 +14,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc index 8535d1ff57..1a595597b6 100644 --- a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc +++ b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc @@ -23,7 +23,7 @@ extern "C" { #include "webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h" #include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h" } -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc index f509af476f..a8a4ead2e3 100644 --- a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc +++ b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc @@ -21,7 +21,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/audio_processing/vad/common.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/test/testsupport/fileutils.h" namespace webrtc { diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc index 8857ee4b4a..d7795621af 100644 --- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc +++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc @@ -14,7 +14,7 @@ #include #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc index 72831c78d6..2b9e589fbd 100644 --- a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc +++ b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc @@ -14,7 +14,7 @@ #include #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" using webrtc::RtcpBandwidthObserver; using webrtc::BitrateObserver; diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h index bb532886c7..5b20287c63 100644 --- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h +++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h @@ -17,8 +17,8 @@ #include -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h index 40061d3ee7..baf8857559 100644 --- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h +++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h @@ -15,7 +15,7 @@ #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" namespace webrtc { diff --git a/webrtc/modules/include/module.h b/webrtc/modules/include/module.h new file mode 100644 index 0000000000..d02aa95dc8 --- /dev/null +++ b/webrtc/modules/include/module.h @@ -0,0 +1,81 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_ +#define WEBRTC_MODULES_INCLUDE_MODULE_H_ + +#include "webrtc/typedefs.h" + +namespace webrtc { + +class ProcessThread; + +class Module { + public: + // Returns the number of milliseconds until the module wants a worker + // thread to call Process. + // This method is called on the same worker thread as Process will + // be called on. + // TODO(tommi): Almost all implementations of this function, need to know + // the current tick count. Consider passing it as an argument. It could + // also improve the accuracy of when the next callback occurs since the + // thread that calls Process() will also have it's tick count reference + // which might not match with what the implementations use. + virtual int64_t TimeUntilNextProcess() = 0; + + // Process any pending tasks such as timeouts. + // Called on a worker thread. + virtual int32_t Process() = 0; + + // This method is called when the module is attached to a *running* process + // thread or detached from one. In the case of detaching, |process_thread| + // will be nullptr. + // + // This method will be called in the following cases: + // + // * Non-null process_thread: + // * ProcessThread::RegisterModule() is called while the thread is running. + // * ProcessThread::Start() is called and RegisterModule has previously + // been called. The thread will be started immediately after notifying + // all modules. + // + // * Null process_thread: + // * ProcessThread::DeRegisterModule() is called while the thread is + // running. + // * ProcessThread::Stop() was called and the thread has been stopped. + // + // NOTE: This method is not called from the worker thread itself, but from + // the thread that registers/deregisters the module or calls Start/Stop. + virtual void ProcessThreadAttached(ProcessThread* process_thread) {} + + protected: + virtual ~Module() {} +}; + +// Reference counted version of the Module interface. +class RefCountedModule : public Module { + public: + // Increase the reference count by one. + // Returns the incremented reference count. + virtual int32_t AddRef() const = 0; + + // Decrease the reference count by one. + // Returns the decreased reference count. + // Returns 0 if the last reference was just released. + // When the reference count reaches 0 the object will self-destruct. + virtual int32_t Release() const = 0; + + protected: + ~RefCountedModule() override = default; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_INCLUDE_MODULE_H_ diff --git a/webrtc/modules/include/module_common_types.h b/webrtc/modules/include/module_common_types.h new file mode 100644 index 0000000000..fb635813cf --- /dev/null +++ b/webrtc/modules/include/module_common_types.h @@ -0,0 +1,810 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ +#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ + +#include +#include // memcpy + +#include +#include + +#include "webrtc/base/constructormagic.h" +#include "webrtc/common_types.h" +#include "webrtc/common_video/rotation.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct RTPAudioHeader { + uint8_t numEnergy; // number of valid entries in arrOfEnergy + uint8_t arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel + bool isCNG; // is this CNG + uint8_t channel; // number of channels 2 = stereo +}; + +const int16_t kNoPictureId = -1; +const int16_t kMaxOneBytePictureId = 0x7F; // 7 bits +const int16_t kMaxTwoBytePictureId = 0x7FFF; // 15 bits +const int16_t kNoTl0PicIdx = -1; +const uint8_t kNoTemporalIdx = 0xFF; +const uint8_t kNoSpatialIdx = 0xFF; +const uint8_t kNoGofIdx = 0xFF; +const size_t kMaxVp9RefPics = 3; +const size_t kMaxVp9FramesInGof = 0xFF; // 8 bits +const size_t kMaxVp9NumberOfSpatialLayers = 8; +const int kNoKeyIdx = -1; + +struct RTPVideoHeaderVP8 { + void InitRTPVideoHeaderVP8() { + nonReference = false; + pictureId = kNoPictureId; + tl0PicIdx = kNoTl0PicIdx; + temporalIdx = kNoTemporalIdx; + layerSync = false; + keyIdx = kNoKeyIdx; + partitionId = 0; + beginningOfPartition = false; + } + + bool nonReference; // Frame is discardable. + int16_t pictureId; // Picture ID index, 15 bits; + // kNoPictureId if PictureID does not exist. + int16_t tl0PicIdx; // TL0PIC_IDX, 8 bits; + // kNoTl0PicIdx means no value provided. + uint8_t temporalIdx; // Temporal layer index, or kNoTemporalIdx. + bool layerSync; // This frame is a layer sync frame. + // Disabled if temporalIdx == kNoTemporalIdx. + int keyIdx; // 5 bits; kNoKeyIdx means not used. + int partitionId; // VP8 partition ID + bool beginningOfPartition; // True if this packet is the first + // in a VP8 partition. Otherwise false +}; + +enum TemporalStructureMode { + kTemporalStructureMode1, // 1 temporal layer structure - i.e., IPPP... + kTemporalStructureMode2, // 2 temporal layers 0-1-0-1... + kTemporalStructureMode3 // 3 temporal layers 0-2-1-2-0-2-1-2... +}; + +struct GofInfoVP9 { + void SetGofInfoVP9(TemporalStructureMode tm) { + switch (tm) { + case kTemporalStructureMode1: + num_frames_in_gof = 1; + temporal_idx[0] = 0; + temporal_up_switch[0] = false; + num_ref_pics[0] = 1; + pid_diff[0][0] = 1; + break; + case kTemporalStructureMode2: + num_frames_in_gof = 2; + temporal_idx[0] = 0; + temporal_up_switch[0] = false; + num_ref_pics[0] = 1; + pid_diff[0][0] = 2; + + temporal_idx[1] = 1; + temporal_up_switch[1] = true; + num_ref_pics[1] = 1; + pid_diff[1][0] = 1; + break; + case kTemporalStructureMode3: + num_frames_in_gof = 4; + temporal_idx[0] = 0; + temporal_up_switch[0] = false; + num_ref_pics[0] = 1; + pid_diff[0][0] = 4; + + temporal_idx[1] = 2; + temporal_up_switch[1] = true; + num_ref_pics[1] = 1; + pid_diff[1][0] = 1; + + temporal_idx[2] = 1; + temporal_up_switch[2] = true; + num_ref_pics[2] = 1; + pid_diff[2][0] = 2; + + temporal_idx[3] = 2; + temporal_up_switch[3] = false; + num_ref_pics[3] = 2; + pid_diff[3][0] = 1; + pid_diff[3][1] = 2; + break; + default: + assert(false); + } + } + + void CopyGofInfoVP9(const GofInfoVP9& src) { + num_frames_in_gof = src.num_frames_in_gof; + for (size_t i = 0; i < num_frames_in_gof; ++i) { + temporal_idx[i] = src.temporal_idx[i]; + temporal_up_switch[i] = src.temporal_up_switch[i]; + num_ref_pics[i] = src.num_ref_pics[i]; + for (size_t r = 0; r < num_ref_pics[i]; ++r) { + pid_diff[i][r] = src.pid_diff[i][r]; + } + } + } + + size_t num_frames_in_gof; + uint8_t temporal_idx[kMaxVp9FramesInGof]; + bool temporal_up_switch[kMaxVp9FramesInGof]; + size_t num_ref_pics[kMaxVp9FramesInGof]; + int16_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics]; +}; + +struct RTPVideoHeaderVP9 { + void InitRTPVideoHeaderVP9() { + inter_pic_predicted = false; + flexible_mode = false; + beginning_of_frame = false; + end_of_frame = false; + ss_data_available = false; + picture_id = kNoPictureId; + max_picture_id = kMaxTwoBytePictureId; + tl0_pic_idx = kNoTl0PicIdx; + temporal_idx = kNoTemporalIdx; + spatial_idx = kNoSpatialIdx; + temporal_up_switch = false; + inter_layer_predicted = false; + gof_idx = kNoGofIdx; + num_ref_pics = 0; + num_spatial_layers = 1; + } + + bool inter_pic_predicted; // This layer frame is dependent on previously + // coded frame(s). + bool flexible_mode; // This frame is in flexible mode. + bool beginning_of_frame; // True if this packet is the first in a VP9 layer + // frame. + bool end_of_frame; // True if this packet is the last in a VP9 layer frame. + bool ss_data_available; // True if SS data is available in this payload + // descriptor. + int16_t picture_id; // PictureID index, 15 bits; + // kNoPictureId if PictureID does not exist. + int16_t max_picture_id; // Maximum picture ID index; either 0x7F or 0x7FFF; + int16_t tl0_pic_idx; // TL0PIC_IDX, 8 bits; + // kNoTl0PicIdx means no value provided. + uint8_t temporal_idx; // Temporal layer index, or kNoTemporalIdx. + uint8_t spatial_idx; // Spatial layer index, or kNoSpatialIdx. + bool temporal_up_switch; // True if upswitch to higher frame rate is possible + // starting from this frame. + bool inter_layer_predicted; // Frame is dependent on directly lower spatial + // layer frame. + + uint8_t gof_idx; // Index to predefined temporal frame info in SS data. + + size_t num_ref_pics; // Number of reference pictures used by this layer + // frame. + int16_t pid_diff[kMaxVp9RefPics]; // P_DIFF signaled to derive the PictureID + // of the reference pictures. + int16_t ref_picture_id[kMaxVp9RefPics]; // PictureID of reference pictures. + + // SS data. + size_t num_spatial_layers; // Always populated. + bool spatial_layer_resolution_present; + uint16_t width[kMaxVp9NumberOfSpatialLayers]; + uint16_t height[kMaxVp9NumberOfSpatialLayers]; + GofInfoVP9 gof; +}; + +// The packetization types that we support: single, aggregated, and fragmented. +enum H264PacketizationTypes { + kH264SingleNalu, // This packet contains a single NAL unit. + kH264StapA, // This packet contains STAP-A (single time + // aggregation) packets. If this packet has an + // associated NAL unit type, it'll be for the + // first such aggregated packet. + kH264FuA, // This packet contains a FU-A (fragmentation + // unit) packet, meaning it is a part of a frame + // that was too large to fit into a single packet. +}; + +struct RTPVideoHeaderH264 { + uint8_t nalu_type; // The NAL unit type. If this is a header for a + // fragmented packet, it's the NAL unit type of + // the original data. If this is the header for an + // aggregated packet, it's the NAL unit type of + // the first NAL unit in the packet. + H264PacketizationTypes packetization_type; +}; + +union RTPVideoTypeHeader { + RTPVideoHeaderVP8 VP8; + RTPVideoHeaderVP9 VP9; + RTPVideoHeaderH264 H264; +}; + +enum RtpVideoCodecTypes { + kRtpVideoNone, + kRtpVideoGeneric, + kRtpVideoVp8, + kRtpVideoVp9, + kRtpVideoH264 +}; +// Since RTPVideoHeader is used as a member of a union, it can't have a +// non-trivial default constructor. +struct RTPVideoHeader { + uint16_t width; // size + uint16_t height; + VideoRotation rotation; + + bool isFirstPacket; // first packet in frame + uint8_t simulcastIdx; // Index if the simulcast encoder creating + // this frame, 0 if not using simulcast. + RtpVideoCodecTypes codec; + RTPVideoTypeHeader codecHeader; +}; +union RTPTypeHeader { + RTPAudioHeader Audio; + RTPVideoHeader Video; +}; + +struct WebRtcRTPHeader { + RTPHeader header; + FrameType frameType; + RTPTypeHeader type; + // NTP time of the capture time in local timebase in milliseconds. + int64_t ntp_time_ms; +}; + +class RTPFragmentationHeader { + public: + RTPFragmentationHeader() + : fragmentationVectorSize(0), + fragmentationOffset(NULL), + fragmentationLength(NULL), + fragmentationTimeDiff(NULL), + fragmentationPlType(NULL) {}; + + ~RTPFragmentationHeader() { + delete[] fragmentationOffset; + delete[] fragmentationLength; + delete[] fragmentationTimeDiff; + delete[] fragmentationPlType; + } + + void CopyFrom(const RTPFragmentationHeader& src) { + if (this == &src) { + return; + } + + if (src.fragmentationVectorSize != fragmentationVectorSize) { + // new size of vectors + + // delete old + delete[] fragmentationOffset; + fragmentationOffset = NULL; + delete[] fragmentationLength; + fragmentationLength = NULL; + delete[] fragmentationTimeDiff; + fragmentationTimeDiff = NULL; + delete[] fragmentationPlType; + fragmentationPlType = NULL; + + if (src.fragmentationVectorSize > 0) { + // allocate new + if (src.fragmentationOffset) { + fragmentationOffset = new size_t[src.fragmentationVectorSize]; + } + if (src.fragmentationLength) { + fragmentationLength = new size_t[src.fragmentationVectorSize]; + } + if (src.fragmentationTimeDiff) { + fragmentationTimeDiff = new uint16_t[src.fragmentationVectorSize]; + } + if (src.fragmentationPlType) { + fragmentationPlType = new uint8_t[src.fragmentationVectorSize]; + } + } + // set new size + fragmentationVectorSize = src.fragmentationVectorSize; + } + + if (src.fragmentationVectorSize > 0) { + // copy values + if (src.fragmentationOffset) { + memcpy(fragmentationOffset, src.fragmentationOffset, + src.fragmentationVectorSize * sizeof(size_t)); + } + if (src.fragmentationLength) { + memcpy(fragmentationLength, src.fragmentationLength, + src.fragmentationVectorSize * sizeof(size_t)); + } + if (src.fragmentationTimeDiff) { + memcpy(fragmentationTimeDiff, src.fragmentationTimeDiff, + src.fragmentationVectorSize * sizeof(uint16_t)); + } + if (src.fragmentationPlType) { + memcpy(fragmentationPlType, src.fragmentationPlType, + src.fragmentationVectorSize * sizeof(uint8_t)); + } + } + } + + void VerifyAndAllocateFragmentationHeader(const size_t size) { + assert(size <= std::numeric_limits::max()); + const uint16_t size16 = static_cast(size); + if (fragmentationVectorSize < size16) { + uint16_t oldVectorSize = fragmentationVectorSize; + { + // offset + size_t* oldOffsets = fragmentationOffset; + fragmentationOffset = new size_t[size16]; + memset(fragmentationOffset + oldVectorSize, 0, + sizeof(size_t) * (size16 - oldVectorSize)); + // copy old values + memcpy(fragmentationOffset, oldOffsets, + sizeof(size_t) * oldVectorSize); + delete[] oldOffsets; + } + // length + { + size_t* oldLengths = fragmentationLength; + fragmentationLength = new size_t[size16]; + memset(fragmentationLength + oldVectorSize, 0, + sizeof(size_t) * (size16 - oldVectorSize)); + memcpy(fragmentationLength, oldLengths, + sizeof(size_t) * oldVectorSize); + delete[] oldLengths; + } + // time diff + { + uint16_t* oldTimeDiffs = fragmentationTimeDiff; + fragmentationTimeDiff = new uint16_t[size16]; + memset(fragmentationTimeDiff + oldVectorSize, 0, + sizeof(uint16_t) * (size16 - oldVectorSize)); + memcpy(fragmentationTimeDiff, oldTimeDiffs, + sizeof(uint16_t) * oldVectorSize); + delete[] oldTimeDiffs; + } + // payload type + { + uint8_t* oldTimePlTypes = fragmentationPlType; + fragmentationPlType = new uint8_t[size16]; + memset(fragmentationPlType + oldVectorSize, 0, + sizeof(uint8_t) * (size16 - oldVectorSize)); + memcpy(fragmentationPlType, oldTimePlTypes, + sizeof(uint8_t) * oldVectorSize); + delete[] oldTimePlTypes; + } + fragmentationVectorSize = size16; + } + } + + uint16_t fragmentationVectorSize; // Number of fragmentations + size_t* fragmentationOffset; // Offset of pointer to data for each + // fragmentation + size_t* fragmentationLength; // Data size for each fragmentation + uint16_t* fragmentationTimeDiff; // Timestamp difference relative "now" for + // each fragmentation + uint8_t* fragmentationPlType; // Payload type of each fragmentation + + private: + RTC_DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader); +}; + +struct RTCPVoIPMetric { + // RFC 3611 4.7 + uint8_t lossRate; + uint8_t discardRate; + uint8_t burstDensity; + uint8_t gapDensity; + uint16_t burstDuration; + uint16_t gapDuration; + uint16_t roundTripDelay; + uint16_t endSystemDelay; + uint8_t signalLevel; + uint8_t noiseLevel; + uint8_t RERL; + uint8_t Gmin; + uint8_t Rfactor; + uint8_t extRfactor; + uint8_t MOSLQ; + uint8_t MOSCQ; + uint8_t RXconfig; + uint16_t JBnominal; + uint16_t JBmax; + uint16_t JBabsMax; +}; + +// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a +// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive +// loss model. The packet masks are defined in +// modules/rtp_rtcp/fec_private_tables_random(bursty).h +enum FecMaskType { + kFecMaskRandom, + kFecMaskBursty, +}; + +// Struct containing forward error correction settings. +struct FecProtectionParams { + int fec_rate; + bool use_uep_protection; + int max_fec_frames; + FecMaskType fec_mask_type; +}; + +// Interface used by the CallStats class to distribute call statistics. +// Callbacks will be triggered as soon as the class has been registered to a +// CallStats object using RegisterStatsObserver. +class CallStatsObserver { + public: + virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0; + + virtual ~CallStatsObserver() {} +}; + +struct VideoContentMetrics { + VideoContentMetrics() + : motion_magnitude(0.0f), + spatial_pred_err(0.0f), + spatial_pred_err_h(0.0f), + spatial_pred_err_v(0.0f) {} + + void Reset() { + motion_magnitude = 0.0f; + spatial_pred_err = 0.0f; + spatial_pred_err_h = 0.0f; + spatial_pred_err_v = 0.0f; + } + float motion_magnitude; + float spatial_pred_err; + float spatial_pred_err_h; + float spatial_pred_err_v; +}; + +/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It + * allows for adding and subtracting frames while keeping track of the resulting + * states. + * + * Notes + * - The total number of samples in |data_| is + * samples_per_channel_ * num_channels_ + * + * - Stereo data is interleaved starting with the left channel. + * + * - The +operator assume that you would never add exactly opposite frames when + * deciding the resulting state. To do this use the -operator. + */ +class AudioFrame { + public: + // Stereo, 32 kHz, 60 ms (2 * 32 * 60) + static const size_t kMaxDataSizeSamples = 3840; + + enum VADActivity { + kVadActive = 0, + kVadPassive = 1, + kVadUnknown = 2 + }; + enum SpeechType { + kNormalSpeech = 0, + kPLC = 1, + kCNG = 2, + kPLCCNG = 3, + kUndefined = 4 + }; + + AudioFrame(); + virtual ~AudioFrame() {} + + // Resets all members to their default state (except does not modify the + // contents of |data_|). + void Reset(); + + // |interleaved_| is not changed by this method. + void UpdateFrame(int id, uint32_t timestamp, const int16_t* data, + size_t samples_per_channel, int sample_rate_hz, + SpeechType speech_type, VADActivity vad_activity, + int num_channels = 1, uint32_t energy = -1); + + AudioFrame& Append(const AudioFrame& rhs); + + void CopyFrom(const AudioFrame& src); + + void Mute(); + + AudioFrame& operator>>=(const int rhs); + AudioFrame& operator+=(const AudioFrame& rhs); + AudioFrame& operator-=(const AudioFrame& rhs); + + int id_; + // RTP timestamp of the first sample in the AudioFrame. + uint32_t timestamp_; + // Time since the first frame in milliseconds. + // -1 represents an uninitialized value. + int64_t elapsed_time_ms_; + // NTP time of the estimated capture time in local timebase in milliseconds. + // -1 represents an uninitialized value. + int64_t ntp_time_ms_; + int16_t data_[kMaxDataSizeSamples]; + size_t samples_per_channel_; + int sample_rate_hz_; + int num_channels_; + SpeechType speech_type_; + VADActivity vad_activity_; + // Note that there is no guarantee that |energy_| is correct. Any user of this + // member must verify that the value is correct. + // TODO(henrike) Remove |energy_|. + // See https://code.google.com/p/webrtc/issues/detail?id=3315. + uint32_t energy_; + bool interleaved_; + + private: + RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame); +}; + +inline AudioFrame::AudioFrame() + : data_() { + Reset(); +} + +inline void AudioFrame::Reset() { + id_ = -1; + // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize + // to an invalid value, or add a new member to indicate invalidity. + timestamp_ = 0; + elapsed_time_ms_ = -1; + ntp_time_ms_ = -1; + samples_per_channel_ = 0; + sample_rate_hz_ = 0; + num_channels_ = 0; + speech_type_ = kUndefined; + vad_activity_ = kVadUnknown; + energy_ = 0xffffffff; + interleaved_ = true; +} + +inline void AudioFrame::UpdateFrame(int id, + uint32_t timestamp, + const int16_t* data, + size_t samples_per_channel, + int sample_rate_hz, + SpeechType speech_type, + VADActivity vad_activity, + int num_channels, + uint32_t energy) { + id_ = id; + timestamp_ = timestamp; + samples_per_channel_ = samples_per_channel; + sample_rate_hz_ = sample_rate_hz; + speech_type_ = speech_type; + vad_activity_ = vad_activity; + num_channels_ = num_channels; + energy_ = energy; + + assert(num_channels >= 0); + const size_t length = samples_per_channel * num_channels; + assert(length <= kMaxDataSizeSamples); + if (data != NULL) { + memcpy(data_, data, sizeof(int16_t) * length); + } else { + memset(data_, 0, sizeof(int16_t) * length); + } +} + +inline void AudioFrame::CopyFrom(const AudioFrame& src) { + if (this == &src) return; + + id_ = src.id_; + timestamp_ = src.timestamp_; + elapsed_time_ms_ = src.elapsed_time_ms_; + ntp_time_ms_ = src.ntp_time_ms_; + samples_per_channel_ = src.samples_per_channel_; + sample_rate_hz_ = src.sample_rate_hz_; + speech_type_ = src.speech_type_; + vad_activity_ = src.vad_activity_; + num_channels_ = src.num_channels_; + energy_ = src.energy_; + interleaved_ = src.interleaved_; + + assert(num_channels_ >= 0); + const size_t length = samples_per_channel_ * num_channels_; + assert(length <= kMaxDataSizeSamples); + memcpy(data_, src.data_, sizeof(int16_t) * length); +} + +inline void AudioFrame::Mute() { + memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); +} + +inline AudioFrame& AudioFrame::operator>>=(const int rhs) { + assert((num_channels_ > 0) && (num_channels_ < 3)); + if ((num_channels_ > 2) || (num_channels_ < 1)) return *this; + + for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) { + data_[i] = static_cast(data_[i] >> rhs); + } + return *this; +} + +inline AudioFrame& AudioFrame::Append(const AudioFrame& rhs) { + // Sanity check + assert((num_channels_ > 0) && (num_channels_ < 3)); + assert(interleaved_ == rhs.interleaved_); + if ((num_channels_ > 2) || (num_channels_ < 1)) return *this; + if (num_channels_ != rhs.num_channels_) return *this; + + if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) { + vad_activity_ = kVadActive; + } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) { + vad_activity_ = kVadUnknown; + } + if (speech_type_ != rhs.speech_type_) { + speech_type_ = kUndefined; + } + + size_t offset = samples_per_channel_ * num_channels_; + for (size_t i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) { + data_[offset + i] = rhs.data_[i]; + } + samples_per_channel_ += rhs.samples_per_channel_; + return *this; +} + +namespace { +inline int16_t ClampToInt16(int32_t input) { + if (input < -0x00008000) { + return -0x8000; + } else if (input > 0x00007FFF) { + return 0x7FFF; + } else { + return static_cast(input); + } +} +} + +inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) { + // Sanity check + assert((num_channels_ > 0) && (num_channels_ < 3)); + assert(interleaved_ == rhs.interleaved_); + if ((num_channels_ > 2) || (num_channels_ < 1)) return *this; + if (num_channels_ != rhs.num_channels_) return *this; + + bool noPrevData = false; + if (samples_per_channel_ != rhs.samples_per_channel_) { + if (samples_per_channel_ == 0) { + // special case we have no data to start with + samples_per_channel_ = rhs.samples_per_channel_; + noPrevData = true; + } else { + return *this; + } + } + + if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) { + vad_activity_ = kVadActive; + } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) { + vad_activity_ = kVadUnknown; + } + + if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined; + + if (noPrevData) { + memcpy(data_, rhs.data_, + sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_); + } else { + // IMPROVEMENT this can be done very fast in assembly + for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) { + int32_t wrap_guard = + static_cast(data_[i]) + static_cast(rhs.data_[i]); + data_[i] = ClampToInt16(wrap_guard); + } + } + energy_ = 0xffffffff; + return *this; +} + +inline AudioFrame& AudioFrame::operator-=(const AudioFrame& rhs) { + // Sanity check + assert((num_channels_ > 0) && (num_channels_ < 3)); + assert(interleaved_ == rhs.interleaved_); + if ((num_channels_ > 2) || (num_channels_ < 1)) return *this; + + if ((samples_per_channel_ != rhs.samples_per_channel_) || + (num_channels_ != rhs.num_channels_)) { + return *this; + } + if ((vad_activity_ != kVadPassive) || rhs.vad_activity_ != kVadPassive) { + vad_activity_ = kVadUnknown; + } + speech_type_ = kUndefined; + + for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) { + int32_t wrap_guard = + static_cast(data_[i]) - static_cast(rhs.data_[i]); + data_[i] = ClampToInt16(wrap_guard); + } + energy_ = 0xffffffff; + return *this; +} + +inline bool IsNewerSequenceNumber(uint16_t sequence_number, + uint16_t prev_sequence_number) { + // Distinguish between elements that are exactly 0x8000 apart. + // If s1>s2 and |s1-s2| = 0x8000: IsNewer(s1,s2)=true, IsNewer(s2,s1)=false + // rather than having IsNewer(s1,s2) = IsNewer(s2,s1) = false. + if (static_cast(sequence_number - prev_sequence_number) == 0x8000) { + return sequence_number > prev_sequence_number; + } + return sequence_number != prev_sequence_number && + static_cast(sequence_number - prev_sequence_number) < 0x8000; +} + +inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) { + // Distinguish between elements that are exactly 0x80000000 apart. + // If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true, + // IsNewer(t2,t1)=false + // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false. + if (static_cast(timestamp - prev_timestamp) == 0x80000000) { + return timestamp > prev_timestamp; + } + return timestamp != prev_timestamp && + static_cast(timestamp - prev_timestamp) < 0x80000000; +} + +inline uint16_t LatestSequenceNumber(uint16_t sequence_number1, + uint16_t sequence_number2) { + return IsNewerSequenceNumber(sequence_number1, sequence_number2) + ? sequence_number1 + : sequence_number2; +} + +inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) { + return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2; +} + +// Utility class to unwrap a sequence number to a larger type, for easier +// handling large ranges. Note that sequence numbers will never be unwrapped +// to a negative value. +class SequenceNumberUnwrapper { + public: + SequenceNumberUnwrapper() : last_seq_(-1) {} + + // Get the unwrapped sequence, but don't update the internal state. + int64_t UnwrapWithoutUpdate(uint16_t sequence_number) { + if (last_seq_ == -1) + return sequence_number; + + uint16_t cropped_last = static_cast(last_seq_); + int64_t delta = sequence_number - cropped_last; + if (IsNewerSequenceNumber(sequence_number, cropped_last)) { + if (delta < 0) + delta += (1 << 16); // Wrap forwards. + } else if (delta > 0 && (last_seq_ + delta - (1 << 16)) >= 0) { + // If sequence_number is older but delta is positive, this is a backwards + // wrap-around. However, don't wrap backwards past 0 (unwrapped). + delta -= (1 << 16); + } + + return last_seq_ + delta; + } + + // Only update the internal state to the specified last (unwrapped) sequence. + void UpdateLast(int64_t last_sequence) { last_seq_ = last_sequence; } + + // Unwrap the sequence number and update the internal state. + int64_t Unwrap(uint16_t sequence_number) { + int64_t unwrapped = UnwrapWithoutUpdate(sequence_number); + UpdateLast(unwrapped); + return unwrapped; + } + + private: + int64_t last_seq_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ diff --git a/webrtc/modules/interface/module.h b/webrtc/modules/interface/module.h index ffd3065a5c..06054c3d22 100644 --- a/webrtc/modules/interface/module.h +++ b/webrtc/modules/interface/module.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef MODULES_INTERFACE_MODULE_H_ -#define MODULES_INTERFACE_MODULE_H_ +#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_ +#define WEBRTC_MODULES_INCLUDE_MODULE_H_ + +#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include") #include "webrtc/typedefs.h" @@ -78,4 +80,4 @@ class RefCountedModule : public Module { } // namespace webrtc -#endif // MODULES_INTERFACE_MODULE_H_ +#endif // WEBRTC_MODULES_INCLUDE_MODULE_H_ diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h index 45e93d8fad..e76a71495b 100644 --- a/webrtc/modules/interface/module_common_types.h +++ b/webrtc/modules/interface/module_common_types.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef MODULE_COMMON_TYPES_H -#define MODULE_COMMON_TYPES_H +#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ +#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ + +#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include") #include #include // memcpy @@ -807,4 +809,4 @@ class SequenceNumberUnwrapper { } // namespace webrtc -#endif // MODULE_COMMON_TYPES_H +#endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn index 05cfb4e555..70abb51f4d 100644 --- a/webrtc/modules/media_file/BUILD.gn +++ b/webrtc/modules/media_file/BUILD.gn @@ -14,8 +14,8 @@ config("media_file_config") { source_set("media_file") { sources = [ - "interface/media_file.h", - "interface/media_file_defines.h", + "include/media_file.h", + "include/media_file_defines.h", "source/media_file_impl.cc", "source/media_file_impl.h", "source/media_file_utility.cc", diff --git a/webrtc/modules/media_file/include/media_file.h b/webrtc/modules/media_file/include/media_file.h new file mode 100644 index 0000000000..22c727e1a8 --- /dev/null +++ b/webrtc/modules/media_file/include/media_file.h @@ -0,0 +1,180 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ +#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ + +#include "webrtc/common_types.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +class MediaFile : public Module +{ +public: + // Factory method. Constructor disabled. id is the identifier for the + // MediaFile instance. + static MediaFile* CreateMediaFile(const int32_t id); + static void DestroyMediaFile(MediaFile* module); + + // Put 10-60ms of audio data from file into the audioBuffer depending on + // codec frame size. dataLengthInBytes is both an input and output + // parameter. As input parameter it indicates the size of audioBuffer. + // As output parameter it indicates the number of bytes written to + // audioBuffer. + // Note: This API only play mono audio but can be used on file containing + // audio with more channels (in which case the audio will be converted to + // mono). + virtual int32_t PlayoutAudioData( + int8_t* audioBuffer, + size_t& dataLengthInBytes) = 0; + + // Put 10-60ms, depending on codec frame size, of audio data from file into + // audioBufferLeft and audioBufferRight. The buffers contain the left and + // right channel of played out stereo audio. + // dataLengthInBytes is both an input and output parameter. As input + // parameter it indicates the size of both audioBufferLeft and + // audioBufferRight. As output parameter it indicates the number of bytes + // written to both audio buffers. + // Note: This API can only be successfully called for WAV files with stereo + // audio. + virtual int32_t PlayoutStereoData( + int8_t* audioBufferLeft, + int8_t* audioBufferRight, + size_t& dataLengthInBytes) = 0; + + // Open the file specified by fileName (relative path is allowed) for + // reading. FileCallback::PlayNotification(..) will be called after + // notificationTimeMs of the file has been played if notificationTimeMs is + // greater than zero. If loop is true the file will be played until + // StopPlaying() is called. When end of file is reached the file is read + // from the start. format specifies the type of file fileName refers to. + // codecInst specifies the encoding of the audio data. Note that + // file formats that contain this information (like WAV files) don't need to + // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero, + // specify what part of the file should be read. From startPointMs ms to + // stopPointMs ms. + // Note: codecInst.channels should be set to 2 for stereo (and 1 for + // mono). Stereo audio is only supported for WAV files. + virtual int32_t StartPlayingAudioFile( + const char* fileName, + const uint32_t notificationTimeMs = 0, + const bool loop = false, + const FileFormats format = kFileFormatPcm16kHzFile, + const CodecInst* codecInst = NULL, + const uint32_t startPointMs = 0, + const uint32_t stopPointMs = 0) = 0; + + // Prepare for playing audio from stream. + // FileCallback::PlayNotification(..) will be called after + // notificationTimeMs of the file has been played if notificationTimeMs is + // greater than zero. format specifies the type of file fileName refers to. + // codecInst specifies the encoding of the audio data. Note that + // file formats that contain this information (like WAV files) don't need to + // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero, + // specify what part of the file should be read. From startPointMs ms to + // stopPointMs ms. + // Note: codecInst.channels should be set to 2 for stereo (and 1 for + // mono). Stereo audio is only supported for WAV files. + virtual int32_t StartPlayingAudioStream( + InStream& stream, + const uint32_t notificationTimeMs = 0, + const FileFormats format = kFileFormatPcm16kHzFile, + const CodecInst* codecInst = NULL, + const uint32_t startPointMs = 0, + const uint32_t stopPointMs = 0) = 0; + + // Stop playing from file or stream. + virtual int32_t StopPlaying() = 0; + + // Return true if playing. + virtual bool IsPlaying() = 0; + + + // Set durationMs to the number of ms that has been played from file. + virtual int32_t PlayoutPositionMs( + uint32_t& durationMs) const = 0; + + // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, + // to file. The audio frame size is determined by the codecInst.pacsize + // parameter of the last sucessfull StartRecordingAudioFile(..) call. + // Note: bufferLength must be exactly one frame. + virtual int32_t IncomingAudioData( + const int8_t* audioBuffer, + const size_t bufferLength) = 0; + + // Open/creates file specified by fileName for writing (relative path is + // allowed). FileCallback::RecordNotification(..) will be called after + // notificationTimeMs of audio data has been recorded if + // notificationTimeMs is greater than zero. + // format specifies the type of file that should be created/opened. + // codecInst specifies the encoding of the audio data. maxSizeBytes + // specifies the number of bytes allowed to be written to file if it is + // greater than zero. + // Note: codecInst.channels should be set to 2 for stereo (and 1 for + // mono). Stereo is only supported for WAV files. + virtual int32_t StartRecordingAudioFile( + const char* fileName, + const FileFormats format, + const CodecInst& codecInst, + const uint32_t notificationTimeMs = 0, + const uint32_t maxSizeBytes = 0) = 0; + + // Prepare for recording audio to stream. + // FileCallback::RecordNotification(..) will be called after + // notificationTimeMs of audio data has been recorded if + // notificationTimeMs is greater than zero. + // format specifies the type of file that stream should correspond to. + // codecInst specifies the encoding of the audio data. + // Note: codecInst.channels should be set to 2 for stereo (and 1 for + // mono). Stereo is only supported for WAV files. + virtual int32_t StartRecordingAudioStream( + OutStream& stream, + const FileFormats format, + const CodecInst& codecInst, + const uint32_t notificationTimeMs = 0) = 0; + + // Stop recording to file or stream. + virtual int32_t StopRecording() = 0; + + // Return true if recording. + virtual bool IsRecording() = 0; + + // Set durationMs to the number of ms that has been recorded to file. + virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0; + + // Return true if recording or playing is stereo. + virtual bool IsStereo() = 0; + + // Register callback to receive media file related notifications. Disables + // callbacks if callback is NULL. + virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0; + + // Set durationMs to the size of the file (in ms) specified by fileName. + // format specifies the type of file fileName refers to. freqInHz specifies + // the sampling frequency of the file. + virtual int32_t FileDurationMs( + const char* fileName, + uint32_t& durationMs, + const FileFormats format, + const uint32_t freqInHz = 16000) = 0; + + // Update codecInst according to the current audio codec being used for + // reading or writing. + virtual int32_t codec_info(CodecInst& codecInst) const = 0; + +protected: + MediaFile() {} + virtual ~MediaFile() {} +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ diff --git a/webrtc/modules/media_file/include/media_file_defines.h b/webrtc/modules/media_file/include/media_file_defines.h new file mode 100644 index 0000000000..05e7e72770 --- /dev/null +++ b/webrtc/modules/media_file/include/media_file_defines.h @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_ +#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_ + +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { +// Callback class for the MediaFile class. +class FileCallback +{ +public: + virtual ~FileCallback(){} + + // This function is called by MediaFile when a file has been playing for + // durationMs ms. id is the identifier for the MediaFile instance calling + // the callback. + virtual void PlayNotification(const int32_t id, + const uint32_t durationMs) = 0; + + // This function is called by MediaFile when a file has been recording for + // durationMs ms. id is the identifier for the MediaFile instance calling + // the callback. + virtual void RecordNotification(const int32_t id, + const uint32_t durationMs) = 0; + + // This function is called by MediaFile when a file has been stopped + // playing. id is the identifier for the MediaFile instance calling the + // callback. + virtual void PlayFileEnded(const int32_t id) = 0; + + // This function is called by MediaFile when a file has been stopped + // recording. id is the identifier for the MediaFile instance calling the + // callback. + virtual void RecordFileEnded(const int32_t id) = 0; + +protected: + FileCallback() {} +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_ diff --git a/webrtc/modules/media_file/interface/media_file.h b/webrtc/modules/media_file/interface/media_file.h index 5b09ad4383..3be5f08251 100644 --- a/webrtc/modules/media_file/interface/media_file.h +++ b/webrtc/modules/media_file/interface/media_file.h @@ -8,13 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_ -#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_ +#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ +#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ + +#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include") #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/media_file/interface/media_file_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -177,4 +179,4 @@ protected: virtual ~MediaFile() {} }; } // namespace webrtc -#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_ +#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_ diff --git a/webrtc/modules/media_file/interface/media_file_defines.h b/webrtc/modules/media_file/interface/media_file_defines.h index ded71a8ca7..345fe547b2 100644 --- a/webrtc/modules/media_file/interface/media_file_defines.h +++ b/webrtc/modules/media_file/interface/media_file_defines.h @@ -8,11 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_ -#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_ +#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_ +#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_ + +#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include") #include "webrtc/engine_configurations.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -48,4 +50,4 @@ protected: FileCallback() {} }; } // namespace webrtc -#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_ +#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_ diff --git a/webrtc/modules/media_file/media_file.gypi b/webrtc/modules/media_file/media_file.gypi index 4ec80c3c52..b3d8968b16 100644 --- a/webrtc/modules/media_file/media_file.gypi +++ b/webrtc/modules/media_file/media_file.gypi @@ -17,8 +17,8 @@ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', ], 'sources': [ - 'interface/media_file.h', - 'interface/media_file_defines.h', + 'include/media_file.h', + 'include/media_file_defines.h', 'source/media_file_impl.cc', 'source/media_file_impl.h', 'source/media_file_utility.cc', diff --git a/webrtc/modules/media_file/source/media_file_impl.h b/webrtc/modules/media_file/source/media_file_impl.h index cdb54d880d..618497955e 100644 --- a/webrtc/modules/media_file/source/media_file_impl.h +++ b/webrtc/modules/media_file/source/media_file_impl.h @@ -12,9 +12,9 @@ #define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_ #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/media_file/interface/media_file.h" -#include "webrtc/modules/media_file/interface/media_file_defines.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/media_file/include/media_file.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" #include "webrtc/modules/media_file/source/media_file_utility.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/media_file/source/media_file_unittest.cc b/webrtc/modules/media_file/source/media_file_unittest.cc index 370d13228a..4a50c2e2d0 100644 --- a/webrtc/modules/media_file/source/media_file_unittest.cc +++ b/webrtc/modules/media_file/source/media_file_unittest.cc @@ -9,7 +9,7 @@ */ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/media_file/interface/media_file.h" +#include "webrtc/modules/media_file/include/media_file.h" #include "webrtc/system_wrappers/include/sleep.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" diff --git a/webrtc/modules/media_file/source/media_file_utility.cc b/webrtc/modules/media_file/source/media_file_utility.cc index 61ae442d0e..fad7fe4079 100644 --- a/webrtc/modules/media_file/source/media_file_utility.cc +++ b/webrtc/modules/media_file/source/media_file_utility.cc @@ -19,7 +19,7 @@ #include "webrtc/common_audio/wav_header.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/file_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/source/media_file_utility.h index 2823ceca8a..65921bd963 100644 --- a/webrtc/modules/media_file/source/media_file_utility.h +++ b/webrtc/modules/media_file/source/media_file_utility.h @@ -15,7 +15,7 @@ #include #include "webrtc/common_types.h" -#include "webrtc/modules/media_file/interface/media_file_defines.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" namespace webrtc { class InStream; diff --git a/webrtc/modules/module_common_types_unittest.cc b/webrtc/modules/module_common_types_unittest.cc index bc0b7a1a5b..acd58476a1 100644 --- a/webrtc/modules/module_common_types_unittest.cc +++ b/webrtc/modules/module_common_types_unittest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "testing/gtest/include/gtest/gtest.h" diff --git a/webrtc/modules/pacing/include/paced_sender.h b/webrtc/modules/pacing/include/paced_sender.h index f142f55173..2c8e36ec85 100644 --- a/webrtc/modules/pacing/include/paced_sender.h +++ b/webrtc/modules/pacing/include/paced_sender.h @@ -16,8 +16,8 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/include/packet_router.h index 9d461d13a9..7a65295f83 100644 --- a/webrtc/modules/pacing/include/packet_router.h +++ b/webrtc/modules/pacing/include/packet_router.h @@ -19,7 +19,7 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/modules/pacing/include/paced_sender.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc index 5d7ae17b23..40887154a4 100644 --- a/webrtc/modules/pacing/paced_sender.cc +++ b/webrtc/modules/pacing/paced_sender.cc @@ -16,7 +16,7 @@ #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/pacing/bitrate_prober.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc index 563773b41f..7af74ec723 100644 --- a/webrtc/modules/pacing/packet_router.cc +++ b/webrtc/modules/pacing/packet_router.cc @@ -12,8 +12,8 @@ #include "webrtc/base/atomicops.h" #include "webrtc/base/checks.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" namespace webrtc { diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc index eecb13757c..f17f797fbb 100644 --- a/webrtc/modules/pacing/packet_router_unittest.cc +++ b/webrtc/modules/pacing/packet_router_unittest.cc @@ -14,7 +14,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/pacing/include/packet_router.