From 3e6db2321ccdc8738c9cecbe9bdab13d4f0f658d Mon Sep 17 00:00:00 2001 From: kjellander Date: Thu, 26 Nov 2015 04:44:54 -0800 Subject: [PATCH] audio_coding: remove "main" directory This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803} --- webrtc/call/call_perf_tests.cc | 2 +- webrtc/modules/audio_coding/BUILD.gn | 42 +- .../{main => }/acm2/acm_codec_database.cc | 4 +- .../{main => }/acm2/acm_codec_database.h | 8 +- .../{main => }/acm2/acm_common_defs.h | 6 +- .../{main => }/acm2/acm_neteq_unittest.cc | 0 .../acm2/acm_receive_test_oldapi.cc | 4 +- .../{main => }/acm2/acm_receive_test_oldapi.h | 6 +- .../{main => }/acm2/acm_receiver.cc | 6 +- .../{main => }/acm2/acm_receiver.h | 14 +- .../acm2/acm_receiver_unittest_oldapi.cc | 6 +- .../{main => }/acm2/acm_resampler.cc | 2 +- .../{main => }/acm2/acm_resampler.h | 6 +- .../{main => }/acm2/acm_send_test_oldapi.cc | 4 +- .../{main => }/acm2/acm_send_test_oldapi.h | 8 +- .../{main => }/acm2/audio_coding_module.cc | 6 +- .../acm2/audio_coding_module_impl.cc | 10 +- .../acm2/audio_coding_module_impl.h | 12 +- .../audio_coding_module_unittest_oldapi.cc | 8 +- .../{main => }/acm2/call_statistics.cc | 2 +- .../{main => }/acm2/call_statistics.h | 6 +- .../acm2/call_statistics_unittest.cc | 2 +- .../{main => }/acm2/codec_manager.cc | 4 +- .../{main => }/acm2/codec_manager.h | 10 +- .../{main => }/acm2/codec_manager_unittest.cc | 2 +- .../{main => }/acm2/initial_delay_manager.cc | 2 +- .../{main => }/acm2/initial_delay_manager.h | 6 +- .../acm2/initial_delay_manager_unittest.cc | 2 +- .../{main => }/acm2/rent_a_codec.cc | 6 +- .../{main => }/acm2/rent_a_codec.h | 8 +- .../{main => }/acm2/rent_a_codec_unittest.cc | 2 +- webrtc/modules/audio_coding/audio_coding.gypi | 185 ++++- .../include/audio_coding_module.h | 741 ++++++++++++++++++ .../include/audio_coding_module_typedefs.h | 51 ++ webrtc/modules/audio_coding/main/acm2/OWNERS | 5 - .../main/audio_coding_module.gypi | 196 ----- .../main/include/audio_coding_module.h | 10 +- .../include/audio_coding_module_typedefs.h | 8 +- .../audio_coding/neteq/audio_decoder_impl.h | 2 +- webrtc/modules/audio_coding/neteq/nack.h | 2 +- .../audio_coding/neteq/nack_unittest.cc | 2 +- .../audio_coding/{main => }/test/ACMTest.h | 6 +- .../audio_coding/{main => }/test/APITest.cc | 6 +- .../audio_coding/{main => }/test/APITest.h | 16 +- .../audio_coding/{main => }/test/Channel.cc | 2 +- .../audio_coding/{main => }/test/Channel.h | 8 +- .../{main => }/test/EncodeDecodeTest.cc | 8 +- .../{main => }/test/EncodeDecodeTest.h | 14 +- .../audio_coding/{main => }/test/PCMFile.cc | 0 .../audio_coding/{main => }/test/PCMFile.h | 6 +- .../{main => }/test/PacketLossTest.cc | 2 +- .../{main => }/test/PacketLossTest.h | 8 +- .../audio_coding/{main => }/test/RTPFile.cc | 0 .../audio_coding/{main => }/test/RTPFile.h | 8 +- .../{main => }/test/SpatialAudio.cc | 2 +- .../{main => }/test/SpatialAudio.h | 16 +- .../{main => }/test/TestAllCodecs.cc | 8 +- .../{main => }/test/TestAllCodecs.h | 12 +- .../{main => }/test/TestRedFec.cc | 6 +- .../audio_coding/{main => }/test/TestRedFec.h | 12 +- .../{main => }/test/TestStereo.cc | 6 +- .../audio_coding/{main => }/test/TestStereo.h | 12 +- .../{main => }/test/TestVADDTX.cc | 6 +- .../audio_coding/{main => }/test/TestVADDTX.h | 14 +- .../audio_coding/{main => }/test/Tester.cc | 22 +- .../{main => }/test/TimedTrace.cc | 0 .../audio_coding/{main => }/test/TimedTrace.h | 6 +- .../{main => }/test/TwoWayCommunication.cc | 4 +- .../{main => }/test/TwoWayCommunication.h | 16 +- .../{main => }/test/delay_test.cc | 12 +- .../audio_coding/{main => }/test/iSACTest.cc | 6 +- .../audio_coding/{main => }/test/iSACTest.h | 16 +- .../test/insert_packet_with_timing.cc | 6 +- .../audio_coding/{main => }/test/opus_test.cc | 8 +- .../audio_coding/{main => }/test/opus_test.h | 16 +- .../{main => }/test/target_delay_unittest.cc | 4 +- .../audio_coding/{main => }/test/utility.cc | 4 +- .../audio_coding/{main => }/test/utility.h | 8 +- webrtc/modules/modules.gyp | 48 +- webrtc/modules/utility/source/coder.h | 2 +- webrtc/voice_engine/channel.h | 2 +- webrtc/voice_engine/voe_base_impl.cc | 2 +- webrtc/voice_engine/voe_codec_impl.cc | 2 +- webrtc/voice_engine/voe_neteq_stats_impl.cc | 2 +- webrtc/voice_engine/voice_engine_impl.cc | 2 +- 85 files changed, 1276 insertions(+), 498 deletions(-) rename webrtc/modules/audio_coding/{main => }/acm2/acm_codec_database.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_codec_database.h (91%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_common_defs.h (82%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_neteq_unittest.cc (100%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_receive_test_oldapi.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_receive_test_oldapi.h (92%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_receiver.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_receiver.h (95%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_receiver_unittest_oldapi.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_resampler.cc (96%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_resampler.h (83%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_send_test_oldapi.cc (97%) rename webrtc/modules/audio_coding/{main => }/acm2/acm_send_test_oldapi.h (91%) rename webrtc/modules/audio_coding/{main => }/acm2/audio_coding_module.cc (92%) rename webrtc/modules/audio_coding/{main => }/acm2/audio_coding_module_impl.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/audio_coding_module_impl.h (95%) rename webrtc/modules/audio_coding/{main => }/acm2/audio_coding_module_unittest_oldapi.cc (99%) rename webrtc/modules/audio_coding/{main => }/acm2/call_statistics.cc (95%) rename webrtc/modules/audio_coding/{main => }/acm2/call_statistics.h (91%) rename webrtc/modules/audio_coding/{main => }/acm2/call_statistics_unittest.cc (95%) rename webrtc/modules/audio_coding/{main => }/acm2/codec_manager.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/codec_manager.h (86%) rename webrtc/modules/audio_coding/{main => }/acm2/codec_manager_unittest.cc (97%) rename webrtc/modules/audio_coding/{main => }/acm2/initial_delay_manager.cc (99%) rename webrtc/modules/audio_coding/{main => }/acm2/initial_delay_manager.h (95%) rename webrtc/modules/audio_coding/{main => }/acm2/initial_delay_manager_unittest.cc (99%) rename webrtc/modules/audio_coding/{main => }/acm2/rent_a_codec.cc (98%) rename webrtc/modules/audio_coding/{main => }/acm2/rent_a_codec.h (96%) rename webrtc/modules/audio_coding/{main => }/acm2/rent_a_codec_unittest.cc (99%) create mode 100644 webrtc/modules/audio_coding/include/audio_coding_module.h create mode 100644 webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h delete mode 100644 webrtc/modules/audio_coding/main/acm2/OWNERS delete mode 100644 webrtc/modules/audio_coding/main/audio_coding_module.gypi rename webrtc/modules/audio_coding/{main => }/test/ACMTest.h (74%) rename webrtc/modules/audio_coding/{main => }/test/APITest.cc (99%) rename webrtc/modules/audio_coding/{main => }/test/APITest.h (87%) rename webrtc/modules/audio_coding/{main => }/test/Channel.cc (99%) rename webrtc/modules/audio_coding/{main => }/test/Channel.h (93%) rename webrtc/modules/audio_coding/{main => }/test/EncodeDecodeTest.cc (97%) rename webrtc/modules/audio_coding/{main => }/test/EncodeDecodeTest.h (86%) rename webrtc/modules/audio_coding/{main => }/test/PCMFile.cc (100%) rename webrtc/modules/audio_coding/{main => }/test/PCMFile.h (90%) rename webrtc/modules/audio_coding/{main => }/test/PacketLossTest.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/PacketLossTest.h (86%) rename webrtc/modules/audio_coding/{main => }/test/RTPFile.cc (100%) rename webrtc/modules/audio_coding/{main => }/test/RTPFile.h (92%) rename webrtc/modules/audio_coding/{main => }/test/SpatialAudio.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/SpatialAudio.h (66%) rename webrtc/modules/audio_coding/{main => }/test/TestAllCodecs.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/TestAllCodecs.h (85%) rename webrtc/modules/audio_coding/{main => }/test/TestRedFec.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/TestRedFec.h (78%) rename webrtc/modules/audio_coding/{main => }/test/TestStereo.cc (99%) rename webrtc/modules/audio_coding/{main => }/test/TestStereo.h (88%) rename webrtc/modules/audio_coding/{main => }/test/TestVADDTX.cc (97%) rename webrtc/modules/audio_coding/{main => }/test/TestVADDTX.h (85%) rename webrtc/modules/audio_coding/{main => }/test/Tester.cc (87%) rename webrtc/modules/audio_coding/{main => }/test/TimedTrace.cc (100%) rename webrtc/modules/audio_coding/{main => }/test/TimedTrace.h (82%) rename webrtc/modules/audio_coding/{main => }/test/TwoWayCommunication.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/TwoWayCommunication.h (69%) rename webrtc/modules/audio_coding/{main => }/test/delay_test.cc (95%) rename webrtc/modules/audio_coding/{main => }/test/iSACTest.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/iSACTest.h (76%) rename webrtc/modules/audio_coding/{main => }/test/insert_packet_with_timing.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/opus_test.cc (97%) rename webrtc/modules/audio_coding/{main => }/test/opus_test.h (72%) rename webrtc/modules/audio_coding/{main => }/test/target_delay_unittest.cc (98%) rename webrtc/modules/audio_coding/{main => }/test/utility.cc (97%) rename webrtc/modules/audio_coding/{main => }/test/utility.h (94%) diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index 69f090a63f..44ecae325e 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -18,7 +18,7 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/call.h" #include "webrtc/call/transport_adapter.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.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" diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 533e848cdc..382ae51576 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -11,10 +11,10 @@ import("../../build/webrtc.gni") source_set("rent_a_codec") { sources = [ - "main/acm2/acm_codec_database.cc", - "main/acm2/acm_codec_database.h", - "main/acm2/rent_a_codec.cc", - "main/acm2/rent_a_codec.h", + "acm2/acm_codec_database.cc", + "acm2/acm_codec_database.h", + "acm2/rent_a_codec.cc", + "acm2/rent_a_codec.h", ] configs += [ "../..:common_config" ] public_configs = [ "../..:common_inherited_config" ] @@ -44,29 +44,29 @@ source_set("rent_a_codec") { config("audio_coding_config") { include_dirs = [ - "main/include", + "include", "../include", ] } source_set("audio_coding") { sources = [ - "main/acm2/acm_common_defs.h", - "main/acm2/acm_receiver.cc", - "main/acm2/acm_receiver.h", - "main/acm2/acm_resampler.cc", - "main/acm2/acm_resampler.h", - "main/acm2/audio_coding_module.cc", - "main/acm2/audio_coding_module_impl.cc", - "main/acm2/audio_coding_module_impl.h", - "main/acm2/call_statistics.cc", - "main/acm2/call_statistics.h", - "main/acm2/codec_manager.cc", - "main/acm2/codec_manager.h", - "main/acm2/initial_delay_manager.cc", - "main/acm2/initial_delay_manager.h", - "main/include/audio_coding_module.h", - "main/include/audio_coding_module_typedefs.h", + "acm2/acm_common_defs.h", + "acm2/acm_receiver.cc", + "acm2/acm_receiver.h", + "acm2/acm_resampler.cc", + "acm2/acm_resampler.h", + "acm2/audio_coding_module.cc", + "acm2/audio_coding_module_impl.cc", + "acm2/audio_coding_module_impl.h", + "acm2/call_statistics.cc", + "acm2/call_statistics.h", + "acm2/codec_manager.cc", + "acm2/codec_manager.h", + "acm2/initial_delay_manager.cc", + "acm2/initial_delay_manager.h", + "include/audio_coding_module.h", + "include/audio_coding_module_typedefs.h", ] defines = [] diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc rename to webrtc/modules/audio_coding/acm2/acm_codec_database.cc index b54fc0bd48..8d4072fae4 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc +++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.cc @@ -15,12 +15,12 @@ // TODO(tlegrand): Change constant input pointers in all functions to constant // references, where appropriate. -#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" +#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h" #include #include "webrtc/base/checks.