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Unified Diff: content/renderer/media/webrtc_audio_capturer.h

Issue 99033003: Enable platform echo cancellation through the AudioRecord path. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Rebase. Created 7 years ago
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Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index 580e9fdcf6ed2ea905ea36682fae0502d6eaecb7..663f63d3e5740e1e21fadf887c80fa1af526b54a 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -55,7 +55,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
int session_id,
const std::string& device_id,
int paired_output_sample_rate,
- int paired_output_frames_per_buffer);
+ int paired_output_frames_per_buffer,
+ bool use_platform_aec);
Ami GONE FROM CHROMIUM 2013/12/06 19:22:53 I'm going to pretty sad if each platform effect me
ajm 2013/12/06 19:31:24 Agreed, I was thinking the same thing after readin
// Add a audio track to the sinks of the capturer.
// WebRtcAudioDeviceImpl calls this method on the main render thread but
@@ -78,7 +79,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
void SetCapturerSource(
const scoped_refptr<media::AudioCapturerSource>& source,
media::ChannelLayout channel_layout,
- float sample_rate);
+ float sample_rate,
+ bool use_platform_aec);
// Called when a stream is connecting to a peer connection. This will set
// up the native buffer size for the stream in order to optimize the
@@ -141,7 +143,8 @@ class CONTENT_EXPORT WebRtcAudioCapturer
// Reconfigures the capturer with a new capture parameters.
// Must be called without holding the lock.
- void Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
+ void Reconfigure(int sample_rate, media::ChannelLayout channel_layout,
+ bool use_platform_aec);
// Starts recording audio.
// Triggered by AddSink() on the main render thread or a Libjingle working

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