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/base/scoped_ptr.h" diff --git a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h index 4bd9d8c7bc..0734cbf255 100644 --- a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h +++ b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h @@ -17,9 +17,9 @@ #include #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc index 3dee305bad..961d6bfb8e 100644 --- a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc +++ b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc @@ -14,7 +14,7 @@ #include #include "webrtc/base/logging.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h index bb69a8a0a1..56e9c14206 100644 --- a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h +++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h @@ -13,7 +13,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc index b7f9f65dbc..5c453705f7 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc +++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc @@ -15,7 +15,7 @@ #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/modules/pacing/include/packet_router.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h index e867ff77a4..98a68b3dcf 100644 --- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h +++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h @@ -15,7 +15,7 @@ #include #include "webrtc/base/criticalsection.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc index f837638474..f2ff7680dd 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc @@ -14,7 +14,7 @@ #include "webrtc/base/common.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h" #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h" #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h" diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h index 64dfa85535..b3df7124e3 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h +++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_BASELINEFILE_H_ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { namespace testing { diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h index 2881eba424..d470324ac3 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h +++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h @@ -16,7 +16,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { namespace testing { diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h index 6b24cf30a6..6b5f04b68f 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h +++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h @@ -24,12 +24,12 @@ #include "webrtc/base/common.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/pacing/include/paced_sender.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" #include "webrtc/modules/remote_bitrate_estimator/test/packet.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/test/random.h" diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc index d77447f1ea..171d196fc0 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc @@ -21,7 +21,7 @@ #include "webrtc/base/common.h" #include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" namespace webrtc { namespace testing { diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h index eee90cf463..bf23d09884 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h +++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h @@ -20,7 +20,7 @@ #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h" #include "webrtc/voice_engine/channel.h" diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc index b18b9f06b9..9599b01933 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc @@ -17,7 +17,7 @@ #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" namespace webrtc { namespace testing { diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc index a02abc6ab8..b7e4f971fa 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc @@ -16,7 +16,7 @@ #include "webrtc/base/common.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" namespace webrtc { namespace testing { diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet.h b/webrtc/modules/remote_bitrate_estimator/test/packet.h index 11885a4544..4a361c4dc2 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/packet.h +++ b/webrtc/modules/remote_bitrate_estimator/test/packet.h @@ -16,7 +16,7 @@ #include #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc index f70c212af7..793e06421f 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc @@ -14,10 +14,10 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/common.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include "webrtc/system_wrappers/include/clock.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc index f1faa49d7e..3bcbc0a071 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc +++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc @@ -15,7 +15,7 @@ #include #include "webrtc/base/checks.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h" #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h" diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h index c42647e2d3..0b3741d3c7 100644 --- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h +++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h @@ -17,7 +17,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h" #include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h" diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc index 9493805a1c..083d1fd469 100644 --- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc +++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc @@ -17,8 +17,8 @@ #include "gflags/gflags.h" #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h" #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "webrtc/test/rtp_file_reader.h" namespace flags { diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc index 19e4a07b4d..4574faf8b7 100644 --- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc +++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc @@ -14,8 +14,8 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "webrtc/test/rtp_file_reader.h" class Observer : public webrtc::RemoteBitrateObserver { diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc index e277481886..bf698728e8 100644 --- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc +++ b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc @@ -14,8 +14,8 @@ #include "webrtc/base/format_macros.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "webrtc/test/rtp_file_reader.h" int main(int argc, char** argv) { diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc index f2e073aa53..332590b1c6 100644 --- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc +++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc @@ -15,7 +15,7 @@ #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h" #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" namespace webrtc { diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h index 58829b072b..93f30e6cee 100644 --- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h +++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h @@ -15,8 +15,8 @@ #include "webrtc/base/criticalsection.h" #include "webrtc/base/thread_annotations.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h" diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc index b2bc646e2d..64d0e55397 100644 --- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc +++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc @@ -18,9 +18,9 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h" #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" -#include "webrtc/modules/utility/interface/mock/mock_process_thread.h" +#include "webrtc/modules/utility/include/mock/mock_process_thread.h" #include "webrtc/system_wrappers/include/clock.h" using ::testing::_; diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index c651424611..6239c0420d 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -10,14 +10,14 @@ import("../../build/webrtc.gni") source_set("rtp_rtcp") { sources = [ - "interface/fec_receiver.h", - "interface/receive_statistics.h", - "interface/remote_ntp_time_estimator.h", - "interface/rtp_header_parser.h", - "interface/rtp_payload_registry.h", - "interface/rtp_receiver.h", - "interface/rtp_rtcp.h", - "interface/rtp_rtcp_defines.h", + "include/fec_receiver.h", + "include/receive_statistics.h", + "include/remote_ntp_time_estimator.h", + "include/rtp_header_parser.h", + "include/rtp_payload_registry.h", + "include/rtp_receiver.h", + "include/rtp_rtcp.h", + "include/rtp_rtcp_defines.h", "mocks/mock_rtp_rtcp.h", "source/bitrate.cc", "source/bitrate.h", diff --git a/webrtc/modules/rtp_rtcp/include/fec_receiver.h b/webrtc/modules/rtp_rtcp/include/fec_receiver.h new file mode 100644 index 0000000000..65e85ad7a5 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/fec_receiver.h @@ -0,0 +1,46 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct FecPacketCounter { + FecPacketCounter() + : num_packets(0), + num_fec_packets(0), + num_recovered_packets(0) {} + + size_t num_packets; // Number of received packets. + size_t num_fec_packets; // Number of received FEC packets. + size_t num_recovered_packets; // Number of recovered media packets using FEC. +}; + +class FecReceiver { + public: + static FecReceiver* Create(RtpData* callback); + + virtual ~FecReceiver() {} + + virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header, + const uint8_t* incoming_rtp_packet, + size_t packet_length, + uint8_t ulpfec_payload_type) = 0; + + virtual int32_t ProcessReceivedFec() = 0; + + virtual FecPacketCounter GetPacketCounter() const = 0; +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ diff --git a/webrtc/modules/rtp_rtcp/include/receive_statistics.h b/webrtc/modules/rtp_rtcp/include/receive_statistics.h new file mode 100644 index 0000000000..b4a7cd0de2 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/receive_statistics.h @@ -0,0 +1,102 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ + +#include + +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class Clock; + +class StreamStatistician { + public: + virtual ~StreamStatistician(); + + virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0; + virtual void GetDataCounters(size_t* bytes_received, + uint32_t* packets_received) const = 0; + + // Gets received stream data counters (includes reset counter values). + virtual void GetReceiveStreamDataCounters( + StreamDataCounters* data_counters) const = 0; + + virtual uint32_t BitrateReceived() const = 0; + + // Returns true if the packet with RTP header |header| is likely to be a + // retransmitted packet, false otherwise. + virtual bool IsRetransmitOfOldPacket(const RTPHeader& header, + int64_t min_rtt) const = 0; + + // Returns true if |sequence_number| is received in order, false otherwise. + virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0; +}; + +typedef std::map StatisticianMap; + +class ReceiveStatistics : public Module { + public: + virtual ~ReceiveStatistics() {} + + static ReceiveStatistics* Create(Clock* clock); + + // Updates the receive statistics with this packet. + virtual void IncomingPacket(const RTPHeader& rtp_header, + size_t packet_length, + bool retransmitted) = 0; + + // Increment counter for number of FEC packets received. + virtual void FecPacketReceived(const RTPHeader& header, + size_t packet_length) = 0; + + // Returns a map of all statisticians which have seen an incoming packet + // during the last two seconds. + virtual StatisticianMap GetActiveStatisticians() const = 0; + + // Returns a pointer to the statistician of an ssrc. + virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0; + + // Sets the max reordering threshold in number of packets. + virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0; + + // Called on new RTCP stats creation. + virtual void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) = 0; + + // Called on new RTP stats creation. + virtual void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) = 0; +}; + +class NullReceiveStatistics : public ReceiveStatistics { + public: + void IncomingPacket(const RTPHeader& rtp_header, + size_t packet_length, + bool retransmitted) override; + void FecPacketReceived(const RTPHeader& header, + size_t packet_length) override; + StatisticianMap GetActiveStatisticians() const override; + StreamStatistician* GetStatistician(uint32_t ssrc) const override; + int64_t TimeUntilNextProcess() override; + int32_t Process() override; + void SetMaxReorderingThreshold(int max_reordering_threshold) override; + void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) override; + void RegisterRtpStatisticsCallback( + StreamDataCountersCallback* callback) override; +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ diff --git a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h new file mode 100644 index 0000000000..56c6e48691 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h @@ -0,0 +1,51 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/system_wrappers/include/rtp_to_ntp.h" + +namespace webrtc { + +class Clock; +class TimestampExtrapolator; + +// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP +// time in local timebase. +// Note that it needs to be trained with at least 2 RTCP SR (by calling +// |UpdateRtcpTimestamp|) before it can be used. +class RemoteNtpTimeEstimator { + public: + explicit RemoteNtpTimeEstimator(Clock* clock); + + ~RemoteNtpTimeEstimator(); + + // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|, + // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|. + bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac, + uint32_t rtp_timestamp); + + // Estimates the NTP timestamp in local timebase from |rtp_timestamp|. + // Returns the NTP timestamp in ms when success. -1 if failed. + int64_t Estimate(uint32_t rtp_timestamp); + + private: + Clock* clock_; + rtc::scoped_ptr ts_extrapolator_; + RtcpList rtcp_list_; + int64_t last_timing_log_ms_; + RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator); +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_cvo.h b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h new file mode 100644 index 0000000000..2e30d898ec --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h @@ -0,0 +1,54 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ + +#include "webrtc/common_video/rotation.h" + +namespace webrtc { + +// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/ +// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is +// the clockwise angle the frames must be rotated in order to display the frames +// correctly if the display is rotated in its natural orientation. +inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) { + switch (rotation) { + case kVideoRotation_0: + return 0; + case kVideoRotation_90: + return 1; + case kVideoRotation_180: + return 2; + case kVideoRotation_270: + return 3; + } + assert(false); + return 0; +} + +inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) { + switch (rotation) { + case 0: + return kVideoRotation_0; + case 1: + return kVideoRotation_90; + case 2: + return kVideoRotation_180; + break; + case 3: + return kVideoRotation_270; + default: + assert(false); + return kVideoRotation_0; + } +} + +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h new file mode 100644 index 0000000000..329de32611 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h @@ -0,0 +1,44 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +struct RTPHeader; + +class RtpHeaderParser { + public: + static RtpHeaderParser* Create(); + virtual ~RtpHeaderParser() {} + + // Returns true if the packet is an RTCP packet, false otherwise. + static bool IsRtcp(const uint8_t* packet, size_t length); + + // Parses the packet and stores the parsed packet in |header|. Returns true on + // success, false otherwise. + // This method is thread-safe in the sense that it can parse multiple packets + // at once. + virtual bool Parse(const uint8_t* packet, + size_t length, + RTPHeader* header) const = 0; + + // Registers an RTP header extension and binds it to |id|. + virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, + uint8_t id) = 0; + + // De-registers an RTP header extension. + virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0; +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h new file mode 100644 index 0000000000..c2f467af35 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h @@ -0,0 +1,193 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" + +namespace webrtc { + +// This strategy deals with the audio/video-specific aspects +// of payload handling. +class RTPPayloadStrategy { + public: + virtual ~RTPPayloadStrategy() {} + + virtual bool CodecsMustBeUnique() const = 0; + + virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const = 0; + + virtual void UpdatePayloadRate(RtpUtility::Payload* payload, + const uint32_t rate) const = 0; + + virtual RtpUtility::Payload* CreatePayloadType( + const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int8_t payloadType, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) const = 0; + + virtual int GetPayloadTypeFrequency( + const RtpUtility::Payload& payload) const = 0; + + static RTPPayloadStrategy* CreateStrategy(const bool handling_audio); + + protected: + RTPPayloadStrategy() {} +}; + +class RTPPayloadRegistry { + public: + // The registry takes ownership of the strategy. + RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy); + ~RTPPayloadRegistry(); + + int32_t RegisterReceivePayload( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const int8_t payload_type, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate, + bool* created_new_payload_type); + + int32_t DeRegisterReceivePayload( + const int8_t payload_type); + + int32_t ReceivePayloadType( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate, + int8_t* payload_type) const; + + bool RtxEnabled() const; + + void SetRtxSsrc(uint32_t ssrc); + + bool GetRtxSsrc(uint32_t* ssrc) const; + + void SetRtxPayloadType(int payload_type, int associated_payload_type); + + bool IsRtx(const RTPHeader& header) const; + + // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t* + // restored_packet, instead of a uint8_t**. + // TODO(noahric): Remove this when all callers have been updated. + bool RestoreOriginalPacket(uint8_t** restored_packet, + const uint8_t* packet, + size_t* packet_length, + uint32_t original_ssrc, + const RTPHeader& header) const; + + bool RestoreOriginalPacket(uint8_t* restored_packet, + const uint8_t* packet, + size_t* packet_length, + uint32_t original_ssrc, + const RTPHeader& header) const; + + bool IsRed(const RTPHeader& header) const; + + // Returns true if the media of this RTP packet is encapsulated within an + // extra header, such as RTX or RED. + bool IsEncapsulated(const RTPHeader& header) const; + + bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const; + + int GetPayloadTypeFrequency(uint8_t payload_type) const; + + bool PayloadTypeToPayload(const uint8_t payload_type, + RtpUtility::Payload*& payload) const; + + void ResetLastReceivedPayloadTypes() { + CriticalSectionScoped cs(crit_sect_.get()); + last_received_payload_type_ = -1; + last_received_media_payload_type_ = -1; + } + + // This sets the payload type of the packets being received from the network + // on the media SSRC. For instance if packets are encapsulated with RED, this + // payload type will be the RED payload type. + void SetIncomingPayloadType(const RTPHeader& header); + + // Returns true if the new media payload type has not changed. + bool ReportMediaPayloadType(uint8_t media_payload_type); + + int8_t red_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return red_payload_type_; + } + int8_t ulpfec_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return ulpfec_payload_type_; + } + int8_t last_received_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return last_received_payload_type_; + } + void set_last_received_payload_type(int8_t last_received_payload_type) { + CriticalSectionScoped cs(crit_sect_.get()); + last_received_payload_type_ = last_received_payload_type; + } + + int8_t last_received_media_payload_type() const { + CriticalSectionScoped cs(crit_sect_.get()); + return last_received_media_payload_type_; + }; + + bool use_rtx_payload_mapping_on_restore() const { + CriticalSectionScoped cs(crit_sect_.get()); + return use_rtx_payload_mapping_on_restore_; + } + + void set_use_rtx_payload_mapping_on_restore(bool val) { + CriticalSectionScoped cs(crit_sect_.get()); + use_rtx_payload_mapping_on_restore_ = val; + } + + private: + // Prunes the payload type map of the specific payload type, if it exists. + void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const size_t payload_name_length, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate); + + bool IsRtxInternal(const RTPHeader& header) const; + + rtc::scoped_ptr crit_sect_; + RtpUtility::PayloadTypeMap payload_type_map_; + rtc::scoped_ptr rtp_payload_strategy_; + int8_t red_payload_type_; + int8_t ulpfec_payload_type_; + int8_t incoming_payload_type_; + int8_t last_received_payload_type_; + int8_t last_received_media_payload_type_; + bool rtx_; + // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that + // only understand one RTX PT is no longer needed. + int rtx_payload_type_; + // Mapping rtx_payload_type_map_[rtx] = associated. + std::map rtx_payload_type_map_; + // When true, use rtx_payload_type_map_ when restoring RTX packets to get the + // correct payload type. + bool use_rtx_payload_mapping_on_restore_; + uint32_t ssrc_rtx_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h new file mode 100644 index 0000000000..d257a30d41 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h @@ -0,0 +1,103 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +class RTPPayloadRegistry; + +class TelephoneEventHandler { + public: + virtual ~TelephoneEventHandler() {} + + // The following three methods implement the TelephoneEventHandler interface. + // Forward DTMFs to decoder for playout. + virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0; + + // Is forwarding of outband telephone events turned on/off? + virtual bool TelephoneEventForwardToDecoder() const = 0; + + // Is TelephoneEvent configured with payload type payload_type + virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0; +}; + +class RtpReceiver { + public: + // Creates a video-enabled RTP receiver. + static RtpReceiver* CreateVideoReceiver( + Clock* clock, + RtpData* incoming_payload_callback, + RtpFeedback* incoming_messages_callback, + RTPPayloadRegistry* rtp_payload_registry); + + // Creates an audio-enabled RTP receiver. + static RtpReceiver* CreateAudioReceiver( + Clock* clock, + RtpAudioFeedback* incoming_audio_feedback, + RtpData* incoming_payload_callback, + RtpFeedback* incoming_messages_callback, + RTPPayloadRegistry* rtp_payload_registry); + + virtual ~RtpReceiver() {} + + // Returns a TelephoneEventHandler if available. + virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0; + + // Registers a receive payload in the payload registry and notifies the media + // receiver strategy. + virtual int32_t RegisterReceivePayload( + const char payload_name[RTP_PAYLOAD_NAME_SIZE], + const int8_t payload_type, + const uint32_t frequency, + const