| Index: content/renderer/media/media_stream_audio_processor.h
|
| diff --git a/content/renderer/media/media_stream_audio_processor.h b/content/renderer/media/media_stream_audio_processor.h
|
| index 4b74ca0a24090d73f7649b93a36dce24d2f9c62b..69143c9ca166f1dd3d113d6a5f0479b4ddb2183a 100644
|
| --- a/content/renderer/media/media_stream_audio_processor.h
|
| +++ b/content/renderer/media/media_stream_audio_processor.h
|
| @@ -13,7 +13,7 @@
|
| #include "base/time/time.h"
|
| #include "content/common/content_export.h"
|
| #include "content/public/common/media_stream_request.h"
|
| -#include "content/renderer/media/aec_dump_message_filter.h"
|
| +#include "content/renderer/media/audio_debug_recorder.h"
|
| #include "content/renderer/media/webrtc_audio_device_impl.h"
|
| #include "media/base/audio_converter.h"
|
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| @@ -46,10 +46,10 @@ using webrtc::AudioProcessorInterface;
|
| // processing components like AGC, AEC and NS. It enables the components based
|
| // on the getUserMedia constraints, processes the data and outputs it in a unit
|
| // of 10 ms data chunk.
|
| -class CONTENT_EXPORT MediaStreamAudioProcessor :
|
| - NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
|
| - NON_EXPORTED_BASE(public AudioProcessorInterface),
|
| - NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
|
| +class CONTENT_EXPORT MediaStreamAudioProcessor
|
| + : NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
|
| + NON_EXPORTED_BASE(public AudioProcessorInterface),
|
| + NON_EXPORTED_BASE(public AudioDebugRecorder::AecDumpDelegate) {
|
| public:
|
| // |playout_data_source| is used to register this class as a sink to the
|
| // WebRtc playout data for processing AEC. If clients do not enable AEC,
|
| @@ -104,7 +104,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
|
|
|
| // AecDumpMessageFilter::AecDumpDelegate implementation.
|
| // Called on the main render thread.
|
| - void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
|
| + void OnAecDumpFile(base::PlatformFile file_handle) override;
|
| void OnDisableAecDump() override;
|
| void OnIpcClosing() override;
|
|
|
| @@ -114,7 +114,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
|
| private:
|
| friend class MediaStreamAudioProcessorTest;
|
| FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
|
| - GetAecDumpMessageFilter);
|
| + GetAudioDebugRecorder);
|
|
|
| // WebRtcPlayoutDataSource::Sink implementation.
|
| void OnPlayoutData(media::AudioBus* audio_bus,
|
| @@ -193,7 +193,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor :
|
| base::subtle::Atomic32 typing_detected_;
|
|
|
| // Communication with browser for AEC dump.
|
| - scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
|
| + scoped_refptr<AudioDebugRecorder> audio_debug_recorder_;
|
|
|
| // Flag to avoid executing Stop() more than once.
|
| bool stopped_;
|
|
|