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" #include "webrtc/system_wrappers/include/trace.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h b/webrtc/modules/audio_coding/acm2/acm_codec_database.h similarity index 91% rename from webrtc/modules/audio_coding/main/acm2/acm_codec_database.h rename to webrtc/modules/audio_coding/acm2/acm_codec_database.h index f9adda0988..9e87238474 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_codec_database.h +++ b/webrtc/modules/audio_coding/acm2/acm_codec_database.h @@ -13,12 +13,12 @@ * codecs. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_ #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" namespace webrtc { @@ -80,4 +80,4 @@ class ACMCodecDB { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_CODEC_DATABASE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/acm2/acm_common_defs.h similarity index 82% rename from webrtc/modules/audio_coding/main/acm2/acm_common_defs.h rename to webrtc/modules/audio_coding/acm2/acm_common_defs.h index 23e3519ed0..483bdd93f1 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h +++ b/webrtc/modules/audio_coding/acm2/acm_common_defs.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_ #include "webrtc/engine_configurations.h" @@ -29,4 +29,4 @@ const int kIsacPacSize960 = 960; } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_COMMON_DEFS_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc b/webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc similarity index 100% rename from webrtc/modules/audio_coding/main/acm2/acm_neteq_unittest.cc rename to webrtc/modules/audio_coding/acm2/acm_neteq_unittest.cc diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc rename to webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc index fdcfdfc22d..bb83e77931 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc +++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h" +#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h" #include #include #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h similarity index 92% rename from webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h rename to webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h index 0b5671fe8c..091513db46 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h +++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_ #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" @@ -91,4 +91,4 @@ class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi { } // namespace test } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVE_TEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/acm_receiver.cc rename to webrtc/modules/audio_coding/acm2/acm_receiver.cc index 6c2893336a..036877ce11 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" +#include "webrtc/modules/audio_coding/acm2/acm_receiver.h" #include // malloc @@ -21,8 +21,8 @@ #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/acm2/call_statistics.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h similarity index 95% rename from webrtc/modules/audio_coding/main/acm2/acm_receiver.h rename to webrtc/modules/audio_coding/acm2/acm_receiver.h index bcedacd14f..d5a644d5c8 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h +++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ #include #include @@ -20,10 +20,10 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/common_audio/vad/include/webrtc_vad.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" -#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/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/acm2/call_statistics.h" +#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" @@ -302,4 +302,4 @@ class AcmReceiver { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc rename to webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc index 8f43ac456a..8076687c14 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc +++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" +#include "webrtc/modules/audio_coding/acm2/acm_receiver.h" #include // std::min #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h" #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/test/test_suite.h" diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc b/webrtc/modules/audio_coding/acm2/acm_resampler.cc similarity index 96% rename from webrtc/modules/audio_coding/main/acm2/acm_resampler.cc rename to webrtc/modules/audio_coding/acm2/acm_resampler.cc index cbcad85f5b..e38cd94a97 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc +++ b/webrtc/modules/audio_coding/acm2/acm_resampler.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" #include #include diff --git a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h b/webrtc/modules/audio_coding/acm2/acm_resampler.h similarity index 83% rename from webrtc/modules/audio_coding/main/acm2/acm_resampler.h rename to webrtc/modules/audio_coding/acm2/acm_resampler.h index a19b0c4569..700fefa274 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_resampler.h +++ b/webrtc/modules/audio_coding/acm2/acm_resampler.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_ #include "webrtc/common_audio/resampler/include/push_resampler.h" #include "webrtc/typedefs.h" @@ -36,4 +36,4 @@ class ACMResampler { } // namespace acm2 } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RESAMPLER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc similarity index 97% rename from webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc rename to webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc index ac38dc011d..3a89a77487 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc +++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h" +#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h" #include #include @@ -17,7 +17,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/checks.h" #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h" #include "webrtc/modules/audio_coding/neteq/tools/packet.h" diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h similarity index 91% rename from webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h rename to webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h index 3e65ec6c2d..ce68196a3f 100644 --- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h +++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_ #include #include "webrtc/base/constructormagic.h" #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" #include "webrtc/system_wrappers/include/clock.h" @@ -88,4 +88,4 @@ class AcmSendTestOldApi : public AudioPacketizationCallback, } // namespace test } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc similarity index 92% rename from webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc rename to webrtc/modules/audio_coding/acm2/audio_coding_module.cc index 889d62092a..034de32cb6 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/base/checks.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/system_wrappers/include/clock.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc rename to webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc index 5d18bda00c..5f61ef695b 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h" +#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h" #include #include @@ -17,10 +17,10 @@ #include "webrtc/base/checks.h" #include "webrtc/base/safe_conversions.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/acm2/call_statistics.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/system_wrappers/include/metrics.h" diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h similarity index 95% rename from webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h rename to webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h index c04ccf9c2f..6006c68f5c 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ #include @@ -18,9 +18,9 @@ #include "webrtc/base/thread_annotations.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h" +#include "webrtc/modules/audio_coding/acm2/acm_receiver.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/acm2/codec_manager.h" namespace webrtc { @@ -277,4 +277,4 @@ class AudioCodingModuleImpl final : public AudioCodingModule { } // namespace acm2 } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc similarity index 99% rename from webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc rename to webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc index f14dcf3dea..39c14a880c 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc +++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc @@ -21,10 +21,10 @@ #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h" #include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h" #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h" +#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" #include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h" #include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h" diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.cc b/webrtc/modules/audio_coding/acm2/call_statistics.cc similarity index 95% rename from webrtc/modules/audio_coding/main/acm2/call_statistics.cc rename to webrtc/modules/audio_coding/acm2/call_statistics.cc index 4c3e9fc393..4441932c8c 100644 --- a/webrtc/modules/audio_coding/main/acm2/call_statistics.cc +++ b/webrtc/modules/audio_coding/acm2/call_statistics.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" +#include "webrtc/modules/audio_coding/acm2/call_statistics.h" #include diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/webrtc/modules/audio_coding/acm2/call_statistics.h similarity index 91% rename from webrtc/modules/audio_coding/main/acm2/call_statistics.h rename to webrtc/modules/audio_coding/acm2/call_statistics.h index e2df9210ff..888afea0a7 100644 --- a/webrtc/modules/audio_coding/main/acm2/call_statistics.h +++ b/webrtc/modules/audio_coding/acm2/call_statistics.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_ #include "webrtc/common_types.h" #include "webrtc/modules/include/module_common_types.h" @@ -60,4 +60,4 @@ class CallStatistics { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc b/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc similarity index 95% rename from webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc rename to webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc index 2bee96465d..9ba0774ce1 100644 --- a/webrtc/modules/audio_coding/main/acm2/call_statistics_unittest.cc +++ b/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc @@ -9,7 +9,7 @@ */ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h" +#include "webrtc/modules/audio_coding/acm2/call_statistics.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc b/webrtc/modules/audio_coding/acm2/codec_manager.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/codec_manager.cc rename to webrtc/modules/audio_coding/acm2/codec_manager.cc index 7796786866..a5a9e09793 100644 --- a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc +++ b/webrtc/modules/audio_coding/acm2/codec_manager.cc @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h" +#include "webrtc/modules/audio_coding/acm2/codec_manager.h" #include "webrtc/base/checks.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/system_wrappers/include/trace.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.h b/webrtc/modules/audio_coding/acm2/codec_manager.h similarity index 86% rename from webrtc/modules/audio_coding/main/acm2/codec_manager.h rename to webrtc/modules/audio_coding/acm2/codec_manager.h index 7670bbd2de..61832e4f10 100644 --- a/webrtc/modules/audio_coding/main/acm2/codec_manager.h +++ b/webrtc/modules/audio_coding/acm2/codec_manager.