uint8_t channels, + const uint32_t rate) = 0; + + // De-registers |payload_type| from the payload registry. + virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0; + + // Parses the media specific parts of an RTP packet and updates the receiver + // state. This for instance means that any changes in SSRC and payload type is + // detected and acted upon. + virtual bool IncomingRtpPacket(const RTPHeader& rtp_header, + const uint8_t* payload, + size_t payload_length, + PayloadUnion payload_specific, + bool in_order) = 0; + + // Returns the currently configured NACK method. + virtual NACKMethod NACK() const = 0; + + // Turn negative acknowledgement (NACK) requests on/off. + virtual void SetNACKStatus(const NACKMethod method) = 0; + + // Gets the last received timestamp. Returns true if a packet has been + // received, false otherwise. + virtual bool Timestamp(uint32_t* timestamp) const = 0; + // Gets the time in milliseconds when the last timestamp was received. + // Returns true if a packet has been received, false otherwise. + virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0; + + // Returns the remote SSRC of the currently received RTP stream. + virtual uint32_t SSRC() const = 0; + + // Returns the current remote CSRCs. + virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0; + + // Returns the current energy of the RTP stream received. + virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0; +}; +} // namespace webrtc + +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h new file mode 100644 index 0000000000..39aeba2f47 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h @@ -0,0 +1,641 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ + +#include +#include + +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" + +namespace webrtc { +// Forward declarations. +class ReceiveStatistics; +class RemoteBitrateEstimator; +class RtpReceiver; +class Transport; +namespace rtcp { +class TransportFeedback; +} + +class RtpRtcp : public Module { + public: + struct Configuration { + Configuration(); + + /* id - Unique identifier of this RTP/RTCP module object + * audio - True for a audio version of the RTP/RTCP module + * object false will create a video version + * clock - The clock to use to read time. If NULL object + * will be using the system clock. + * incoming_data - Callback object that will receive the incoming + * data. May not be NULL; default callback will do + * nothing. + * incoming_messages - Callback object that will receive the incoming + * RTP messages. May not be NULL; default callback + * will do nothing. + * outgoing_transport - Transport object that will be called when packets + * are ready to be sent out on the network + * intra_frame_callback - Called when the receiver request a intra frame. + * bandwidth_callback - Called when we receive a changed estimate from + * the receiver of out stream. + * audio_messages - Telephone events. May not be NULL; default + * callback will do nothing. + * remote_bitrate_estimator - Estimates the bandwidth available for a set of + * streams from the same client. + * paced_sender - Spread any bursts of packets into smaller + * bursts to minimize packet loss. + */ + bool audio; + bool receiver_only; + Clock* clock; + ReceiveStatistics* receive_statistics; + Transport* outgoing_transport; + RtcpIntraFrameObserver* intra_frame_callback; + RtcpBandwidthObserver* bandwidth_callback; + TransportFeedbackObserver* transport_feedback_callback; + RtcpRttStats* rtt_stats; + RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer; + RtpAudioFeedback* audio_messages; + RemoteBitrateEstimator* remote_bitrate_estimator; + RtpPacketSender* paced_sender; + TransportSequenceNumberAllocator* transport_sequence_number_allocator; + BitrateStatisticsObserver* send_bitrate_observer; + FrameCountObserver* send_frame_count_observer; + SendSideDelayObserver* send_side_delay_observer; + }; + + /* + * Create a RTP/RTCP module object using the system clock. + * + * configuration - Configuration of the RTP/RTCP module. + */ + static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); + + /************************************************************************** + * + * Receiver functions + * + ***************************************************************************/ + + virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, + size_t incoming_packet_length) = 0; + + virtual void SetRemoteSSRC(uint32_t ssrc) = 0; + + /************************************************************************** + * + * Sender + * + ***************************************************************************/ + + /* + * set MTU + * + * size - Max transfer unit in bytes, default is 1500 + * + * return -1 on failure else 0 + */ + virtual int32_t SetMaxTransferUnit(uint16_t size) = 0; + + /* + * set transtport overhead + * default is IPv4 and UDP with no encryption + * + * TCP - true for TCP false UDP + * IPv6 - true for IP version 6 false for version 4 + * authenticationOverhead - number of bytes to leave for an + * authentication header + * + * return -1 on failure else 0 + */ + virtual int32_t SetTransportOverhead( + bool TCP, + bool IPV6, + uint8_t authenticationOverhead = 0) = 0; + + /* + * Get max payload length + * + * A combination of the configuration MaxTransferUnit and + * TransportOverhead. + * Does not account FEC/ULP/RED overhead if FEC is enabled. + * Does not account for RTP headers + */ + virtual uint16_t MaxPayloadLength() const = 0; + + /* + * Get max data payload length + * + * A combination of the configuration MaxTransferUnit, headers and + * TransportOverhead. + * Takes into account FEC/ULP/RED overhead if FEC is enabled. + * Takes into account RTP headers + */ + virtual uint16_t MaxDataPayloadLength() const = 0; + + /* + * set codec name and payload type + * + * return -1 on failure else 0 + */ + virtual int32_t RegisterSendPayload( + const CodecInst& voiceCodec) = 0; + + /* + * set codec name and payload type + * + * return -1 on failure else 0 + */ + virtual int32_t RegisterSendPayload( + const VideoCodec& videoCodec) = 0; + + /* + * Unregister a send payload + * + * payloadType - payload type of codec + * + * return -1 on failure else 0 + */ + virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0; + + /* + * (De)register RTP header extension type and id. + * + * return -1 on failure else 0 + */ + virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type, + uint8_t id) = 0; + + virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; + + /* + * get start timestamp + */ + virtual uint32_t StartTimestamp() const = 0; + + /* + * configure start timestamp, default is a random number + * + * timestamp - start timestamp + */ + virtual void SetStartTimestamp(uint32_t timestamp) = 0; + + /* + * Get SequenceNumber + */ + virtual uint16_t SequenceNumber() const = 0; + + /* + * Set SequenceNumber, default is a random number + */ + virtual void SetSequenceNumber(uint16_t seq) = 0; + + // Returns true if the ssrc matched this module, false otherwise. + virtual bool SetRtpStateForSsrc(uint32_t ssrc, + const RtpState& rtp_state) = 0; + virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0; + + /* + * Get SSRC + */ + virtual uint32_t SSRC() const = 0; + + /* + * configure SSRC, default is a random number + */ + virtual void SetSSRC(uint32_t ssrc) = 0; + + /* + * Set CSRC + * + * csrcs - vector of CSRCs + */ + virtual void SetCsrcs(const std::vector& csrcs) = 0; + + /* + * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination + * of values of the enumerator RtxMode. + */ + virtual void SetRtxSendStatus(int modes) = 0; + + /* + * Get status of sending RTX (RFC 4588). The returned value can be + * a combination of values of the enumerator RtxMode. + */ + virtual int RtxSendStatus() const = 0; + + // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, + // only the SSRC is set. + virtual void SetRtxSsrc(uint32_t ssrc) = 0; + + // Sets the payload type to use when sending RTX packets. Note that this + // doesn't enable RTX, only the payload type is set. + virtual void SetRtxSendPayloadType(int payload_type, + int associated_payload_type) = 0; + + // Gets the payload type pair of (RTX, associated) to use when sending RTX + // packets. + virtual std::pair RtxSendPayloadType() const = 0; + + /* + * sends kRtcpByeCode when going from true to false + * + * sending - on/off + * + * return -1 on failure else 0 + */ + virtual int32_t SetSendingStatus(bool sending) = 0; + + /* + * get send status + */ + virtual bool Sending() const = 0; + + /* + * Starts/Stops media packets, on by default + * + * sending - on/off + */ + virtual void SetSendingMediaStatus(bool sending) = 0; + + /* + * get send status + */ + virtual bool SendingMedia() const = 0; + + /* + * get sent bitrate in Kbit/s + */ + virtual void BitrateSent(uint32_t* totalRate, + uint32_t* videoRate, + uint32_t* fecRate, + uint32_t* nackRate) const = 0; + + /* + * Used by the codec module to deliver a video or audio frame for + * packetization. + * + * frameType - type of frame to send + * payloadType - payload type of frame to send + * timestamp - timestamp of frame to send + * payloadData - payload buffer of frame to send + * payloadSize - size of payload buffer to send + * fragmentation - fragmentation offset data for fragmented frames such + * as layers or RED + * + * return -1 on failure else 0 + */ + virtual int32_t SendOutgoingData( + FrameType frameType, + int8_t payloadType, + uint32_t timeStamp, + int64_t capture_time_ms, + const uint8_t* payloadData, + size_t payloadSize, + const RTPFragmentationHeader* fragmentation = NULL, + const RTPVideoHeader* rtpVideoHdr = NULL) = 0; + + virtual bool TimeToSendPacket(uint32_t ssrc, + uint16_t sequence_number, + int64_t capture_time_ms, + bool retransmission) = 0; + + virtual size_t TimeToSendPadding(size_t bytes) = 0; + + // Called on generation of new statistics after an RTP send. + virtual void RegisterSendChannelRtpStatisticsCallback( + StreamDataCountersCallback* callback) = 0; + virtual StreamDataCountersCallback* + GetSendChannelRtpStatisticsCallback() const = 0; + + /************************************************************************** + * + * RTCP + * + ***************************************************************************/ + + /* + * Get RTCP status + */ + virtual RtcpMode RTCP() const = 0; + + /* + * configure RTCP status i.e on(compound or non- compound)/off + * + * method - RTCP method to use + */ + virtual void SetRTCPStatus(RtcpMode method) = 0; + + /* + * Set RTCP CName (i.e unique identifier) + * + * return -1 on failure else 0 + */ + virtual int32_t SetCNAME(const char* c_name) = 0; + + /* + * Get remote CName + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteCNAME(uint32_t remoteSSRC, + char cName[RTCP_CNAME_SIZE]) const = 0; + + /* + * Get remote NTP + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteNTP( + uint32_t *ReceivedNTPsecs, + uint32_t *ReceivedNTPfrac, + uint32_t *RTCPArrivalTimeSecs, + uint32_t *RTCPArrivalTimeFrac, + uint32_t *rtcp_timestamp) const = 0; + + /* + * AddMixedCNAME + * + * return -1 on failure else 0 + */ + virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0; + + /* + * RemoveMixedCNAME + * + * return -1 on failure else 0 + */ + virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0; + + /* + * Get RoundTripTime + * + * return -1 on failure else 0 + */ + virtual int32_t RTT(uint32_t remoteSSRC, + int64_t* RTT, + int64_t* avgRTT, + int64_t* minRTT, + int64_t* maxRTT) const = 0; + + /* + * Force a send of a RTCP packet + * periodic SR and RR are triggered via the process function + * + * return -1 on failure else 0 + */ + virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0; + + /* + * Force a send of a RTCP packet with more than one packet type. + * periodic SR and RR are triggered via the process function + * + * return -1 on failure else 0 + */ + virtual int32_t SendCompoundRTCP( + const std::set& rtcpPacketTypes) = 0; + + /* + * Good state of RTP receiver inform sender + */ + virtual int32_t SendRTCPReferencePictureSelection( + const uint64_t pictureID) = 0; + + /* + * Send a RTCP Slice Loss Indication (SLI) + * 6 least significant bits of pictureID + */ + virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0; + + /* + * Statistics of the amount of data sent + * + * return -1 on failure else 0 + */ + virtual int32_t DataCountersRTP( + size_t* bytesSent, + uint32_t* packetsSent) const = 0; + + /* + * Get send statistics for the RTP and RTX stream. + */ + virtual void GetSendStreamDataCounters( + StreamDataCounters* rtp_counters, + StreamDataCounters* rtx_counters) const = 0; + + /* + * Get packet loss statistics for the RTP stream. + */ + virtual void GetRtpPacketLossStats( + bool outgoing, + uint32_t ssrc, + struct RtpPacketLossStats* loss_stats) const = 0; + + /* + * Get received RTCP sender info + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0; + + /* + * Get received RTCP report block + * + * return -1 on failure else 0 + */ + virtual int32_t RemoteRTCPStat( + std::vector* receiveBlocks) const = 0; + + /* + * (APP) Application specific data + * + * return -1 on failure else 0 + */ + virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType, + uint32_t name, + const uint8_t* data, + uint16_t length) = 0; + /* + * (XR) VOIP metric + * + * return -1 on failure else 0 + */ + virtual int32_t SetRTCPVoIPMetrics( + const RTCPVoIPMetric* VoIPMetric) = 0; + + /* + * (XR) Receiver Reference Time Report + */ + virtual void SetRtcpXrRrtrStatus(bool enable) = 0; + + virtual bool RtcpXrRrtrStatus() const = 0; + + /* + * (REMB) Receiver Estimated Max Bitrate + */ + virtual bool REMB() const = 0; + + virtual void SetREMBStatus(bool enable) = 0; + + virtual void SetREMBData(uint32_t bitrate, + const std::vector& ssrcs) = 0; + + /* + * (TMMBR) Temporary Max Media Bit Rate + */ + virtual bool TMMBR() const = 0; + + virtual void SetTMMBRStatus(bool enable) = 0; + + /* + * (NACK) + */ + + /* + * TODO(holmer): Propagate this API to VideoEngine. + * Returns the currently configured selective retransmission settings. + */ + virtual int SelectiveRetransmissions() const = 0; + + /* + * TODO(holmer): Propagate this API to VideoEngine. + * Sets the selective retransmission settings, which will decide which + * packets will be retransmitted if NACKed. Settings are constructed by + * combining the constants in enum RetransmissionMode with bitwise OR. + * All packets are retransmitted if kRetransmitAllPackets is set, while no + * packets are retransmitted if kRetransmitOff is set. + * By default all packets except FEC packets are retransmitted. For VP8 + * with temporal scalability only base layer packets are retransmitted. + * + * Returns -1 on failure, otherwise 0. + */ + virtual int SetSelectiveRetransmissions(uint8_t settings) = 0; + + /* + * Send a Negative acknowledgement packet + * + * return -1 on failure else 0 + */ + virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0; + + /* + * Store the sent packets, needed to answer to a Negative acknowledgement + * requests + */ + virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0; + + // Returns true if the module is configured to store packets. + virtual bool StorePackets() const = 0; + + // Called on receipt of RTCP report block from remote side. + virtual void RegisterRtcpStatisticsCallback( + RtcpStatisticsCallback* callback) = 0; + virtual RtcpStatisticsCallback* + GetRtcpStatisticsCallback() = 0; + // BWE feedback packets. + virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; + + /************************************************************************** + * + * Audio + * + ***************************************************************************/ + + /* + * set audio packet size, used to determine when it's time to send a DTMF + * packet in silence (CNG) + * + * return -1 on failure else 0 + */ + virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0; + + /* + * Send a TelephoneEvent tone using RFC 2833 (4733) + * + * return -1 on failure else 0 + */ + virtual int32_t SendTelephoneEventOutband(uint8_t key, + uint16_t time_ms, + uint8_t level) = 0; + + /* + * Set payload type for Redundant Audio Data RFC 2198 + * + * return -1 on failure else 0 + */ + virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0; + + /* + * Get payload type for Redundant Audio Data RFC 2198 + * + * return -1 on failure else 0 + */ + virtual int32_t SendREDPayloadType( + int8_t& payloadType) const = 0; + + /* + * Store the audio level in dBov for header-extension-for-audio-level- + * indication. + * This API shall be called before transmision of an RTP packet to ensure + * that the |level| part of the extended RTP header is updated. + * + * return -1 on failure else 0. + */ + virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0; + + /************************************************************************** + * + * Video + * + ***************************************************************************/ + + /* + * Set the target send bitrate + */ + virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0; + + /* + * Turn on/off generic FEC + */ + virtual void SetGenericFECStatus(bool enable, + uint8_t payload_type_red, + uint8_t payload_type_fec) = 0; + + /* + * Get generic FEC setting + */ + virtual void GenericFECStatus(bool& enable, + uint8_t& payloadTypeRED, + uint8_t& payloadTypeFEC) = 0; + + + virtual int32_t SetFecParameters( + const FecProtectionParams* delta_params, + const FecProtectionParams* key_params) = 0; + + /* + * Set method for requestion a new key frame + * + * return -1 on failure else 0 + */ + virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; + + /* + * send a request for a keyframe + * + * return -1 on failure else 0 + */ + virtual int32_t RequestKeyFrame() = 0; +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h new file mode 100644 index 0000000000..6373de28e7 --- /dev/null +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h @@ -0,0 +1,440 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ + +#include +#include + +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/system_wrappers/include/clock.h" +#include "webrtc/typedefs.h" + +#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination +#define IP_PACKET_SIZE 1500 // we assume ethernet +#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10 +#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds + +namespace webrtc { +namespace rtcp { +class TransportFeedback; +} + +const int kVideoPayloadTypeFrequency = 90000; + +// Minimum RTP header size in bytes. +const uint8_t kRtpHeaderSize = 12; + +struct AudioPayload +{ + uint32_t frequency; + uint8_t channels; + uint32_t rate; +}; + +struct VideoPayload +{ + RtpVideoCodecTypes videoCodecType; + uint32_t maxRate; +}; + +union PayloadUnion +{ + AudioPayload Audio; + VideoPayload Video; +}; + +enum RTPAliveType +{ + kRtpDead = 0, + kRtpNoRtp = 1, + kRtpAlive = 2 +}; + +enum ProtectionType { + kUnprotectedPacket, + kProtectedPacket +}; + +enum StorageType { + kDontRetransmit, + kAllowRetransmission +}; + +enum RTPExtensionType { + kRtpExtensionNone, + kRtpExtensionTransmissionTimeOffset, + kRtpExtensionAudioLevel, + kRtpExtensionAbsoluteSendTime, + kRtpExtensionVideoRotation, + kRtpExtensionTransportSequenceNumber, +}; + +enum RTCPAppSubTypes +{ + kAppSubtypeBwe = 0x00 +}; + +// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up. +enum RTCPPacketType : uint32_t { + kRtcpReport = 0x0001, + kRtcpSr = 0x0002, + kRtcpRr = 0x0004, + kRtcpSdes = 0x0008, + kRtcpBye = 0x0010, + kRtcpPli = 0x0020, + kRtcpNack = 0x0040, + kRtcpFir = 0x0080, + kRtcpTmmbr = 0x0100, + kRtcpTmmbn = 0x0200, + kRtcpSrReq = 0x0400, + kRtcpXrVoipMetric = 0x0800, + kRtcpApp = 0x1000, + kRtcpSli = 0x4000, + kRtcpRpsi = 0x8000, + kRtcpRemb = 0x10000, + kRtcpTransmissionTimeOffset = 0x20000, + kRtcpXrReceiverReferenceTime = 0x40000, + kRtcpXrDlrrReportBlock = 0x80000, + kRtcpTransportFeedback = 0x100000, +}; + +enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp }; + +enum RtpRtcpPacketType +{ + kPacketRtp = 0, + kPacketKeepAlive = 1 +}; + +enum NACKMethod +{ + kNackOff = 0, + kNackRtcp = 2 +}; + +enum RetransmissionMode : uint8_t { + kRetransmitOff = 0x0, + kRetransmitFECPackets = 0x1, + kRetransmitBaseLayer = 0x2, + kRetransmitHigherLayers = 0x4, + kRetransmitAllPackets = 0xFF +}; + +enum RtxMode { + kRtxOff = 0x0, + kRtxRetransmitted = 0x1, // Only send retransmissions over RTX. + kRtxRedundantPayloads = 0x2 // Preventively send redundant payloads + // instead of padding. +}; + +const size_t kRtxHeaderSize = 2; + +struct RTCPSenderInfo +{ + uint32_t NTPseconds; + uint32_t NTPfraction; + uint32_t RTPtimeStamp; + uint32_t sendPacketCount; + uint32_t sendOctetCount; +}; + +struct RTCPReportBlock { + RTCPReportBlock() + : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0), + extendedHighSeqNum(0), jitter(0), lastSR(0), + delaySinceLastSR(0) {} + + RTCPReportBlock(uint32_t remote_ssrc, + uint32_t source_ssrc, + uint8_t fraction_lost, + uint32_t cumulative_lost, + uint32_t extended_high_sequence_number, + uint32_t jitter, + uint32_t last_sender_report, + uint32_t delay_since_last_sender_report) + : remoteSSRC(remote_ssrc), + sourceSSRC(source_ssrc), + fractionLost(fraction_lost), + cumulativeLost(cumulative_lost), + extendedHighSeqNum(extended_high_sequence_number), + jitter(jitter), + lastSR(last_sender_report), + delaySinceLastSR(delay_since_last_sender_report) {} + + // Fields as described by RFC 3550 6.4.2. + uint32_t remoteSSRC; // SSRC of sender of this report. + uint32_t sourceSSRC; // SSRC of the RTP packet sender. + uint8_t fractionLost; + uint32_t cumulativeLost; // 24 bits valid. + uint32_t extendedHighSeqNum; + uint32_t jitter; + uint32_t lastSR; + uint32_t delaySinceLastSR; +}; + +struct RtcpReceiveTimeInfo { + // Fields as described by RFC 3611 4.5. + uint32_t sourceSSRC; + uint32_t lastRR; + uint32_t delaySinceLastRR; +}; + +typedef std::list ReportBlockList; + +struct RtpState { + RtpState() + : sequence_number(0), + start_timestamp(0), + timestamp(0), + capture_time_ms(-1), + last_timestamp_time_ms(-1), + media_has_been_sent(false) {} + uint16_t sequence_number; + uint32_t start_timestamp; + uint32_t timestamp; + int64_t capture_time_ms; + int64_t last_timestamp_time_ms; + bool media_has_been_sent; +}; + +class RtpData +{ +public: + virtual ~RtpData() {} + + virtual int32_t OnReceivedPayloadData( + const uint8_t* payloadData, + const size_t payloadSize, + const WebRtcRTPHeader* rtpHeader) = 0; + + virtual bool OnRecoveredPacket(const uint8_t* packet, + size_t packet_length) = 0; +}; + +class RtpFeedback +{ +public: + virtual ~RtpFeedback() {} + + // Receiving payload change or SSRC change. (return success!) + /* + * channels - number of channels in codec (1 = mono, 2 = stereo) + */ + virtual int32_t OnInitializeDecoder( + const int8_t payloadType, + const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int frequency, + const