diff --git a/.gn b/.gn index 5e3c6ca9af..e1ed9ae2b8 100644 --- a/.gn +++ b/.gn @@ -42,7 +42,6 @@ check_targets = [ "//system_wrappers/*", "//test/*", "//video/*", - "//voice_engine/*", "//third_party/libyuv/*", ] diff --git a/BUILD.gn b/BUILD.gn index 4e6f01b2fe..bb5c953bc1 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -63,7 +63,6 @@ if (!build_with_chromium) { "video:screenshare_loopback", "video:sv_loopback", "video:video_loopback", - "voice_engine:voice_engine_unittests", ] if (is_android) { deps += [ @@ -342,7 +341,6 @@ if (!build_with_chromium) { "sdk", "system_wrappers:system_wrappers_default", "video", - "voice_engine", ] if (build_with_mozilla) { diff --git a/WATCHLISTS b/WATCHLISTS index ed0a614009..b6a57031b5 100644 --- a/WATCHLISTS +++ b/WATCHLISTS @@ -23,8 +23,7 @@ 'filepath': '^[^/]*$|^webrtc/[^/]*$|^webrtc/build/.*', }, 'documented_interfaces': { - 'filepath': '^webrtc/[^/]*\.h$|'\ - 'webrtc/voice_engine/include/.*', + 'filepath': '^webrtc/[^/]*\.h$', }, 'build_files': { 'filepath': '\.gyp$|\.gypi$|Android\.mk$', @@ -50,9 +49,6 @@ 'video': { 'filepath': 'webrtc/video/.*', }, - 'voice_engine': { - 'filepath': 'webrtc/voice_engine/.*', - }, 'common_audio': { 'filepath': 'webrtc/common_audio/.*', }, @@ -126,20 +122,11 @@ 'call': ['mflodman@webrtc.org', 'solenberg@webrtc.org', 'stefan@webrtc.org'], - 'media_engine': ['solenberg@webrtc.org'], 'video': ['mflodman@webrtc.org', 'stefan@webrtc.org', 'video-team@agora.io', 'yujie.mao@webrtc.org', 'zhengzhonghou@agora.io'], - 'voice_engine': ['alessiob@webrtc.org', - 'andrew@webrtc.org', - 'audio-team@agora.io', - 'henrika@webrtc.org', - 'henrik.lundin@webrtc.org', - 'minyue@webrtc.org', - 'peah@webrtc.org', - 'solenberg@webrtc.org'], 'video_capture': ['mflodman@webrtc.org', 'perkj@webrtc.org', 'sdk-team@agora.io', diff --git a/api/DEPS b/api/DEPS index a537633981..5ac116c34e 100644 --- a/api/DEPS +++ b/api/DEPS @@ -7,10 +7,6 @@ include_rules = [ ] specific_include_rules = { - "peerconnection_jni\.cc": [ - "+voice_engine", - ], - # TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h # has moved to api/. "peerconnectioninterface\.h": [ diff --git a/audio/BUILD.gn b/audio/BUILD.gn index d3af836947..2eee20be23 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -14,6 +14,8 @@ if (is_android) { rtc_static_library("audio") { sources = [ + "audio_level.cc", + "audio_level.h", "audio_receive_stream.cc", "audio_receive_stream.h", "audio_send_stream.cc", @@ -22,11 +24,19 @@ rtc_static_library("audio") { "audio_state.h", "audio_transport_impl.cc", "audio_transport_impl.h", + "channel.cc", + "channel.h", + "channel_proxy.cc", + "channel_proxy.h", "conversion.h", "null_audio_poller.cc", "null_audio_poller.h", + "remix_resample.cc", + "remix_resample.h", "time_interval.cc", "time_interval.h", + "transport_feedback_packet_loss_tracker.cc", + "transport_feedback_packet_loss_tracker.h", ] if (!build_with_chromium && is_clang) { @@ -36,15 +46,23 @@ rtc_static_library("audio") { deps = [ "..:webrtc_common", + "../api:array_view", "../api:audio_mixer_api", "../api:call_api", + "../api:libjingle_peerconnection_api", "../api:optional", + "../api:transport_api", "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs:builtin_audio_encoder_factory", "../call:bitrate_allocator", "../call:call_interfaces", "../call:rtp_interfaces", "../common_audio", + "../logging:rtc_event_log_api", + "../modules:module_api", + "../modules/audio_coding", + "../modules/audio_coding:audio_format_conversion", + "../modules/audio_coding:audio_network_adaptor_config", "../modules/audio_coding:cng", "../modules/audio_device", "../modules/audio_processing", @@ -53,14 +71,16 @@ rtc_static_library("audio") { "../modules/pacing:pacing", "../modules/remote_bitrate_estimator:remote_bitrate_estimator", "../modules/rtp_rtcp", + "../modules/rtp_rtcp:rtp_rtcp_format", + "../modules/utility", "../rtc_base:checks", + "../rtc_base:rate_limiter", "../rtc_base:rtc_base", "../rtc_base:rtc_base_approved", "../rtc_base:rtc_task_queue", "../system_wrappers", "../system_wrappers:field_trial_api", - "../voice_engine", - "../voice_engine:audio_level", + "../system_wrappers:metrics_api", "utility:audio_frame_operations", ] } @@ -94,7 +114,10 @@ if (rtc_include_tests) { "audio_send_stream_tests.cc", "audio_send_stream_unittest.cc", "audio_state_unittest.cc", + "mock_voe_channel_proxy.h", + "remix_resample_unittest.cc", "time_interval_unittest.cc", + "transport_feedback_packet_loss_tracker_unittest.cc", ] deps = [ ":audio", @@ -104,7 +127,9 @@ if (rtc_include_tests) { "../call:mock_rtp_interfaces", "../call:rtp_interfaces", "../call:rtp_receiver", + "../common_audio", "../logging:mocks", + "../modules:module_api", "../modules/audio_device:mock_audio_device", "../modules/audio_mixer:audio_mixer_impl", "../modules/audio_processing:audio_processing_statistics", @@ -116,6 +141,7 @@ if (rtc_include_tests) { "../modules/pacing:pacing", "../modules/rtp_rtcp:mock_rtp_rtcp", "../modules/rtp_rtcp:rtp_rtcp_format", + "../rtc_base:checks", "../rtc_base:rtc_base_approved", "../rtc_base:rtc_base_tests_utils", "../rtc_base:rtc_task_queue", @@ -125,7 +151,6 @@ if (rtc_include_tests) { "../test:rtp_test_utils", "../test:test_common", "../test:test_support", - "../voice_engine", "utility:utility_tests", "//testing/gmock", "//testing/gtest", diff --git a/audio/DEPS b/audio/DEPS index 70e33469df..8bb1f80805 100644 --- a/audio/DEPS +++ b/audio/DEPS @@ -5,14 +5,15 @@ include_rules = [ "+modules/audio_coding", "+modules/audio_device", "+modules/audio_mixer", + "+modules/audio_processing", "+modules/audio_processing/include", "+modules/bitrate_controller", "+modules/congestion_controller", "+modules/pacing", "+modules/remote_bitrate_estimator", "+modules/rtp_rtcp", + "+modules/utility", "+system_wrappers", - "+voice_engine", ] specific_include_rules = { diff --git a/voice_engine/audio_level.cc b/audio/audio_level.cc similarity index 98% rename from voice_engine/audio_level.cc rename to audio/audio_level.cc index 57b485546b..fe2a240a10 100644 --- a/voice_engine/audio_level.cc +++ b/audio/audio_level.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "voice_engine/audio_level.h" +#include "audio/audio_level.h" #include "common_audio/signal_processing/include/signal_processing_library.h" #include "modules/include/module_common_types.h" diff --git a/voice_engine/audio_level.h b/audio/audio_level.h similarity index 91% rename from voice_engine/audio_level.h rename to audio/audio_level.h index a1951edba9..883641ac79 100644 --- a/voice_engine/audio_level.h +++ b/audio/audio_level.h @@ -8,12 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef VOICE_ENGINE_AUDIO_LEVEL_H_ -#define VOICE_ENGINE_AUDIO_LEVEL_H_ +#ifndef AUDIO_AUDIO_LEVEL_H_ +#define AUDIO_AUDIO_LEVEL_H_ #include "rtc_base/criticalsection.h" #include "rtc_base/thread_annotations.h" -#include "typedefs.h" // NOLINT(build/include) namespace webrtc { @@ -57,4 +56,4 @@ class AudioLevel { } // namespace voe } // namespace webrtc -#endif // VOICE_ENGINE_AUDIO_LEVEL_H_ +#endif // AUDIO_AUDIO_LEVEL_H_ diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 294dad0094..45ffe34f14 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -16,6 +16,7 @@ #include "api/call/audio_sink.h" #include "audio/audio_send_stream.h" #include "audio/audio_state.h" +#include "audio/channel_proxy.h" #include "audio/conversion.h" #include "call/rtp_stream_receiver_controller_interface.h" #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" @@ -24,7 +25,6 @@ #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/timeutils.h" -#include "voice_engine/channel_proxy.h" namespace webrtc { diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc index c8318aa967..fa663fe7ab 100644 --- a/audio/audio_receive_stream_unittest.cc +++ b/audio/audio_receive_stream_unittest.cc @@ -15,6 +15,7 @@ #include "api/test/mock_audio_mixer.h" #include "audio/audio_receive_stream.h" #include "audio/conversion.h" +#include "audio/mock_voe_channel_proxy.h" #include "call/rtp_stream_receiver_controller.h" #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" #include "modules/audio_device/include/mock_audio_device.h" @@ -24,7 +25,6 @@ #include "modules/rtp_rtcp/source/byte_io.h" #include "test/gtest.h" #include "test/mock_audio_decoder_factory.h" -#include "test/mock_voe_channel_proxy.h" namespace webrtc { namespace test { diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc index d207dbd578..80c2c6b435 100644 --- a/audio/audio_send_stream.cc +++ b/audio/audio_send_stream.cc @@ -15,6 +15,7 @@ #include #include "audio/audio_state.h" +#include "audio/channel_proxy.h" #include "audio/conversion.h" #include "call/rtp_transport_controller_send_interface.h" #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h" @@ -28,7 +29,6 @@ #include "rtc_base/task_queue.h" #include "rtc_base/timeutils.h" #include "system_wrappers/include/field_trial.h" -#include "voice_engine/channel_proxy.h" namespace webrtc { namespace internal { diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h index bab8f80de5..093ca4621a 100644 --- a/audio/audio_send_stream.h +++ b/audio/audio_send_stream.h @@ -15,6 +15,7 @@ #include #include "audio/time_interval.h" +#include "audio/transport_feedback_packet_loss_tracker.h" #include "call/audio_send_stream.h" #include "call/audio_state.h" #include "call/bitrate_allocator.h" @@ -22,7 +23,6 @@ #include "rtc_base/constructormagic.h" #include "rtc_base/race_checker.h" #include "rtc_base/thread_checker.h" -#include "voice_engine/transport_feedback_packet_loss_tracker.h" namespace webrtc { class RtcEventLog; diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index 4644bf471e..925f93e3cf 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -15,6 +15,7 @@ #include "audio/audio_send_stream.h" #include "audio/audio_state.h" #include "audio/conversion.h" +#include "audio/mock_voe_channel_proxy.h" #include "call/fake_rtp_transport_controller_send.h" #include "call/rtp_transport_controller_send_interface.h" #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" @@ -32,7 +33,6 @@ #include "test/gtest.h" #include "test/mock_audio_encoder.h" #include "test/mock_audio_encoder_factory.h" -#include "test/mock_voe_channel_proxy.h" namespace webrtc { namespace test { diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc index 30ffc6de3c..f9b0311ccb 100644 --- a/audio/audio_transport_impl.cc +++ b/audio/audio_transport_impl.cc @@ -14,10 +14,10 @@ #include #include +#include "audio/remix_resample.h" #include "audio/utility/audio_frame_operations.h" #include "call/audio_send_stream.h" #include "rtc_base/logging.h" -#include "voice_engine/utility.h" namespace webrtc { diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h index 8a316a586c..4e6e0470f1 100644 --- a/audio/audio_transport_impl.h +++ b/audio/audio_transport_impl.h @@ -14,6 +14,7 @@ #include #include "api/audio/audio_mixer.h" +#include "audio/audio_level.h" #include "common_audio/resampler/include/push_resampler.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_processing/include/audio_processing.h" @@ -22,7 +23,6 @@ #include "rtc_base/criticalsection.h" #include "rtc_base/scoped_ref_ptr.h" #include "rtc_base/thread_annotations.h" -#include "voice_engine/audio_level.h" namespace webrtc { diff --git a/voice_engine/channel.cc b/audio/channel.cc similarity index 99% rename from voice_engine/channel.cc rename to audio/channel.cc index efc76bdcab..1799e7aae1 100644 --- a/voice_engine/channel.cc +++ b/audio/channel.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "voice_engine/channel.h" +#include "audio/channel.h" #include #include @@ -365,10 +365,7 @@ bool Channel::SendRtp(const uint8_t* data, return false; } - uint8_t* bufferToSendPtr = (uint8_t*)data; - size_t bufferLength = len; - - if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) { + if (!