diff --git a/.gn b/.gn index 75e2587cee..3a744bf7bd 100644 --- a/.gn +++ b/.gn @@ -21,6 +21,9 @@ 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 676160ae65..01d4ea5689 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -306,7 +306,6 @@ 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", @@ -596,8 +595,6 @@ 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", diff --git a/webrtc/modules/audio_mixer/audio_mixer.h b/webrtc/modules/audio_mixer/audio_mixer.h index 78cd4e5c79..eeeb193b3a 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/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/modules/modules.gyp b/webrtc/modules/modules.gyp index 094204f723..68f7a5112e 100644 --- a/webrtc/modules/modules.gyp +++ b/webrtc/modules/modules.gyp @@ -358,7 +358,6 @@ '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', @@ -599,8 +598,6 @@ '<(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 af7e6ef46e..933478d434 100644 --- a/webrtc/modules/modules_unittests.isolate +++ b/webrtc/modules/modules_unittests.isolate @@ -110,8 +110,6 @@ '<(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 5437e4f5f7..c3c9f0a923 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -11,18 +11,10 @@ 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/modules/utility/utility.gypi b/webrtc/modules/utility/utility.gypi index 6e11f1654d..2c4e20f0da 100644 --- a/webrtc/modules/utility/utility.gypi +++ b/webrtc/modules/utility/utility.gypi @@ -20,19 +20,11 @@ ], '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 e330bab4cc..ed34637e38 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -9,6 +9,74 @@ 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", @@ -89,6 +157,8 @@ source_set("voice_engine") { } deps = [ + ":file_player", + ":file_recorder", ":level_indicator", "..:rtc_event_log", "..:webrtc_common", @@ -129,6 +199,7 @@ 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", @@ -144,10 +215,15 @@ 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 34e5c5aff5..10de18ac65 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/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_network.h" #include "webrtc/voice_engine/level_indicator.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 f2ae43eb10..ab724e5cec 100644 --- a/webrtc/modules/utility/source/coder.cc +++ b/webrtc/voice_engine/coder.cc @@ -8,10 +8,11 @@ * 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/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 5f441904be..41a7c59bbf 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/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 b064e3021b..898d66cd4d 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 "webrtc/common_types.h" #include "webrtc/engine_configurations.h" @@ -83,4 +83,5 @@ protected: }; } // 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_impl.cc b/webrtc/voice_engine/file_player_impl.cc similarity index 99% rename from webrtc/modules/utility/source/file_player_impl.cc rename to webrtc/voice_engine/file_player_impl.cc index e783a7eca8..c1239d36e5 100644 --- a/webrtc/modules/utility/source/file_player_impl.cc +++ b/webrtc/voice_engine/file_player_impl.cc @@ -8,7 +8,8 @@ * be found in the AUTHORS file in the root of the source tree. */ -#include "webrtc/modules/utility/source/file_player_impl.h" +#include "webrtc/voice_engine/file_player_impl.h" + #include "webrtc/system_wrappers/include/logging.h" namespace webrtc { diff --git a/webrtc/modules/utility/source/file_player_impl.h b/webrtc/voice_engine/file_player_impl.h similarity index 89% rename from webrtc/modules/utility/source/file_player_impl.h rename to webrtc/voice_engine/file_player_impl.h index 62887da13b..82d7daf47c 100644 --- a/webrtc/modules/utility/source/file_player_impl.h +++ b/webrtc/voice_engine/file_player_impl.h @@ -8,18 +8,18 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_ -#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_ +#ifndef WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_H_ +#define WEBRTC_VOICE_ENGINE_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 @@ -75,4 +75,5 @@ private: float _scaling; }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_ + +#endif // WEBRTC_VOICE_ENGINE_FILE_PLAYER_IMPL_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 58471e5e8d..dd440fb750 100644 --- a/webrtc/modules/utility/source/file_player_unittests.cc +++ b/webrtc/voice_engine/file_player_unittests.cc @@ -10,8 +10,6 @@ // Unit tests for FilePlayer. -#include "webrtc/modules/utility/include/file_player.h" - #include #include @@ -20,6 +18,7 @@ #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/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 92c91bd4b0..001a449b6a 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 "webrtc/common_types.h" #include "webrtc/engine_configurations.h" @@ -61,4 +61,5 @@ protected: }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_ + +#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_H_ diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/voice_engine/file_recorder_impl.cc similarity index 99% rename from webrtc/modules/utility/source/file_recorder_impl.cc rename to webrtc/voice_engine/file_recorder_impl.cc index 82b37f0118..bfdc01d7a5 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.cc +++ b/webrtc/voice_engine/file_recorder_impl.cc @@ -8,9 +8,10 @@ * 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/modules/utility/source/file_recorder_impl.h b/webrtc/voice_engine/file_recorder_impl.h similarity index 89% rename from webrtc/modules/utility/source/file_recorder_impl.h rename to webrtc/voice_engine/file_recorder_impl.h index a9dd3a8863..67af742f41 100644 --- a/webrtc/modules/utility/source/file_recorder_impl.h +++ b/webrtc/voice_engine/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_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ -#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ +#ifndef WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_ +#define WEBRTC_VOICE_ENGINE_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). @@ -76,4 +76,5 @@ private: Resampler _audioResampler; }; } // namespace webrtc -#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ + +#endif // WEBRTC_VOICE_ENGINE_FILE_RECORDER_IMPL_H_ diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h index ae2f53fdb9..9bf3b35c93 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/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 483af0518a..ebd90a7acd 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/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" diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp index 912b5228c1..8103f8394f 100644 --- a/webrtc/voice_engine/voice_engine.gyp +++ b/webrtc/voice_engine/voice_engine.gyp @@ -29,6 +29,8 @@ '<(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': [ @@ -94,6 +96,53 @@ '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', @@ -121,6 +170,7 @@ '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', @@ -136,6 +186,7 @@ ], 'sources': [ 'channel_unittest.cc', + 'file_player_unittests.cc', 'network_predictor_unittest.cc', 'transmit_mixer_unittest.cc', 'utility_unittest.cc', @@ -152,6 +203,12 @@ '<(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 0d55515f99..5541c4af07 100644 --- a/webrtc/voice_engine/voice_engine_unittests.isolate +++ b/webrtc/voice_engine/voice_engine_unittests.isolate @@ -19,5 +19,13 @@ ], }, }], + ['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', + ], + }, + }], ], }