Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(218)

Side by Side Diff: content/renderer/media/webrtc/peer_connection_remote_audio_source.h

Issue 1966043006: Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
(Empty)
1 // Copyright 2015 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
4
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
7
8 #include <memory>
9
10 #include "base/memory/ref_counted.h"
11 #include "base/synchronization/lock.h"
12 #include "content/renderer/media/media_stream_audio_source.h"
13 #include "content/renderer/media/media_stream_audio_track.h"
14 #include "third_party/webrtc/api/mediastreaminterface.h"
15
16 namespace media {
17 class AudioBus;
18 }
19
20 namespace content {
21
22 // PeerConnectionRemoteAudioTrack is a WebRTC specific implementation of an
23 // audio track whose data is sourced from a PeerConnection.
24 class PeerConnectionRemoteAudioTrack final
25 : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
26 public:
27 explicit PeerConnectionRemoteAudioTrack(
28 scoped_refptr<webrtc::AudioTrackInterface> track_interface);
29 ~PeerConnectionRemoteAudioTrack() final;
30
31 // If |track| is an instance of PeerConnectionRemoteAudioTrack, return a
32 // type-casted pointer to it. Otherwise, return null.
33 static PeerConnectionRemoteAudioTrack* From(MediaStreamAudioTrack* track);
34
35 webrtc::AudioTrackInterface* track_interface() const {
36 return track_interface_.get();
37 }
38
39 // MediaStreamAudioTrack override.
40 void SetEnabled(bool enabled) override;
41
42 private:
43 // MediaStreamAudioTrack overrides.
44 void* GetClassIdentifier() const final;
45
46 const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
47
48 // In debug builds, check that all methods that could cause object graph
49 // or data flow changes are being called on the main thread.
50 base::ThreadChecker thread_checker_;
51
52 DISALLOW_COPY_AND_ASSIGN(PeerConnectionRemoteAudioTrack);
53 };
54
55 // Represents the audio provided by the receiving end of a PeerConnection.
56 class PeerConnectionRemoteAudioSource final
57 : NON_EXPORTED_BASE(public MediaStreamAudioSource),
58 NON_EXPORTED_BASE(protected webrtc::AudioTrackSinkInterface) {
59 public:
60 explicit PeerConnectionRemoteAudioSource(
61 scoped_refptr<webrtc::AudioTrackInterface> track_interface);
62 ~PeerConnectionRemoteAudioSource() final;
63
64 protected:
65 // MediaStreamAudioSource implementation.
66 std::unique_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
67 const std::string& id) final;
68 bool EnsureSourceIsStarted() final;
69 void EnsureSourceIsStopped() final;
70
71 // webrtc::AudioTrackSinkInterface implementation.
72 void OnData(const void* audio_data, int bits_per_sample, int sample_rate,
73 size_t number_of_channels, size_t number_of_frames) final;
74
75 private:
76 // Interface to the implementation that calls OnData().
77 const scoped_refptr<webrtc::AudioTrackInterface> track_interface_;
78
79 // In debug builds, check that all methods that could cause object graph
80 // or data flow changes are being called on the main thread.
81 base::ThreadChecker thread_checker_;
82
83 // True if |this| is receiving an audio flow as a sink of the remote
84 // PeerConnection via |track_interface_|.
85 bool is_sink_of_peer_connection_;
86
87 // Buffer for converting from interleaved signed-integer PCM samples to the
88 // planar float format. Only used on the thread that calls OnData().
89 std::unique_ptr<media::AudioBus> audio_bus_;
90
91 // In debug builds, use a "try lock" to sanity-check that there are no
92 // concurrent calls to OnData(). See notes in OnData() implementation.
93 #ifndef NDEBUG
94 base::Lock single_audio_thread_guard_;
95 #endif
96
97 DISALLOW_COPY_AND_ASSIGN(PeerConnectionRemoteAudioSource);
98 };
99
100 } // namespace content
101
102 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_REMOTE_AUDIO_SOURCE_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698