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Unified Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: REBASE + Workaround to ensure MediaStreamAudioProcessor is destroyed on the main thread. Created 4 years, 7 months ago
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Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
deleted file mode 100644
index 72b80194b08ed09a01673c98c4ed9816aa4e6d74..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
+++ /dev/null
@@ -1,107 +0,0 @@
-// Copyright 2014 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_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
-
-#include <vector>
-
-#include "base/memory/ref_counted.h"
-#include "base/memory/scoped_vector.h"
-#include "base/single_thread_task_runner.h"
-#include "base/synchronization/lock.h"
-#include "content/common/content_export.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "third_party/webrtc/api/mediastreamtrack.h"
-#include "third_party/webrtc/media/base/audiorenderer.h"
-
-namespace cricket {
-class AudioRenderer;
-}
-
-namespace webrtc {
-class AudioSourceInterface;
-class AudioProcessorInterface;
-}
-
-namespace content {
-
-class MediaStreamAudioProcessor;
-class WebRtcAudioSinkAdapter;
-class WebRtcLocalAudioTrack;
-
-// Provides an implementation of the webrtc::AudioTrackInterface that can be
-// bound/unbound to/from a MediaStreamAudioTrack. In other words, this is an
-// adapter that sits between the media stream object graph and WebRtc's object
-// graph and proxies between the two.
-class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
- : NON_EXPORTED_BASE(
- public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
- public:
- static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
- const std::string& label,
- webrtc::AudioSourceInterface* track_source);
-
- WebRtcLocalAudioTrackAdapter(
- const std::string& label,
- webrtc::AudioSourceInterface* track_source,
- scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
-
- ~WebRtcLocalAudioTrackAdapter() override;
-
- void Initialize(WebRtcLocalAudioTrack* owner);
-
- // Set the object that provides shared access to the current audio signal
- // level. This method may only be called once, before the audio data flow
- // starts, and before any calls to GetSignalLevel() might be made.
- void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
-
- // Method called by the WebRtcLocalAudioTrack to set the processor that
- // applies signal processing on the data of the track.
- // This class will keep a reference of the |processor|.
- // Called on the main render thread.
- // This method may only be called once, before the audio data flow starts, and
- // before any calls to GetAudioProcessor() might be made.
- void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
-
- // webrtc::MediaStreamTrack implementation.
- std::string kind() const override;
- bool set_enabled(bool enable) override;
-
- private:
- // webrtc::AudioTrackInterface implementation.
- void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
- void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
- bool GetSignalLevel(int* level) override;
- rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
- override;
- webrtc::AudioSourceInterface* GetSource() const override;
-
- // Weak reference.
- WebRtcLocalAudioTrack* owner_;
-
- // The source of the audio track which handles the audio constraints.
- // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
- rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
-
- // Libjingle's signaling thread.
- const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
-
- // The audio processsor that applies audio processing on the data of audio
- // track. This must be set before calls to GetAudioProcessor() are made.
- scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
-
- // A vector of the peer connection sink adapters which receive the audio data
- // from the audio track.
- ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
-
- // Thread-safe accessor to current audio signal level. This must be set
- // before calls to GetSignalLevel() are made.
- scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
-};
-
-} // namespace content
-
-#endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_

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