From 9774447b8f30a531caea824b5d9b208abf182e7c Mon Sep 17 00:00:00 2001 From: kwiberg Date: Tue, 10 Jan 2017 01:12:51 -0800 Subject: [PATCH] Move FilePlayer and FileRecorder to Voice Engine Because Voice Engine was the only user. (We have tried to land this many times before. I'm hoping that this time all external dependencies on these files will really be gone.) BUG=none Review-Url: https://codereview.webrtc.org/2622493002 Cr-Commit-Position: refs/heads/master@{#15978} --- .gn | 3 + webrtc/modules/BUILD.gn | 3 - webrtc/modules/utility/BUILD.gn | 6 -- webrtc/voice_engine/BUILD.gn | 67 +++++++++++++++++++ webrtc/voice_engine/channel.h | 4 +- .../utility/source => voice_engine}/coder.cc | 3 +- .../utility/source => voice_engine}/coder.h | 6 +- .../source => voice_engine}/file_player.cc | 4 +- .../include => voice_engine}/file_player.h | 6 +- .../file_player_unittests.cc | 2 +- .../source => voice_engine}/file_recorder.cc | 4 +- .../include => voice_engine}/file_recorder.h | 6 +- webrtc/voice_engine/output_mixer.h | 2 +- webrtc/voice_engine/transmit_mixer.h | 4 +- 14 files changed, 91 insertions(+), 29 deletions(-) rename webrtc/{modules/utility/source => voice_engine}/coder.cc (98%) rename webrtc/{modules/utility/source => voice_engine}/coder.h (93%) rename webrtc/{modules/utility/source => voice_engine}/file_player.cc (99%) rename webrtc/{modules/utility/include => voice_engine}/file_player.h (94%) rename webrtc/{modules/utility/source => voice_engine}/file_player_unittests.cc (98%) rename webrtc/{modules/utility/source => voice_engine}/file_recorder.cc (98%) rename webrtc/{modules/utility/include => voice_engine}/file_recorder.h (91%) diff --git a/.gn b/.gn index 3e0fcf275e..77394c6440 100644 --- a/.gn +++ b/.gn @@ -42,6 +42,9 @@ check_targets = [ "//webrtc/modules/remote_bitrate_estimator/*", "//webrtc/stats:rtc_stats", "//webrtc/voice_engine", + "//webrtc/voice_engine:audio_coder", + "//webrtc/voice_engine:file_player", + "//webrtc/voice_engine:file_recorder", "//webrtc/voice_engine:level_indicator", ] diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index bbb5744422..37b5b12ff3 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -231,8 +231,6 @@ if (rtc_include_tests) { "//resources/synthetic-trace.rx", "//resources/tmobile-downlink.rx", "//resources/tmobile-uplink.rx", - "//resources/utility/encapsulated_pcm16b_8khz.wav", - "//resources/utility/encapsulated_pcmu_8khz.wav", "//resources/verizon3g-downlink.rx", "//resources/verizon3g-uplink.rx", "//resources/verizon4g-downlink.rx", @@ -483,7 +481,6 @@ if (rtc_include_tests) { "rtp_rtcp/test/testAPI/test_api_audio.cc", "rtp_rtcp/test/testAPI/test_api_rtcp.cc", "rtp_rtcp/test/testAPI/test_api_video.cc", - "utility/source/file_player_unittests.cc", "utility/source/process_thread_impl_unittest.cc", "video_coding/codecs/test/packet_manipulator_unittest.cc", "video_coding/codecs/test/stats_unittest.cc", diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index edd428fe09..48cbf793ed 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -11,15 +11,9 @@ import("../../build/webrtc.gni") rtc_static_library("utility") { sources = [ "include/audio_frame_operations.h", - "include/file_player.h", - "include/file_recorder.h", "include/helpers_android.h", "include/jvm_android.h", "include/process_thread.h", - "source/coder.cc", - "source/coder.h", - "source/file_player.cc", - "source/file_recorder.cc", "source/helpers_android.cc", "source/jvm_android.cc", "source/process_thread_impl.cc", diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index 6413c1842a..0591999def 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -8,6 +8,64 @@ import("../build/webrtc.gni") +rtc_static_library("audio_coder") { + sources = [ + "coder.cc", + "coder.h", + ] + deps = [ + "..:webrtc_common", + "../modules/audio_coding", + "../modules/audio_coding:audio_format_conversion", + "../modules/audio_coding:builtin_audio_decoder_factory", + "../modules/audio_coding:rent_a_codec", + ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } +} + +rtc_static_library("file_player") { + sources = [ + "file_player.cc", + "file_player.h", + ] + deps = [ + ":audio_coder", + "..:webrtc_common", + "../common_audio", + "../modules/media_file", + "../system_wrappers", + ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } +} + +rtc_static_library("file_recorder") { + sources = [ + "file_recorder.cc", + "file_recorder.h", + ] + deps = [ + ":audio_coder", + "..:webrtc_common", + "../base:rtc_base_approved", + "../common_audio", + "../modules/media_file:media_file", + "../system_wrappers", + ] + + if (!build_with_chromium && is_clang) { + # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). + suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] + } +} + rtc_static_library("voice_engine") { sources = [ "channel.cc", @@ -85,6 +143,8 @@ rtc_static_library("voice_engine") { "../modules/audio_coding", ] deps = [ + ":file_player", + ":file_recorder", ":level_indicator", "..:webrtc_common", "../api:audio_mixer_api", @@ -191,6 +251,7 @@ if (rtc_include_tests) { ":voice_engine", "//testing/gmock", "//testing/gtest", + "//third_party/gflags", "//webrtc/common_audio", "//webrtc/modules/audio_coding", "//webrtc/modules/audio_conference_mixer", @@ -210,6 +271,7 @@ if (rtc_include_tests) { sources = [ "channel_unittest.cc", + "file_player_unittests.cc", "test/channel_transport/udp_socket_manager_unittest.cc", "test/channel_transport/udp_socket_wrapper_unittest.cc", "test/channel_transport/udp_transport_unittest.cc", @@ -223,6 +285,11 @@ if (rtc_include_tests) { "voice_engine_fixture.h", ] + data = [ + "//resources/utility/encapsulated_pcm16b_8khz.wav", + "//resources/utility/encapsulated_pcmu_8khz.wav", + ] + if (is_win) { defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ] diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index 01907d269b..11e109e3c8 100644 --- a/webrtc/voice_engine/channel.h +++ b/webrtc/voice_engine/channel.h @@ -28,8 +28,8 @@ #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" -#include "webrtc/modules/utility/include/file_player.h" -#include "webrtc/modules/utility/include/file_recorder.h" +#include "webrtc/voice_engine/file_player.h" +#include "webrtc/voice_engine/file_recorder.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/include/voe_network.h" diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/voice_engine/coder.cc similarity index 98% rename from webrtc/modules/utility/source/coder.cc rename to webrtc/voice_engine/coder.cc index 71f969097f..82eb2487b6 100644 --- a/webrtc/modules/utility/source/coder.cc +++ b/webrtc/voice_engine/coder.cc @@ -8,11 +8,12 @@ * be found in the AUTHORS file in the root of the source tree. */ +#include "webrtc/voice_engine/coder.h" + #include "webrtc/common_types.h" #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h" #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h" #include "webrtc/modules/include/module_common_types.h" -#include "webrtc/modules/utility/source/coder.h" namespace webrtc { namespace { diff --git a/webrtc/modules/utility/source/coder.h b/webrtc/voice_engine/coder.h similarity index 93% rename from webrtc/modules/utility/source/coder.h rename to webrtc/voice_engine/coder.h index 4855a00465..5e16b0af66 100644 --- a/webrtc/modules/utility/source/coder.h +++ b/webrtc/voice_engine/coder.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ -#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ +#ifndef WEBRTC_VOICE_ENGINE_CODER_H_ +#define WEBRTC_VOICE_ENGINE_CODER_H_ #include @@ -65,4 +65,4 @@ class AudioCoder : public AudioPacketizationCallback { }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ +#endif // WEBRTC_VOICE_ENGINE_CODER_H_ diff --git a/webrtc/modules/utility/source/file_player.cc b/webrtc/voice_engine/file_player.cc similarity index 99% rename from webrtc/modules/utility/source/file_player.cc rename to webrtc/voice_engine/file_player.cc index fa5bc3d2e4..b581d5235b 100644 --- a/webrtc/modules/utility/source/file_player.cc +++ b/webrtc/voice_engine/file_player.cc @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/voice_engine/file_player.h" #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" #include "webrtc/modules/media_file/media_file.h" #include "webrtc/modules/media_file/media_file_defines.h" -#include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine/coder.h" namespace webrtc { diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/voice_engine/file_player.h similarity index 94% rename from webrtc/modules/utility/include/file_player.h rename to webrtc/voice_engine/file_player.h index 1adbb9d70e..956016f07f 100644 --- a/webrtc/modules/utility/include/file_player.h +++ b/webrtc/voice_engine/file_player.