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 fa4c8ff71fe1e7c77c3e4fc75a98c7149b9d8bef..d001734661e42b4770a5f189c45a13fe58b4b14a 100644 |
--- a/content/renderer/media/media_stream_audio_processor.h |
+++ b/content/renderer/media/media_stream_audio_processor.h |
@@ -5,6 +5,8 @@ |
#ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
#define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_ |
+#include <memory> |
+ |
#include "base/atomicops.h" |
#include "base/files/file.h" |
#include "base/gtest_prod_util.h" |
@@ -19,6 +21,7 @@ |
#include "content/public/common/media_stream_request.h" |
#include "content/renderer/media/aec_dump_message_filter.h" |
#include "content/renderer/media/audio_repetition_detector.h" |
+#include "content/renderer/media/media_stream_audio_processor_options.h" |
#include "content/renderer/media/webrtc_audio_device_impl.h" |
#include "media/base/audio_converter.h" |
#include "third_party/webrtc/api/mediastreaminterface.h" |
@@ -34,10 +37,6 @@ |
#endif |
#endif |
-namespace blink { |
-class WebMediaConstraints; |
-} |
- |
namespace media { |
class AudioBus; |
class AudioParameters; |
@@ -70,10 +69,8 @@ class CONTENT_EXPORT MediaStreamAudioProcessor : |
// |
// Threading note: The constructor assumes it is being run on the main render |
// thread. |
- MediaStreamAudioProcessor( |
- const blink::WebMediaConstraints& constraints, |
- const MediaStreamDevice::AudioDeviceParameters& input_params, |
- WebRtcPlayoutDataSource* playout_data_source); |
+ MediaStreamAudioProcessor(const AudioProcessingProperties& properties, |
+ WebRtcPlayoutDataSource* playout_data_source); |
// Called when the format of the capture data has changed. |
// Called on the main render thread. The caller is responsible for stopping |
@@ -126,15 +123,10 @@ class CONTENT_EXPORT MediaStreamAudioProcessor : |
void OnIpcClosing() override; |
// Returns true if MediaStreamAudioProcessor would modify the audio signal, |
- // based on the |constraints| and |effects_flags| parsed from a user media |
- // request. If the audio signal would not be modified, there is no need to |
- // instantiate a MediaStreamAudioProcessor and feed audio through it. Doing so |
- // would waste a non-trivial amount of memory and CPU resources. |
- // |
- // See media::AudioParameters::PlatformEffectsMask for interpretation of |
- // |effects_flags|. |
- static bool WouldModifyAudio(const blink::WebMediaConstraints& constraints, |
- int effects_flags); |
+ // based on |properties|. If the audio signal would not be modified, there is |
+ // no need to instantiate a MediaStreamAudioProcessor and feed audio through |
+ // it. Doing so would waste a non-trivial amount of memory and CPU resources. |
+ static bool WouldModifyAudio(const AudioProcessingProperties& properties); |
protected: |
~MediaStreamAudioProcessor() override; |
@@ -158,8 +150,7 @@ class CONTENT_EXPORT MediaStreamAudioProcessor : |
// Helper to initialize the WebRtc AudioProcessing. |
void InitializeAudioProcessingModule( |
- const blink::WebMediaConstraints& constraints, |
- const MediaStreamDevice::AudioDeviceParameters& input_params); |
+ const AudioProcessingProperties& properties); |
// Helper to initialize the capture converter. |
void InitializeCaptureFifo(const media::AudioParameters& input_format); |