uint8_t channels, + const uint32_t rate) = 0; + + virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0; + + virtual void OnIncomingCSRCChanged(const uint32_t CSRC, + const bool added) = 0; +}; + +class RtpAudioFeedback { + public: + virtual void OnPlayTelephoneEvent(const uint8_t event, + const uint16_t lengthMs, + const uint8_t volume) = 0; + + protected: + virtual ~RtpAudioFeedback() {} +}; + +class RtcpIntraFrameObserver { + public: + virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0; + + virtual void OnReceivedSLI(uint32_t ssrc, + uint8_t picture_id) = 0; + + virtual void OnReceivedRPSI(uint32_t ssrc, + uint64_t picture_id) = 0; + + virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0; + + virtual ~RtcpIntraFrameObserver() {} +}; + +class RtcpBandwidthObserver { + public: + // REMB or TMMBR + virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0; + + virtual void OnReceivedRtcpReceiverReport( + const ReportBlockList& report_blocks, + int64_t rtt, + int64_t now_ms) = 0; + + virtual ~RtcpBandwidthObserver() {} +}; + +struct PacketInfo { + PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number) + : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {} + + PacketInfo(int64_t arrival_time_ms, + int64_t send_time_ms, + uint16_t sequence_number, + size_t payload_size, + bool was_paced) + : PacketInfo(-1, + arrival_time_ms, + send_time_ms, + sequence_number, + payload_size, + was_paced) {} + + PacketInfo(int64_t creation_time_ms, + int64_t arrival_time_ms, + int64_t send_time_ms, + uint16_t sequence_number, + size_t payload_size, + bool was_paced) + : creation_time_ms(creation_time_ms), + arrival_time_ms(arrival_time_ms), + send_time_ms(send_time_ms), + sequence_number(sequence_number), + payload_size(payload_size), + was_paced(was_paced) {} + + // Time corresponding to when this object was created. + int64_t creation_time_ms; + // Time corresponding to when the packet was received. Timestamped with the + // receiver's clock. + int64_t arrival_time_ms; + // Time corresponding to when the packet was sent, timestamped with the + // sender's clock. + int64_t send_time_ms; + // Packet identifier, incremented with 1 for every packet generated by the + // sender. + uint16_t sequence_number; + // Size of the packet excluding RTP headers. + size_t payload_size; + // True if the packet was paced out by the pacer. + bool was_paced; +}; + +class TransportFeedbackObserver { + public: + TransportFeedbackObserver() {} + virtual ~TransportFeedbackObserver() {} + + // Note: Transport-wide sequence number as sequence number. Arrival time + // must be set to 0. + virtual void AddPacket(uint16_t sequence_number, + size_t length, + bool was_paced) = 0; + + virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0; +}; + +class RtcpRttStats { + public: + virtual void OnRttUpdate(int64_t rtt) = 0; + + virtual int64_t LastProcessedRtt() const = 0; + + virtual ~RtcpRttStats() {}; +}; + +// Null object version of RtpFeedback. +class NullRtpFeedback : public RtpFeedback { + public: + virtual ~NullRtpFeedback() {} + + int32_t OnInitializeDecoder(const int8_t payloadType, + const char payloadName[RTP_PAYLOAD_NAME_SIZE], + const int frequency, + const uint8_t channels, + const uint32_t rate) override { + return 0; + } + + void OnIncomingSSRCChanged(const uint32_t ssrc) override {} + void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {} +}; + +// Null object version of RtpData. +class NullRtpData : public RtpData { + public: + virtual ~NullRtpData() {} + + int32_t OnReceivedPayloadData(const uint8_t* payloadData, + const size_t payloadSize, + const WebRtcRTPHeader* rtpHeader) override { + return 0; + } + + bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override { + return true; + } +}; + +// Null object version of RtpAudioFeedback. +class NullRtpAudioFeedback : public RtpAudioFeedback { + public: + virtual ~NullRtpAudioFeedback() {} + + void OnPlayTelephoneEvent(const uint8_t event, + const uint16_t lengthMs, + const uint8_t volume) override {} +}; + +// Statistics about packet loss for a single directional connection. All values +// are totals since the connection initiated. +struct RtpPacketLossStats { + // The number of packets lost in events where no adjacent packets were also + // lost. + uint64_t single_packet_loss_count; + // The number of events in which more than one adjacent packet was lost. + uint64_t multiple_packet_loss_event_count; + // The number of packets lost in events where more than one adjacent packet + // was lost. + uint64_t multiple_packet_loss_packet_count; +}; + +class RtpPacketSender { + public: + RtpPacketSender() {} + virtual ~RtpPacketSender() {} + + enum Priority { + kHighPriority = 0, // Pass through; will be sent immediately. + kNormalPriority = 2, // Put in back of the line. + kLowPriority = 3, // Put in back of the low priority line. + }; + // Low priority packets are mixed with the normal priority packets + // while we are paused. + + // Returns true if we send the packet now, else it will add the packet + // information to the queue and call TimeToSendPacket when it's time to send. + virtual void InsertPacket(Priority priority, + uint32_t ssrc, + uint16_t sequence_number, + int64_t capture_time_ms, + size_t bytes, + bool retransmission) = 0; +}; + +class TransportSequenceNumberAllocator { + public: + TransportSequenceNumberAllocator() {} + virtual ~TransportSequenceNumberAllocator() {} + + virtual uint16_t AllocateSequenceNumber() = 0; +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h index 3608165dab..ec7ec399d9 100644 --- a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h +++ b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include") #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" @@ -43,4 +45,4 @@ class FecReceiver { virtual FecPacketCounter GetPacketCounter() const = 0; }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h index 6bd5cd846e..183eb2e98a 100644 --- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h +++ b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h @@ -8,13 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") #include -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -99,4 +101,4 @@ class NullReceiveStatistics : public ReceiveStatistics { }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h index 0ffba212a6..d271c9bbd5 100644 --- a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h +++ b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") #include "webrtc/base/scoped_ptr.h" #include "webrtc/system_wrappers/include/rtp_to_ntp.h" @@ -48,4 +50,4 @@ class RemoteNtpTimeEstimator { } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h index c7a0268ef0..6d942cbc92 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h @@ -7,8 +7,10 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") #include "webrtc/common_video/rotation.h" @@ -51,4 +53,4 @@ inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) { } } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h index 2809996b25..9b2cabc546 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h @@ -7,10 +7,12 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -41,4 +43,4 @@ class RtpHeaderParser { virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0; }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h index 313bef1112..4994eb50a2 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" @@ -190,4 +192,4 @@ class RTPPayloadRegistry { } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h index 2fb8ac5d61..db5f46e430 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h @@ -8,10 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") + +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -100,4 +102,4 @@ class RtpReceiver { }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h index f907408573..d004f67498 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h @@ -8,14 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") #include #include -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { // Forward declarations. @@ -638,4 +640,4 @@ class RtpRtcp : public Module { virtual int32_t RequestKeyFrame() = 0; }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h index 6936352aca..ac3954ba9b 100644 --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h @@ -8,13 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ +#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ +#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ + +#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.") #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/typedefs.h" @@ -437,4 +439,4 @@ class TransportSequenceNumberAllocator { }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_ diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h index bc4aec8967..ca5434ee6d 100644 --- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h +++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h @@ -13,9 +13,9 @@ #include "testing/gmock/include/gmock/gmock.h" -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi index e35a75cd0f..57bea0b2dc 100644 --- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi +++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi @@ -17,14 +17,14 @@ ], 'sources': [ # Common - 'interface/fec_receiver.h', - 'interface/receive_statistics.h', - 'interface/remote_ntp_time_estimator.h', - 'interface/rtp_header_parser.h', - 'interface/rtp_payload_registry.h', - 'interface/rtp_receiver.h', - 'interface/rtp_rtcp.h', - 'interface/rtp_rtcp_defines.h', + 'include/fec_receiver.h', + 'include/receive_statistics.h', + 'include/remote_ntp_time_estimator.h', + 'include/rtp_header_parser.h', + 'include/rtp_payload_registry.h', + 'include/rtp_receiver.h', + 'include/rtp_rtcp.h', + 'include/rtp_rtcp_defines.h', 'source/bitrate.cc', 'source/bitrate.h', 'source/byte_io.h', diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h index 24db39b902..b79f6ba2f4 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h +++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h @@ -14,8 +14,8 @@ // This header is included to get the nested declaration of Packet structure. #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc index f64b537a52..bb22e1d580 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc @@ -15,8 +15,8 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h" diff --git a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h index e1791adba3..a5de924c89 100644 --- a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h +++ b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc index aad418f015..7a9a8beecf 100644 --- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc +++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc @@ -18,7 +18,7 @@ #include #include "webrtc/base/logging.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h" diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h index f92f014db3..9ba6ce0438 100644 --- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h +++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h @@ -15,7 +15,7 @@ #include #include "webrtc/base/scoped_ref_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/ref_count.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h index f577cbaad1..549821f808 100644 --- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h +++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_ #include "testing/gmock/include/gmock/gmock.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc index 07a3693507..d95c9636d9 100644 --- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc @@ -16,12 +16,12 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/transport.h" using namespace webrtc; diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h index fe42990fe9..0cd80125da 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_ -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc index 5b522281df..c265c17c04 100644 --- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc @@ -11,7 +11,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include "webrtc/system_wrappers/include/clock.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc index 74fc9cdc56..ccc15ec417 100644 --- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc +++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" +#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" #include "webrtc/base/logging.h" #include "webrtc/system_wrappers/include/clock.h" diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc index bc9cf2ee39..797c7883a9 100644 --- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc @@ -11,7 +11,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" +#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" #include "webrtc/system_wrappers/include/clock.h" using ::testing::_; diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h index 3c34957c36..2956bc7bea 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h @@ -18,7 +18,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h index 4cc1f38479..ad6fd166f2 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h @@ -15,7 +15,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h index 272397675b..fc43de5e70 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h @@ -16,7 +16,7 @@ #include #include "webrtc/base/thread_annotations.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h index 37b7b88370..a697fd2e30 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h @@ -14,7 +14,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" // RTCPReportBlock +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPReportBlock #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h index 9ec928363b..1bdd6d1db9 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h @@ -20,8 +20,8 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h index f05d512919..e1f8737ec5 100644 --- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h +++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h @@ -14,7 +14,7 @@ #include // size_t, ptrdiff_t #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h index 18225f9bb4..3519499248 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h @@ -14,8 +14,8 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc index aeef44364a..c422577c81 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc @@ -11,7 +11,7 @@ #include #include "webrtc/base/logging.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc index 1a14b5554a..d29e3d4f21 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc @@ -13,7 +13,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc index 39b64c6ffa..b47e9b9359 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc @@ -11,7 +11,7 @@ #include #include "webrtc/base/logging.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h index 63db349c74..d62ecba85f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h @@ -30,7 +30,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h index 2fe963251f..668476833d 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h @@ -19,7 +19,7 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_ #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h index abce7e7791..3feca4392a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h @@ -25,7 +25,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h index 7be3c2e5c4..263b2d790f 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h @@ -13,7 +13,7 @@ #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc index 520cf7a962..ca37750621 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc @@ -15,7 +15,7 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc index 266bad8858..82c813fd76 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc @@ -7,7 +7,7 @@ * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h index e97d11eeaa..cece640eae 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h @@ -16,8 +16,8 @@ #include #include "webrtc/base/thread_annotations.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc index 00a6ac7ed2..d39991e6ba 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc @@ -12,7 +12,7 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/video_engine/vie_defines.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc index 38d1450c23..a5bfd71664 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "webrtc/base/logging.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc index 0b9bf2751e..e20c94e7fc 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h index 176852e01e..b2413f349a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h @@ -14,8 +14,8 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc index e1ebd0c8bb..bd7bd1b82e 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc @@ -16,8 +16,8 @@ #include #include "webrtc/base/logging.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h index 46741d59b4..4ec35315d2 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h @@ -12,8 +12,8 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h index 37c3e6e49a..0e0a34a4a4 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h @@ -12,8 +12,8 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc index 1af2d4857e..53051dd321 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc @@ -16,8 +16,8 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h index 23128df6e1..56f761a2e1 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/bitrate.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h index c9b6686c0a..ada66673ad 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h @@ -15,7 +15,7 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc index 4c94764ee6..a5e2076a21 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc @@ -12,8 +12,8 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h" #include "webrtc/system_wrappers/include/scoped_vector.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc index 50f476829d..dc544fbe69 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc @@ -16,7 +16,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h index a134370c76..8fc0696067 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h @@ -17,7 +17,7 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/bitrate.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc index 1fc9a89ce1..f5df5b3b4a 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc @@ -14,7 +14,7 @@ #include //memcpy #include "webrtc/base/trace_event.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc index fde6d47ceb..332a0f81f9 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc @@ -17,9 +17,9 @@ #include "webrtc/base/buffer.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc index 66062771de..0209510d84 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc @@ -18,7 +18,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h index f412542d86..03ed6000da 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h @@ -16,7 +16,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/bitrate.h" #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" #include "webrtc/modules/rtp_rtcp/source/producer_fec.h" diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h index af20f97e82..1c615e66e7 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h @@ -13,8 +13,8 @@ #include // size_t, ptrdiff_t -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h index 53b678f3b9..4716dfb317 100644 --- a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h +++ b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h @@ -14,7 +14,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc index 711be4a623..2d70289d3a 100644 --- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc +++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc @@ -14,7 +14,7 @@ #include #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/channel_transport/udp_transport.h" diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h index bd83962fa3..59742b2329 100644 --- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h +++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h @@ -13,7 +13,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/thread_wrapper.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc index 47f2880638..c1d91cdbc3 100644 --- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc +++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc @@ -13,7 +13,7 @@ #include #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/event_wrapper.h" diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h index f394412a73..02fbe0a936 100644 --- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h +++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h @@ -11,8 +11,8 @@ #ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ #define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_ -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/thread_wrapper.h" #include "webrtc/test/channel_transport/udp_transport.