_transportPtr->SendRtp(data, len, options)) { RTC_LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed"; return false; } @@ -383,10 +380,7 @@ bool Channel::SendRtcp(const uint8_t* data, size_t len) { return false; } - uint8_t* bufferToSendPtr = (uint8_t*)data; - size_t bufferLength = len; - - int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength); + int n = _transportPtr->SendRtcp(data, len); if (n < 0) { RTC_LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed"; return false; diff --git a/voice_engine/channel.h b/audio/channel.h similarity index 98% rename from voice_engine/channel.h rename to audio/channel.h index 3d6dd8f226..c5d243cac4 100644 --- a/voice_engine/channel.h +++ b/audio/channel.h @@ -8,16 +8,20 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef VOICE_ENGINE_CHANNEL_H_ -#define VOICE_ENGINE_CHANNEL_H_ +#ifndef AUDIO_CHANNEL_H_ +#define AUDIO_CHANNEL_H_ +#include #include +#include +#include #include "api/audio/audio_mixer.h" #include "api/audio_codecs/audio_encoder.h" #include "api/call/audio_sink.h" #include "api/call/transport.h" #include "api/optional.h" +#include "audio/audio_level.h" #include "common_types.h" // NOLINT(build/include) #include "modules/audio_coding/include/audio_coding_module.h" #include "modules/audio_processing/rms_level.h" @@ -29,7 +33,6 @@ #include "rtc_base/event.h" #include "rtc_base/task_queue.h" #include "rtc_base/thread_checker.h" -#include "voice_engine/audio_level.h" namespace rtc { class TimestampWrapAroundHandler; @@ -410,4 +413,4 @@ class Channel } // namespace voe } // namespace webrtc -#endif // VOICE_ENGINE_CHANNEL_H_ +#endif // AUDIO_CHANNEL_H_ diff --git a/voice_engine/channel_proxy.cc b/audio/channel_proxy.cc similarity index 99% rename from voice_engine/channel_proxy.cc rename to audio/channel_proxy.cc index 70564d4c91..cdfe0902b5 100644 --- a/voice_engine/channel_proxy.cc +++ b/audio/channel_proxy.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "voice_engine/channel_proxy.h" +#include "audio/channel_proxy.h" #include diff --git a/voice_engine/channel_proxy.h b/audio/channel_proxy.h similarity index 97% rename from voice_engine/channel_proxy.h rename to audio/channel_proxy.h index ef81174c0e..f5f603ba75 100644 --- a/voice_engine/channel_proxy.h +++ b/audio/channel_proxy.h @@ -8,9 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef VOICE_ENGINE_CHANNEL_PROXY_H_ -#define VOICE_ENGINE_CHANNEL_PROXY_H_ +#ifndef AUDIO_CHANNEL_PROXY_H_ +#define AUDIO_CHANNEL_PROXY_H_ +#include #include #include #include @@ -18,11 +19,11 @@ #include "api/audio/audio_mixer.h" #include "api/audio_codecs/audio_encoder.h" #include "api/rtpreceiverinterface.h" +#include "audio/channel.h" #include "call/rtp_packet_sink_interface.h" #include "rtc_base/constructormagic.h" #include "rtc_base/race_checker.h" #include "rtc_base/thread_checker.h" -#include "voice_engine/channel.h" namespace webrtc { @@ -141,4 +142,4 @@ class ChannelProxy : public RtpPacketSinkInterface { } // namespace voe } // namespace webrtc -#endif // VOICE_ENGINE_CHANNEL_PROXY_H_ +#endif // AUDIO_CHANNEL_PROXY_H_ diff --git a/test/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h similarity index 95% rename from test/mock_voe_channel_proxy.h rename to audio/mock_voe_channel_proxy.h index 29635bac1c..d512ad45e0 100644 --- a/test/mock_voe_channel_proxy.h +++ b/audio/mock_voe_channel_proxy.h @@ -8,14 +8,17 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef TEST_MOCK_VOE_CHANNEL_PROXY_H_ -#define TEST_MOCK_VOE_CHANNEL_PROXY_H_ +#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ +#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ +#include +#include #include +#include +#include "audio/channel_proxy.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "test/gmock.h" -#include "voice_engine/channel_proxy.h" namespace webrtc { namespace test { @@ -101,4 +104,4 @@ class MockVoEChannelProxy : public voe::ChannelProxy { } // namespace test } // namespace webrtc -#endif // TEST_MOCK_VOE_CHANNEL_PROXY_H_ +#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_ diff --git a/voice_engine/utility.cc b/audio/remix_resample.cc similarity index 99% rename from voice_engine/utility.cc rename to audio/remix_resample.cc index 939870245f..52a491fdd9 100644 --- a/voice_engine/utility.cc +++ b/audio/remix_resample.