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_ #include @@ -17,8 +17,8 @@ #include "webrtc/base/optional.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/base/thread_checker.h" -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/common_types.h" namespace webrtc { @@ -78,4 +78,4 @@ class CodecManager final { } // namespace acm2 } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CODEC_MANAGER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc similarity index 97% rename from webrtc/modules/audio_coding/main/acm2/codec_manager_unittest.cc rename to webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc index e930ca1ea5..c09f256298 100644 --- a/webrtc/modules/audio_coding/main/acm2/codec_manager_unittest.cc +++ b/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc @@ -10,7 +10,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h" -#include "webrtc/modules/audio_coding/main/acm2/codec_manager.h" +#include "webrtc/modules/audio_coding/acm2/codec_manager.h" namespace webrtc { namespace acm2 { diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc similarity index 99% rename from webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc rename to webrtc/modules/audio_coding/acm2/initial_delay_manager.cc index 786fb2e527..0c31b83eb3 100644 --- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc +++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" +#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h similarity index 95% rename from webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h rename to webrtc/modules/audio_coding/acm2/initial_delay_manager.h index 6b50dd07f8..32dd1260f1 100644 --- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h +++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/include/module_common_types.h" @@ -117,4 +117,4 @@ class InitialDelayManager { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc similarity index 99% rename from webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc rename to webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc index e973593eb4..d86d221851 100644 --- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager_unittest.cc +++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc @@ -11,7 +11,7 @@ #include #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h" +#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc similarity index 98% rename from webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc rename to webrtc/modules/audio_coding/acm2/rent_a_codec.cc index 229d367f61..480024934c 100644 --- a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.cc +++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/base/logging.h" #include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h" @@ -34,8 +34,8 @@ #ifdef WEBRTC_CODEC_RED #include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h" #endif -#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" namespace webrtc { namespace acm2 { diff --git a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.h b/webrtc/modules/audio_coding/acm2/rent_a_codec.h similarity index 96% rename from webrtc/modules/audio_coding/main/acm2/rent_a_codec.h rename to webrtc/modules/audio_coding/acm2/rent_a_codec.h index 45d46bb057..7035104dd4 100644 --- a/webrtc/modules/audio_coding/main/acm2/rent_a_codec.h +++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_ #include #include @@ -20,7 +20,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_coding/codecs/audio_decoder.h" #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/typedefs.h" #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) @@ -246,4 +246,4 @@ class RentACodec { } // namespace acm2 } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_RENT_A_CODEC_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/rent_a_codec_unittest.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc similarity index 99% rename from webrtc/modules/audio_coding/main/acm2/rent_a_codec_unittest.cc rename to webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc index ae6c98b34c..11c4bcb292 100644 --- a/webrtc/modules/audio_coding/main/acm2/rent_a_codec_unittest.cc +++ b/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc @@ -11,7 +11,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/arraysize.h" #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h" -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" namespace webrtc { namespace acm2 { diff --git a/webrtc/modules/audio_coding/audio_coding.gypi b/webrtc/modules/audio_coding/audio_coding.gypi index bc3c48d075..abdb1915c3 100644 --- a/webrtc/modules/audio_coding/audio_coding.gypi +++ b/webrtc/modules/audio_coding/audio_coding.gypi @@ -19,12 +19,195 @@ 'codecs/isac/isacfix.gypi', 'codecs/pcm16b/pcm16b.gypi', 'codecs/red/red.gypi', - 'main/audio_coding_module.gypi', 'neteq/neteq.gypi', ], + 'variables': { + 'audio_coding_dependencies': [ + 'cng', + 'g711', + 'pcm16b', + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + ], + 'audio_coding_defines': [], + 'conditions': [ + ['include_opus==1', { + 'audio_coding_dependencies': ['webrtc_opus',], + 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',], + }], + ['build_with_mozilla==0', { + 'conditions': [ + ['target_arch=="arm"', { + 'audio_coding_dependencies': ['isac_fix',], + 'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',], + }, { + 'audio_coding_dependencies': ['isac',], + 'audio_coding_defines': ['WEBRTC_CODEC_ISAC',], + }], + ], + 'audio_coding_dependencies': ['g722',], + 'audio_coding_defines': ['WEBRTC_CODEC_G722',], + }], + ['build_with_mozilla==0 and build_with_chromium==0', { + 'audio_coding_dependencies': ['ilbc', 'red',], + 'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',], + }], + ], + }, + 'targets': [ + { + 'target_name': 'rent_a_codec', + 'type': 'static_library', + 'defines': [ + '<@(audio_coding_defines)', + ], + 'dependencies': [ + '<(webrtc_root)/common.gyp:webrtc_common', + ], + 'include_dirs': [ + '<(webrtc_root)', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + '<(webrtc_root)', + ], + }, + 'sources': [ + 'acm2/acm_codec_database.cc', + 'acm2/acm_codec_database.h', + 'acm2/rent_a_codec.cc', + 'acm2/rent_a_codec.h', + ], + }, + { + 'target_name': 'audio_coding_module', + 'type': 'static_library', + 'defines': [ + '<@(audio_coding_defines)', + ], + 'dependencies': [ + '<@(audio_coding_dependencies)', + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/webrtc.gyp:rtc_event_log', + 'neteq', + 'rent_a_codec', + ], + 'include_dirs': [ + 'include', + '../include', + '<(webrtc_root)', + ], + 'direct_dependent_settings': { + 'include_dirs': [ + 'include', + '../include', + '<(webrtc_root)', + ], + }, + 'conditions': [ + ['include_opus==1', { + 'export_dependent_settings': ['webrtc_opus'], + }], + ], + 'sources': [ + 'acm2/acm_common_defs.h', + 'acm2/acm_receiver.cc', + 'acm2/acm_receiver.h', + 'acm2/acm_resampler.cc', + 'acm2/acm_resampler.h', + 'acm2/audio_coding_module.cc', + 'acm2/audio_coding_module_impl.cc', + 'acm2/audio_coding_module_impl.h', + 'acm2/call_statistics.cc', + 'acm2/call_statistics.h', + 'acm2/codec_manager.cc', + 'acm2/codec_manager.h', + 'acm2/initial_delay_manager.cc', + 'acm2/initial_delay_manager.h', + 'include/audio_coding_module.h', + 'include/audio_coding_module_typedefs.h', + ], + }, + ], 'conditions': [ ['include_opus==1', { 'includes': ['codecs/opus/opus.gypi',], }], + ['include_tests==1', { + 'targets': [ + { + 'target_name': 'acm_receive_test', + 'type': 'static_library', + 'defines': [ + '<@(audio_coding_defines)', + ], + 'dependencies': [ + '<@(audio_coding_dependencies)', + 'audio_coding_module', + 'neteq_unittest_tools', + '<(DEPTH)/testing/gtest.gyp:gtest', + ], + 'sources': [ + 'acm2/acm_receive_test_oldapi.cc', + 'acm2/acm_receive_test_oldapi.h', + ], + }, # acm_receive_test + { + 'target_name': 'acm_send_test', + 'type': 'static_library', + 'defines': [ + '<@(audio_coding_defines)', + ], + 'dependencies': [ + '<@(audio_coding_dependencies)', + 'audio_coding_module', + 'neteq_unittest_tools', + '<(DEPTH)/testing/gtest.gyp:gtest', + ], + 'sources': [ + 'acm2/acm_send_test_oldapi.cc', + 'acm2/acm_send_test_oldapi.h', + ], + }, # acm_send_test + { + 'target_name': 'delay_test', + 'type': 'executable', + 'dependencies': [ + 'audio_coding_module', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/test/test.gyp:test_support', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + ], + 'sources': [ + 'test/delay_test.cc', + 'test/Channel.cc', + 'test/PCMFile.cc', + 'test/utility.cc', + ], + }, # delay_test + { + 'target_name': 'insert_packet_with_timing', + 'type': 'executable', + 'dependencies': [ + 'audio_coding_module', + '<(DEPTH)/testing/gtest.gyp:gtest', + '<(webrtc_root)/common.gyp:webrtc_common', + '<(webrtc_root)/test/test.gyp:test_support', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', + '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', + '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', + ], + 'sources': [ + 'test/insert_packet_with_timing.cc', + 'test/Channel.cc', + 'test/PCMFile.cc', + ], + }, # delay_test + ], + }], ], } diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h new file mode 100644 index 0000000000..844bd57cd1 --- /dev/null +++ b/webrtc/modules/audio_coding/include/audio_coding_module.h @@ -0,0 +1,741 @@ +/* + * 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_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ + +#include + +#include "webrtc/base/optional.h" +#include "webrtc/common_types.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/neteq/include/neteq.h" +#include "webrtc/modules/include/module.h" +#include "webrtc/system_wrappers/include/clock.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +// forward declarations +struct CodecInst; +struct WebRtcRTPHeader; +class AudioDecoder; +class AudioEncoder; +class AudioFrame; +class RTPFragmentationHeader; + +#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz + +// Callback class used for sending data ready to be packetized +class AudioPacketizationCallback { + public: + virtual ~AudioPacketizationCallback() {} + + virtual int32_t SendData(FrameType frame_type, + uint8_t payload_type, + uint32_t timestamp, + const uint8_t* payload_data, + size_t payload_len_bytes, + const RTPFragmentationHeader* fragmentation) = 0; +}; + +// Callback class used for reporting VAD decision +class ACMVADCallback { + public: + virtual ~ACMVADCallback() {} + + virtual int32_t InFrameType(FrameType frame_type) = 0; +}; + +class AudioCodingModule { + protected: + AudioCodingModule() {} + + public: + struct Config { + Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) { + // Post-decode VAD is disabled by default in NetEq, however, Audio + // Conference Mixer relies on VAD decisions and fails without them. + neteq_config.enable_post_decode_vad = true; + } + + int id; + NetEq::Config neteq_config; + Clock* clock; + }; + + /////////////////////////////////////////////////////////////////////////// + // Creation and destruction of a ACM. + // + // The second method is used for testing where a simulated clock can be + // injected into ACM. ACM will take the ownership of the object clock and + // delete it when destroyed. + // + static AudioCodingModule* Create(int id); + static AudioCodingModule* Create(int id, Clock* clock); + static AudioCodingModule* Create(const Config& config); + virtual ~AudioCodingModule() = default; + + /////////////////////////////////////////////////////////////////////////// + // Utility functions + // + + /////////////////////////////////////////////////////////////////////////// + // uint8_t NumberOfCodecs() + // Returns number of supported codecs. + // + // Return value: + // number of supported codecs. + /// + static int NumberOfCodecs(); + + /////////////////////////////////////////////////////////////////////////// + // int32_t Codec() + // Get supported codec with list number. + // + // Input: + // -list_id : list number. + // + // Output: + // -codec : a structure where the parameters of the codec, + // given by list number is written to. + // + // Return value: + // -1 if the list number (list_id) is invalid. + // 0 if succeeded. + // + static int Codec(int list_id, CodecInst* codec); + + /////////////////////////////////////////////////////////////////////////// + // int32_t Codec() + // Get supported codec with the given codec name, sampling frequency, and + // a given number of channels. + // + // Input: + // -payload_name : name of the codec. + // -sampling_freq_hz : sampling frequency of the codec. Note! for RED + // a sampling frequency of -1 is a valid input. + // -channels : number of channels ( 1 - mono, 2 - stereo). + // + // Output: + // -codec : a structure where the function returns the + // default parameters of the codec. + // + // Return value: + // -1 if no codec matches the given parameters. + // 0 if succeeded. + // + static int Codec(const char* payload_name, CodecInst* codec, + int sampling_freq_hz, int channels); + + /////////////////////////////////////////////////////////////////////////// + // int32_t Codec() + // + // Returns the list number of the given codec name, sampling frequency, and + // a given number of channels. + // + // Input: + // -payload_name : name of the codec. + // -sampling_freq_hz : sampling frequency of the codec. Note! for RED + // a sampling frequency of -1 is a valid input. + // -channels : number of channels ( 1 - mono, 2 - stereo). + // + // Return value: + // if the codec is found, the index of the codec in the list, + // -1 if the codec is not found. + // + static int Codec(const char* payload_name, int sampling_freq_hz, + int channels); + + /////////////////////////////////////////////////////////////////////////// + // bool IsCodecValid() + // Checks the validity of the parameters of the given codec. + // + // Input: + // -codec : the structure which keeps the parameters of the + // codec. + // + // Return value: + // true if the parameters are valid, + // false if any parameter is not valid. + // + static bool IsCodecValid(const CodecInst& codec); + + /////////////////////////////////////////////////////////////////////////// + // Sender + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t RegisterSendCodec() + // Registers a codec, specified by |send_codec|, as sending codec. + // This API can be called multiple of times to register Codec. The last codec + // registered overwrites the previous ones. + // The API can also be used to change payload type for CNG and RED, which are + // registered by default to default payload types. + // Note that registering CNG and RED won't overwrite speech codecs. + // This API can be called to set/change the send payload-type, frame-size + // or encoding rate (if applicable for the codec). + // + // Note: If a stereo codec is registered as send codec, VAD/DTX will + // automatically be turned off, since it is not supported for stereo sending. + // + // Note: If a secondary encoder is already registered, and the new send-codec + // has a sampling rate that does not match the secondary encoder, the + // secondary encoder will be unregistered. + // + // Input: + // -send_codec : Parameters of the codec to be registered, c.f. + // common_types.h for the definition of + // CodecInst. + // + // Return value: + // -1 if failed to initialize, + // 0 if succeeded. + // + virtual int32_t RegisterSendCodec(const CodecInst& send_codec) = 0; + + // Registers |external_speech_encoder| as encoder. The new encoder will + // replace any previously registered speech encoder (internal or external). + virtual void RegisterExternalSendCodec( + AudioEncoder* external_speech_encoder) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t SendCodec() + // Get parameters for the codec currently registered as send codec. + // + // Return value: + // The send codec, or nothing if we don't have one + // + virtual rtc::Optional SendCodec() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t SendFrequency() + // Get the sampling frequency of the current encoder in Hertz. + // + // Return value: + // positive; sampling frequency [Hz] of the current encoder. + // -1 if an error has happened. + // + virtual int32_t SendFrequency() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // Sets the bitrate to the specified value in bits/sec. If the value is not + // supported by the codec, it will choose another appropriate value. + virtual void SetBitRate(int bitrate_bps) = 0; + + // int32_t RegisterTransportCallback() + // Register a transport callback which will be called to deliver + // the encoded buffers whenever Process() is called and a + // bit-stream is ready. + // + // Input: + // -transport : pointer to the callback class + // transport->SendData() is called whenever + // Process() is called and bit-stream is ready + // to deliver. + // + // Return value: + // -1 if the transport callback could not be registered + // 0 if registration is successful. + // + virtual int32_t RegisterTransportCallback( + AudioPacketizationCallback* transport) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t Add10MsData() + // Add 10MS of raw (PCM) audio data and encode it. If the sampling + // frequency of the audio does not match the sampling frequency of the + // current encoder ACM will resample the audio. If an encoded packet was + // produced, it will be delivered via the callback object registered using + // RegisterTransportCallback, and the return value from this function will + // be the number of bytes encoded. + // + // Input: + // -audio_frame : the input audio frame, containing raw audio + // sampling frequency etc., + // c.f. module_common_types.h for definition of + // AudioFrame. + // + // Return value: + // >= 0 number of bytes encoded. + // -1 some error occurred. + // + virtual int32_t Add10MsData(const AudioFrame& audio_frame) = 0; + + /////////////////////////////////////////////////////////////////////////// + // (RED) Redundant Coding + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t SetREDStatus() + // configure RED status i.e. on/off. + // + // RFC 2198 describes a solution which has a single payload type which + // signifies a packet with redundancy. That packet then becomes a container, + // encapsulating multiple payloads into a single RTP packet. + // Such a scheme is flexible, since any amount of redundancy may be + // encapsulated within a single packet. There is, however, a small overhead + // since each encapsulated payload must be preceded by a header indicating + // the type of data enclosed. + // + // Input: + // -enable_red : if true RED is enabled, otherwise RED is + // disabled. + // + // Return value: + // -1 if failed to set RED status, + // 0 if succeeded. + // + virtual int32_t SetREDStatus(bool enable_red) = 0; + + /////////////////////////////////////////////////////////////////////////// + // bool REDStatus() + // Get RED status + // + // Return value: + // true if RED is enabled, + // false if RED is disabled. + // + virtual bool REDStatus() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // (FEC) Forward Error Correction (codec internal) + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t SetCodecFEC() + // Configures codec internal FEC status i.e. on/off. No effects on codecs that + // do not provide internal FEC. + // + // Input: + // -enable_fec : if true FEC will be enabled otherwise the FEC is + // disabled. + // + // Return value: + // -1 if failed, or the codec does not support FEC + // 0 if succeeded. + // + virtual int SetCodecFEC(bool enable_codec_fec) = 0; + + /////////////////////////////////////////////////////////////////////////// + // bool CodecFEC() + // Gets status of codec internal FEC. + // + // Return value: + // true if FEC is enabled, + // false if FEC is disabled. + // + virtual bool CodecFEC() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int SetPacketLossRate() + // Sets expected packet loss rate for encoding. Some encoders provide packet + // loss gnostic encoding to make stream less sensitive to packet losses, + // through e.g., FEC. No effects on codecs that do not provide such encoding. + // + // Input: + // -packet_loss_rate : expected packet loss rate (0 -- 100 inclusive). + // + // Return value + // -1 if failed to set packet loss rate, + // 0 if succeeded. + // + virtual int SetPacketLossRate(int packet_loss_rate) = 0; + + /////////////////////////////////////////////////////////////////////////// + // (VAD) Voice Activity Detection + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t SetVAD() + // If DTX is enabled & the codec does not have internal DTX/VAD + // WebRtc VAD will be automatically enabled and |enable_vad| is ignored. + // + // If DTX is disabled but VAD is enabled no DTX packets are send, + // regardless of whether the codec has internal DTX/VAD or not. In this + // case, WebRtc VAD is running to label frames as active/in-active. + // + // NOTE! VAD/DTX is not supported when sending stereo. + // + // Inputs: + // -enable_dtx : if true DTX is enabled, + // otherwise DTX is disabled. + // -enable_vad : if true VAD is enabled, + // otherwise VAD is disabled. + // -vad_mode : determines the aggressiveness of VAD. A more + // aggressive mode results in more frames labeled + // as in-active, c.f. definition of + // ACMVADMode in audio_coding_module_typedefs.h + // for valid values. + // + // Return value: + // -1 if failed to set up VAD/DTX, + // 0 if succeeded. + // + virtual int32_t SetVAD(const bool enable_dtx = true, + const bool enable_vad = false, + const ACMVADMode vad_mode = VADNormal) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t VAD() + // Get VAD status. + // + // Outputs: + // -dtx_enabled : is set to true if DTX is enabled, otherwise + // is set to false. + // -vad_enabled : is set to true if VAD is enabled, otherwise + // is set to false. + // -vad_mode : is set to the current aggressiveness of VAD. + // + // Return value: + // -1 if fails to retrieve the setting of DTX/VAD, + // 0 if succeeded. + // + virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled, + ACMVADMode* vad_mode) const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t RegisterVADCallback() + // Call this method to register a callback function which is called + // any time that ACM encounters an empty frame. That is a frame which is + // recognized inactive. Depending on the codec WebRtc VAD or internal codec + // VAD is employed to identify a frame as active/inactive. + // + // Input: + // -vad_callback : pointer to a callback function. + // + // Return value: + // -1 if failed to register the callback function. + // 0 if the callback function is registered successfully. + // + virtual int32_t RegisterVADCallback(ACMVADCallback* vad_callback) = 0; + + /////////////////////////////////////////////////////////////////////////// + // Receiver + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t InitializeReceiver() + // Any decoder-related state of ACM will be initialized to the + // same state when ACM is created. This will not interrupt or + // effect encoding functionality of ACM. ACM would lose all the + // decoding-related settings by calling this function. + // For instance, all registered codecs are deleted and have to be + // registered again. + // + // Return value: + // -1 if failed to initialize, + // 0 if succeeded. + // + virtual int32_t InitializeReceiver() = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t ReceiveFrequency() + // Get sampling frequency of the last received payload. + // + // Return value: + // non-negative the sampling frequency in Hertz. + // -1 if an error has occurred. + // + virtual int32_t ReceiveFrequency() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t PlayoutFrequency() + // Get sampling frequency of audio played out. + // + // Return value: + // the sampling frequency in Hertz. + // + virtual int32_t PlayoutFrequency() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t RegisterReceiveCodec() + // Register possible decoders, can be called multiple times for + // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED. + // + // Input: + // -receive_codec : parameters of the codec to be registered, c.f. + // common_types.h for the definition of + // CodecInst. + // + // Return value: + // -1 if failed to register the codec + // 0 if the codec registered successfully. + // + virtual int RegisterReceiveCodec(const CodecInst& receive_codec) = 0; + + virtual int RegisterExternalReceiveCodec(int rtp_payload_type, + AudioDecoder* external_decoder, + int sample_rate_hz, + int num_channels) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t UnregisterReceiveCodec() + // Unregister the codec currently registered with a specific payload type + // from the list of possible receive codecs. + // + // Input: + // -payload_type : The number representing the payload type to + // unregister. + // + // Output: + // -1 if fails to unregister. + // 0 if the given codec is successfully unregistered. + // + virtual int UnregisterReceiveCodec( + uint8_t payload_type) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t ReceiveCodec() + // Get the codec associated with last received payload. + // + // Output: + // -curr_receive_codec : parameters of the codec associated with the last + // received payload, c.f. common_types.h for + // the definition of CodecInst. + // + // Return value: + // -1 if failed to retrieve the codec, + // 0 if the codec is successfully retrieved. + // + virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t IncomingPacket() + // Call this function to insert a parsed RTP packet into ACM. + // + // Inputs: + // -incoming_payload : received payload. + // -payload_len_bytes : the length of payload in bytes. + // -rtp_info : the relevant information retrieved from RTP + // header. + // + // Return value: + // -1 if failed to push in the payload + // 0 if payload is successfully pushed in. + // + virtual int32_t IncomingPacket(const uint8_t* incoming_payload, + const size_t payload_len_bytes, + const WebRtcRTPHeader& rtp_info) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t IncomingPayload() + // Call this API to push incoming payloads when there is no rtp-info. + // The rtp-info will be created in ACM. One usage for this API is when + // pre-encoded files are pushed in ACM + // + // Inputs: + // -incoming_payload : received payload. + // -payload_len_byte : the length, in bytes, of the received payload. + // -payload_type : the payload-type. This specifies which codec has + // to be used to decode the payload. + // -timestamp : send timestamp of the payload. ACM starts with + // a random value and increment it by the + // packet-size, which is given when the codec in + // question is registered by RegisterReceiveCodec(). + // Therefore, it is essential to have the timestamp + // if the frame-size differ from the registered + // value or if the incoming payload contains DTX + // packets. + // + // Return value: + // -1 if failed to push in the payload + // 0 if payload is successfully pushed in. + // + virtual int32_t IncomingPayload(const uint8_t* incoming_payload, + const size_t payload_len_byte, + const uint8_t payload_type, + const uint32_t timestamp = 0) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int SetMinimumPlayoutDelay() + // Set a minimum for the playout delay, used for lip-sync. NetEq maintains + // such a delay unless channel condition yields to a higher delay. + // + // Input: + // -time_ms : minimum delay in milliseconds. + // + // Return value: + // -1 if failed to set the delay, + // 0 if the minimum delay is set. + // + virtual int SetMinimumPlayoutDelay(int time_ms) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int SetMaximumPlayoutDelay() + // Set a maximum for the playout delay + // + // Input: + // -time_ms : maximum delay in milliseconds. + // + // Return value: + // -1 if failed to set the delay, + // 0 if the maximum delay is set. + // + virtual int SetMaximumPlayoutDelay(int time_ms) = 0; + + // + // The shortest latency, in milliseconds, required by jitter buffer. This + // is computed based on inter-arrival times and playout mode of NetEq. The + // actual delay is the maximum of least-required-delay and the minimum-delay + // specified by SetMinumumPlayoutDelay() API. + // + virtual int LeastRequiredDelayMs() const = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t PlayoutTimestamp() + // The send timestamp of an RTP packet is associated with the decoded + // audio of the packet in question. This function returns the timestamp of + // the latest audio obtained by calling PlayoutData10ms(). + // + // Input: + // -timestamp : a reference to a uint32_t to receive the + // timestamp. + // Return value: + // 0 if the output is a correct timestamp. + // -1 if failed to output the correct timestamp. + // + // TODO(tlegrand): Change function to return the timestamp. + virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int32_t PlayoutData10Ms( + // Get 10 milliseconds of raw audio data for playout, at the given sampling + // frequency. ACM will perform a resampling if required. + // + // Input: + // -desired_freq_hz : the desired sampling frequency, in Hertz, of the + // output audio. If set to -1, the function returns + // the audio at the current sampling frequency. + // + // Output: + // -audio_frame : output audio frame which contains raw audio data + // and other relevant parameters, c.f. + // module_common_types.h for the definition of + // AudioFrame. + // + // Return value: + // -1 if the function fails, + // 0 if the function succeeds. + // + virtual int32_t PlayoutData10Ms(int32_t desired_freq_hz, + AudioFrame* audio_frame) = 0; + + /////////////////////////////////////////////////////////////////////////// + // Codec specific + // + + /////////////////////////////////////////////////////////////////////////// + // int SetOpusApplication() + // Sets the intended application if current send codec is Opus. Opus uses this + // to optimize the encoding for applications like VOIP and music. Currently, + // two modes are supported: kVoip and kAudio. + // + // Input: + // - application : intended application. + // + // Return value: + // -1 if current send codec is not Opus or error occurred in setting the + // Opus application mode. + // 0 if the Opus application mode is successfully set. + // + virtual int SetOpusApplication(OpusApplicationMode application) = 0; + + /////////////////////////////////////////////////////////////////////////// + // int SetOpusMaxPlaybackRate() + // If current send codec is Opus, informs it about maximum playback rate the + // receiver will render. Opus can use this information to optimize the bit + // rate and increase the computation efficiency. + // + // Input: + // -frequency_hz : maximum playback rate in Hz. + // + // Return value: + // -1 if current send codec is not Opus or + // error occurred in setting the maximum playback rate, + // 0 if maximum bandwidth is set successfully. + // + virtual int SetOpusMaxPlaybackRate(int frequency_hz) = 0; + + /////////////////////////////////////////////////////////////////////////// + // EnableOpusDtx() + // Enable the DTX, if current send codec is Opus. + // + // Return value: + // -1 if current send codec is not Opus or error occurred in enabling the + // Opus DTX. + // 0 if Opus DTX is enabled successfully. + // + virtual int EnableOpusDtx() = 0; + + /////////////////////////////////////////////////////////////////////////// + // int DisableOpusDtx() + // If current send codec is Opus, disables its internal DTX. + // + // Return value: + // -1 if current send codec is not Opus or error occurred in disabling DTX. + // 0 if Opus DTX is disabled successfully. + // + virtual int DisableOpusDtx() = 0; + + /////////////////////////////////////////////////////////////////////////// + // statistics + // + + /////////////////////////////////////////////////////////////////////////// + // int32_t GetNetworkStatistics() + // Get network statistics. Note that the internal statistics of NetEq are + // reset by this call. + // + // Input: + // -network_statistics : a structure that contains network statistics. + // + // Return value: + // -1 if failed to set the network statistics, + // 0 if statistics are set successfully. + // + virtual int32_t GetNetworkStatistics( + NetworkStatistics* network_statistics) = 0; + + // + // Enable NACK and set the maximum size of the NACK list. If NACK is already + // enable then the maximum NACK list size is modified accordingly. + // + // If the sequence number of last received packet is N, the sequence numbers + // of NACK list are in the range of [N - |max_nack_list_size|, N). + // + // |max_nack_list_size| should be positive (none zero) and less than or + // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1 + // is returned. 0 is returned at success. + // + virtual int EnableNack(size_t max_nack_list_size) = 0; + + // Disable NACK. + virtual void DisableNack() = 0; + + // + // Get a list of packets to be retransmitted. |round_trip_time_ms| is an + // estimate of the round-trip-time (in milliseconds). Missing packets which + // will be playout in a shorter time than the round-trip-time (with respect + // to the time this API is called) will not be included in the list. + // + // Negative |round_trip_time_ms| results is an error message and empty list + // is returned. + // + virtual std::vector GetNackList( + int64_t round_trip_time_ms) const = 0; + + virtual void GetDecodingCallStatistics( + AudioDecodingCallStats* call_stats) const = 0; +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ diff --git a/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h b/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h new file mode 100644 index 0000000000..280d6bffa2 --- /dev/null +++ b/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.