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h index 73334a8b26..f55b4b5461 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h @@ -11,12 +11,12 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/transport.h" namespace webrtc { diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc index 745386d485..36456f125e 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc @@ -15,8 +15,8 @@ #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" using namespace webrtc; diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc index e9d81122b1..8faa2dd4f4 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc @@ -14,9 +14,9 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc index 30a6a1c303..16ea540bd5 100644 --- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc +++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc @@ -15,9 +15,9 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h" #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index 163515c466..6704cd6d9a 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -10,12 +10,12 @@ import("../../build/webrtc.gni") source_set("utility") { sources = [ - "interface/audio_frame_operations.h", - "interface/file_player.h", - "interface/file_recorder.h", - "interface/helpers_android.h", - "interface/jvm_android.h", - "interface/process_thread.h", + "include/audio_frame_operations.h", + "include/file_player.h", + "include/file_recorder.h", + "include/helpers_android.h", + "include/jvm_android.h", + "include/process_thread.h", "source/audio_frame_operations.cc", "source/coder.cc", "source/coder.h", diff --git a/webrtc/modules/utility/include/audio_frame_operations.h b/webrtc/modules/utility/include/audio_frame_operations.h new file mode 100644 index 0000000000..1551d86894 --- /dev/null +++ b/webrtc/modules/utility/include/audio_frame_operations.h @@ -0,0 +1,58 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ + +#include "webrtc/typedefs.h" + +namespace webrtc { + +class AudioFrame; + +// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h. +// Change reference parameters to pointers. Consider using a namespace rather +// than a class. +class AudioFrameOperations { + public: + // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place + // operation, meaning src_audio and dst_audio must point to different + // buffers. It is the caller's responsibility to ensure that |dst_audio| is + // sufficiently large. + static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel, + int16_t* dst_audio); + // |frame.num_channels_| will be updated. This version checks for sufficient + // buffer size and that |num_channels_| is mono. + static int MonoToStereo(AudioFrame* frame); + + // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place + // operation, meaning |src_audio| and |dst_audio| may point to the same + // buffer. + static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel, + int16_t* dst_audio); + // |frame.num_channels_| will be updated. This version checks that + // |num_channels_| is stereo. + static int StereoToMono(AudioFrame* frame); + + // Swap the left and right channels of |frame|. Fails silently if |frame| is + // not stereo. + static void SwapStereoChannels(AudioFrame* frame); + + // Zeros out the audio and sets |frame.energy| to zero. + static void Mute(AudioFrame& frame); + + static int Scale(float left, float right, AudioFrame& frame); + + static int ScaleWithSat(float scale, AudioFrame& frame); +}; + +} // namespace webrtc + +#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h new file mode 100644 index 0000000000..4ca134a669 --- /dev/null +++ b/webrtc/modules/utility/include/file_player.h @@ -0,0 +1,111 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ + +#include "webrtc/common_types.h" +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" +#include "webrtc/video_frame.h" + +namespace webrtc { +class FileCallback; + +class FilePlayer +{ +public: + // The largest decoded frame size in samples (60ms with 32kHz sample rate). + enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32}; + enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2}; + + // Note: will return NULL for unsupported formats. + static FilePlayer* CreateFilePlayer(const uint32_t instanceID, + const FileFormats fileFormat); + + static void DestroyFilePlayer(FilePlayer* player); + + // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples| + // will be set to the number of samples read (not the number of samples per + // channel). + virtual int Get10msAudioFromFile( + int16_t* outBuffer, + size_t& lengthInSamples, + int frequencyInHz) = 0; + + // Register callback for receiving file playing notifications. + virtual int32_t RegisterModuleFileCallback( + FileCallback* callback) = 0; + + // API for playing audio from fileName to channel. + // Note: codecInst is used for pre-encoded files. + virtual int32_t StartPlayingFile( + const char* fileName, + bool loop, + uint32_t startPosition, + float volumeScaling, + uint32_t notification, + uint32_t stopPosition = 0, + const CodecInst* codecInst = NULL) = 0; + + // Note: codecInst is used for pre-encoded files. + virtual int32_t StartPlayingFile( + InStream& sourceStream, + uint32_t startPosition, + float volumeScaling, + uint32_t notification, + uint32_t stopPosition = 0, + const CodecInst* codecInst = NULL) = 0; + + virtual int32_t StopPlayingFile() = 0; + + virtual bool IsPlayingFile() const = 0; + + virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0; + + // Set audioCodec to the currently used audio codec. + virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0; + + virtual int32_t Frequency() const = 0; + + // Note: scaleFactor is in the range [0.0 - 2.0] + virtual int32_t SetAudioScaling(float scaleFactor) = 0; + + // Return the time in ms until next video frame should be pulled (by + // calling GetVideoFromFile(..)). + // Note: this API reads one video frame from file. This means that it should + // be called exactly once per GetVideoFromFile(..) API call. + virtual int32_t TimeUntilNextVideoFrame() { return -1;} + + virtual int32_t StartPlayingVideoFile( + const char* /*fileName*/, + bool /*loop*/, + bool /*videoOnly*/) { return -1;} + + virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const + {return -1;} + + virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; } + + // Same as GetVideoFromFile(). videoFrame will have the resolution specified + // by the width outWidth and height outHeight in pixels. + virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/, + const uint32_t /*outWidth*/, + const uint32_t /*outHeight*/) { + return -1; + } + +protected: + virtual ~FilePlayer() {} + +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ diff --git a/webrtc/modules/utility/include/file_recorder.h b/webrtc/modules/utility/include/file_recorder.h new file mode 100644 index 0000000000..0c285d442f --- /dev/null +++ b/webrtc/modules/utility/include/file_recorder.h @@ -0,0 +1,84 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ + +#include "webrtc/common_types.h" +#include "webrtc/engine_configurations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" +#include "webrtc/system_wrappers/include/tick_util.h" +#include "webrtc/typedefs.h" +#include "webrtc/video_frame.h" + +namespace webrtc { + +class FileRecorder +{ +public: + + // Note: will return NULL for unsupported formats. + static FileRecorder* CreateFileRecorder(const uint32_t instanceID, + const FileFormats fileFormat); + + static void DestroyFileRecorder(FileRecorder* recorder); + + virtual int32_t RegisterModuleFileCallback( + FileCallback* callback) = 0; + + virtual FileFormats RecordingFileFormat() const = 0; + + virtual int32_t StartRecordingAudioFile( + const char* fileName, + const CodecInst& codecInst, + uint32_t notification) = 0; + + virtual int32_t StartRecordingAudioFile( + OutStream& destStream, + const CodecInst& codecInst, + uint32_t notification) = 0; + + // Stop recording. + // Note: this API is for both audio and video. + virtual int32_t StopRecording() = 0; + + // Return true if recording. + // Note: this API is for both audio and video. + virtual bool IsRecording() const = 0; + + virtual int32_t codec_info(CodecInst& codecInst) const = 0; + + // Write frame to file. Frame should contain 10ms of un-ecoded audio data. + virtual int32_t RecordAudioToFile( + const AudioFrame& frame, + const TickTime* playoutTS = NULL) = 0; + + // Open/create the file specified by fileName for writing audio/video data + // (relative path is allowed). audioCodecInst specifies the encoding of the + // audio data. videoCodecInst specifies the encoding of the video data. + // Only video data will be recorded if videoOnly is true. amrFormat + // specifies the amr/amrwb storage format. + // Note: the file format is AVI. + virtual int32_t StartRecordingVideoFile( + const char* fileName, + const CodecInst& audioCodecInst, + const VideoCodec& videoCodecInst, + bool videoOnly = false) = 0; + + // Record the video frame in videoFrame to AVI file. + virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) = 0; + +protected: + virtual ~FileRecorder() {} + +}; +} // namespace webrtc +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ diff --git a/webrtc/modules/utility/include/helpers_android.h b/webrtc/modules/utility/include/helpers_android.h new file mode 100644 index 0000000000..2840ca965e --- /dev/null +++ b/webrtc/modules/utility/include/helpers_android.h @@ -0,0 +1,87 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_ + +#include +#include + +// Abort the process if |jni| has a Java exception pending. +// TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h. +#define CHECK_EXCEPTION(jni) \ + RTC_CHECK(!jni->ExceptionCheck()) \ + << (jni->ExceptionDescribe(), jni->ExceptionClear(), "") + +namespace webrtc { + +// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached. +JNIEnv* GetEnv(JavaVM* jvm); + +// Return a |jlong| that will correctly convert back to |ptr|. This is needed +// because the alternative (of silently passing a 32-bit pointer to a vararg +// function expecting a 64-bit param) picks up garbage in the high 32 bits. +jlong PointerTojlong(void* ptr); + +// JNIEnv-helper methods that wraps the API which uses the JNI interface +// pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java +// exception is thrown while calling the method. +jmethodID GetMethodID( + JNIEnv* jni, jclass c, const char* name, const char* signature); + +jmethodID GetStaticMethodID( + JNIEnv* jni, jclass c, const char* name, const char* signature); + +jclass FindClass(JNIEnv* jni, const char* name); + +jobject NewGlobalRef(JNIEnv* jni, jobject o); + +void DeleteGlobalRef(JNIEnv* jni, jobject o); + +// Return thread ID as a string. +std::string GetThreadId(); + +// Return thread ID as string suitable for debug logging. +std::string GetThreadInfo(); + +// Attach thread to JVM if necessary and detach at scope end if originally +// attached. +class AttachThreadScoped { + public: + explicit AttachThreadScoped(JavaVM* jvm); + ~AttachThreadScoped(); + JNIEnv* env(); + + private: + bool attached_; + JavaVM* jvm_; + JNIEnv* env_; +}; + +// Scoped holder for global Java refs. +template // T is jclass, jobject, jintArray, etc. +class ScopedGlobalRef { + public: + ScopedGlobalRef(JNIEnv* jni, T obj) + : jni_(jni), obj_(static_cast(NewGlobalRef(jni, obj))) {} + ~ScopedGlobalRef() { + DeleteGlobalRef(jni_, obj_); + } + T operator*() const { + return obj_; + } + private: + JNIEnv* jni_; + T obj_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_ diff --git a/webrtc/modules/utility/include/helpers_ios.h b/webrtc/modules/utility/include/helpers_ios.h new file mode 100644 index 0000000000..a5a07ace17 --- /dev/null +++ b/webrtc/modules/utility/include/helpers_ios.h @@ -0,0 +1,59 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_ + +#if defined(WEBRTC_IOS) + +#include + +namespace webrtc { +namespace ios { + +bool CheckAndLogError(BOOL success, NSError* error); + +std::string StdStringFromNSString(NSString* nsString); + +// Return thread ID as a string. +std::string GetThreadId(); + +// Return thread ID as string suitable for debug logging. +std::string GetThreadInfo(); + +// Returns [NSThread currentThread] description as string. +// Example: {number = 1, name = main} +std::string GetCurrentThreadDescription(); + +std::string GetAudioSessionCategory(); + +// Returns the current name of the operating system. +std::string GetSystemName(); + +// Returns the current version of the operating system. +std::string GetSystemVersion(); + +// Returns the version of the operating system as a floating point value. +float GetSystemVersionAsFloat(); + +// Returns the device type. +// Examples: ”iPhone” and ”iPod touch”. +std::string GetDeviceType(); + +// Returns a more detailed device name. +// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus". +std::string GetDeviceName(); + +} // namespace ios +} // namespace webrtc + +#endif // defined(WEBRTC_IOS) + +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_ diff --git a/webrtc/modules/utility/include/jvm_android.h b/webrtc/modules/utility/include/jvm_android.h new file mode 100644 index 0000000000..f527dff632 --- /dev/null +++ b/webrtc/modules/utility/include/jvm_android.h @@ -0,0 +1,185 @@ +/* + * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ + +#include +#include + +#include "webrtc/base/scoped_ptr.h" +#include "webrtc/base/thread_checker.h" +#include "webrtc/modules/utility/include/helpers_android.h" + +namespace webrtc { + +// The JNI interface pointer (JNIEnv) is valid only in the current thread. +// Should another thread need to access the Java VM, it must first call +// AttachCurrentThread() to attach itself to the VM and obtain a JNI interface +// pointer. The native thread remains attached to the VM until it calls +// DetachCurrentThread() to detach. +class AttachCurrentThreadIfNeeded { + public: + AttachCurrentThreadIfNeeded(); + ~AttachCurrentThreadIfNeeded(); + + private: + rtc::ThreadChecker thread_checker_; + bool attached_; +}; + +// This class is created by the NativeRegistration class and is used to wrap +// the actual Java object handle (jobject) on which we can call methods from +// C++ in to Java. See example in JVM for more details. +// TODO(henrika): extend support for type of function calls. +class GlobalRef { + public: + GlobalRef(JNIEnv* jni, jobject object); + ~GlobalRef(); + + jboolean CallBooleanMethod(jmethodID methodID, ...); + jint CallIntMethod(jmethodID methodID, ...); + void CallVoidMethod(jmethodID methodID, ...); + + private: + JNIEnv* const jni_; + const jobject j_object_; +}; + +// Wraps the jclass object on which we can call GetMethodId() functions to +// query method IDs. +class JavaClass { + public: + JavaClass(JNIEnv* jni, jclass clazz) : jni_(jni), j_class_(clazz) {} + ~JavaClass() {} + + jmethodID GetMethodId(const char* name, const char* signature); + jmethodID GetStaticMethodId(const char* name, const char* signature); + jobject CallStaticObjectMethod(jmethodID methodID, ...); + + protected: + JNIEnv* const jni_; + jclass const j_class_; +}; + +// Adds support of the NewObject factory method to the JavaClass class. +// See example in JVM for more details on how to use it. +class NativeRegistration : public JavaClass { + public: + NativeRegistration(JNIEnv* jni, jclass clazz); + ~NativeRegistration(); + + rtc::scoped_ptr NewObject( + const char* name, const char* signature, ...); + + private: + JNIEnv* const jni_; +}; + +// This class is created by the JVM class and is used to expose methods that +// needs the JNI interface pointer but its main purpose is to create a +// NativeRegistration object given name of a Java class and a list of native +// methods. See example in JVM for more details. +class JNIEnvironment { + public: + explicit JNIEnvironment(JNIEnv* jni); + ~JNIEnvironment(); + + // Registers native methods with the Java class specified by |name|. + // Note that the class name must be one of the names in the static + // |loaded_classes| array defined in jvm_android.cc. + // This method must be called on the construction thread. + rtc::scoped_ptr RegisterNatives( + const char* name, const JNINativeMethod *methods, int num_methods); + + // Converts from Java string to std::string. + // This method must be called on the construction thread. + std::string JavaToStdString(const jstring& j_string); + + private: + rtc::ThreadChecker thread_checker_; + JNIEnv* const jni_; +}; + +// Main class for working with Java from C++ using JNI in WebRTC. +// +// Example usage: +// +// // At initialization (e.g. in JNI_OnLoad), call JVM::Initialize. +// JNIEnv* jni = ::base::android::AttachCurrentThread(); +// JavaVM* jvm = NULL; +// jni->GetJavaVM(&jvm); +// jobject context = ::base::android::GetApplicationContext(); +// webrtc::JVM::Initialize(jvm, context); +// +// // Header (.h) file of example class called User. +// rtc::scoped_ptr env; +// rtc::scoped_ptr reg; +// rtc::scoped_ptr obj; +// +// // Construction (in .cc file) of User class. +// User::User() { +// // Calling thread must be attached to the JVM. +// env = JVM::GetInstance()->environment(); +// reg = env->RegisterNatives("org/webrtc/WebRtcTest", ,); +// obj = reg->NewObject("", ,); +// } +// +// // Each User method can now use |reg| and |obj| and call Java functions +// // in WebRtcTest.java, e.g. boolean init() {}. +// bool User::Foo() { +// jmethodID id = reg->GetMethodId("init", "()Z"); +// return obj->CallBooleanMethod(id); +// } +// +// // And finally, e.g. in JNI_OnUnLoad, call JVM::Uninitialize. +// JVM::Uninitialize(); +class JVM { + public: + // Stores global handles to the Java VM interface and the application context. + // Should be called once on a thread that is attached to the JVM. + static void Initialize(JavaVM* jvm, jobject context); + // Clears handles stored in Initialize(). Must be called on same thread as + // Initialize(). + static void Uninitialize(); + // Gives access to the global Java VM interface pointer, which then can be + // used to create a valid JNIEnvironment object or to get a JavaClass object. + static JVM* GetInstance(); + + // Creates a JNIEnvironment object. + // This method returns a NULL pointer if AttachCurrentThread() has not been + // called successfully. Use the AttachCurrentThreadIfNeeded class if needed. + rtc::scoped_ptr environment(); + + // Returns a JavaClass object given class |name|. + // Note that the class name must be one of the names in the static + // |loaded_classes| array defined in jvm_android.cc. + // This method must be called on the construction thread. + JavaClass GetClass(const char* name); + + // TODO(henrika): can we make these private? + JavaVM* jvm() const { return jvm_; } + jobject context() const { return context_; } + + protected: + JVM(JavaVM* jvm, jobject context); + ~JVM(); + + private: + JNIEnv* jni() const { return GetEnv(jvm_); } + + rtc::ThreadChecker thread_checker_; + JavaVM* const jvm_; + jobject context_; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ diff --git a/webrtc/modules/utility/include/mock/mock_process_thread.h b/webrtc/modules/utility/include/mock/mock_process_thread.h new file mode 100644 index 0000000000..56d92f4527 --- /dev/null +++ b/webrtc/modules/utility/include/mock/mock_process_thread.h @@ -0,0 +1,38 @@ +/* + * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_ + +#include "webrtc/modules/utility/include/process_thread.h" + +#include "testing/gmock/include/gmock/gmock.h" + +namespace webrtc { + +class MockProcessThread : public ProcessThread { + public: + MOCK_METHOD0(Start, void()); + MOCK_METHOD0(Stop, void()); + MOCK_METHOD1(WakeUp, void(Module* module)); + MOCK_METHOD1(PostTask, void(ProcessTask* task)); + MOCK_METHOD1(RegisterModule, void(Module* module)); + MOCK_METHOD1(DeRegisterModule, void(Module* module)); + + // MOCK_METHOD1 gets confused with mocking this method, so we work around it + // by overriding the method from the interface and forwarding the call to a + // mocked, simpler method. + void PostTask(rtc::scoped_ptr task) override { + PostTask(task.get()); + } +}; + +} // namespace webrtc +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_ diff --git a/webrtc/modules/utility/include/process_thread.h b/webrtc/modules/utility/include/process_thread.h new file mode 100644 index 0000000000..285a5ea587 --- /dev/null +++ b/webrtc/modules/utility/include/process_thread.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ + +#include "webrtc/typedefs.h" +#include "webrtc/base/scoped_ptr.h" + +namespace webrtc { +class Module; + +class ProcessTask { + public: + ProcessTask() {} + virtual ~ProcessTask() {} + + virtual void Run() = 0; +}; + +class ProcessThread { + public: + virtual ~ProcessThread(); + + static rtc::scoped_ptr Create(const char* thread_name); + + // Starts the worker thread. Must be called from the construction thread. + virtual void Start() = 0; + + // Stops the worker thread. Must be called from the construction thread. + virtual void Stop() = 0; + + // Wakes the thread up to give a module a chance to do processing right + // away. This causes the worker thread to wake up and requery the specified + // module for when it should be called back. (Typically the module should + // return 0 from TimeUntilNextProcess on the worker thread at that point). + // Can be called on any thread. + virtual void WakeUp(Module* module) = 0; + + // Queues a task object to run on the worker thread. Ownership of the + // task object is transferred to the ProcessThread and the object will + // either be deleted after running on the worker thread, or on the + // construction thread of the ProcessThread instance, if the task did not + // get a chance to run (e.g. posting the task while shutting down or when + // the thread never runs). + virtual void PostTask(rtc::scoped_ptr task) = 0; + + // Adds a module that will start to receive callbacks on the worker thread. + // Can be called from any thread. + virtual void RegisterModule(Module* module) = 0; + + // Removes a previously registered module. + // Can be called from any thread. + virtual void DeRegisterModule(Module* module) = 0; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ diff --git a/webrtc/modules/utility/interface/audio_frame_operations.h b/webrtc/modules/utility/interface/audio_frame_operations.h index c2af68ab1b..017352afc6 100644 --- a/webrtc/modules/utility/interface/audio_frame_operations.h +++ b/webrtc/modules/utility/interface/audio_frame_operations.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_ -#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #include "webrtc/typedefs.h" @@ -55,4 +57,4 @@ class AudioFrameOperations { } // namespace webrtc -#endif // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_ +#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_ diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/interface/file_player.h index 44f03e475a..d6737e1918 100644 --- a/webrtc/modules/utility/interface/file_player.h +++ b/webrtc/modules/utility/interface/file_player.h @@ -8,12 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" #include "webrtc/video_frame.h" @@ -108,4 +110,4 @@ protected: }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ diff --git a/webrtc/modules/utility/interface/file_recorder.h b/webrtc/modules/utility/interface/file_recorder.h index f2ce785368..c7e26f6b7f 100644 --- a/webrtc/modules/utility/interface/file_recorder.h +++ b/webrtc/modules/utility/interface/file_recorder.h @@ -8,13 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/media_file/interface/media_file_defines.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/typedefs.h" #include "webrtc/video_frame.h" @@ -81,4 +83,4 @@ protected: }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ diff --git a/webrtc/modules/utility/interface/helpers_android.h b/webrtc/modules/utility/interface/helpers_android.h index 5c73fe4566..80dd676faa 100644 --- a/webrtc/modules/utility/interface/helpers_android.h +++ b/webrtc/modules/utility/interface/helpers_android.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #include #include @@ -84,4 +86,4 @@ class ScopedGlobalRef { } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_ diff --git a/webrtc/modules/utility/interface/helpers_ios.h b/webrtc/modules/utility/interface/helpers_ios.h index a5edee0279..6013ee9c14 100644 --- a/webrtc/modules/utility/interface/helpers_ios.h +++ b/webrtc/modules/utility/interface/helpers_ios.