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author | Alex Mineer <amineer@chromium.org> | 2014-11-14 09:13:38 -0800 |
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committer | Alex Mineer <amineer@chromium.org> | 2014-11-14 17:14:33 +0000 |
commit | e5e683b7c41ca32b42486b0de4bbcdd106ee9e59 (patch) | |
tree | 6e2e71a98648356f4d0c1dcde9e3811dcf98504b | |
parent | ddcb7a2971fd021bec923e9377f7f1093a28bf20 (diff) | |
download | chromium_src-e5e683b7c41ca32b42486b0de4bbcdd106ee9e59.zip chromium_src-e5e683b7c41ca32b42486b0de4bbcdd106ee9e59.tar.gz chromium_src-e5e683b7c41ca32b42486b0de4bbcdd106ee9e59.tar.bz2 |
Revert "Merge 663413002 to M39:"
This reverts commit 255ee24a13c30cdd0e6cd75db897f03f35bc0674.
R=tommi@chromium.org
Revert of Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome (patchset #4 id:60001 of https://codereview.chromium.org/663413002/)
Reason for revert:
This CL broke the configuration of AudioProcessing, we have to revert it to fix the echo issues it introduces.
Original issue's description:
> Reland 597923002: Fix the way how we create webrtc::AudioProcessing in Chrome.
>
> The original review thread is in https://codereview.chromium.org/588523002/
>
> Fix the way how we create webrtc::AudioProcessing in Chrome.
>
> TBR=tommi@chromium.org,maruel@chromium.org
>
> BUG=415935
> TEST=all webrtc tests in all bots + manual test to verify the agc loggings exist.
Review URL: https://codereview.chromium.org/730563002
(cherry picked from commit e78feb7865c15d596a428ba892f842c653995615)
Cr-Original-Commit-Position: refs/branch-heads/2171@{#419}
Cr-Original-Branched-From: 267aeeb8d85c8503a7fd12bd14654b8ea78d3974-refs/heads/master@{#297060}
Cr-Commit-Position: refs/branch-heads/2171_62@{#2}
Cr-Branched-From: 6ac5b3b7a7861707e68ff88605ec7e2e1dd3f941-refs/branch-heads/2171@{#415}
Cr-Branched-From: 267aeeb8d85c8503a7fd12bd14654b8ea78d3974-refs/heads/master@{#297060}
-rw-r--r-- | build/android/pylib/gtest/setup.py | 1 | ||||
-rw-r--r-- | content/content_tests.gypi | 15 | ||||
-rw-r--r-- | content/content_unittests.isolate | 14 | ||||
-rw-r--r-- | content/renderer/media/media_stream_audio_processor.cc | 3 | ||||
-rw-r--r-- | content/renderer/media/media_stream_audio_processor_options.cc | 11 | ||||
-rw-r--r-- | third_party/libjingle/BUILD.gn | 1 | ||||
-rw-r--r-- | third_party/libjingle/libjingle.gyp | 1 | ||||
-rw-r--r-- | third_party/libjingle/overrides/init_webrtc.cc | 22 | ||||
-rw-r--r-- | third_party/libjingle/overrides/init_webrtc.h | 13 | ||||
-rw-r--r-- | third_party/libjingle/overrides/initialize_module.cc | 6 |
10 files changed, 13 insertions, 74 deletions
diff --git a/build/android/pylib/gtest/setup.py b/build/android/pylib/gtest/setup.py index 2782859..6a9e65b 100644 --- a/build/android/pylib/gtest/setup.py +++ b/build/android/pylib/gtest/setup.py @@ -118,7 +118,6 @@ def _GenerateDepsDirUsingIsolate(suite_name, isolate_file_path=None): '--config-variable', 'component', 'static_library', '--config-variable', 'fastbuild', '0', '--config-variable', 'icu_use_data_file_flag', '1', - '--config-variable', 'libpeer_target_type', 'static_library', # TODO(maruel): This may not be always true. '--config-variable', 'target_arch', 'arm', '--config-variable', 'use_openssl', '0', diff --git a/content/content_tests.gypi b/content/content_tests.gypi index ba9da06..e311974 100644 --- a/content/content_tests.gypi +++ b/content/content_tests.gypi @@ -838,21 +838,6 @@ '../third_party/webrtc/modules/modules.gyp:desktop_capture', ], }], - ['enable_webrtc==1 and OS=="mac"', { - 'variables': { - 'libpeer_target_type%': 'static_library', - }, - 'conditions': [ - ['libpeer_target_type!