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_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ + +#include + +#include "webrtc/modules/include/module_common_types.h" +#include "webrtc/typedefs.h" + +namespace webrtc { + +/////////////////////////////////////////////////////////////////////////// +// enum ACMVADMode +// An enumerator for aggressiveness of VAD +// -VADNormal : least aggressive mode. +// -VADLowBitrate : more aggressive than "VADNormal" to save on +// bit-rate. +// -VADAggr : an aggressive mode. +// -VADVeryAggr : the most agressive mode. +// +enum ACMVADMode { + VADNormal = 0, + VADLowBitrate = 1, + VADAggr = 2, + VADVeryAggr = 3 +}; + +/////////////////////////////////////////////////////////////////////////// +// +// Enumeration of Opus mode for intended application. +// +// kVoip : optimized for voice signals. +// kAudio : optimized for non-voice signals like music. +// +enum OpusApplicationMode { + kVoip = 0, + kAudio = 1, +}; + +} // namespace webrtc + +#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ diff --git a/webrtc/modules/audio_coding/main/acm2/OWNERS b/webrtc/modules/audio_coding/main/acm2/OWNERS deleted file mode 100644 index 3ee6b4bf5f..0000000000 --- a/webrtc/modules/audio_coding/main/acm2/OWNERS +++ /dev/null @@ -1,5 +0,0 @@ - -# 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 *.gyp=* -per-file *.gypi=* diff --git a/webrtc/modules/audio_coding/main/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/audio_coding_module.gypi deleted file mode 100644 index 061ffaa212..0000000000 --- a/webrtc/modules/audio_coding/main/audio_coding_module.gypi +++ /dev/null @@ -1,196 +0,0 @@ -# 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. - -{ - 'variables': { - 'audio_coding_dependencies': [ - 'cng', - 'g711', - 'pcm16b', - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - ], - 'audio_coding_defines': [], - 'conditions': [ - ['include_opus==1', { - 'audio_coding_dependencies': ['webrtc_opus',], - 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',], - }], - ['build_with_mozilla==0', { - 'conditions': [ - ['target_arch=="arm"', { - 'audio_coding_dependencies': ['isac_fix',], - 'audio_coding_defines': ['WEBRTC_CODEC_ISACFX',], - }, { - 'audio_coding_dependencies': ['isac',], - 'audio_coding_defines': ['WEBRTC_CODEC_ISAC',], - }], - ], - 'audio_coding_dependencies': ['g722',], - 'audio_coding_defines': ['WEBRTC_CODEC_G722',], - }], - ['build_with_mozilla==0 and build_with_chromium==0', { - 'audio_coding_dependencies': ['ilbc', 'red',], - 'audio_coding_defines': ['WEBRTC_CODEC_ILBC', 'WEBRTC_CODEC_RED',], - }], - ], - }, - 'targets': [ - { - 'target_name': 'rent_a_codec', - 'type': 'static_library', - 'defines': [ - '<@(audio_coding_defines)', - ], - 'dependencies': [ - '<(webrtc_root)/common.gyp:webrtc_common', - ], - 'include_dirs': [ - '<(webrtc_root)', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - '<(webrtc_root)', - ], - }, - 'sources': [ - 'acm2/acm_codec_database.cc', - 'acm2/acm_codec_database.h', - 'acm2/rent_a_codec.cc', - 'acm2/rent_a_codec.h', - ], - }, - { - 'target_name': 'audio_coding_module', - 'type': 'static_library', - 'defines': [ - '<@(audio_coding_defines)', - ], - 'dependencies': [ - '<@(audio_coding_dependencies)', - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/webrtc.gyp:rtc_event_log', - 'neteq', - 'rent_a_codec', - ], - 'include_dirs': [ - 'include', - '../../include', - '<(webrtc_root)', - ], - 'direct_dependent_settings': { - 'include_dirs': [ - 'include', - '../../include', - '<(webrtc_root)', - ], - }, - 'conditions': [ - ['include_opus==1', { - 'export_dependent_settings': ['webrtc_opus'], - }], - ], - 'sources': [ - 'acm2/acm_common_defs.h', - 'acm2/acm_receiver.cc', - 'acm2/acm_receiver.h', - 'acm2/acm_resampler.cc', - 'acm2/acm_resampler.h', - 'acm2/audio_coding_module.cc', - 'acm2/audio_coding_module_impl.cc', - 'acm2/audio_coding_module_impl.h', - 'acm2/call_statistics.cc', - 'acm2/call_statistics.h', - 'acm2/codec_manager.cc', - 'acm2/codec_manager.h', - 'acm2/initial_delay_manager.cc', - 'acm2/initial_delay_manager.h', - 'include/audio_coding_module.h', - 'include/audio_coding_module_typedefs.h', - ], - }, - ], - 'conditions': [ - ['include_tests==1', { - 'targets': [ - { - 'target_name': 'acm_receive_test', - 'type': 'static_library', - 'defines': [ - '<@(audio_coding_defines)', - ], - 'dependencies': [ - '<@(audio_coding_dependencies)', - 'audio_coding_module', - 'neteq_unittest_tools', - '<(DEPTH)/testing/gtest.gyp:gtest', - ], - 'sources': [ - 'acm2/acm_receive_test_oldapi.cc', - 'acm2/acm_receive_test_oldapi.h', - ], - }, # acm_receive_test - { - 'target_name': 'acm_send_test', - 'type': 'static_library', - 'defines': [ - '<@(audio_coding_defines)', - ], - 'dependencies': [ - '<@(audio_coding_dependencies)', - 'audio_coding_module', - 'neteq_unittest_tools', - '<(DEPTH)/testing/gtest.gyp:gtest', - ], - 'sources': [ - 'acm2/acm_send_test_oldapi.cc', - 'acm2/acm_send_test_oldapi.h', - ], - }, # acm_send_test - { - 'target_name': 'delay_test', - 'type': 'executable', - 'dependencies': [ - 'audio_coding_module', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/test/test.gyp:test_support', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', - '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', - ], - 'sources': [ - 'test/delay_test.cc', - 'test/Channel.cc', - 'test/PCMFile.cc', - 'test/utility.cc', - ], - }, # delay_test - { - 'target_name': 'insert_packet_with_timing', - 'type': 'executable', - 'dependencies': [ - 'audio_coding_module', - '<(DEPTH)/testing/gtest.gyp:gtest', - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/test/test.gyp:test_support', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default', - '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', - ], - 'sources': [ - 'test/insert_packet_with_timing.cc', - 'test/Channel.cc', - 'test/PCMFile.cc', - ], - }, # delay_test - ], - }], - ], -} 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 fc3ddd5486..03f408776a 100644 --- a/webrtc/modules/audio_coding/main/include/audio_coding_module.h +++ b/webrtc/modules/audio_coding/main/include/audio_coding_module.h @@ -8,14 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_ + +#pragma message("WARNING: audio_coding/main/include is DEPRECATED; use audio_coding/include") #include #include "webrtc/base/optional.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/modules/audio_coding/neteq/include/neteq.h" #include "webrtc/modules/include/module.h" #include "webrtc/system_wrappers/include/clock.h" @@ -738,4 +740,4 @@ class AudioCodingModule { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_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 1ca6f9d13c..e1ec30a5c1 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 @@ -8,8 +8,10 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ + +#pragma message("WARNING: audio_coding/main/include is DEPRECATED; use audio_coding/include") #include @@ -48,4 +50,4 @@ enum OpusApplicationMode { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_ diff --git a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h index d1aae4a258..bc8bdd9626 100644 --- a/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h +++ b/webrtc/modules/audio_coding/neteq/audio_decoder_impl.h @@ -20,7 +20,7 @@ #ifdef WEBRTC_CODEC_G722 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" #endif -#include "webrtc/modules/audio_coding/main/acm2/rent_a_codec.h" +#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/neteq/nack.h b/webrtc/modules/audio_coding/neteq/nack.h index 116b7e2192..17fef46464 100644 --- a/webrtc/modules/audio_coding/neteq/nack.h +++ b/webrtc/modules/audio_coding/neteq/nack.h @@ -15,7 +15,7 @@ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" #include "webrtc/test/testsupport/gtest_prod_util.h" // diff --git a/webrtc/modules/audio_coding/neteq/nack_unittest.cc b/webrtc/modules/audio_coding/neteq/nack_unittest.cc index 853af94ede..53b19dc50f 100644 --- a/webrtc/modules/audio_coding/neteq/nack_unittest.cc +++ b/webrtc/modules/audio_coding/neteq/nack_unittest.cc @@ -17,7 +17,7 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/typedefs.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" namespace webrtc { namespace { diff --git a/webrtc/modules/audio_coding/main/test/ACMTest.h b/webrtc/modules/audio_coding/test/ACMTest.h similarity index 74% rename from webrtc/modules/audio_coding/main/test/ACMTest.h rename to webrtc/modules/audio_coding/test/ACMTest.h index f73961f5e5..d7e87d34ba 100644 --- a/webrtc/modules/audio_coding/main/test/ACMTest.h +++ b/webrtc/modules/audio_coding/test/ACMTest.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_ class ACMTest { public: @@ -18,4 +18,4 @@ class ACMTest { virtual void Perform() = 0; }; -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ACMTEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_ diff --git a/webrtc/modules/audio_coding/main/test/APITest.cc b/webrtc/modules/audio_coding/test/APITest.cc similarity index 99% rename from webrtc/modules/audio_coding/main/test/APITest.cc rename to webrtc/modules/audio_coding/test/APITest.cc index 88ad7e2a76..59a5a3aea6 100644 --- a/webrtc/modules/audio_coding/main/test/APITest.cc +++ b/webrtc/modules/audio_coding/test/APITest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/APITest.h" +#include "webrtc/modules/audio_coding/test/APITest.h" #include #include @@ -24,8 +24,8 @@ #include "webrtc/common.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/modules/audio_coding/main/test/APITest.h b/webrtc/modules/audio_coding/test/APITest.h similarity index 87% rename from webrtc/modules/audio_coding/main/test/APITest.h rename to webrtc/modules/audio_coding/test/APITest.h index d4c5b1ecdd..a1937c2b00 100644 --- a/webrtc/modules/audio_coding/main/test/APITest.h +++ b/webrtc/modules/audio_coding/test/APITest.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" @@ -160,4 +160,4 @@ class APITest : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_APITEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_ diff --git a/webrtc/modules/audio_coding/main/test/Channel.cc b/webrtc/modules/audio_coding/test/Channel.cc similarity index 99% rename from webrtc/modules/audio_coding/main/test/Channel.cc rename to webrtc/modules/audio_coding/test/Channel.cc index 02bd783a38..31521fe1e3 100644 --- a/webrtc/modules/audio_coding/main/test/Channel.cc +++ b/webrtc/modules/audio_coding/test/Channel.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/Channel.h" +#include "webrtc/modules/audio_coding/test/Channel.h" #include #include diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h similarity index 93% rename from webrtc/modules/audio_coding/main/test/Channel.h rename to webrtc/modules/audio_coding/test/Channel.h index ff6937ec08..b047aa9909 100644 --- a/webrtc/modules/audio_coding/main/test/Channel.h +++ b/webrtc/modules/audio_coding/test/Channel.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ #include -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/typedefs.h" @@ -127,4 +127,4 @@ class Channel : public AudioPacketizationCallback { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_CHANNEL_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_ diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc similarity index 97% rename from webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc rename to webrtc/modules/audio_coding/test/EncodeDecodeTest.cc index d68e57532e..ef45705e20 100644 --- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc +++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h" +#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" #include #include @@ -17,9 +17,9 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h similarity index 86% rename from webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h rename to webrtc/modules/audio_coding/test/EncodeDecodeTest.h index 4ad92cec15..3881062219 100644 --- a/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h +++ b/webrtc/modules/audio_coding/test/EncodeDecodeTest.h @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_ #include #include -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/RTPFile.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/RTPFile.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -120,4 +120,4 @@ class EncodeDecodeTest : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_ diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/test/PCMFile.cc similarity index 100% rename from webrtc/modules/audio_coding/main/test/PCMFile.cc rename to webrtc/modules/audio_coding/test/PCMFile.cc diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/test/PCMFile.h similarity index 90% rename from webrtc/modules/audio_coding/main/test/PCMFile.h rename to webrtc/modules/audio_coding/test/PCMFile.h index 785ed667fe..9365180208 100644 --- a/webrtc/modules/audio_coding/main/test/PCMFile.h +++ b/webrtc/modules/audio_coding/test/PCMFile.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_ #include #include @@ -65,4 +65,4 @@ class PCMFile { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PCMFILE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_ diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc b/webrtc/modules/audio_coding/test/PacketLossTest.