| // Copyright 2015 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_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_ |
| #define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_ |
| |
| #include <memory> |
| |
| #include "base/memory/ref_counted.h" |
| #include "base/synchronization/lock.h" |
| #include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_source.h" |
| #include "third_party/blink/public/platform/modules/mediastream/media_stream_audio_track.h" |
| #include "third_party/webrtc/api/media_stream_interface.h" |
| |
| namespace media { |
| class AudioBus; |
| } |
| |
| namespace content { |
| |
| // PeerConnectionRemoteAudioTrack is a WebRTC specific implementation of an |
| // audio track whose data is sourced from a PeerConnection. |
| class PeerConnectionRemoteAudioTrack final |
| : public blink::MediaStreamAudioTrack { |
| public: |
| explicit PeerConnectionRemoteAudioTrack( |
| scoped_refptr<webrtc::AudioTrackInterface> track_interface); |
| ~PeerConnectionRemoteAudioTrack() final; |
| |
| // If |track| is an instance of PeerConnectionRemoteAudioTrack, return a |
| // type-casted pointer to it. Otherwise, return null. |
| static PeerConnectionRemoteAudioTrack* From( |
| blink::MediaStreamAudioTrack* track); |
| |
| webrtc::AudioTrackInterface* track_interface() const { |
| return track_interface_.get(); |
| } |
| |
| // MediaStreamAudioTrack override. |
| void SetEnabled(bool enabled) override; |
| |
| private: |
| // MediaStreamAudioTrack overrides. |
| void* GetClassIdentifier() const final; |
| |
| const scoped_refptr<webrtc::AudioTrackInterface> track_interface_; |
| |
| // In debug builds, check that all methods that could cause object graph |
| // or data flow changes are being called on the main thread. |
| base::ThreadChecker thread_checker_; |
| |
| DISALLOW_COPY_AND_ASSIGN(PeerConnectionRemoteAudioTrack); |
| }; |
| |
| // Represents the audio provided by the receiving end of a PeerConnection. |
| class PeerConnectionRemoteAudioSource final |
| : public blink::MediaStreamAudioSource, |
| protected webrtc::AudioTrackSinkInterface { |
| public: |
| explicit PeerConnectionRemoteAudioSource( |
| scoped_refptr<webrtc::AudioTrackInterface> track_interface); |
| ~PeerConnectionRemoteAudioSource() final; |
| |
| protected: |
| // MediaStreamAudioSource implementation. |
| std::unique_ptr<blink::MediaStreamAudioTrack> CreateMediaStreamAudioTrack( |
| const std::string& id) final; |
| bool EnsureSourceIsStarted() final; |
| void EnsureSourceIsStopped() final; |
| |
| // webrtc::AudioTrackSinkInterface implementation. |
| void OnData(const void* audio_data, int bits_per_sample, int sample_rate, |
| size_t number_of_channels, size_t number_of_frames) final; |
| |
| private: |
| // Interface to the implementation that calls OnData(). |
| const scoped_refptr<webrtc::AudioTrackInterface> track_interface_; |
| |
| // In debug builds, check that all methods that could cause object graph |
| // or data flow changes are being called on the main thread. |
| base::ThreadChecker thread_checker_; |
| |
| // True if |this| is receiving an audio flow as a sink of the remote |
| // PeerConnection via |track_interface_|. |
| bool is_sink_of_peer_connection_; |
| |
| // Buffer for converting from interleaved signed-integer PCM samples to the |
| // planar float format. Only used on the thread that calls OnData(). |
| std::unique_ptr<media::AudioBus> audio_bus_; |
| |
| // In debug builds, use a "try lock" to sanity-check that there are no |
| // concurrent calls to OnData(). See notes in OnData() implementation. |
| #ifndef NDEBUG |
| base::Lock single_audio_thread_guard_; |
| #endif |
| |
| DISALLOW_COPY_AND_ASSIGN(PeerConnectionRemoteAudioSource); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_ |