MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc/processed_local_audio_source.h

Index: content/renderer/media/webrtc/processed_local_audio_source.h
diff --git a/content/renderer/media/webrtc/processed_local_audio_source.h b/content/renderer/media/webrtc/processed_local_audio_source.h
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+// Copyright 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_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_
+
+#include "base/memory/ref_counted.h"
+#include "base/synchronization/lock.h"
+#include "content/common/media/media_stream_options.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.h"
+#include "content/renderer/media/media_stream_audio_source.h"
+#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
+#include "media/base/audio_capturer_source.h"
+#include "third_party/webrtc/api/mediastreaminterface.h"
+#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
+
+namespace media {
+class AudioInputDevice;
+}  // namespace media
+
+namespace content {
+
+class MediaStreamAudioProcessor;
+
+// Represents a local source of audio data that is routed through the WebRTC
+// audio pipeline for post-processing.  Owns a media::AudioInputDevice and the
+// MediaStreamProcessor that modifies its audio.  Modified audio is delivered to
+// WebRtcLocalAudioTracks.
+class CONTENT_EXPORT ProcessedLocalAudioSource
+    : NON_EXPORTED_BASE(public MediaStreamAudioSource),
+      NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
+ public:
+  // |consumer_render_frame_id| references the RenderFrame that will consume the
+  // audio data.  Audio parameters and (optionally) a pre-existing audio session
+  // ID are read from |device_info|.  |factory| must outlive this instance.
+  ProcessedLocalAudioSource(int consumer_render_frame_id,
+                            const StreamDeviceInfo& device_info,
+                            PeerConnectionDependencyFactory* factory);
+
+  ~ProcessedLocalAudioSource() final;
+
+  // If |source| is an instance of ProcessedLocalAudioSource, return a
+  // type-casted pointer to it.  Otherwise, return null.
+  static ProcessedLocalAudioSource* From(MediaStreamAudioSource* source);
+
+  // Non-browser unit tests cannot provide RenderFrame implementations at
+  // run-time.  This is used to skip the otherwise mandatory check for a valid
+  // render frame ID when the source is started.
+  void SetAllowInvalidRenderFrameIdForTesting(bool allowed) {
+    allow_invalid_render_frame_id_for_testing_ = allowed;
+  }
+
+  // Gets/Sets source constraints.  Using this is optional, but must be done
+  // before the first call to ConnectToTrack().
+  blink::WebMediaConstraints source_constraints() const { return constraints_; }
+  void SetSourceConstraints(const blink::WebMediaConstraints& constraints);
+
+  // Not valid until after the source is started (when the first track is
+  // connected).
+  webrtc::AudioSourceInterface* rtc_source() const { return rtc_source_.get(); }
+
+  // Thread-safe volume accessors used by WebRtcAudioDeviceImpl.
+  void SetVolume(int volume);
+  int Volume() const;
+  int MaxVolume() const;
+
+  // Thread-safe accessor for querying the audio format prior to processing.
+  // TODO(phoglund): Think over the implications of this accessor and if we can
+  // remove it.
+  media::AudioParameters GetInputAudioParameters() const;
+
+ protected:
+  // MediaStreamAudioSource implementation.
+  void* GetClassIdentifier() const final;
+  void DoStopSource() final;
+  scoped_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
+      const std::string& id) final;
+  bool EnsureSourceIsStarted() final;
+
+  // AudioCapturerSource::CaptureCallback implementation.
+  void Capture(const media::AudioBus* audio_bus,
+               int audio_delay_milliseconds,
+               double volume,
+               bool key_pressed) override;
+  void OnCaptureError(const std::string& message) override;
+
+ private:
+  // The RenderFrame that will consume the audio data.  Used when creating
+  // AudioInputDevices via the AudioDeviceFactory.
+  const int consumer_render_frame_id_;
+
+  PeerConnectionDependencyFactory* const pc_factory_;
+
+  // Constraints used when initializing and starting the source.
+  blink::WebMediaConstraints constraints_;
+
+  // Audio processor doing processing like FIFO, AGC, AEC and NS.  Its output
+  // data is in a unit of 10 ms data chunk.  Created by EnsureSourceIsStarted().
+  scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
+
+  // The device created by the AudioDeviceFactory in EnsureSourceIsStarted().
+  // This is set once, and should never be changed again to avoid a
+  // thread-race condition in the SetVolume() method.
+  scoped_refptr<media::AudioInputDevice> input_device_;
+
+  // Holder for WebRTC audio pipeline objects.  Created in
+  // EnsureSourceIsStarted().
+  scoped_refptr<webrtc::AudioSourceInterface> rtc_source_;
+
+  // Protects data elements from concurrent access when using the volume methods.
+  mutable base::Lock volume_lock_;
+
+  // Stores latest microphone volume received in a CaptureData() callback.
+  // Range is [0, 255].
+  int exposed_volume_;
+
+  // Used to calculate the signal level that shows in the UI.
+  MediaStreamAudioLevelCalculator level_calculator_;
+
+  bool allow_invalid_render_frame_id_for_testing_;
+
+  DISALLOW_COPY_AND_ASSIGN(ProcessedLocalAudioSource);
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PROCESSED_LOCAL_AUDIO_SOURCE_H_