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

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: NOT FOR REVIEW -- This will be broken-up across multiple CLs. Created 4 years, 10 months ago
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Index: content/renderer/media/webaudio_media_stream_source.h
diff --git a/content/renderer/media/webaudio_capturer_source.h b/content/renderer/media/webaudio_media_stream_source.h
similarity index 25%
rename from content/renderer/media/webaudio_capturer_source.h
rename to content/renderer/media/webaudio_media_stream_source.h
index b0ee262ccd7c3c06135b9252b3b394db8c0dc7dd..ae854542830144415c79671b6e2fa5fff31b021b 100644
--- a/content/renderer/media/webaudio_capturer_source.h
+++ b/content/renderer/media/webaudio_media_stream_source.h
@@ -1,100 +1,77 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Copyright (c) 2016 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
-#ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
-#define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
+#ifndef CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_
-#include <stddef.h>
-
-#include "base/macros.h"
-#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"
+#include "base/memory/scoped_ptr.h"
+#include "base/time/time.h"
+#include "content/renderer/media/media_stream_audio_source.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_rechunker.h"
#include "third_party/WebKit/public/platform/WebAudioDestinationConsumer.h"
#include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
#include "third_party/WebKit/public/platform/WebVector.h"
namespace content {
-class WebRtcLocalAudioTrack;
-
-// WebAudioCapturerSource is the missing link between
-// WebAudio's MediaStreamAudioDestinationNode and WebRtcLocalAudioTrack.
-//
-// 1. WebKit calls the setFormat() method setting up the basic stream format
-// (channels, and sample-rate).
-// 2. consumeAudio() is called periodically by WebKit which dispatches the
-// audio stream to the WebRtcLocalAudioTrack::Capture() method.
-class WebAudioCapturerSource
- : public base::RefCountedThreadSafe<WebAudioCapturerSource>,
- public blink::WebAudioDestinationConsumer {
+// Implements the WebAudioDestinationConsumer interface to provide a source of
+// audio data (i.e., the output from a graph of WebAudio nodes) to one or more
+// MediaStreamAudioTracks. Audio data is transported directly to the tracks in
+// 10 ms chunks.
+class WebAudioMediaStreamSource
+ : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+ public blink::WebAudioDestinationConsumer {
public:
- explicit WebAudioCapturerSource(
+ explicit WebAudioMediaStreamSource(
const blink::WebMediaStreamSource& blink_source);
+ ~WebAudioMediaStreamSource() final;
+
+ private:
// WebAudioDestinationConsumer implementation.
- // setFormat() is called early on, so that we can configure the audio track.
+ //
+ // Note: Blink ensures setFormat() and consumeAudio() are not called
+ // concurrently across threads, but these methods could be called on any
+ // thread.
void setFormat(size_t number_of_channels, float sample_rate) override;
- // MediaStreamAudioDestinationNode periodically calls consumeAudio().
- // Called on the WebAudio audio thread.
void consumeAudio(const blink::WebVector<const float*>& audio_data,
size_t number_of_frames) 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);
+ // Called by AudioRechunker zero or more times during the call to
+ // consumeAudio(). Delivers re-chunked audio data to the tracks.
+ void DeliverRechunkedAudio(const media::AudioBus& audio_bus,
+ base::TimeDelta reference_timestamp);
- // Called when the media audio track is stopping.
- void Stop();
+ // MediaStreamAudioSource implementation.
+ void DoStopSource() final;
+ bool EnsureSourceIsStarted() final;
- protected:
- friend class base::RefCountedThreadSafe<WebAudioCapturerSource>;
- ~WebAudioCapturerSource() override;
-
- private:
// Removes this object from a blink::WebMediaStreamSource with which it
// might be registered. The goal is to avoid dangling pointers.
void removeFromBlinkSource();
- // 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_;
-
- media::AudioParameters params_;
+ // This object registers and de-registers as an audio consumer of a
+ // blink::WebMediaStreamSource.
+ blink::WebMediaStreamSource blink_source_;
- // Flag to help notify the |track_| when the audio format has changed.
- bool audio_format_changed_;
+ // True while this WebAudioMediaStreamSource is registered with
+ // |blink_source_| and is consuming audio.
+ bool is_started_;
- // Wraps data coming from HandleCapture().
+ // An adapter used for providing audio to |rechunker_|.
scoped_ptr<media::AudioBus> wrapper_bus_;
- // Bus for reading from FIFO and calling the CaptureCallback.
- scoped_ptr<media::AudioBus> capture_bus_;
-
- // Handles mismatch between WebAudio buffer size and WebRTC.
- scoped_ptr<media::AudioFifo> fifo_;
-
- // Synchronizes HandleCapture() with AudioCapturerSource calls.
- base::Lock lock_;
- bool started_;
-
- // This object registers with a blink::WebMediaStreamSource. We keep track of
- // that in order to be able to deregister before stopping the audio track.
- blink::WebMediaStreamSource blink_source_;
+ // Takes in the audio data passed to consumeAudio() and re-chunks it into 10
+ // ms chunks for the tracks. This ensures each chunk of audio delivered to
+ // the tracks has the same buffer size, even if audio is provided in
+ // varying-sized chunks.
+ media::AudioRechunker rechunker_;
- DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
+ DISALLOW_COPY_AND_ASSIGN(WebAudioMediaStreamSource);
};
} // namespace content
-#endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
+#endif // CONTENT_RENDERER_MEDIA_WEBAUDIO_MEDIA_STREAM_SOURCE_H_
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