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/PacketLossTest.cc rename to webrtc/modules/audio_coding/test/PacketLossTest.cc index f7c96faacd..ad3e83403e 100644 --- a/webrtc/modules/audio_coding/main/test/PacketLossTest.cc +++ b/webrtc/modules/audio_coding/test/PacketLossTest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h" +#include "webrtc/modules/audio_coding/test/PacketLossTest.h" #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common.h" diff --git a/webrtc/modules/audio_coding/main/test/PacketLossTest.h b/webrtc/modules/audio_coding/test/PacketLossTest.h similarity index 86% rename from webrtc/modules/audio_coding/main/test/PacketLossTest.h rename to webrtc/modules/audio_coding/test/PacketLossTest.h index d25dea264f..f3570ae1ca 100644 --- a/webrtc/modules/audio_coding/main/test/PacketLossTest.h +++ b/webrtc/modules/audio_coding/test/PacketLossTest.h @@ -8,12 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h" +#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" namespace webrtc { @@ -64,4 +64,4 @@ class PacketLossTest : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_ diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.cc b/webrtc/modules/audio_coding/test/RTPFile.cc similarity index 100% rename from webrtc/modules/audio_coding/main/test/RTPFile.cc rename to webrtc/modules/audio_coding/test/RTPFile.cc diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/test/RTPFile.h similarity index 92% rename from webrtc/modules/audio_coding/main/test/RTPFile.h rename to webrtc/modules/audio_coding/test/RTPFile.h index 6bad755af9..696d41ebd2 100644 --- a/webrtc/modules/audio_coding/main/test/RTPFile.h +++ b/webrtc/modules/audio_coding/test/RTPFile.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_ #include #include -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" #include "webrtc/typedefs.h" @@ -123,4 +123,4 @@ class RTPFile : public RTPStream { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_ diff --git a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc b/webrtc/modules/audio_coding/test/SpatialAudio.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/SpatialAudio.cc rename to webrtc/modules/audio_coding/test/SpatialAudio.cc index 17d4fc88b2..c9f8080826 100644 --- a/webrtc/modules/audio_coding/main/test/SpatialAudio.cc +++ b/webrtc/modules/audio_coding/test/SpatialAudio.cc @@ -14,7 +14,7 @@ #include #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/test/SpatialAudio.h" +#include "webrtc/modules/audio_coding/test/SpatialAudio.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/SpatialAudio.h b/webrtc/modules/audio_coding/test/SpatialAudio.h similarity index 66% rename from webrtc/modules/audio_coding/main/test/SpatialAudio.h rename to webrtc/modules/audio_coding/test/SpatialAudio.h index fc258977f3..3548cc98eb 100644 --- a/webrtc/modules/audio_coding/main/test/SpatialAudio.h +++ b/webrtc/modules/audio_coding/test/SpatialAudio.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" #define MAX_FILE_NAME_LENGTH_BYTE 500 @@ -44,4 +44,4 @@ class SpatialAudio : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_SPATIALAUDIO_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_SPATIALAUDIO_H_ diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc b/webrtc/modules/audio_coding/test/TestAllCodecs.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/TestAllCodecs.cc rename to webrtc/modules/audio_coding/test/TestAllCodecs.cc index e9e4f2bcea..21ce7c11aa 100644 --- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc +++ b/webrtc/modules/audio_coding/test/TestAllCodecs.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h" +#include "webrtc/modules/audio_coding/test/TestAllCodecs.h" #include #include @@ -18,9 +18,9 @@ #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/typedefs.h" diff --git a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h b/webrtc/modules/audio_coding/test/TestAllCodecs.h similarity index 85% rename from webrtc/modules/audio_coding/main/test/TestAllCodecs.h rename to webrtc/modules/audio_coding/test/TestAllCodecs.h index 1cdc0cba98..e79bd69faa 100644 --- a/webrtc/modules/audio_coding/main/test/TestAllCodecs.h +++ b/webrtc/modules/audio_coding/test/TestAllCodecs.h @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" #include "webrtc/typedefs.h" namespace webrtc { @@ -81,4 +81,4 @@ class TestAllCodecs : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTALLCODECS_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_ diff --git a/webrtc/modules/audio_coding/main/test/TestRedFec.cc b/webrtc/modules/audio_coding/test/TestRedFec.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/TestRedFec.cc rename to webrtc/modules/audio_coding/test/TestRedFec.cc index 0627ae2d74..d54402657f 100644 --- a/webrtc/modules/audio_coding/main/test/TestRedFec.cc +++ b/webrtc/modules/audio_coding/test/TestRedFec.cc @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/TestRedFec.h" +#include "webrtc/modules/audio_coding/test/TestRedFec.h" #include #include "webrtc/common.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/TestRedFec.h b/webrtc/modules/audio_coding/test/TestRedFec.h similarity index 78% rename from webrtc/modules/audio_coding/main/test/TestRedFec.h rename to webrtc/modules/audio_coding/test/TestRedFec.h index ac0b6cdfc7..6343d8e374 100644 --- a/webrtc/modules/audio_coding/main/test/TestRedFec.h +++ b/webrtc/modules/audio_coding/test/TestRedFec.h @@ -8,14 +8,14 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" namespace webrtc { @@ -48,4 +48,4 @@ class TestRedFec : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TESTREDFEC_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_ diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.cc b/webrtc/modules/audio_coding/test/TestStereo.cc similarity index 99% rename from webrtc/modules/audio_coding/main/test/TestStereo.cc rename to webrtc/modules/audio_coding/test/TestStereo.cc index bb38fac738..19f027b058 100644 --- a/webrtc/modules/audio_coding/main/test/TestStereo.cc +++ b/webrtc/modules/audio_coding/test/TestStereo.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/TestStereo.h" +#include "webrtc/modules/audio_coding/test/TestStereo.h" #include @@ -17,8 +17,8 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/TestStereo.h b/webrtc/modules/audio_coding/test/TestStereo.h similarity index 88% rename from webrtc/modules/audio_coding/main/test/TestStereo.h rename to webrtc/modules/audio_coding/test/TestStereo.h index b56e995272..4526be6960 100644 --- a/webrtc/modules/audio_coding/main/test/TestStereo.h +++ b/webrtc/modules/audio_coding/test/TestStereo.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ #include #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" #define PCMA_AND_PCMU @@ -114,4 +114,4 @@ class TestStereo : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_ diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc b/webrtc/modules/audio_coding/test/TestVADDTX.cc similarity index 97% rename from webrtc/modules/audio_coding/main/test/TestVADDTX.cc rename to webrtc/modules/audio_coding/test/TestVADDTX.cc index bba7b91c40..98b1224762 100644 --- a/webrtc/modules/audio_coding/main/test/TestVADDTX.cc +++ b/webrtc/modules/audio_coding/test/TestVADDTX.cc @@ -8,13 +8,13 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h" +#include "webrtc/modules/audio_coding/test/TestVADDTX.h" #include #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/test/testsupport/fileutils.h" namespace webrtc { diff --git a/webrtc/modules/audio_coding/main/test/TestVADDTX.h b/webrtc/modules/audio_coding/test/TestVADDTX.h similarity index 85% rename from webrtc/modules/audio_coding/main/test/TestVADDTX.h rename to webrtc/modules/audio_coding/test/TestVADDTX.h index 07596e2e86..1e7f0ef4d7 100644 --- a/webrtc/modules/audio_coding/main/test/TestVADDTX.h +++ b/webrtc/modules/audio_coding/test/TestVADDTX.h @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" namespace webrtc { @@ -99,4 +99,4 @@ class TestOpusDtx final : public TestVadDtx { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTVADDTX_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_ diff --git a/webrtc/modules/audio_coding/main/test/Tester.cc b/webrtc/modules/audio_coding/test/Tester.cc similarity index 87% rename from webrtc/modules/audio_coding/main/test/Tester.cc rename to webrtc/modules/audio_coding/test/Tester.cc index 7302e5dcbe..3ff3dd8cd4 100644 --- a/webrtc/modules/audio_coding/main/test/Tester.cc +++ b/webrtc/modules/audio_coding/test/Tester.cc @@ -13,17 +13,17 @@ #include #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/APITest.h" -#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h" -#include "webrtc/modules/audio_coding/main/test/iSACTest.h" -#include "webrtc/modules/audio_coding/main/test/opus_test.h" -#include "webrtc/modules/audio_coding/main/test/PacketLossTest.h" -#include "webrtc/modules/audio_coding/main/test/TestAllCodecs.h" -#include "webrtc/modules/audio_coding/main/test/TestRedFec.h" -#include "webrtc/modules/audio_coding/main/test/TestStereo.h" -#include "webrtc/modules/audio_coding/main/test/TestVADDTX.h" -#include "webrtc/modules/audio_coding/main/test/TwoWayCommunication.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/APITest.h" +#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h" +#include "webrtc/modules/audio_coding/test/iSACTest.h" +#include "webrtc/modules/audio_coding/test/opus_test.h" +#include "webrtc/modules/audio_coding/test/PacketLossTest.h" +#include "webrtc/modules/audio_coding/test/TestAllCodecs.h" +#include "webrtc/modules/audio_coding/test/TestRedFec.h" +#include "webrtc/modules/audio_coding/test/TestStereo.h" +#include "webrtc/modules/audio_coding/test/TestVADDTX.h" +#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" #include "webrtc/test/testsupport/gtest_disable.h" diff --git a/webrtc/modules/audio_coding/main/test/TimedTrace.cc b/webrtc/modules/audio_coding/test/TimedTrace.cc similarity index 100% rename from webrtc/modules/audio_coding/main/test/TimedTrace.cc rename to webrtc/modules/audio_coding/test/TimedTrace.cc diff --git a/webrtc/modules/audio_coding/main/test/TimedTrace.h b/webrtc/modules/audio_coding/test/TimedTrace.h similarity index 82% rename from webrtc/modules/audio_coding/main/test/TimedTrace.h rename to webrtc/modules/audio_coding/test/TimedTrace.h index ef9609a267..0793eb0c0c 100644 --- a/webrtc/modules/audio_coding/main/test/TimedTrace.h +++ b/webrtc/modules/audio_coding/test/TimedTrace.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef TIMED_TRACE_H -#define TIMED_TRACE_H +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_ #include "webrtc/typedefs.h" @@ -33,4 +33,4 @@ class TimedTrace { }; -#endif +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TIMEDTRACE_H_ diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc rename to webrtc/modules/audio_coding/test/TwoWayCommunication.cc index 725cbf74d7..56e136bd34 100644 --- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc +++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.cc @@ -21,8 +21,8 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/engine_configurations.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h b/webrtc/modules/audio_coding/test/TwoWayCommunication.h similarity index 69% rename from webrtc/modules/audio_coding/main/test/TwoWayCommunication.h rename to webrtc/modules/audio_coding/test/TwoWayCommunication.h index bf969fe683..77639935da 100644 --- a/webrtc/modules/audio_coding/main/test/TwoWayCommunication.h +++ b/webrtc/modules/audio_coding/test/TwoWayCommunication.h @@ -8,15 +8,15 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_ #include "webrtc/base/scoped_ptr.