From 9d7eb13c404fcb610f1b9babe46dbefe7cadc4e3 Mon Sep 17 00:00:00 2001 From: kwiberg Date: Tue, 16 Aug 2016 04:08:30 -0700 Subject: [PATCH] Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #3 id:40001 of https://codereview.webrtc.org/2247033003/ ) Reason for revert: Reverting, because it turns out that third-party code was using webrtc::FilePlayer. I'm not at all sure that this is something WebRTC ought to be exporting, but since we did export it, we have to live with it for now. Original issue's description: > Move FilePlayer and FileRecorder to Voice Engine > > Because Voice Engine was the only user. > > (This has been landed twice before, as > https://codereview.webrtc.org/2037623002 and > https://codereview.webrtc.org/2240163002. Third time's a charm!) > > NOPRESUBMIT=True > TBR=kjellander@webrtc.org > > Committed: https://crrev.com/427ce3d86f6328dc994f84a15c28bb7bfbaa46ef > Cr-Commit-Position: refs/heads/master@{#13777} TBR= # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2245413002 Cr-Commit-Position: refs/heads/master@{#13779} --- .gn | 3 - webrtc/modules/BUILD.gn | 3 + webrtc/modules/audio_mixer/audio_mixer.h | 2 +- webrtc/modules/modules.gyp | 3 + webrtc/modules/modules_unittests.isolate | 2 + webrtc/modules/utility/BUILD.gn | 8 ++ .../utility/include}/file_player.h | 7 +- .../utility/include}/file_recorder.h | 7 +- .../utility/source}/coder.cc | 3 +- .../utility/source}/coder.h | 6 +- .../utility/source}/file_player_impl.cc | 3 +- .../utility/source}/file_player_impl.h | 11 ++- .../utility/source}/file_player_unittests.cc | 3 +- .../utility/source}/file_recorder_impl.cc | 3 +- .../utility/source}/file_recorder_impl.h | 11 ++- webrtc/modules/utility/utility.gypi | 8 ++ webrtc/voice_engine/BUILD.gn | 76 ------------------- webrtc/voice_engine/channel.h | 4 +- webrtc/voice_engine/output_mixer.h | 2 +- webrtc/voice_engine/transmit_mixer.h | 4 +- webrtc/voice_engine/voice_engine.gyp | 57 -------------- .../voice_engine_unittests.isolate | 8 -- 22 files changed, 54 insertions(+), 180 deletions(-) rename webrtc/{voice_engine => modules/utility/include}/file_player.h (94%) rename webrtc/{voice_engine => modules/utility/include}/file_recorder.h (91%) rename webrtc/{voice_engine => modules/utility/source}/coder.cc (98%) rename webrtc/{voice_engine => modules/utility/source}/coder.h (93%) rename webrtc/{voice_engine => modules/utility/source}/file_player_impl.cc (99%) rename webrtc/{voice_engine => modules/utility/source}/file_player_impl.h (89%) rename webrtc/{voice_engine => modules/utility/source}/file_player_unittests.cc (98%) rename webrtc/{voice_engine => modules/utility/source}/file_recorder_impl.cc (99%) rename webrtc/{voice_engine => modules/utility/source}/file_recorder_impl.h (89%) diff --git a/.gn b/.gn index 3a744bf7bd..75e2587cee 100644 --- a/.gn +++ b/.gn @@ -21,9 +21,6 @@ secondary_source = "//build/secondary/" # TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done. check_targets = [ "//webrtc/modules/audio_device/*", - "//webrtc/voice_engine:audio_coder", - "//webrtc/voice_engine:file_player", - "//webrtc/voice_engine:file_recorder", "//webrtc/voice_engine:level_indicator", "//webrtc/modules/audio_coding:isac_fix_test", "//webrtc/modules/audio_mixer:audio_conference_mixer", diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index 01d4ea5689..676160ae65 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -306,6 +306,7 @@ if (rtc_include_tests) { "rtp_rtcp/test/testAPI/test_api_rtcp.cc", "rtp_rtcp/test/testAPI/test_api_video.cc", "utility/source/audio_frame_operations_unittest.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", @@ -595,6 +596,8 @@ 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", diff --git a/webrtc/modules/audio_mixer/audio_mixer.h b/webrtc/modules/audio_mixer/audio_mixer.h index eeeb193b3a..78cd4e5c79 100644 --- a/webrtc/modules/audio_mixer/audio_mixer.h +++ b/webrtc/modules/audio_mixer/audio_mixer.h @@ -16,7 +16,7 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_mixer/new_audio_conference_mixer.h" #include "webrtc/modules/audio_mixer/audio_mixer_defines.h" -#include "webrtc/voice_engine/file_recorder.