="static_library"', { - 'copies': [{ - 'destination': '<(PRODUCT_DIR)/Libraries', - 'files': [ - '<(PRODUCT_DIR)/libpeerconnection.so', - ], - }], - }], - ], - }], ['enable_webrtc==1 and chromeos==1', { 'sources': [ 'browser/media/capture/desktop_capture_device_aura_unittest.cc', diff --git a/content/content_unittests.isolate b/content/content_unittests.isolate index 87a4ee9..7f4f9c4 100644 --- a/content/content_unittests.isolate +++ b/content/content_unittests.isolate @@ -56,13 +56,6 @@ ], }, }], - ['OS=="linux" and libpeer_target_type=="loadable_module"', { - 'variables': { - 'isolate_dependency_tracked': [ - '<(PRODUCT_DIR)/lib/libpeerconnection.so', - ], - }, - }], ['OS=="mac"', { 'variables': { 'command': [ @@ -98,13 +91,6 @@ ], }, }], - ['OS=="win" and libpeer_target_type=="loadable_module"', { - 'variables': { - 'isolate_dependency_tracked': [ - '<(PRODUCT_DIR)/libpeerconnection.dll', - ], - }, - }], ], 'includes': [ '../base/base.isolate', diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc index ac41187..4efc507 100644 --- a/content/renderer/media/media_stream_audio_processor.cc +++ b/content/renderer/media/media_stream_audio_processor.cc @@ -19,7 +19,6 @@ #include "media/base/audio_fifo.h" #include "media/base/channel_layout.h" #include "third_party/WebKit/public/platform/WebMediaConstraints.h" -#include "third_party/libjingle/overrides/init_webrtc.h" #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h" #include "third_party/webrtc/modules/audio_processing/typing_detection.h" @@ -424,7 +423,7 @@ void MediaStreamAudioProcessor::InitializeAudioProcessingModule( #endif // Create and configure the webrtc::AudioProcessing. - audio_processing_.reset(CreateWebRtcAudioProcessing(config)); + audio_processing_.reset(webrtc::AudioProcessing::Create(config)); // Enable the audio processing components. if (echo_cancellation) { diff --git a/content/renderer/media/media_stream_audio_processor_options.cc b/content/renderer/media/media_stream_audio_processor_options.cc index 8462a65..e386308 100644 --- a/content/renderer/media/media_stream_audio_processor_options.cc +++ b/content/renderer/media/media_stream_audio_processor_options.cc @@ -251,10 +251,15 @@ void EnableTypingDetection(AudioProcessing* audio_processing, void StartEchoCancellationDump(AudioProcessing* audio_processing, base::File aec_dump_file) { DCHECK(aec_dump_file.IsValid()); - if (audio_processing->StartDebugRecordingForPlatformFile( - aec_dump_file.TakePlatformFile())) { - DLOG(ERROR) << "Fail to start AEC debug recording"; + + FILE* stream = base::FileToFILE(aec_dump_file.Pass(), "w"); + if (!stream) { + LOG(ERROR) << "Failed to open AEC dump file"; + return; } + + if (audio_processing->StartDebugRecording(stream)) + DLOG(ERROR) << "Fail to start AEC debug recording"; } void StopEchoCancellationDump(AudioProcessing* audio_processing) { diff --git a/third_party/libjingle/BUILD.gn b/third_party/libjingle/BUILD.gn index 69a744e..02c2bf9 100644 --- a/third_party/libjingle/BUILD.gn +++ b/third_party/libjingle/BUILD.gn @@ -548,7 +548,6 @@ if (enable_webrtc) { deps = [ ":libjingle_webrtc_common", "//third_party/webrtc", - "//third_party/webrtc/modules/audio_processing", "//third_party/webrtc/system_wrappers", "//third_party/webrtc/voice_engine", ] diff --git a/third_party/libjingle/libjingle.gyp b/third_party/libjingle/libjingle.gyp index 63a5230..0eaf86c 100644 --- a/third_party/libjingle/libjingle.gyp +++ b/third_party/libjingle/libjingle.gyp @@ -589,7 +589,6 @@ '<(libjingle_source)/talk/media/webrtc/webrtcvoiceengine.h', ], 'dependencies': [ - '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing', '<(DEPTH)/third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers', '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine', '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc', diff --git a/third_party/libjingle/overrides/init_webrtc.cc b/third_party/libjingle/overrides/init_webrtc.cc index 0004d8e..ab89d58 100644 --- a/third_party/libjingle/overrides/init_webrtc.cc +++ b/third_party/libjingle/overrides/init_webrtc.cc @@ -11,8 +11,6 @@ #include "base/metrics/field_trial.h" #include "base/native_library.h" #include "base/path_service.h" -#include "third_party/webrtc/common.h" -#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/logging.h" @@ -55,13 +53,6 @@ bool InitializeWebRtcModule() { return true; } -webrtc::AudioProcessing* CreateWebRtcAudioProcessing( - const webrtc::Config& config) { - // libpeerconnection is being compiled as a static lib, use - // webrtc::AudioProcessing directly. - return webrtc::AudioProcessing::Create(config); -} - #else // !LIBPEERCONNECTION_LIB // When being compiled as a shared library, we need to bridge the gap between @@ -71,7 +62,6 @@ webrtc::AudioProcessing* CreateWebRtcAudioProcessing( // Global function pointers to the factory functions in the shared library. CreateWebRtcMediaEngineFunction g_create_webrtc_media_engine = NULL; DestroyWebRtcMediaEngineFunction g_destroy_webrtc_media_engine = NULL; -CreateWebRtcAudioProcessingFunction g_create_webrtc_audio_processing = NULL; // Returns the full or relative path to the libpeerconnection module depending // on what platform we're on. @@ -145,8 +135,8 @@ bool InitializeWebRtcModule() { &AddTraceEvent, &g_create_webrtc_media_engine, &g_destroy_webrtc_media_engine, - &init_diagnostic_logging, - &g_create_webrtc_audio_processing); + &init_diagnostic_logging); + if (init_ok) rtc::SetExtraLoggingInit(init_diagnostic_logging); return init_ok; @@ -170,12 +160,4 @@ void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) { g_destroy_webrtc_media_engine(media_engine); } -webrtc::AudioProcessing* CreateWebRtcAudioProcessing( - const webrtc::Config& config) { - // The same as CreateWebRtcMediaEngine(), we call InitializeWebRtcModule here - // for convenience of tests. - InitializeWebRtcModule(); - return g_create_webrtc_audio_processing(config); -} - #endif // LIBPEERCONNECTION_LIB diff --git a/third_party/libjingle/overrides/init_webrtc.h b/third_party/libjingle/overrides/init_webrtc.h index 4d06e9e..c5c190c 100644 --- a/third_party/libjingle/overrides/init_webrtc.h +++ b/third_party/libjingle/overrides/init_webrtc.h @@ -23,8 +23,6 @@ class WebRtcVideoEncoderFactory; namespace webrtc { class AudioDeviceModule; -class AudioProcessing; -class Config; } // namespace webrtc typedef std::string (*FieldTrialFindFullName)(const std::string& trial_name); @@ -41,9 +39,6 @@ typedef void (*DestroyWebRtcMediaEngineFunction)( typedef void (*InitDiagnosticLoggingDelegateFunctionFunction)( void (*DelegateFunction)(const std::string&)); -typedef webrtc::AudioProcessing* (*CreateWebRtcAudioProcessingFunction)( - const webrtc::Config& config); - // A typedef for the main initialize function in libpeerconnection. // This will initialize logging in the module with the proper arguments // as well as provide pointers back to a couple webrtc factory functions. @@ -61,8 +56,7 @@ typedef bool (*InitializeModuleFunction)( webrtc::AddTraceEventPtr trace_add_trace_event, CreateWebRtcMediaEngineFunction* create_media_engine, DestroyWebRtcMediaEngineFunction* destroy_media_engine, - InitDiagnosticLoggingDelegateFunctionFunction* init_diagnostic_logging, - CreateWebRtcAudioProcessingFunction* create_audio_processing); + InitDiagnosticLoggingDelegateFunctionFunction* init_diagnostic_logging); #if !defined(LIBPEERCONNECTION_IMPLEMENTATION) // Load and initialize the shared WebRTC module (libpeerconnection). @@ -71,11 +65,6 @@ typedef bool (*InitializeModuleFunction)( // If not called explicitly, this function will still be called from the main // CreateWebRtcMediaEngine factory function the first time it is called. bool InitializeWebRtcModule(); - -// Return a webrtc::AudioProcessing object. -webrtc::AudioProcessing* CreateWebRtcAudioProcessing( - const webrtc::Config& config); - #endif #endif // THIRD_PARTY_LIBJINGLE_OVERRIDES_INIT_WEBRTC_H_ diff --git a/third_party/libjingle/overrides/initialize_module.cc b/third_party/libjingle/overrides/initialize_module.cc index b84e2d8..ce11567 100644 --- a/third_party/libjingle/overrides/initialize_module.cc +++ b/third_party/libjingle/overrides/initialize_module.cc @@ -8,7 +8,6 @@ #include "base/logging.h" #include "init_webrtc.h" #include "talk/media/webrtc/webrtcmediaengine.h" -#include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/base/basictypes.h" #include "webrtc/base/logging.h" @@ -72,9 +71,7 @@ bool InitializeModule(const CommandLine& command_line, CreateWebRtcMediaEngineFunction* create_media_engine, DestroyWebRtcMediaEngineFunction* destroy_media_engine, InitDiagnosticLoggingDelegateFunctionFunction* - init_diagnostic_logging, - CreateWebRtcAudioProcessingFunction* - create_audio_processing) { + init_diagnostic_logging) { #if !defined(OS_MACOSX) && !defined(OS_ANDROID) g_alloc = alloc; g_dealloc = dealloc; @@ -85,7 +82,6 @@ bool InitializeModule(const CommandLine& command_line, *create_media_engine = &CreateWebRtcMediaEngine; *destroy_media_engine = &DestroyWebRtcMediaEngine; *init_diagnostic_logging = &rtc::InitDiagnosticLoggingDelegateFunction; - *create_audio_processing = &webrtc::AudioProcessing::Create; if (CommandLine::Init(0, NULL)) { #if !defined(OS_WIN) |