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" namespace webrtc { @@ -57,4 +57,4 @@ class TwoWayCommunication : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TWOWAYCOMMUNICATION_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_ diff --git a/webrtc/modules/audio_coding/main/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc similarity index 95% rename from webrtc/modules/audio_coding/main/test/delay_test.cc rename to webrtc/modules/audio_coding/test/delay_test.cc index ce08c0f4a2..a8c137f501 100644 --- a/webrtc/modules/audio_coding/main/test/delay_test.cc +++ b/webrtc/modules/audio_coding/test/delay_test.cc @@ -19,12 +19,12 @@ #include "webrtc/common.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.cc b/webrtc/modules/audio_coding/test/iSACTest.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/iSACTest.cc rename to webrtc/modules/audio_coding/test/iSACTest.cc index 203e12b6a2..09744b13c6 100644 --- a/webrtc/modules/audio_coding/main/test/iSACTest.cc +++ b/webrtc/modules/audio_coding/test/iSACTest.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/iSACTest.h" +#include "webrtc/modules/audio_coding/test/iSACTest.h" #include #include @@ -23,8 +23,8 @@ #include #endif -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/tick_util.h" #include "webrtc/system_wrappers/include/trace.h" diff --git a/webrtc/modules/audio_coding/main/test/iSACTest.h b/webrtc/modules/audio_coding/test/iSACTest.h similarity index 76% rename from webrtc/modules/audio_coding/main/test/iSACTest.h rename to webrtc/modules/audio_coding/test/iSACTest.h index 0693d935e1..c5bb515437 100644 --- a/webrtc/modules/audio_coding/main/test/iSACTest.h +++ b/webrtc/modules/audio_coding/test/iSACTest.h @@ -8,18 +8,18 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_ #include #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/utility.h" #define MAX_FILE_NAME_LENGTH_BYTE 500 #define NO_OF_CLIENTS 15 @@ -76,4 +76,4 @@ class ISACTest : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ISACTEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_ diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc rename to webrtc/modules/audio_coding/test/insert_packet_with_timing.cc index 857381d250..481df55ffd 100644 --- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc +++ b/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc @@ -14,9 +14,9 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#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/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.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/opus_test.cc b/webrtc/modules/audio_coding/test/opus_test.cc similarity index 97% rename from webrtc/modules/audio_coding/main/test/opus_test.cc rename to webrtc/modules/audio_coding/test/opus_test.cc index 27cc40aa3c..3372a2a4a0 100644 --- a/webrtc/modules/audio_coding/main/test/opus_test.cc +++ b/webrtc/modules/audio_coding/test/opus_test.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/audio_coding/main/test/opus_test.h" +#include "webrtc/modules/audio_coding/test/opus_test.h" #include @@ -18,9 +18,9 @@ #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h" -#include "webrtc/modules/audio_coding/main/test/TestStereo.h" -#include "webrtc/modules/audio_coding/main/test/utility.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h" +#include "webrtc/modules/audio_coding/test/TestStereo.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/opus_test.h b/webrtc/modules/audio_coding/test/opus_test.h similarity index 72% rename from webrtc/modules/audio_coding/main/test/opus_test.h rename to webrtc/modules/audio_coding/test/opus_test.h index 0b96009241..090c8fa9dd 100644 --- a/webrtc/modules/audio_coding/main/test/opus_test.h +++ b/webrtc/modules/audio_coding/test/opus_test.h @@ -8,18 +8,18 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_ #include #include "webrtc/base/scoped_ptr.h" #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" -#include "webrtc/modules/audio_coding/main/test/ACMTest.h" -#include "webrtc/modules/audio_coding/main/test/Channel.h" -#include "webrtc/modules/audio_coding/main/test/PCMFile.h" -#include "webrtc/modules/audio_coding/main/test/TestStereo.h" +#include "webrtc/modules/audio_coding/acm2/acm_resampler.h" +#include "webrtc/modules/audio_coding/test/ACMTest.h" +#include "webrtc/modules/audio_coding/test/Channel.h" +#include "webrtc/modules/audio_coding/test/PCMFile.h" +#include "webrtc/modules/audio_coding/test/TestStereo.h" namespace webrtc { @@ -54,4 +54,4 @@ class OpusTest : public ACMTest { } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_OPUS_TEST_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_ diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/test/target_delay_unittest.cc similarity index 98% rename from webrtc/modules/audio_coding/main/test/target_delay_unittest.cc rename to webrtc/modules/audio_coding/test/target_delay_unittest.cc index afc0e10225..d7c0411c92 100644 --- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc +++ b/webrtc/modules/audio_coding/test/target_delay_unittest.cc @@ -12,8 +12,8 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/pcm16b/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/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/test/utility.h" #include "webrtc/modules/include/module_common_types.h" #include "webrtc/system_wrappers/include/sleep.h" #include "webrtc/test/testsupport/fileutils.h" diff --git a/webrtc/modules/audio_coding/main/test/utility.cc b/webrtc/modules/audio_coding/test/utility.cc similarity index 97% rename from webrtc/modules/audio_coding/main/test/utility.cc rename to webrtc/modules/audio_coding/test/utility.cc index 34af5e703f..89368bce51 100644 --- a/webrtc/modules/audio_coding/main/test/utility.cc +++ b/webrtc/modules/audio_coding/test/utility.cc @@ -18,8 +18,8 @@ #include "testing/gtest/include/gtest/gtest.h" #include "webrtc/common.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" -#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/acm2/acm_common_defs.h" #define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13 diff --git a/webrtc/modules/audio_coding/main/test/utility.h b/webrtc/modules/audio_coding/test/utility.h similarity index 94% rename from webrtc/modules/audio_coding/main/test/utility.h rename to webrtc/modules/audio_coding/test/utility.h index e936ec1cdd..23869be7ed 100644 --- a/webrtc/modules/audio_coding/main/test/utility.h +++ b/webrtc/modules/audio_coding/test/utility.h @@ -8,11 +8,11 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_ -#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_ +#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_ +#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_ #include "testing/gtest/include/gtest/gtest.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" namespace webrtc { @@ -136,4 +136,4 @@ void UseNewAcm(webrtc::Config* config); } // namespace webrtc -#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_UTILITY_H_ +#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_ diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 0c981e691f..599a931fb9 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -71,24 +71,24 @@ '<@(audio_coding_defines)', ], 'sources': [ - 'audio_coding/main/test/APITest.cc', - 'audio_coding/main/test/Channel.cc', - 'audio_coding/main/test/EncodeDecodeTest.cc', - 'audio_coding/main/test/PCMFile.cc', - 'audio_coding/main/test/PacketLossTest.cc', - 'audio_coding/main/test/RTPFile.cc', - 'audio_coding/main/test/SpatialAudio.cc', - 'audio_coding/main/test/TestAllCodecs.cc', - 'audio_coding/main/test/TestRedFec.cc', - 'audio_coding/main/test/TestStereo.cc', - 'audio_coding/main/test/TestVADDTX.cc', - 'audio_coding/main/test/Tester.cc', - 'audio_coding/main/test/TimedTrace.cc', - 'audio_coding/main/test/TwoWayCommunication.cc', - 'audio_coding/main/test/iSACTest.cc', - 'audio_coding/main/test/opus_test.cc', - 'audio_coding/main/test/target_delay_unittest.cc', - 'audio_coding/main/test/utility.cc', + 'audio_coding/test/APITest.cc', + 'audio_coding/test/Channel.cc', + 'audio_coding/test/EncodeDecodeTest.cc', + 'audio_coding/test/PCMFile.cc', + 'audio_coding/test/PacketLossTest.cc', + 'audio_coding/test/RTPFile.cc', + 'audio_coding/test/SpatialAudio.cc', + 'audio_coding/test/TestAllCodecs.cc', + 'audio_coding/test/TestRedFec.cc', + 'audio_coding/test/TestStereo.cc', + 'audio_coding/test/TestVADDTX.cc', + 'audio_coding/test/Tester.cc', + 'audio_coding/test/TimedTrace.cc', + 'audio_coding/test/TwoWayCommunication.cc', + 'audio_coding/test/iSACTest.cc', + 'audio_coding/test/opus_test.cc', + 'audio_coding/test/target_delay_unittest.cc', + 'audio_coding/test/utility.cc', 'rtp_rtcp/test/testFec/test_fec.cc', 'video_coding/codecs/test/videoprocessor_integrationtest.cc', 'video_coding/codecs/vp8/test/vp8_impl_unittest.cc', @@ -156,12 +156,12 @@ ], 'sources': [ 'audio_coding/codecs/cng/audio_encoder_cng_unittest.cc', - 'audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc', - 'audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc', - 'audio_coding/main/acm2/call_statistics_unittest.cc', - 'audio_coding/main/acm2/codec_manager_unittest.cc', - 'audio_coding/main/acm2/initial_delay_manager_unittest.cc', - 'audio_coding/main/acm2/rent_a_codec_unittest.cc', + 'audio_coding/acm2/acm_receiver_unittest_oldapi.cc', + 'audio_coding/acm2/audio_coding_module_unittest_oldapi.cc', + 'audio_coding/acm2/call_statistics_unittest.cc', + 'audio_coding/acm2/codec_manager_unittest.cc', + 'audio_coding/acm2/initial_delay_manager_unittest.cc', + 'audio_coding/acm2/rent_a_codec_unittest.cc', 'audio_coding/codecs/cng/cng_unittest.cc', 'audio_coding/codecs/isac/fix/source/filters_unittest.cc', 'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc', diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/modules/utility/source/coder.h index 4270e9b380..abfa87efe1 100644 --- a/webrtc/modules/utility/source/coder.h +++ b/webrtc/modules/utility/source/coder.h @@ -13,7 +13,7 @@ #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_types.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/typedefs.h" namespace webrtc { diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index ba18aaa8dd..0e509d2fd0 100644 --- a/webrtc/voice_engine/channel.h +++ b/webrtc/voice_engine/channel.h @@ -14,7 +14,7 @@ #include "webrtc/base/scoped_ptr.h" #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_coding/include/audio_coding_module.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" diff --git a/webrtc/voice_engine/voe_base_impl.cc b/webrtc/voice_engine/voe_base_impl.cc index 677e9b1cf9..2b5587ddef 100644 --- a/webrtc/voice_engine/voe_base_impl.cc +++ b/webrtc/voice_engine/voe_base_impl.cc @@ -13,7 +13,7 @@ #include "webrtc/base/format_macros.h" #include "webrtc/common.h" #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/modules/audio_device/audio_device_impl.h" #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" diff --git a/webrtc/voice_engine/voe_codec_impl.cc b/webrtc/voice_engine/voe_codec_impl.cc index 3ab02a6ebb..162f1c2efb 100644 --- a/webrtc/voice_engine/voe_codec_impl.cc +++ b/webrtc/voice_engine/voe_codec_impl.cc @@ -10,7 +10,7 @@ #include "webrtc/voice_engine/voe_codec_impl.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/voice_engine/channel.h" diff --git a/webrtc/voice_engine/voe_neteq_stats_impl.cc b/webrtc/voice_engine/voe_neteq_stats_impl.cc index 00e04d8f99..807325b4f8 100644 --- a/webrtc/voice_engine/voe_neteq_stats_impl.cc +++ b/webrtc/voice_engine/voe_neteq_stats_impl.cc @@ -10,7 +10,7 @@ #include "webrtc/voice_engine/voe_neteq_stats_impl.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/voice_engine/channel.h" diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc index d9c574457b..8df05cc012 100644 --- a/webrtc/voice_engine/voice_engine_impl.cc +++ b/webrtc/voice_engine/voice_engine_impl.cc @@ -16,7 +16,7 @@ #endif #include "webrtc/base/checks.h" -#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h" +#include "webrtc/modules/audio_coding/include/audio_coding_module.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/trace.h" #include "webrtc/voice_engine/channel_proxy.h"