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "voice_engine/utility.h" +#include "audio/remix_resample.h" #include "audio/utility/audio_frame_operations.h" #include "common_audio/resampler/include/push_resampler.h" diff --git a/voice_engine/utility.h b/audio/remix_resample.h similarity index 87% rename from voice_engine/utility.h rename to audio/remix_resample.h index dc23e1667d..ddd8086957 100644 --- a/voice_engine/utility.h +++ b/audio/remix_resample.h @@ -8,15 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -/* - * Contains functions often used by different parts of VoiceEngine. - */ - -#ifndef VOICE_ENGINE_UTILITY_H_ -#define VOICE_ENGINE_UTILITY_H_ +#ifndef AUDIO_REMIX_RESAMPLE_H_ +#define AUDIO_REMIX_RESAMPLE_H_ #include "common_audio/resampler/include/push_resampler.h" -#include "typedefs.h" // NOLINT(build/include) namespace webrtc { @@ -48,4 +43,4 @@ void RemixAndResample(const int16_t* src_data, } // namespace voe } // namespace webrtc -#endif // VOICE_ENGINE_UTILITY_H_ +#endif // AUDIO_REMIX_RESAMPLE_H_ diff --git a/voice_engine/utility_unittest.cc b/audio/remix_resample_unittest.cc similarity index 99% rename from voice_engine/utility_unittest.cc rename to audio/remix_resample_unittest.cc index c798582d4b..753584b46d 100644 --- a/voice_engine/utility_unittest.cc +++ b/audio/remix_resample_unittest.cc @@ -10,12 +10,12 @@ #include +#include "audio/remix_resample.h" #include "common_audio/resampler/include/push_resampler.h" #include "modules/include/module_common_types.h" #include "rtc_base/arraysize.h" #include "rtc_base/format_macros.h" #include "test/gtest.h" -#include "voice_engine/utility.h" namespace webrtc { namespace voe { diff --git a/voice_engine/transport_feedback_packet_loss_tracker.cc b/audio/transport_feedback_packet_loss_tracker.cc similarity index 99% rename from voice_engine/transport_feedback_packet_loss_tracker.cc rename to audio/transport_feedback_packet_loss_tracker.cc index 774faf5219..101b6b4881 100644 --- a/voice_engine/transport_feedback_packet_loss_tracker.cc +++ b/audio/transport_feedback_packet_loss_tracker.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "voice_engine/transport_feedback_packet_loss_tracker.h" +#include "audio/transport_feedback_packet_loss_tracker.h" #include #include diff --git a/voice_engine/transport_feedback_packet_loss_tracker.h b/audio/transport_feedback_packet_loss_tracker.h similarity index 96% rename from voice_engine/transport_feedback_packet_loss_tracker.h rename to audio/transport_feedback_packet_loss_tracker.h index d7420785f7..7e73210327 100644 --- a/voice_engine/transport_feedback_packet_loss_tracker.h +++ b/audio/transport_feedback_packet_loss_tracker.h @@ -8,10 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_ -#define VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_ +#ifndef AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_ +#define AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_ #include +#include #include "api/optional.h" #include "modules/include/module_common_types.h" @@ -138,4 +139,4 @@ class TransportFeedbackPacketLossTracker final { } // namespace webrtc -#endif // VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_ +#endif // AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_ diff --git a/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc b/audio/transport_feedback_packet_loss_tracker_unittest.cc similarity index 99% rename from voice_engine/transport_feedback_packet_loss_tracker_unittest.cc rename to audio/transport_feedback_packet_loss_tracker_unittest.cc index 55626bedd0..8f8fe05d7a 100644 --- a/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc +++ b/audio/transport_feedback_packet_loss_tracker_unittest.cc @@ -13,12 +13,12 @@ #include #include +#include "audio/transport_feedback_packet_loss_tracker.