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 e2033d63ee7ecf79340c692c8fd77675e85f39b9..8d28008ae785c94940ac35c00f2d3802d079852b 100644 |
--- a/content/renderer/media/media_stream_audio_processor.h |
+++ b/content/renderer/media/media_stream_audio_processor.h |
@@ -11,6 +11,7 @@ |
#include "base/threading/thread_checker.h" |
#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/webrtc_audio_device_impl.h" |
#include "media/base/audio_converter.h" |
@@ -52,9 +53,10 @@ class CONTENT_EXPORT MediaStreamAudioProcessor : |
// |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, |
// |playout_data_source| won't be used. |
- MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints, |
- int effects, |
- WebRtcPlayoutDataSource* playout_data_source); |
+ MediaStreamAudioProcessor( |
+ const blink::WebMediaConstraints& constraints, |
+ const MediaStreamDevice::AudioDeviceParameters& input_params, |
+ 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 |
@@ -125,7 +127,8 @@ class CONTENT_EXPORT MediaStreamAudioProcessor : |
// Helper to initialize the WebRtc AudioProcessing. |
void InitializeAudioProcessingModule( |
- const blink::WebMediaConstraints& constraints, int effects); |
+ const blink::WebMediaConstraints& constraints, |
+ const MediaStreamDevice::AudioDeviceParameters& input_params); |
// Helper to initialize the capture converter. |
void InitializeCaptureFifo(const media::AudioParameters& input_format); |