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/voice_engine/level_indicator.h" #include "webrtc/voice_engine/voice_engine_defines.h" diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp index 68f7a5112e..094204f723 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -358,6 +358,7 @@ 'rtp_rtcp/test/testAPI/test_api_rtcp.cc', 'rtp_rtcp/test/testAPI/test_api_video.cc', 'utility/source/audio_frame_operations_unittest.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', @@ -598,6 +599,8 @@ '<(DEPTH)/resources/synthetic-trace.rx', '<(DEPTH)/resources/tmobile-downlink.rx', '<(DEPTH)/resources/tmobile-uplink.rx', + '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav', + '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav', '<(DEPTH)/resources/verizon3g-downlink.rx', '<(DEPTH)/resources/verizon3g-uplink.rx', '<(DEPTH)/resources/verizon4g-downlink.rx', diff --git a/webrtc/modules/modules_unittests.isolate b/webrtc/modules/modules_unittests.isolate index 933478d434..af7e6ef46e 100644 --- a/webrtc/modules/modules_unittests.isolate +++ b/webrtc/modules/modules_unittests.isolate @@ -110,6 +110,8 @@ '<(DEPTH)/resources/synthetic-trace.rx', '<(DEPTH)/resources/tmobile-downlink.rx', '<(DEPTH)/resources/tmobile-uplink.rx', + '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav', + '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav', '<(DEPTH)/resources/verizon3g-downlink.rx', '<(DEPTH)/resources/verizon3g-uplink.rx', '<(DEPTH)/resources/verizon4g-downlink.rx', diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index c3c9f0a923..5437e4f5f7 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -11,10 +11,18 @@ import("../../build/webrtc.gni") source_set("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/audio_frame_operations.cc", + "source/coder.cc", + "source/coder.h", + "source/file_player_impl.cc", + "source/file_player_impl.h", + "source/file_recorder_impl.cc", + "source/file_recorder_impl.h", "source/helpers_android.cc", "source/helpers_ios.mm", "source/jvm_android.cc", diff --git a/webrtc/voice_engine/file_player.h b/webrtc/modules/utility/include/file_player.h similarity index 94% rename from webrtc/voice_engine/file_player.h rename to webrtc/modules/utility/include/file_player.h index 898d66cd4d..b064e3021b 100644 --- a/webrtc/voice_engine/file_player.h +++ b/webrtc/modules/utility/include/file_player.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_ -#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" @@ -83,5 +83,4 @@ protected: }; } // namespace webrtc - -#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_ diff --git a/webrtc/voice_engine/file_recorder.h b/webrtc/modules/utility/include/file_recorder.h similarity index 91% rename from webrtc/voice_engine/file_recorder.h rename to webrtc/modules/utility/include/file_recorder.h index 001a449b6a..92c91bd4b0 100644 --- a/webrtc/voice_engine/file_recorder.h +++ b/webrtc/modules/utility/include/file_recorder.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ -#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ +#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ +#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" @@ -61,5 +61,4 @@ protected: }; } // namespace webrtc - -#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ +#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ diff --git a/webrtc/voice_engine/coder.cc b/webrtc/modules/utility/source/coder.cc similarity index 98% rename from webrtc/voice_engine/coder.cc rename to webrtc/modules/utility/source/coder.cc index ab724e5cec..f2ae43eb10 100644 --- a/webrtc/voice_engine/coder.cc +++ b/webrtc/modules/utility/source/coder.cc @@ -8,11 +8,10 @@ * 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/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/voice_engine/coder.h b/webrtc/modules/utility/source/coder.h similarity index 93% rename from webrtc/voice_engine/coder.h rename to webrtc/modules/utility/source/coder.h index 41a7c59bbf..5f441904be 100644 --- a/webrtc/voice_engine/coder.h +++ b/webrtc/modules/utility/source/coder.h @@ -8,8 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VOICE_ENGINE_CODER_H_ -#define WEBRTC_VOICE_ENGINE_CODER_H_ +#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ +#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ #include @@ -65,4 +65,4 @@ class AudioCoder : public AudioPacketizationCallback { }; } // namespace webrtc -#endif // WEBRTC_VOICE_ENGINE_CODER_H_ +#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_ diff --git a/webrtc/voice_engine/file_player_impl.cc b/webrtc/modules/utility/source/file_player_impl.cc similarity index 99% rename from webrtc/voice_engine/file_player_impl.cc rename to webrtc/modules/utility/source/file_player_impl.cc index c1239d36e5..e783a7eca8 100644 --- a/webrtc/voice_engine/file_player_impl.cc +++ b/webrtc/modules/utility/source/file_player_impl.cc @@ -8,8 +8,7 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/voice_engine/file_player_impl.h" - +#include "webrtc/modules/utility/source/file_player_impl.h" #include "webrtc/system_wrappers/include/logging.h" namespace webrtc { diff --git a/webrtc/voice_engine/file_player_impl.h b/webrtc/modules/utility/source/file_player_impl.h similarity index 89% rename from webrtc/voice_engine/file_player_impl.h rename to webrtc/modules/utility/source/file_player_impl.h index 82d7daf47c..62887da13b 100644 --- a/webrtc/voice_engine/file_player_impl.h +++ b/webrtc/modules/utility/source/file_player_impl.h @@ -8,18 +8,18 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_ -#define WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_ +#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_ +#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_ #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/common_types.h" #include "webrtc/engine_configurations.h" #include "webrtc/modules/media_file/media_file.h" #include "webrtc/modules/media_file/media_file_defines.h" +#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/critical_section_wrapper.h" #include "webrtc/typedefs.h" -#include "webrtc/voice_engine/coder.h" -#include "webrtc/voice_engine/file_player.h" namespace webrtc { class FilePlayerImpl : public FilePlayer @@ -75,5 +75,4 @@ private: float _scaling; }; } // namespace webrtc - -#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_ +#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_ diff --git a/webrtc/voice_engine/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc similarity index 98% rename from webrtc/voice_engine/file_player_unittests.cc rename to webrtc/modules/utility/source/file_player_unittests.cc index dd440fb750..58471e5e8d 100644 --- a/webrtc/voice_engine/file_player_unittests.cc +++ b/webrtc/modules/utility/source/file_player_unittests.cc @@ -10,6 +10,8 @@ // Unit tests for FilePlayer. +#include "webrtc/modules/utility/include/file_player.h" + #include #include @@ -18,7 +20,6 @@ #include "webrtc/base/md5digest.h" #include "webrtc/base/stringencode.h" #include "webrtc/test/testsupport/fileutils.h" -#include "webrtc/voice_engine/file_player.h" DEFINE_bool(file_player_output, false, "Generate reference files."); diff --git a/webrtc/voice_engine/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc similarity index 99% rename from webrtc/voice_engine/file_recorder_impl.cc rename to webrtc/modules/utility/source/file_recorder_impl.cc index bfdc01d7a5..82b37f0118 100644 --- a/webrtc/voice_engine/file_recorder_impl.cc +++ b/webrtc/modules/utility/source/file_recorder_impl.cc @@ -8,10 +8,9 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/voice_engine/file_recorder_impl.h" - #include "webrtc/engine_configurations.h" #include "webrtc/modules/media_file/media_file.h" +#include "webrtc/modules/utility/source/file_recorder_impl.h" #include "webrtc/system_wrappers/include/logging.h" namespace webrtc { diff --git a/webrtc/voice_engine/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h similarity index 89% rename from webrtc/voice_engine/file_recorder_impl.h rename to webrtc/modules/utility/source/file_recorder_impl.h index 67af742f41..a9dd3a8863 100644 --- a/webrtc/voice_engine/file_recorder_impl.h +++ b/webrtc/modules/utility/source/file_recorder_impl.h @@ -12,8 +12,8 @@ // multiple file formats. The unencoded input data is written to file in the // encoded format specified. -#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_ -#define WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_ +#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ +#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ #include @@ -24,10 +24,10 @@ #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/include/file_recorder.h" +#include "webrtc/modules/utility/source/coder.h" #include "webrtc/system_wrappers/include/event_wrapper.h" #include "webrtc/typedefs.h" -#include "webrtc/voice_engine/coder.h" -#include "webrtc/voice_engine/file_recorder.h" namespace webrtc { // The largest decoded frame size in samples (60ms with 32kHz sample rate). @@ -76,5 +76,4 @@ private: Resampler _audioResampler; }; } // namespace webrtc - -#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_ +#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ diff --git a/webrtc/modules/utility/utility.gypi b/webrtc/modules/utility/utility.gypi index 2c4e20f0da..6e11f1654d 100644 --- a/webrtc/modules/utility/utility.gypi +++ b/webrtc/modules/utility/utility.gypi @@ -20,11 +20,19 @@ ], 'sources': [ 'include/audio_frame_operations.h', + 'include/file_player.h', + 'include/file_recorder.h', 'include/helpers_android.h', 'include/helpers_ios.h', 'include/jvm_android.h', 'include/process_thread.h', 'source/audio_frame_operations.cc', + 'source/coder.cc', + 'source/coder.h', + 'source/file_player_impl.cc', + 'source/file_player_impl.h', + 'source/file_recorder_impl.cc', + 'source/file_recorder_impl.h', 'source/helpers_android.cc', 'source/helpers_ios.mm', 'source/jvm_android.cc', diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index ed34637e38..e330bab4cc 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -9,74 +9,6 @@ import("../build/webrtc.gni") import("//testing/test.gni") -source_set("audio_coder") { - sources = [ - "coder.cc", - "coder.h", - ] - configs += [ "..:common_config" ] - public_configs = [ "..:common_inherited_config" ] - deps = [ - "..:webrtc_common", - "../modules/audio_coding:audio_coding", - "../modules/audio_coding:builtin_audio_decoder_factory", - "../modules/audio_coding:rent_a_codec", - ] - - if (is_clang) { - # Suppress warnings from Chrome's Clang plugins. - # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. - configs -= [ "//build/config/clang:find_bad_constructs" ] - } -} - -source_set("file_player") { - sources = [ - "file_player.h", - "file_player_impl.cc", - "file_player_impl.h", - ] - configs += [ "..:common_config" ] - public_configs = [ "..:common_inherited_config" ] - deps = [ - ":audio_coder", - "..:webrtc_common", - "../common_audio:common_audio", - "../modules/media_file:media_file", - "../system_wrappers:system_wrappers", - ] - - if (is_clang) { - # Suppress warnings from Chrome's Clang plugins. - # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. - configs -= [ "//build/config/clang:find_bad_constructs" ] - } -} - -source_set("file_recorder") { - sources = [ - "file_recorder.h", - "file_recorder_impl.cc", - "file_recorder_impl.h", - ] - configs += [ "..:common_config" ] - public_configs = [ "..:common_inherited_config" ] - deps = [ - ":audio_coder", - "..:webrtc_common", - "../base:rtc_base_approved", - "../common_audio:common_audio", - "../modules/media_file:media_file", - "../system_wrappers:system_wrappers", - ] - - if (is_clang) { - # Suppress warnings from Chrome's Clang plugins. - # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. - configs -= [ "//build/config/clang:find_bad_constructs" ] - } -} - source_set("voice_engine") { sources = [ "channel.cc", @@ -157,8 +89,6 @@ source_set("voice_engine") { } deps = [ - ":file_player", - ":file_recorder", ":level_indicator", "..:rtc_event_log", "..:webrtc_common", @@ -199,7 +129,6 @@ 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", @@ -215,15 +144,10 @@ if (rtc_include_tests) { if (is_android) { deps += [ "//testing/android/native_test:native_test_native_code" ] shard_timeout = 900 - data = [ - "//resources/utility/encapsulated_pcm16b_8khz.wav", - "//resources/utility/encapsulated_pcmu_8khz.wav", - ] } sources = [ "channel_unittest.cc", - "file_player_unittests.cc", "network_predictor_unittest.cc", "transmit_mixer_unittest.cc", "utility_unittest.cc", diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h index 10de18ac65..34e5c5aff5 100644 --- a/webrtc/voice_engine/channel.h +++ b/webrtc/voice_engine/channel.h @@ -26,8 +26,8 @@ #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/voice_engine/file_player.h" -#include "webrtc/voice_engine/file_recorder.h" +#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/voice_engine/include/voe_audio_processing.h" #include "webrtc/voice_engine/include/voe_network.h" #include "webrtc/voice_engine/level_indicator.h" diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h index 9bf3b35c93..ae2f53fdb9 100644 --- a/webrtc/voice_engine/output_mixer.h +++ b/webrtc/voice_engine/output_mixer.h @@ -16,7 +16,7 @@ #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/voice_engine/file_recorder.h" +#include "webrtc/modules/utility/include/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 ebd90a7acd..483af0518a 100644 --- a/webrtc/voice_engine/transmit_mixer.h +++ b/webrtc/voice_engine/transmit_mixer.h @@ -16,8 +16,8 @@ #include "webrtc/common_types.h" #include "webrtc/modules/audio_processing/typing_detection.h" #include "webrtc/modules/include/module_common_types.h" -#include "webrtc/voice_engine/file_player.h" -#include "webrtc/voice_engine/file_recorder.h" +#include "webrtc/modules/utility/include/file_player.h" +#include "webrtc/modules/utility/include/file_recorder.h" #include "webrtc/voice_engine/include/voe_base.h" #include "webrtc/voice_engine/level_indicator.h" #include "webrtc/voice_engine/monitor_module.h" diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp index 8103f8394f..912b5228c1 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/webrtc/voice_engine/voice_engine.gyp @@ -29,8 +29,6 @@ '<(webrtc_root)/modules/modules.gyp:webrtc_utility', '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', '<(webrtc_root)/webrtc.gyp:rtc_event_log', - 'file_player', - 'file_recorder', 'level_indicator', ], 'export_dependent_settings': [ @@ -96,53 +94,6 @@ 'voice_engine_impl.h', ], }, - { - 'target_name': 'audio_coder', - 'type': 'static_library', - 'sources': [ - 'coder.cc', - 'coder.h', - ], - 'dependencies': [ - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/modules/modules.gyp:audio_coding_module', - '<(webrtc_root)/modules/modules.gyp:builtin_audio_decoder_factory', - '<(webrtc_root)/modules/modules.gyp:rent_a_codec', - ], - }, - { - 'target_name': 'file_player', - 'type': 'static_library', - 'sources': [ - 'file_player.h', - 'file_player_impl.cc', - 'file_player_impl.h', - ], - 'dependencies': [ - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', - '<(webrtc_root)/modules/modules.gyp:media_file', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - 'audio_coder', - ], - }, - { - 'target_name': 'file_recorder', - 'type': 'static_library', - 'sources': [ - 'file_recorder.h', - 'file_recorder_impl.cc', - 'file_recorder_impl.h', - ], - 'dependencies': [ - '<(webrtc_root)/base/base.gyp:rtc_base_approved', - '<(webrtc_root)/common.gyp:webrtc_common', - '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', - '<(webrtc_root)/modules/modules.gyp:media_file', - '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', - 'audio_coder', - ], - }, { 'target_name': 'level_indicator', 'type': 'static_library', @@ -170,7 +121,6 @@ 'voice_engine', '<(DEPTH)/testing/gmock.gyp:gmock', '<(DEPTH)/testing/gtest.gyp:gtest', - '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', # The rest are to satisfy the unittests' include chain. # This would be unnecessary if we used qualified includes. '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', @@ -186,7 +136,6 @@ ], 'sources': [ 'channel_unittest.cc', - 'file_player_unittests.cc', 'network_predictor_unittest.cc', 'transmit_mixer_unittest.cc', 'utility_unittest.cc', @@ -203,12 +152,6 @@ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code', ], }], - ['OS=="ios"', { - 'mac_bundle_resources': [ - '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav', - '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav', - ], - }], ], }, { diff --git a/webrtc/voice_engine/voice_engine_unittests.isolate b/webrtc/voice_engine/voice_engine_unittests.isolate index 5541c4af07..0d55515f99 100644 --- a/webrtc/voice_engine/voice_engine_unittests.isolate +++ b/webrtc/voice_engine/voice_engine_unittests.isolate @@ -19,13 +19,5 @@ ], }, }], - ['OS=="linux" or OS=="mac" or OS=="win" or OS=="android"', { - 'variables': { - 'files': [ - '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav', - '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav', - ], - }, - }], ], }