Revert of Move FilePlayer and FileRecorder to Voice Engine (patchset #6 id:100001 of https://codereview.webrtc.org/2240163002/ ) Reason for revert: Breaks downstream code, so revert again. Yay. Original issue's description: > Move FilePlayer and FileRecorder to Voice Engine > > Because Voice Engine was the only user. > > (This is a re-land of https://codereview.webrtc.org/2037623002, which > had to be reverted.) > > NOPRESUBMIT=True > > Committed: https://crrev.com/dc65ea29b3270ad418050658ad962ddd33ee70c1 > Cr-Commit-Position: refs/heads/master@{#13757} TBR=perkj@webrtc.org,kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2245153002 Cr-Commit-Position: refs/heads/master@{#13758}
diff --git a/.gn b/.gn index bf1a0b5..e151f7c 100644 --- a/.gn +++ b/.gn
@@ -19,12 +19,7 @@ # their includes checked for proper dependencies when you run either # "gn check" or "gn gen --check". # TODO(kjellander): Keep adding paths to this list as work in webrtc:5589 is done. -check_targets = [ - "//webrtc/voice_engine:audio_coder", - "//webrtc/voice_engine:file_player", - "//webrtc/voice_engine:file_recorder", - "//webrtc/voice_engine:level_indicator", -] +check_targets = [ "//webrtc/voice_engine:level_indicator" ] # These are the list of GN files that run exec_script. This whitelist exists # to force additional review for new uses of exec_script, which is strongly
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index 01d4ea5..676160a 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn
@@ -306,6 +306,7 @@ "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", @@ -595,6 +596,8 @@ "//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 eeeb193..78cd4e5 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 68f7a51..094204f 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 933478d..af7e6ef 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 c3c9f0a..5437e4f 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn
@@ -11,10 +11,18 @@ 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 93% rename from webrtc/voice_engine/file_player.h rename to webrtc/modules/utility/include/file_player.h index 898d66c..b064e30 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 @@ }; } // 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 001a449..92c91bd 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 @@ }; } // 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 ab724e5..f2ae43e 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 92% rename from webrtc/voice_engine/coder.h rename to webrtc/modules/utility/source/coder.h index 41a7c59..5f44190 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 <memory> @@ -65,4 +65,4 @@ }; } // 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 c1239d3..e783a7e 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 88% rename from webrtc/voice_engine/file_player_impl.h rename to webrtc/modules/utility/source/file_player_impl.h index 82d7daf..62887da 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 @@ 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 dd440fb..58471e5 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 <stdio.h> #include <string> @@ -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 98% rename from webrtc/voice_engine/file_recorder_impl.cc rename to webrtc/modules/utility/source/file_recorder_impl.cc index bfdc01d..82b37f0 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 67af742..a9dd3a8 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 <list> @@ -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 @@ 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 2c4e20f..6e11f16 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 85084cf..e330bab 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 = [ - "../modules/audio_coding:audio_coding", - "../modules/audio_coding:builtin_audio_decoder_factory", - "../modules/audio_coding:rent_a_codec", - "..:webrtc_common", - ] - - 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 = [ - "../common_audio:common_audio", - "../modules/media_file:media_file", - "../system_wrappers:system_wrappers", - "..:webrtc_common", - ":audio_coder", - ] - - 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 = [ - "../base:rtc_base_approved", - "../common_audio:common_audio", - "../modules/media_file:media_file", - "../system_wrappers:system_wrappers", - "..:webrtc_common", - ":audio_coder", - ] - - 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 @@ } deps = [ - ":file_player", - ":file_recorder", ":level_indicator", "..:rtc_event_log", "..:webrtc_common", @@ -199,7 +129,6 @@ ":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 (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 10de18a..34e5c5a 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 9bf3b35..ae2f53f 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 ebd90a7..483af05 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 336bb3a..912b522 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': [ @@ -97,38 +95,6 @@ ], }, { - 'target_name': 'audio_coder', - 'type': 'static_library', - 'sources': [ - 'coder.cc', - 'coder.h', - ], - }, - { - 'target_name': 'file_player', - 'type': 'static_library', - 'sources': [ - 'file_player.h', - 'file_player_impl.cc', - 'file_player_impl.h', - ], - 'dependencies': [ - 'audio_coder', - ], - }, - { - 'target_name': 'file_recorder', - 'type': 'static_library', - 'sources': [ - 'file_recorder.h', - 'file_recorder_impl.cc', - 'file_recorder_impl.h', - ], - 'dependencies': [ - 'audio_coder', - ], - }, - { 'target_name': 'level_indicator', 'type': 'static_library', 'dependencies': [ @@ -155,7 +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', @@ -171,7 +136,6 @@ ], 'sources': [ 'channel_unittest.cc', - 'file_player_unittests.cc', 'network_predictor_unittest.cc', 'transmit_mixer_unittest.cc', 'utility_unittest.cc', @@ -188,12 +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 5541c4a..0d55515 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', - ], - }, - }], ], }