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" #include "rtc_base/checks.h" #include "test/gmock.h" #include "test/gtest.h" -#include "voice_engine/transport_feedback_packet_loss_tracker.h" namespace webrtc { diff --git a/call/BUILD.gn b/call/BUILD.gn index d7ed4d95a0..03ed9b417d 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -271,7 +271,6 @@ if (rtc_include_tests) { "../test:test_support", "../test:video_test_common", "../video", - "../voice_engine", "//testing/gtest", ] if (!build_with_chromium && is_clang) { diff --git a/call/DEPS b/call/DEPS index 7622e24116..307a26e3b0 100644 --- a/call/DEPS +++ b/call/DEPS @@ -11,7 +11,6 @@ include_rules = [ "+modules/rtp_rtcp", "+modules/utility", "+system_wrappers", - "+voice_engine", "+video", ] diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index f8d787cb50..4667498434 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -98,9 +98,6 @@ class AudioReceiveStream { Transport* rtcp_send_transport = nullptr; - // TODO(solenberg): Remove once clients don't use it anymore. - int voe_channel_id = -1; - // NetEq settings. size_t jitter_buffer_max_packets = 50; bool jitter_buffer_fast_accelerate = false; diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h index 03f32b79e6..908da11f6d 100644 --- a/call/audio_send_stream.h +++ b/call/audio_send_stream.h @@ -87,9 +87,6 @@ class AudioSendStream { // the entire life of the AudioSendStream and is owned by the API client. Transport* send_transport = nullptr; - // TODO(solenberg): Remove once clients don't use it anymore. - int voe_channel_id = -1; - // Bitrate limits used for variable audio bitrate streams. Set both to -1 to // disable audio bitrate adaptation. // Note: This is still an experimental feature and not ready for real usage. diff --git a/call/audio_state.h b/call/audio_state.h index a8e57f0f3f..9c40187068 100644 --- a/call/audio_state.h +++ b/call/audio_state.h @@ -19,16 +19,12 @@ namespace webrtc { class AudioDeviceModule; class AudioProcessing; class AudioTransport; -class VoiceEngine; // AudioState holds the state which must be shared between multiple instances of // webrtc::Call for audio processing purposes. class AudioState : public rtc::RefCountInterface { public: struct Config { - // TODO(solenberg): Remove once clients don't use it anymore. - VoiceEngine* voice_engine = nullptr; - // The audio mixer connected to active receive streams. One per // AudioState. rtc::scoped_refptr audio_mixer; diff --git a/media/BUILD.gn b/media/BUILD.gn index b7ea5af833..bfd91ce718 100644 --- a/media/BUILD.gn +++ b/media/BUILD.gn @@ -317,7 +317,6 @@ rtc_static_library("rtc_audio_video") { "../system_wrappers", "../system_wrappers:field_trial_api", "../system_wrappers:metrics_api", - "../voice_engine", ] } @@ -616,7 +615,6 @@ if (rtc_include_tests) { "../test:audio_codec_mocks", "../test:test_support", "../test:video_test_common", - "../voice_engine:voice_engine", ] } } diff --git a/media/DEPS b/media/DEPS index 7a266a23fa..99e62aab08 100644 --- a/media/DEPS +++ b/media/DEPS @@ -14,7 +14,6 @@ include_rules = [ "+pc", "+sound", "+system_wrappers", - "+voice_engine", "+usrsctplib", "+third_party/libyuv", ] diff --git a/modules/audio_mixer/DEPS b/modules/audio_mixer/DEPS index 51608ad47d..de2271a6b5 100644 --- a/modules/audio_mixer/DEPS +++ b/modules/audio_mixer/DEPS @@ -10,5 +10,4 @@ include_rules = [ "+modules/rtp_rtcp", "+modules/utility", "+system_wrappers", - "+voice_engine", ] diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn index 6bca26dfca..2fe3cc2bf5 100644 --- a/modules/remote_bitrate_estimator/BUILD.gn +++ b/modules/remote_bitrate_estimator/BUILD.gn @@ -155,7 +155,6 @@ if (rtc_include_tests) { "../../system_wrappers:field_trial_api", "../../test:perf_test", "../../test:test_support", - "../../voice_engine", "../bitrate_controller", "../congestion_controller", "../congestion_controller:delay_based_bwe", diff --git a/modules/remote_bitrate_estimator/DEPS b/modules/remote_bitrate_estimator/DEPS index d6a2fbafc9..8499c25613 100644 --- a/modules/remote_bitrate_estimator/DEPS +++ b/modules/remote_bitrate_estimator/DEPS @@ -2,9 +2,3 @@ include_rules = [ "+logging/rtc_event_log", "+system_wrappers", ] - -specific_include_rules = { - "nada\.h": [ - "+voice_engine", - ], -} diff --git a/modules/remote_bitrate_estimator/test/estimators/nada.h b/modules/remote_bitrate_estimator/test/estimators/nada.h index 590175c784..f00a6d3690 100644 --- a/modules/remote_bitrate_estimator/test/estimators/nada.h +++ b/modules/remote_bitrate_estimator/test/estimators/nada.h @@ -24,7 +24,6 @@ #include "modules/include/module_common_types.h" #include "modules/remote_bitrate_estimator/test/bwe.h" #include "rtc_base/constructormagic.h" -#include "voice_engine/channel.h" namespace webrtc { diff --git a/pc/DEPS b/pc/DEPS index 1992af013e..24cb242575 100644 --- a/pc/DEPS +++ b/pc/DEPS @@ -18,7 +18,6 @@ include_rules = [ specific_include_rules = { "androidtestinitializer\.cc": [ "+base/android", # Allowed only for Android tests. - "+voice_engine", ], "srtpfilter_unittest\.cc": [ "+crypto", diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn index b57f48ddb8..fa4a8f9df2 100644 --- a/sdk/android/BUILD.gn +++ b/sdk/android/BUILD.gn @@ -111,7 +111,6 @@ rtc_static_library("audio_jni") { "../../api/audio_codecs:builtin_audio_encoder_factory", "../../modules/audio_processing:audio_processing", "../../rtc_base:rtc_base_approved", - "../../voice_engine:voice_engine", ] } diff --git a/sdk/android/src/jni/DEPS b/sdk/android/src/jni/DEPS index db95db649f..1cf4ba1916 100644 --- a/sdk/android/src/jni/DEPS +++ b/sdk/android/src/jni/DEPS @@ -12,5 +12,4 @@ include_rules = [ "+modules/video_coding", "+pc", "+system_wrappers/include", - "+voice_engine/include/voe_base.h", ] diff --git a/test/BUILD.gn b/test/BUILD.gn index c058b5c023..a83e8bb216 100644 --- a/test/BUILD.gn +++ b/test/BUILD.gn @@ -567,7 +567,6 @@ rtc_source_set("test_common") { "layer_filtering_transport.cc", "layer_filtering_transport.h", "mock_transport.h", - "mock_voe_channel_proxy.h", "null_transport.cc", "null_transport.h", "rtp_rtcp_observer.h", @@ -628,7 +627,6 @@ rtc_source_set("test_common") { "../system_wrappers", "../system_wrappers:field_trial_api", "../video", - "../voice_engine", "//testing/gmock", "//testing/gtest", ] diff --git a/test/DEPS b/test/DEPS index 1cf56a8484..71d3d25764 100644 --- a/test/DEPS +++ b/test/DEPS @@ -16,7 +16,6 @@ include_rules = [ "+modules/video_coding", "+sdk", "+system_wrappers", - "+voice_engine", "+third_party/libyuv", ] diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn index 43df61899a..0ccabfc179 100644 --- a/test/fuzzers/BUILD.gn +++ b/test/fuzzers/BUILD.gn @@ -416,9 +416,9 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") { ] deps = [ "../../api:array_view", + "../../audio", "../../modules/rtp_rtcp:rtp_rtcp_format", "../../rtc_base:rtc_base_approved", - "../../voice_engine", ] } rtc_static_library("audio_processing_fuzzer_helper") { diff --git a/test/fuzzers/DEPS b/test/fuzzers/DEPS index cdbb5660e9..222cd0202e 100644 --- a/test/fuzzers/DEPS +++ b/test/fuzzers/DEPS @@ -1,3 +1,3 @@ include_rules = [ - "+webrtc", + "+audio", ] diff --git a/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc b/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc index 31bc1639e2..2168d219f9 100644 --- a/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc +++ b/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc @@ -11,10 +11,10 @@ #include #include "api/array_view.h" +#include "audio/transport_feedback_packet_loss_tracker.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" -#include "voice_engine/transport_feedback_packet_loss_tracker.h" namespace webrtc { diff --git a/video/BUILD.gn b/video/BUILD.gn index 31ad45cbd0..85026918ab 100644 --- a/video/BUILD.gn +++ b/video/BUILD.gn @@ -94,7 +94,6 @@ rtc_static_library("video") { "../rtc_base:sequenced_task_checker", "../rtc_base:weak_ptr", "../system_wrappers", - "../voice_engine", ] if (!build_with_mozilla) { @@ -130,7 +129,6 @@ if (rtc_include_tests) { "../test:test_support_test_artifacts", "../test:video_test_common", "../test:video_test_support", - "../voice_engine", ] if (!build_with_chromium && is_clang) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). diff --git a/video/DEPS b/video/DEPS index b94455edd0..288ecfdfe1 100644 --- a/video/DEPS +++ b/video/DEPS @@ -15,5 +15,4 @@ include_rules = [ "+modules/video_capture", "+modules/video_processing", "+system_wrappers", - "+voice_engine", ] diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn deleted file mode 100644 index d0de9c64af..0000000000 --- a/voice_engine/BUILD.gn +++ /dev/null @@ -1,148 +0,0 @@ -# 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. - -import("../webrtc.gni") - -visibility = [ ":*" ] - -rtc_static_library("voice_engine") { - visibility += [ "*" ] - sources = [ - "channel.cc", - "channel.h", - "channel_proxy.cc", - "channel_proxy.h", - "transport_feedback_packet_loss_tracker.cc", - "transport_feedback_packet_loss_tracker.h", - "utility.cc", - "utility.h", - ] - - if (is_win) { - cflags = [ - # TODO(kjellander): Bug 261: fix this warning. - "/wd4373", # Virtual function override. - ] - } - - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - - deps = [ - ":audio_level", - "..:webrtc_common", - "../:typedefs", - "../api:array_view", - "../api:audio_mixer_api", - "../api:call_api", - "../api:libjingle_peerconnection_api", - "../api:optional", - "../api:refcountedbase", - "../api:transport_api", - "../api/audio_codecs:audio_codecs_api", - "../audio/utility:audio_frame_operations", - "../call:rtp_interfaces", - "../common_audio", - "../logging:rtc_event_log_api", - "../modules:module_api", - "../modules/audio_coding", - "../modules/audio_coding:audio_format_conversion", - "../modules/audio_coding:audio_network_adaptor_config", - "../modules/audio_device", - "../modules/audio_processing", - "../modules/bitrate_controller", - "../modules/media_file", - "../modules/pacing", - "../modules/rtp_rtcp", - "../modules/rtp_rtcp:rtp_rtcp_format", - "../modules/utility", - "../rtc_base:checks", - "../rtc_base:rate_limiter", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", - "../system_wrappers", - "../system_wrappers:field_trial_api", - "../system_wrappers:metrics_api", - ] -} - -rtc_static_library("audio_level") { - visibility += [ - ":voice_engine", - "../audio:audio", - ] - sources = [ - "audio_level.cc", - "audio_level.h", - ] - - deps = [ - "..:webrtc_common", - "../:typedefs", - "../common_audio", - "../modules:module_api", - "../rtc_base:rtc_base_approved", - ] -} - -if (rtc_include_tests) { - rtc_test("voice_engine_unittests") { - visibility += webrtc_default_visibility - deps = [ - ":voice_engine", - "../api/audio_codecs:builtin_audio_decoder_factory", - "../common_audio", - "../modules:module_api", - "../modules/audio_coding", - "../modules/audio_device", - "../modules/audio_processing", - "../modules/media_file", - "../modules/rtp_rtcp:rtp_rtcp_format", - "../modules/utility", - "../modules/video_capture:video_capture", - "../rtc_base:checks", - "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_base_tests_utils", - "../system_wrappers", - "../test:test_common", - "../test:test_main", - "../test:video_test_common", - "//testing/gmock", - "//testing/gtest", - ] - - if (is_android) { - deps += [ "//testing/android/native_test:native_test_native_code" ] - shard_timeout = 900 - } - - sources = [ - "transport_feedback_packet_loss_tracker_unittest.cc", - "utility_unittest.cc", - ] - - data = [ - "../resources/utility/encapsulated_pcm16b_8khz.wav", - "../resources/utility/encapsulated_pcmu_8khz.wav", - ] - - if (is_win) { - cflags = [ - # TODO(kjellander): Bug 261: fix this warning. - "/wd4373", # Virtual function override. - ] - } - - if (!build_with_chromium && is_clang) { - # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). - suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] - } - } -} diff --git a/voice_engine/DEPS b/voice_engine/DEPS deleted file mode 100644 index c8e9a1cac6..0000000000 --- a/voice_engine/DEPS +++ /dev/null @@ -1,14 +0,0 @@ -include_rules = [ - "+audio/utility/audio_frame_operations.h", - "+call", - "+common_audio", - "+logging/rtc_event_log", - "+modules/audio_coding", - "+modules/audio_device", - "+modules/audio_processing", - "+modules/media_file", - "+modules/pacing", - "+modules/rtp_rtcp", - "+modules/utility", - "+system_wrappers", -] diff --git a/voice_engine/OWNERS b/voice_engine/OWNERS deleted file mode 100644 index 0430ede769..0000000000 --- a/voice_engine/OWNERS +++ /dev/null @@ -1,10 +0,0 @@ -henrikg@webrtc.org -henrika@webrtc.org -niklas.enbom@webrtc.org -solenberg@webrtc.org - - -# These are for the common case of adding or renaming files. If you're doing -# structural changes, please get a review from a reviewer in this file. -per-file *.gn=* -per-file *.gni=*