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #if defined(WEBRTC_IOS) @@ -56,4 +58,4 @@ std::string GetDeviceName(); #endif // defined(WEBRTC_IOS) -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_ diff --git a/webrtc/modules/utility/interface/jvm_android.h b/webrtc/modules/utility/interface/jvm_android.h index 0744fdbf12..a417c1b942 100644 --- a/webrtc/modules/utility/interface/jvm_android.h +++ b/webrtc/modules/utility/interface/jvm_android.h @@ -8,15 +8,17 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_ -#define WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #include #include #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/modules/utility/interface/helpers_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" namespace webrtc { @@ -182,4 +184,4 @@ class JVM { } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_ diff --git a/webrtc/modules/utility/interface/mock/mock_process_thread.h b/webrtc/modules/utility/interface/mock/mock_process_thread.h index fd108a8354..c494d4c0ba 100644 --- a/webrtc/modules/utility/interface/mock/mock_process_thread.h +++ b/webrtc/modules/utility/interface/mock/mock_process_thread.h @@ -8,10 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_ -#include "webrtc/modules/utility/interface/process_thread.h" +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") + +#include "webrtc/modules/utility/include/process_thread.h" #include "testing/gmock/include/gmock/gmock.h" @@ -35,4 +37,4 @@ class MockProcessThread : public ProcessThread { }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_ diff --git a/webrtc/modules/utility/interface/process_thread.h b/webrtc/modules/utility/interface/process_thread.h index 451a5a301b..1a7a8750cd 100644 --- a/webrtc/modules/utility/interface/process_thread.h +++ b/webrtc/modules/utility/interface/process_thread.h @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_ -#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ + +#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include") #include "webrtc/typedefs.h" #include "webrtc/base/scoped_ptr.h" @@ -63,4 +65,4 @@ class ProcessThread { } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_ diff --git a/webrtc/modules/utility/source/audio_frame_operations.cc b/webrtc/modules/utility/source/audio_frame_operations.cc index c07ca1fdf6..fe09d7972f 100644 --- a/webrtc/modules/utility/source/audio_frame_operations.cc +++ b/webrtc/modules/utility/source/audio_frame_operations.cc @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" namespace webrtc { diff --git a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc index c278cdddcd..fff8f4407b 100644 --- a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc +++ b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc @@ -10,8 +10,8 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" namespace webrtc { namespace { diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc index 4ec5f9b4e2..18b690dc67 100644 --- a/webrtc/modules/utility/source/coder.cc +++ b/webrtc/modules/utility/source/coder.cc @@ -9,7 +9,7 @@ */ #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/utility/source/coder.h" namespace webrtc { diff --git a/webrtc/modules/utility/source/file_player_impl.h b/webrtc/modules/utility/source/file_player_impl.h index f411db9151..2dfe68273e 100644 --- a/webrtc/modules/utility/source/file_player_impl.h +++ b/webrtc/modules/utility/source/file_player_impl.h @@ -14,9 +14,9 @@ #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/media_file/interface/media_file.h" -#include "webrtc/modules/media_file/interface/media_file_defines.h" -#include "webrtc/modules/utility/interface/file_player.h" +#include "webrtc/modules/media_file/include/media_file.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" +#include "webrtc/modules/utility/include/file_player.h" #include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc index 4b65acdeef..754e1242d3 100644 --- a/webrtc/modules/utility/source/file_player_unittests.cc +++ b/webrtc/modules/utility/source/file_player_unittests.cc @@ -10,7 +10,7 @@ // Unit tests for FilePlayer. -#include "webrtc/modules/utility/interface/file_player.h" +#include "webrtc/modules/utility/include/file_player.h" #include #include diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc index 13926deb4a..cbd7c08d6b 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.cc +++ b/webrtc/modules/utility/source/file_recorder_impl.cc @@ -10,7 +10,7 @@ #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/media_file/interface/media_file.h" +#include "webrtc/modules/media_file/include/media_file.h" #include "webrtc/modules/utility/source/file_recorder_impl.h" #include "webrtc/system_wrappers/include/logging.h" diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h index 8ea96bdad4..01072cdc3d 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.h +++ b/webrtc/modules/utility/source/file_recorder_impl.h @@ -20,10 +20,10 @@ #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/media_file/interface/media_file.h" -#include "webrtc/modules/media_file/interface/media_file_defines.h" -#include "webrtc/modules/utility/interface/file_recorder.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/media_file/include/media_file.h" +#include "webrtc/modules/media_file/include/media_file_defines.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/thread_wrapper.h" diff --git a/webrtc/modules/utility/source/helpers_android.cc b/webrtc/modules/utility/source/helpers_android.cc index 25652f237e..aea35f8d5a 100644 --- a/webrtc/modules/utility/source/helpers_android.cc +++ b/webrtc/modules/utility/source/helpers_android.cc @@ -9,7 +9,7 @@ */ #include "webrtc/base/checks.h" -#include "webrtc/modules/utility/interface/helpers_android.h" +#include "webrtc/modules/utility/include/helpers_android.h" #include #include diff --git a/webrtc/modules/utility/source/helpers_ios.mm b/webrtc/modules/utility/source/helpers_ios.mm index 90b7c8f605..2d0ac098c1 100644 --- a/webrtc/modules/utility/source/helpers_ios.mm +++ b/webrtc/modules/utility/source/helpers_ios.mm @@ -18,7 +18,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/utility/interface/helpers_ios.h" +#include "webrtc/modules/utility/include/helpers_ios.h" namespace webrtc { namespace ios { diff --git a/webrtc/modules/utility/source/jvm_android.cc b/webrtc/modules/utility/source/jvm_android.cc index 648c1685ea..eb37fda040 100644 --- a/webrtc/modules/utility/source/jvm_android.cc +++ b/webrtc/modules/utility/source/jvm_android.cc @@ -10,7 +10,7 @@ #include -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" #include "webrtc/base/checks.h" diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc index 04fa88739f..9c739de8fc 100644 --- a/webrtc/modules/utility/source/process_thread_impl.cc +++ b/webrtc/modules/utility/source/process_thread_impl.cc @@ -11,7 +11,7 @@ #include "webrtc/modules/utility/source/process_thread_impl.h" #include "webrtc/base/checks.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/modules/utility/source/process_thread_impl.h b/webrtc/modules/utility/source/process_thread_impl.h index 4e5861b41e..0a956654f6 100644 --- a/webrtc/modules/utility/source/process_thread_impl.h +++ b/webrtc/modules/utility/source/process_thread_impl.h @@ -16,7 +16,7 @@ #include "webrtc/base/criticalsection.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/thread_wrapper.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/webrtc/modules/utility/source/process_thread_impl_unittest.cc index e080545312..34ed5c5a38 100644 --- a/webrtc/modules/utility/source/process_thread_impl_unittest.cc +++ b/webrtc/modules/utility/source/process_thread_impl_unittest.cc @@ -10,7 +10,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/modules/utility/source/process_thread_impl.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/modules/utility/utility.gypi b/webrtc/modules/utility/utility.gypi index 38c9e3ebd9..e5b0a4d9c0 100644 --- a/webrtc/modules/utility/utility.gypi +++ b/webrtc/modules/utility/utility.gypi @@ -18,13 +18,13 @@ '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', ], 'sources': [ - 'interface/audio_frame_operations.h', - 'interface/file_player.h', - 'interface/file_recorder.h', - 'interface/helpers_android.h', - 'interface/helpers_ios.h', - 'interface/jvm_android.h', - 'interface/process_thread.h', + 'include/audio_frame_operations.h', + 'include/file_player.h', + 'include/file_recorder.h', + 'include/helpers_android.h', + 'include/helpers_ios.h', + 'include/jvm_android.h', + 'include/process_thread.h', 'source/audio_frame_operations.cc', 'source/coder.cc', 'source/coder.h', diff --git a/webrtc/modules/video_capture/include/video_capture.h b/webrtc/modules/video_capture/include/video_capture.h index 09b4502115..a032d3cc18 100644 --- a/webrtc/modules/video_capture/include/video_capture.h +++ b/webrtc/modules/video_capture/include/video_capture.h @@ -12,7 +12,7 @@ #define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_ #include "webrtc/common_video/rotation.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/modules/video_capture/include/video_capture_defines.h" namespace webrtc { diff --git a/webrtc/modules/video_capture/include/video_capture_defines.h b/webrtc/modules/video_capture/include/video_capture_defines.h index 1dee4fa814..f62ddadccf 100644 --- a/webrtc/modules/video_capture/include/video_capture_defines.h +++ b/webrtc/modules/video_capture/include/video_capture_defines.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_ #define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" #include "webrtc/video_frame.h" diff --git a/webrtc/modules/video_capture/test/video_capture_unittest.cc b/webrtc/modules/video_capture/test/video_capture_unittest.cc index 2b8786b0fe..3ec164b37e 100644 --- a/webrtc/modules/video_capture/test/video_capture_unittest.cc +++ b/webrtc/modules/video_capture/test/video_capture_unittest.cc @@ -17,7 +17,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/modules/video_capture/include/video_capture.h" #include "webrtc/modules/video_capture/include/video_capture_factory.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/video_capture/video_capture_impl.cc b/webrtc/modules/video_capture/video_capture_impl.cc index 4046181505..90730cd984 100644 --- a/webrtc/modules/video_capture/video_capture_impl.cc +++ b/webrtc/modules/video_capture/video_capture_impl.cc @@ -14,7 +14,7 @@ #include "webrtc/base/trace_event.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_capture/video_capture_config.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h index 230dea94a0..bdb079bd36 100644 --- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h +++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h @@ -19,7 +19,7 @@ #include #include "webrtc/base/buffer.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h index 6363ab7332..5115c4bc9e 100644 --- a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h +++ b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h @@ -14,7 +14,7 @@ #include #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/codecs/interface/video_error_codes.h" #include "webrtc/typedefs.h" #include "webrtc/video_decoder.h" diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc index da6008ba3d..47f0cf38ce 100644 --- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc +++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc @@ -13,7 +13,7 @@ #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h" diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc index 029ccd1f27..4f4e435712 100644 --- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc +++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc @@ -24,7 +24,7 @@ #include "webrtc/common.h" #include "webrtc/common_types.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h" #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h" diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc index 97f08462bf..7ebe1a275d 100644 --- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc +++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc @@ -26,7 +26,7 @@ #include "webrtc/base/trace_event.h" #include "webrtc/common.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h index 67f7b635cb..94a7b19198 100644 --- a/webrtc/modules/video_coding/main/interface/video_coding.h +++ b/webrtc/modules/video_coding/main/interface/video_coding.h @@ -21,8 +21,8 @@ #include #endif -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/video_frame.h" diff --git a/webrtc/modules/video_coding/main/interface/video_coding_defines.h b/webrtc/modules/video_coding/main/interface/video_coding_defines.h index fd38d64415..c35bbc1f73 100644 --- a/webrtc/modules/video_coding/main/interface/video_coding_defines.h +++ b/webrtc/modules/video_coding/main/interface/video_coding_defines.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_ #define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" #include "webrtc/video_frame.h" diff --git a/webrtc/modules/video_coding/main/source/codec_timer.h b/webrtc/modules/video_coding/main/source/codec_timer.h index 9268e8d817..cb7e813ba0 100644 --- a/webrtc/modules/video_coding/main/source/codec_timer.h +++ b/webrtc/modules/video_coding/main/source/codec_timer.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_ #define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc diff --git a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc b/webrtc/modules/video_coding/main/source/content_metrics_processing.cc index 757ffb0e46..ae5e633316 100644 --- a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc +++ b/webrtc/modules/video_coding/main/source/content_metrics_processing.cc @@ -12,7 +12,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" namespace webrtc { diff --git a/webrtc/modules/video_coding/main/source/decoding_state.cc b/webrtc/modules/video_coding/main/source/decoding_state.cc index cc92f1c83f..a3da7c65d8 100644 --- a/webrtc/modules/video_coding/main/source/decoding_state.cc +++ b/webrtc/modules/video_coding/main/source/decoding_state.cc @@ -10,7 +10,7 @@ #include "webrtc/modules/video_coding/main/source/decoding_state.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/frame_buffer.h" #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" #include "webrtc/modules/video_coding/main/source/packet.h" diff --git a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc index feae701a65..30b57862d5 100644 --- a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc +++ b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc @@ -11,7 +11,7 @@ #include #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/decoding_state.h" #include "webrtc/modules/video_coding/main/source/frame_buffer.h" #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.h b/webrtc/modules/video_coding/main/source/encoded_frame.h index 608578c35d..dc6bbb8445 100644 --- a/webrtc/modules/video_coding/main/source/encoded_frame.h +++ b/webrtc/modules/video_coding/main/source/encoded_frame.h @@ -15,7 +15,7 @@ #include "webrtc/common_types.h" #include "webrtc/common_video/interface/video_image.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" diff --git a/webrtc/modules/video_coding/main/source/frame_buffer.h b/webrtc/modules/video_coding/main/source/frame_buffer.h index ab4ff6574e..ee38a2e798 100644 --- a/webrtc/modules/video_coding/main/source/frame_buffer.h +++ b/webrtc/modules/video_coding/main/source/frame_buffer.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_ #define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/source/encoded_frame.h" #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.h b/webrtc/modules/video_coding/main/source/generic_decoder.h index 09929e64f4..c1298bb3af 100644 --- a/webrtc/modules/video_coding/main/source/generic_decoder.h +++ b/webrtc/modules/video_coding/main/source/generic_decoder.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_ #define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" #include "webrtc/modules/video_coding/main/source/encoded_frame.h" #include "webrtc/modules/video_coding/main/source/timestamp_map.h" diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc index bfdd7867d9..bc63411346 100644 --- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc +++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc @@ -16,7 +16,7 @@ #include "webrtc/base/checks.h" #include "webrtc/base/trace_event.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/source/frame_buffer.h" #include "webrtc/modules/video_coding/main/source/inter_frame_delay.h" diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h index f4a3638f7d..9bde97cafc 100644 --- a/webrtc/modules/video_coding/main/source/jitter_buffer.h +++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h @@ -18,7 +18,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/thread_annotations.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/modules/video_coding/main/source/decoding_state.h" diff --git a/webrtc/modules/video_coding/main/source/media_opt_util.cc b/webrtc/modules/video_coding/main/source/media_opt_util.cc index 51decbed97..002958ecad 100644 --- a/webrtc/modules/video_coding/main/source/media_opt_util.cc +++ b/webrtc/modules/video_coding/main/source/media_opt_util.cc @@ -15,7 +15,7 @@ #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/modules/video_coding/main/source/fec_tables_xor.h" diff --git a/webrtc/modules/video_coding/main/source/media_optimization.h b/webrtc/modules/video_coding/main/source/media_optimization.h index c4feeff743..6ea5c01412 100644 --- a/webrtc/modules/video_coding/main/source/media_optimization.h +++ b/webrtc/modules/video_coding/main/source/media_optimization.h @@ -14,7 +14,7 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/source/media_opt_util.h" #include "webrtc/modules/video_coding/main/source/qm_select.h" diff --git a/webrtc/modules/video_coding/main/source/packet.cc b/webrtc/modules/video_coding/main/source/packet.cc index fd5a6abb8c..34384483be 100644 --- a/webrtc/modules/video_coding/main/source/packet.cc +++ b/webrtc/modules/video_coding/main/source/packet.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/packet.h" #include diff --git a/webrtc/modules/video_coding/main/source/packet.h b/webrtc/modules/video_coding/main/source/packet.h index 80bf532502..040cc88e72 100644 --- a/webrtc/modules/video_coding/main/source/packet.h +++ b/webrtc/modules/video_coding/main/source/packet.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_ #define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/video_coding/main/source/qm_select.cc b/webrtc/modules/video_coding/main/source/qm_select.cc index e86d0755c0..be8fcfc122 100644 --- a/webrtc/modules/video_coding/main/source/qm_select.cc +++ b/webrtc/modules/video_coding/main/source/qm_select.cc @@ -12,7 +12,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/source/qm_select_data.h" diff --git a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc b/webrtc/modules/video_coding/main/source/qm_select_unittest.cc index 6abc0d3099..518df34716 100644 --- a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc +++ b/webrtc/modules/video_coding/main/source/qm_select_unittest.cc @@ -15,7 +15,7 @@ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/qm_select.h" namespace webrtc { diff --git a/webrtc/modules/video_coding/main/source/session_info.h b/webrtc/modules/video_coding/main/source/session_info.h index 88071e19d5..37e9768660 100644 --- a/webrtc/modules/video_coding/main/source/session_info.h +++ b/webrtc/modules/video_coding/main/source/session_info.h @@ -13,7 +13,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/source/packet.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/video_coding/main/source/session_info_unittest.cc b/webrtc/modules/video_coding/main/source/session_info_unittest.cc index 58c352d3fc..8d57287c93 100644 --- a/webrtc/modules/video_coding/main/source/session_info_unittest.cc +++ b/webrtc/modules/video_coding/main/source/session_info_unittest.cc @@ -11,7 +11,7 @@ #include #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/packet.h" #include "webrtc/modules/video_coding/main/source/session_info.h" diff --git a/webrtc/modules/video_coding/main/source/timestamp_map.cc b/webrtc/modules/video_coding/main/source/timestamp_map.cc index c68a5af7ba..d11f9491ea 100644 --- a/webrtc/modules/video_coding/main/source/timestamp_map.cc +++ b/webrtc/modules/video_coding/main/source/timestamp_map.cc @@ -11,7 +11,7 @@ #include #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/source/timestamp_map.h" namespace webrtc { diff --git a/webrtc/modules/video_coding/main/test/receiver_tests.h b/webrtc/modules/video_coding/main/test/receiver_tests.h index 6d7b7beeb5..5335c998e7 100644 --- a/webrtc/modules/video_coding/main/test/receiver_tests.h +++ b/webrtc/modules/video_coding/main/test/receiver_tests.h @@ -12,8 +12,8 @@ #define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_ #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/test/test_util.h" #include "webrtc/modules/video_coding/main/test/video_source.h" diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc index 6717cf227d..5fed3b18c3 100644 --- a/webrtc/modules/video_coding/main/test/rtp_player.cc +++ b/webrtc/modules/video_coding/main/test/rtp_player.cc @@ -15,10 +15,10 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/video_coding/main/source/internal_defines.h" #include "webrtc/modules/video_coding/main/test/test_util.h" #include "webrtc/system_wrappers/include/clock.h" diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/main/test/rtp_player.h index 7459231416..a2ecadd615 100644 --- a/webrtc/modules/video_coding/main/test/rtp_player.h +++ b/webrtc/modules/video_coding/main/test/rtp_player.h @@ -14,7 +14,7 @@ #include #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" namespace webrtc { diff --git a/webrtc/modules/video_coding/main/test/test_util.h b/webrtc/modules/video_coding/main/test/test_util.h index 27f66fe011..68b2c23b77 100644 --- a/webrtc/modules/video_coding/main/test/test_util.h +++ b/webrtc/modules/video_coding/main/test/test_util.h @@ -18,7 +18,7 @@ #include #include "webrtc/base/constructormagic.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/system_wrappers/include/event_wrapper.h" diff --git a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc index 2d874cd1bd..d930805bb4 100644 --- a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc +++ b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc @@ -14,7 +14,7 @@ #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/video_coding/main/test/test_util.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/video_processing/main/interface/video_processing.h b/webrtc/modules/video_processing/main/interface/video_processing.h index 30af99fb8e..377a098229 100644 --- a/webrtc/modules/video_processing/main/interface/video_processing.h +++ b/webrtc/modules/video_processing/main/interface/video_processing.h @@ -18,8 +18,8 @@ #ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H #define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H -#include "webrtc/modules/interface/module.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_processing/main/interface/video_processing_defines.h" #include "webrtc/video_frame.h" diff --git a/webrtc/modules/video_processing/main/source/content_analysis.h b/webrtc/modules/video_processing/main/source/content_analysis.h index 510c1b4a55..5b0767ac04 100644 --- a/webrtc/modules/video_processing/main/source/content_analysis.h +++ b/webrtc/modules/video_processing/main/source/content_analysis.