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

Issue 23691038: Switch LiveAudio to source provider solution. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: addressed Tommi's comments and added unittest Created 7 years, 3 months ago
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Index: content/renderer/media/webaudio_capturer_source.h
diff --git a/content/renderer/media/webaudio_capturer_source.h b/content/renderer/media/webaudio_capturer_source.h
index 68bd05fbe03b8e396c436ce2426853b79e8f7dab..b438a4083ee1d5ff208a8c132bf0ac8346ce0247 100644
--- a/content/renderer/media/webaudio_capturer_source.h
+++ b/content/renderer/media/webaudio_capturer_source.h
@@ -7,6 +7,7 @@
#include "base/memory/ref_counted.h"
#include "base/synchronization/lock.h"
+#include "base/threading/thread_checker.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_capturer_source.h"
#include "media/base/audio_fifo.h"
@@ -15,50 +16,59 @@
namespace content {
-class WebRtcAudioCapturer;
+class WebRtcLocalAudioTrack;
+class WebRtcLocalAudioSourceProvider;
// WebAudioCapturerSource is the missing link between
-// WebAudio's MediaStreamAudioDestinationNode and WebRtcAudioCapturer.
+// WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
//
// 1. WebKit calls the setFormat() method setting up the basic stream format
-// (channels, and sample-rate). At this time, it dispatches this information
-// to the WebRtcAudioCapturer by calling its SetCapturerSource() method.
-// 2. Initialize() is called, where we should get back the same
-// stream format information as (1). We also get the CaptureCallback here.
-// 3. consumeAudio() is called periodically by WebKit which dispatches the
-// audio stream to the CaptureCallback::Capture() method.
+// (channels, and sample-rate).
+// 2. consumeAudio() is called periodically by WebKit which dispatches the
+// audio stream to the WebRtcLocalAudioTrack::Capture() method.
class WebAudioCapturerSource
- : public media::AudioCapturerSource,
- public WebKit::WebAudioDestinationConsumer {
+ : public base::RefCountedThreadSafe<WebAudioCapturerSource>,
+ public WebKit::WebAudioDestinationConsumer {
public:
- explicit WebAudioCapturerSource(WebRtcAudioCapturer* capturer);
+ WebAudioCapturerSource();
// WebAudioDestinationConsumer implementation.
- // setFormat() is called early on, so that we can configure the capturer.
+ // setFormat() is called early on, so that we can configure the audio track.
virtual void setFormat(size_t number_of_channels, float sample_rate) OVERRIDE;
// MediaStreamAudioDestinationNode periodically calls consumeAudio().
+ // Called on the WebAudio audio thread.
virtual void consumeAudio(const WebKit::WebVector<const float*>& audio_data,
size_t number_of_frames) OVERRIDE;
- // AudioCapturerSource implementation.
- virtual void Initialize(
- const media::AudioParameters& params,
- media::AudioCapturerSource::CaptureCallback* callback,
- int session_id) OVERRIDE;
+ // Called when the WebAudioCapturerSource is hooking to a media audio track.
+ // |track| is the sink of the data flow. |source_provider| is the source of
+ // the data flow where stream information like delay, volume, key_pressed,
+ // is stored.
+ void Start(WebRtcLocalAudioTrack* track,
+ WebRtcLocalAudioSourceProvider* source_provider);
- virtual void Start() OVERRIDE;
- virtual void Stop() OVERRIDE;
- virtual void SetVolume(double volume) OVERRIDE { }
- virtual void SetAutomaticGainControl(bool enable) OVERRIDE { }
+ // Called when the media audio track is stopping.
+ void Stop();
- private:
+ protected:
+ friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
virtual ~WebAudioCapturerSource();
- WebRtcAudioCapturer* capturer_;
+ private:
+ // Used to DCHECK that some methods are called on the correct thread.
+ base::ThreadChecker thread_checker_;
+
+ // The audio track this WebAudioCapturerSource is feeding data to.
+ // WebRtcLocalAudioTrack is reference counted, and owning this object.
+ // To avoid circular reference, a raw pointer is kept here.
+ WebRtcLocalAudioTrack* track_;
+
+ // A raw pointer to the source provider to get audio processing params like
+ // delay, volume, key_pressed information.
+ // This |source_provider_| is guaranteed to outlive this object.
+ WebRtcLocalAudioSourceProvider* source_provider_;
- int set_format_channels_;
media::AudioParameters params_;
- media::AudioCapturerSource::CaptureCallback* callback_;
// Wraps data coming from HandleCapture().
scoped_ptr<media::AudioBus> wrapper_bus_;

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