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ -#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ +#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_ +#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_ #include @@ -77,4 +77,4 @@ class FilePlayer { virtual int32_t SetAudioScaling(float scaleFactor) = 0; }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ +#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_ diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/voice_engine/file_player_unittests.cc similarity index 98% rename from webrtc/modules/utility/source/file_player_unittests.cc rename to webrtc/voice_engine/file_player_unittests.cc index cc7865d150..d41f4044b4 100644 --- a/webrtc/modules/utility/source/file_player_unittests.cc +++ b/webrtc/voice_engine/file_player_unittests.cc @@ -10,7 +10,7 @@ // Unit tests for FilePlayer. -#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/voice_engine/file_player.h" #include diff --git a/webrtc/modules/utility/source/file_recorder.cc b/webrtc/voice_engine/file_recorder.cc similarity index 98% rename from webrtc/modules/utility/source/file_recorder.cc rename to webrtc/voice_engine/file_recorder.cc index 34bd3945fa..9a0edb057b 100644 --- a/webrtc/modules/utility/source/file_recorder.cc +++ b/webrtc/voice_engine/file_recorder.cc @@ -8,7 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/utility/include/file_recorder.h" +#include "webrtc/voice_engine/file_recorder.h" #include @@ -18,10 +18,10 @@ #include "webrtc/modules/include/module_common_types.h" #include "webrtc/modules/media_file/media_file.h" #include "webrtc/modules/media_file/media_file_defines.h" -#include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/system_wrappers/include/logging.h" #include "webrtc/typedefs.h" +#include "webrtc/voice_engine/coder.h" namespace webrtc { diff --git a/webrtc/modules/utility/include/file_recorder.h b/webrtc/voice_engine/file_recorder.h similarity index 91% rename from webrtc/modules/utility/include/file_recorder.h rename to webrtc/voice_engine/file_recorder.h index 6d4d321354..c4195d0eba 100644 --- a/webrtc/modules/utility/include/file_recorder.h +++ b/webrtc/voice_engine/file_recorder.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ -#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ +#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ +#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ #include @@ -54,4 +54,4 @@ class FileRecorder { }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ +#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h index cc8dddfbfc..5e3e2e708e 100644 --- a/webrtc/voice_engine/output_mixer.h +++ b/webrtc/voice_engine/output_mixer.h @@ -18,7 +18,7 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h" #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h" -#include "webrtc/modules/utility/include/file_recorder.h" +#include "webrtc/voice_engine/file_recorder.h" #include "webrtc/voice_engine/level_indicator.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h index 3147c1036f..13ddaa5492 100644 --- a/webrtc/voice_engine/transmit_mixer.h +++ b/webrtc/voice_engine/transmit_mixer.h @@ -18,8 +18,8 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_processing/typing_detection.h" #include "webrtc/modules/include/module_common_types.h" -#include "webrtc/modules/utility/include/file_player.h" -#include "webrtc/modules/utility/include/file_recorder.h" +#include "webrtc/voice_engine/file_player.h" +#include "webrtc/voice_engine/file_recorder.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/level_indicator.h" #include "webrtc/voice_engine/monitor_module.h"