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H #define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_processing/main/interface/video_processing_defines.h" #include "webrtc/typedefs.h" #include "webrtc/video_frame.h" diff --git a/webrtc/modules/video_processing/main/source/spatial_resampler.h b/webrtc/modules/video_processing/main/source/spatial_resampler.h index f965a40a83..9a2d8f58c9 100644 --- a/webrtc/modules/video_processing/main/source/spatial_resampler.h +++ b/webrtc/modules/video_processing/main/source/spatial_resampler.h @@ -13,7 +13,7 @@ #include "webrtc/typedefs.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_processing/main/interface/video_processing_defines.h" #include "webrtc/common_video/libyuv/include/scaler.h" diff --git a/webrtc/modules/video_processing/main/source/video_decimator.h b/webrtc/modules/video_processing/main/source/video_decimator.h index 3d4573caf8..c052c30afa 100644 --- a/webrtc/modules/video_processing/main/source/video_decimator.h +++ b/webrtc/modules/video_processing/main/source/video_decimator.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H #define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/video_render/external/video_render_external_impl.h b/webrtc/modules/video_render/external/video_render_external_impl.h index 9230e60acc..a8b663fff7 100644 --- a/webrtc/modules/video_render/external/video_render_external_impl.h +++ b/webrtc/modules/video_render/external/video_render_external_impl.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_ #define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_ -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/video_render/i_video_render.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/video_render/include/video_render.h b/webrtc/modules/video_render/include/video_render.h index 51fcce10c3..edd2302696 100644 --- a/webrtc/modules/video_render/include/video_render.h +++ b/webrtc/modules/video_render/include/video_render.h @@ -20,7 +20,7 @@ * */ -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/modules/video_render/include/video_render_defines.h" namespace webrtc { diff --git a/webrtc/modules/video_render/include/video_render_defines.h b/webrtc/modules/video_render/include/video_render_defines.h index f8f48035ea..1602d66065 100644 --- a/webrtc/modules/video_render/include/video_render_defines.h +++ b/webrtc/modules/video_render/include/video_render_defines.h @@ -13,7 +13,7 @@ #include "webrtc/common_types.h" #include "webrtc/common_video/interface/incoming_video_stream.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/modules/video_render/test/testAPI/testAPI.cc b/webrtc/modules/video_render/test/testAPI/testAPI.cc index 256d031c0c..4de554c475 100644 --- a/webrtc/modules/video_render/test/testAPI/testAPI.cc +++ b/webrtc/modules/video_render/test/testAPI/testAPI.cc @@ -32,8 +32,8 @@ #endif #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/modules/video_render/include/video_render.h" #include "webrtc/modules/video_render/include/video_render_defines.h" #include "webrtc/system_wrappers/include/sleep.h" diff --git a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm index dd57397c73..a8808dd85f 100644 --- a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm +++ b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm @@ -20,8 +20,8 @@ #import "webrtc/modules/video_render/mac/cocoa_render_view.h" #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/modules/video_render/include/video_render.h" #include "webrtc/modules/video_render/include/video_render_defines.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc index 5533a4cd46..9cf02edb2b 100644 --- a/webrtc/test/layer_filtering_transport.cc +++ b/webrtc/test/layer_filtering_transport.cc @@ -9,7 +9,8 @@ */ #include "webrtc/base/checks.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/source/byte_io.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/test/layer_filtering_transport.h" diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h index 89b6dd06bd..c7e463e164 100644 --- a/webrtc/test/rtp_rtcp_observer.h +++ b/webrtc/test/rtp_rtcp_observer.h @@ -16,7 +16,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/criticalsection.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/test/constants.h" #include "webrtc/test/direct_transport.h" #include "webrtc/typedefs.h" diff --git a/webrtc/tools/agc/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc index 18e7c6dad8..3f8235e641 100644 --- a/webrtc/tools/agc/activity_metric.cc +++ b/webrtc/tools/agc/activity_metric.cc @@ -24,7 +24,7 @@ #include "webrtc/modules/audio_processing/vad/common.h" #include "webrtc/modules/audio_processing/vad/pitch_based_vad.h" #include "webrtc/modules/audio_processing/vad/standalone_vad.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" static const int kAgcAnalWindowSamples = 100; static const double kDefaultActivityThreshold = 0.3; diff --git a/webrtc/tools/agc/test_utils.cc b/webrtc/tools/agc/test_utils.cc index 81819c598e..a0ed74732d 100644 --- a/webrtc/tools/agc/test_utils.cc +++ b/webrtc/tools/agc/test_utils.cc @@ -14,7 +14,7 @@ #include -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" namespace webrtc { diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc index 70efe3b9ed..e2dce3f3f8 100644 --- a/webrtc/video/rampup_tests.cc +++ b/webrtc/video/rampup_tests.cc @@ -16,10 +16,10 @@ #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h" #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h" #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/thread_wrapper.h" diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc index f54909e55f..484924872b 100644 --- a/webrtc/video/replay.cc +++ b/webrtc/video/replay.cc @@ -20,7 +20,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/sleep.h" #include "webrtc/test/encoder_settings.h" diff --git a/webrtc/video/video_capture_input.cc b/webrtc/video/video_capture_input.cc index 42bc65f05f..834849c4c6 100644 --- a/webrtc/video/video_capture_input.cc +++ b/webrtc/video/video_capture_input.cc @@ -13,8 +13,8 @@ #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/modules/video_capture/include/video_capture_factory.h" #include "webrtc/modules/video_processing/main/interface/video_processing.h" #include "webrtc/modules/video_render/include/video_render_defines.h" diff --git a/webrtc/video/video_capture_input_unittest.cc b/webrtc/video/video_capture_input_unittest.cc index e8bc2ad1c9..12ef076c6d 100644 --- a/webrtc/video/video_capture_input_unittest.cc +++ b/webrtc/video/video_capture_input_unittest.cc @@ -15,7 +15,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common.h" -#include "webrtc/modules/utility/interface/mock/mock_process_thread.h" +#include "webrtc/modules/utility/include/mock/mock_process_thread.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/ref_count.h" diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc index 333f00dc43..ebceb5642b 100644 --- a/webrtc/video/video_quality_test.cc +++ b/webrtc/video/video_quality_test.cc @@ -22,6 +22,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/call.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/system_wrappers/include/cpu_info.h" #include "webrtc/test/layer_filtering_transport.h" diff --git a/webrtc/video/video_send_stream.h b/webrtc/video/video_send_stream.h index 36a87e3514..6b0d30213a 100644 --- a/webrtc/video/video_send_stream.h +++ b/webrtc/video/video_send_stream.h @@ -17,7 +17,7 @@ #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/video/encoded_frame_callback_adapter.h" #include "webrtc/video/send_statistics_proxy.h" diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc index 59011a6162..e19dc48fab 100644 --- a/webrtc/video/video_send_stream_tests.cc +++ b/webrtc/video/video_send_stream_tests.cc @@ -20,8 +20,8 @@ #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" #include "webrtc/frame_callback.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h" diff --git a/webrtc/video_engine/call_stats.cc b/webrtc/video_engine/call_stats.cc index 0b71cc346c..4d5338c5fc 100644 --- a/webrtc/video_engine/call_stats.cc +++ b/webrtc/video_engine/call_stats.cc @@ -12,7 +12,7 @@ #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/video_engine/call_stats.h b/webrtc/video_engine/call_stats.h index a17330a7c1..d0a0b5313a 100644 --- a/webrtc/video_engine/call_stats.h +++ b/webrtc/video_engine/call_stats.h @@ -15,7 +15,7 @@ #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" namespace webrtc { diff --git a/webrtc/video_engine/call_stats_unittest.cc b/webrtc/video_engine/call_stats_unittest.cc index 4fb88df338..bfba5cb72a 100644 --- a/webrtc/video_engine/call_stats_unittest.cc +++ b/webrtc/video_engine/call_stats_unittest.cc @@ -12,7 +12,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/video_engine/call_stats.h" diff --git a/webrtc/video_engine/encoder_state_feedback.cc b/webrtc/video_engine/encoder_state_feedback.cc index 1c376b2820..16ee8b95fc 100644 --- a/webrtc/video_engine/encoder_state_feedback.cc +++ b/webrtc/video_engine/encoder_state_feedback.cc @@ -13,7 +13,7 @@ #include #include "webrtc/base/checks.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/video_engine/vie_encoder.h" diff --git a/webrtc/video_engine/encoder_state_feedback_unittest.cc b/webrtc/video_engine/encoder_state_feedback_unittest.cc index 9787acc144..ea252dc48a 100644 --- a/webrtc/video_engine/encoder_state_feedback_unittest.cc +++ b/webrtc/video_engine/encoder_state_feedback_unittest.cc @@ -20,8 +20,8 @@ #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" #include "webrtc/modules/pacing/include/paced_sender.h" #include "webrtc/modules/pacing/include/packet_router.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" -#include "webrtc/modules/utility/interface/mock/mock_process_thread.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "webrtc/modules/utility/include/mock/mock_process_thread.h" #include "webrtc/video_engine/payload_router.h" #include "webrtc/video_engine/vie_encoder.h" diff --git a/webrtc/video_engine/overuse_frame_detector.h b/webrtc/video_engine/overuse_frame_detector.h index aff4b43025..0deb5ba83c 100644 --- a/webrtc/video_engine/overuse_frame_detector.h +++ b/webrtc/video_engine/overuse_frame_detector.h @@ -17,7 +17,7 @@ #include "webrtc/base/exp_filter.h" #include "webrtc/base/thread_annotations.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" namespace webrtc { diff --git a/webrtc/video_engine/payload_router.cc b/webrtc/video_engine/payload_router.cc index 3af3d4829e..85b294bfdf 100644 --- a/webrtc/video_engine/payload_router.cc +++ b/webrtc/video_engine/payload_router.cc @@ -11,8 +11,8 @@ #include "webrtc/video_engine/payload_router.h" #include "webrtc/base/checks.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" namespace webrtc { diff --git a/webrtc/video_engine/payload_router_unittest.cc b/webrtc/video_engine/payload_router_unittest.cc index de391576d8..acaa4006c4 100644 --- a/webrtc/video_engine/payload_router_unittest.cc +++ b/webrtc/video_engine/payload_router_unittest.cc @@ -14,7 +14,7 @@ #include "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" #include "webrtc/video_engine/payload_router.h" diff --git a/webrtc/video_engine/report_block_stats.h b/webrtc/video_engine/report_block_stats.h index dadcc9d410..dd430729fb 100644 --- a/webrtc/video_engine/report_block_stats.h +++ b/webrtc/video_engine/report_block_stats.h @@ -15,7 +15,7 @@ #include #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc index 9140cf1510..b4e83320d8 100644 --- a/webrtc/video_engine/vie_channel.cc +++ b/webrtc/video_engine/vie_channel.cc @@ -21,9 +21,9 @@ #include "webrtc/frame_callback.h" #include "webrtc/modules/pacing/include/paced_sender.h" #include "webrtc/modules/pacing/include/packet_router.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_processing/main/interface/video_processing.h" #include "webrtc/modules/video_render/include/video_render_defines.h" diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h index b1c1bbaa9d..20eae7d298 100644 --- a/webrtc/video_engine/vie_channel.h +++ b/webrtc/video_engine/vie_channel.h @@ -16,8 +16,8 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/scoped_ref_ptr.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc index 0f4a5a14f5..96ed2f9afd 100644 --- a/webrtc/video_engine/vie_encoder.cc +++ b/webrtc/video_engine/vie_encoder.cc @@ -22,7 +22,7 @@ #include "webrtc/frame_callback.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" #include "webrtc/modules/pacing/include/paced_sender.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" diff --git a/webrtc/video_engine/vie_encoder.h b/webrtc/video_engine/vie_encoder.h index 54aacdbfa9..872fbb669d 100644 --- a/webrtc/video_engine/vie_encoder.h +++ b/webrtc/video_engine/vie_encoder.h @@ -20,7 +20,7 @@ #include "webrtc/common_types.h" #include "webrtc/frame_callback.h" #include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h" #include "webrtc/modules/video_processing/main/interface/video_processing.h" #include "webrtc/typedefs.h" diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc index 2e3b588302..c99fa76e7b 100644 --- a/webrtc/video_engine/vie_receiver.cc +++ b/webrtc/video_engine/vie_receiver.cc @@ -14,14 +14,14 @@ #include "webrtc/base/logging.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" -#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/metrics.h" diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h index cd069eaa5b..20a9627beb 100644 --- a/webrtc/video_engine/vie_receiver.h +++ b/webrtc/video_engine/vie_receiver.h @@ -15,8 +15,8 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "webrtc/typedefs.h" #include "webrtc/video_engine/vie_defines.h" diff --git a/webrtc/video_engine/vie_remb.cc b/webrtc/video_engine/vie_remb.cc index b347f2ee00..3901d6d6e9 100644 --- a/webrtc/video_engine/vie_remb.cc +++ b/webrtc/video_engine/vie_remb.cc @@ -14,8 +14,8 @@ #include -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/video_engine/vie_remb.h b/webrtc/video_engine/vie_remb.h index 9f38259ca8..6a79ffe652 100644 --- a/webrtc/video_engine/vie_remb.h +++ b/webrtc/video_engine/vie_remb.h @@ -16,9 +16,9 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" namespace webrtc { diff --git a/webrtc/video_engine/vie_remb_unittest.cc b/webrtc/video_engine/vie_remb_unittest.cc index 3289c4b822..b6fbf29bf2 100644 --- a/webrtc/video_engine/vie_remb_unittest.cc +++ b/webrtc/video_engine/vie_remb_unittest.cc @@ -17,9 +17,9 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" -#include "webrtc/modules/utility/interface/mock/mock_process_thread.h" +#include "webrtc/modules/utility/include/mock/mock_process_thread.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/video_engine/vie_remb.h" diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc index e7327eb103..b2e2713d92 100644 --- a/webrtc/video_engine/vie_sync_module.cc +++ b/webrtc/video_engine/vie_sync_module.cc @@ -12,8 +12,8 @@ #include "webrtc/base/logging.h" #include "webrtc/base/trace_event.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" #include "webrtc/modules/video_coding/main/interface/video_coding.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/video_engine/stream_synchronization.h" diff --git a/webrtc/video_engine/vie_sync_module.h b/webrtc/video_engine/vie_sync_module.h index 2a343b8ea9..dcd8072095 100644 --- a/webrtc/video_engine/vie_sync_module.h +++ b/webrtc/video_engine/vie_sync_module.h @@ -15,7 +15,7 @@ #define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/video_engine/stream_synchronization.h" #include "webrtc/voice_engine/include/voe_video_sync.h" diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc index ed47b57fa8..0a3e69654e 100644 --- a/webrtc/voice_engine/channel.cc +++ b/webrtc/voice_engine/channel.cc @@ -19,13 +19,13 @@ #include "webrtc/config.h" #include "webrtc/modules/audio_device/include/audio_device.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index 0f7b5435a9..ba18aaa8dd 100644 --- a/webrtc/voice_engine/channel.h +++ b/webrtc/voice_engine/channel.h @@ -15,14 +15,14 @@ #include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" #include "webrtc/modules/audio_processing/rms_level.h" #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" -#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" -#include "webrtc/modules/utility/interface/file_player.h" -#include "webrtc/modules/utility/interface/file_recorder.h" +#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" +#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/voice_engine/dtmf_inband.h" #include "webrtc/voice_engine/dtmf_inband_queue.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" diff --git a/webrtc/voice_engine/level_indicator.cc b/webrtc/voice_engine/level_indicator.cc index 059b15f927..68a837edb9 100644 --- a/webrtc/voice_engine/level_indicator.cc +++ b/webrtc/voice_engine/level_indicator.cc @@ -9,7 +9,7 @@ */ #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/voice_engine/level_indicator.h" diff --git a/webrtc/voice_engine/monitor_module.h b/webrtc/voice_engine/monitor_module.h index 42ea74d7e2..fe915b320b 100644 --- a/webrtc/voice_engine/monitor_module.h +++ b/webrtc/voice_engine/monitor_module.h @@ -11,7 +11,7 @@ #ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H #define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H -#include "webrtc/modules/interface/module.h" +#include "webrtc/modules/include/module.h" #include "webrtc/typedefs.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc index 31b429c498..1b4d2e23b3 100644 --- a/webrtc/voice_engine/output_mixer.cc +++ b/webrtc/voice_engine/output_mixer.cc @@ -11,7 +11,7 @@ #include "webrtc/voice_engine/output_mixer.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/file_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h index 71e55e4885..8c0a2de53b 100644 --- a/webrtc/voice_engine/output_mixer.h +++ b/webrtc/voice_engine/output_mixer.h @@ -13,9 +13,9 @@ #include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h" -#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h" -#include "webrtc/modules/utility/interface/file_recorder.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h" +#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/voice_engine/dtmf_inband.h" #include "webrtc/voice_engine/level_indicator.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h index 311bfa063d..9c3d4b1c57 100644 --- a/webrtc/voice_engine/shared_data.h +++ b/webrtc/voice_engine/shared_data.h @@ -14,7 +14,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_device/include/audio_device.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" -#include "webrtc/modules/utility/interface/process_thread.h" +#include "webrtc/modules/utility/include/process_thread.h" #include "webrtc/voice_engine/channel_manager.h" #include "webrtc/voice_engine/statistics.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h index 7b45e6d3e1..fb430eba4d 100644 --- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h +++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h @@ -19,7 +19,7 @@ #include "webrtc/base/basictypes.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/thread_wrapper.h" diff --git a/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc b/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc index b4daba5afd..4f86010a18 100644 --- a/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc +++ b/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc @@ -9,7 +9,7 @@ */ #include "webrtc/base/arraysize.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/voice_engine/include/voe_external_media.h" #include "webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h" #include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h" diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc index 780a7f7ba9..1dc15dff49 100644 --- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc +++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/system_wrappers/include/atomic32.h" #include "webrtc/system_wrappers/include/sleep.h" #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h" diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc index 94592cf616..b237414b81 100644 --- a/webrtc/voice_engine/transmit_mixer.cc +++ b/webrtc/voice_engine/transmit_mixer.cc @@ -11,7 +11,7 @@ #include "webrtc/voice_engine/transmit_mixer.h" #include "webrtc/base/format_macros.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h index 714efb48dc..071d91d77a 100644 --- a/webrtc/voice_engine/transmit_mixer.h +++ b/webrtc/voice_engine/transmit_mixer.h @@ -15,9 +15,9 @@ #include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_processing/typing_detection.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/file_player.h" -#include "webrtc/modules/utility/interface/file_recorder.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/level_indicator.h" #include "webrtc/voice_engine/monitor_module.h" diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc index 7bc7e0e963..b04163834c 100644 --- a/webrtc/voice_engine/utility.cc +++ b/webrtc/voice_engine/utility.cc @@ -13,8 +13,8 @@ #include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_types.h" -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/modules/utility/interface/audio_frame_operations.h" +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/modules/utility/include/audio_frame_operations.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc index 226e38366d..921c3e5085 100644 --- a/webrtc/voice_engine/utility_unittest.cc +++ b/webrtc/voice_engine/utility_unittest.cc @@ -13,7 +13,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/format_macros.h" #include "webrtc/common_audio/resampler/include/push_resampler.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/voice_engine/utility.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h index f0ac959dcd..5cd0acdb72 100644 --- a/webrtc/voice_engine/voe_base_impl.h +++ b/webrtc/voice_engine/voe_base_impl.h @@ -13,7 +13,7 @@ #include "webrtc/voice_engine/include/voe_base.h" -#include "webrtc/modules/interface/module_common_types.h" +#include "webrtc/modules/include/module_common_types.h" #include "webrtc/voice_engine/shared_data.h" namespace webrtc { diff --git a/webrtc/voice_engine/voe_file_impl.cc b/webrtc/voice_engine/voe_file_impl.cc index 7927f9ff05..c2d5887ce9 100644 --- a/webrtc/voice_engine/voe_file_impl.cc +++ b/webrtc/voice_engine/voe_file_impl.cc @@ -10,7 +10,7 @@ #include "webrtc/voice_engine/voe_file_impl.h" -#include "webrtc/modules/media_file/interface/media_file.h" +#include "webrtc/modules/media_file/include/media_file.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/file_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc index c8761bc38d..814d619589 100644 --- a/webrtc/voice_engine/voice_engine_impl.cc +++ b/webrtc/voice_engine/voice_engine_impl.cc @@ -12,7 +12,7 @@ #include "webrtc/modules/audio_device/android/audio_device_template.h" #include "webrtc/modules/audio_device/android/audio_record_jni.h" #include "webrtc/modules/audio_device/android/audio_track_jni.h" -#include "webrtc/modules/utility/interface/jvm_android.h" +#include "webrtc/modules/utility/include/jvm_android.h" #endif #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"