Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(35)

Side by Side Diff: content/renderer/media/webrtc_local_audio_track.cc

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
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 (c) 2012 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 #include "content/renderer/media/webrtc_local_audio_track.h"
6
7 #include <stdint.h>
8
9 #include <limits>
10
11 #include "content/public/renderer/media_stream_audio_sink.h"
12 #include "content/renderer/media/media_stream_audio_level_calculator.h"
13 #include "content/renderer/media/media_stream_audio_processor.h"
14 #include "content/renderer/media/media_stream_audio_sink_owner.h"
15 #include "content/renderer/media/media_stream_audio_track_sink.h"
16 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
17
18 namespace content {
19
20 WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
21 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter)
22 : MediaStreamAudioTrack(true), adapter_(std::move(adapter)) {
23 signal_thread_checker_.DetachFromThread();
24 DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
25
26 adapter_->Initialize(this);
27 }
28
29 WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
30 DCHECK(main_render_thread_checker_.CalledOnValidThread());
31 DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
32 // Ensure the track is stopped.
33 MediaStreamAudioTrack::Stop();
34 }
35
36 media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
37 DCHECK(main_render_thread_checker_.CalledOnValidThread());
38 base::AutoLock auto_lock(lock_);
39 return audio_parameters_;
40 }
41
42 void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
43 base::TimeTicks estimated_capture_time) {
44 DCHECK(capture_thread_checker_.CalledOnValidThread());
45 DCHECK(!estimated_capture_time.is_null());
46
47 SinkList::ItemList sinks;
48 SinkList::ItemList sinks_to_notify_format;
49 {
50 base::AutoLock auto_lock(lock_);
51 sinks = sinks_.Items();
52 sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
53 }
54
55 // Notify the tracks on when the format changes. This will do nothing if
56 // |sinks_to_notify_format| is empty. Note that accessing |audio_parameters_|
57 // without holding the |lock_| is valid since |audio_parameters_| is only
58 // changed on the current thread.
59 for (const auto& sink : sinks_to_notify_format)
60 sink->OnSetFormat(audio_parameters_);
61
62 // Feed the data to the sinks.
63 // TODO(jiayl): we should not pass the real audio data down if the track is
64 // disabled. This is currently done so to feed input to WebRTC typing
65 // detection and should be changed when audio processing is moved from
66 // WebRTC to the track.
67 for (const auto& sink : sinks)
68 sink->OnData(audio_bus, estimated_capture_time);
69 }
70
71 void WebRtcLocalAudioTrack::OnSetFormat(
72 const media::AudioParameters& params) {
73 DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
74 // If the source is restarted, we might have changed to another capture
75 // thread.
76 capture_thread_checker_.DetachFromThread();
77 DCHECK(capture_thread_checker_.CalledOnValidThread());
78
79 base::AutoLock auto_lock(lock_);
80 audio_parameters_ = params;
81 // Remember to notify all sinks of the new format.
82 sinks_.TagAll();
83 }
84
85 void WebRtcLocalAudioTrack::SetLevel(
86 scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
87 adapter_->SetLevel(std::move(level));
88 }
89
90 void WebRtcLocalAudioTrack::SetAudioProcessor(
91 scoped_refptr<MediaStreamAudioProcessor> processor) {
92 adapter_->SetAudioProcessor(std::move(processor));
93 }
94
95 void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
96 // This method is called from webrtc, on the signaling thread, when the local
97 // description is set and from the main thread from WebMediaPlayerMS::load
98 // (via WebRtcLocalAudioRenderer::Start).
99 DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
100 signal_thread_checker_.CalledOnValidThread());
101 DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
102 base::AutoLock auto_lock(lock_);
103
104 // Verify that |sink| is not already added to the list.
105 DCHECK(!sinks_.Contains(
106 MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
107
108 // Create (and add to the list) a new MediaStreamAudioTrackSink
109 // which owns the |sink| and delagates all calls to the
110 // MediaStreamAudioSink interface. It will be tagged in the list, so
111 // we remember to call OnSetFormat() on the new sink.
112 scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
113 new MediaStreamAudioSinkOwner(sink));
114 sinks_.AddAndTag(sink_owner.get());
115 }
116
117 void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
118 // See AddSink for additional context. When local audio is stopped from
119 // webrtc, we'll be called here on the signaling thread.
120 DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
121 signal_thread_checker_.CalledOnValidThread());
122 DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
123
124 scoped_refptr<MediaStreamAudioTrackSink> removed_item;
125 {
126 base::AutoLock auto_lock(lock_);
127 removed_item = sinks_.Remove(
128 MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
129 }
130
131 // Clear the delegate to ensure that no more capture callbacks will
132 // be sent to this sink. Also avoids a possible crash which can happen
133 // if this method is called while capturing is active.
134 if (removed_item.get())
135 removed_item->Reset();
136 }
137
138 void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
139 DCHECK(main_render_thread_checker_.CalledOnValidThread());
140 if (adapter_.get())
141 adapter_->set_enabled(enabled);
142 }
143
144 void WebRtcLocalAudioTrack::OnStop() {
145 DCHECK(main_render_thread_checker_.CalledOnValidThread());
146 DVLOG(1) << "WebRtcLocalAudioTrack::OnStop()";
147
148 // Protect the pointers using the lock when accessing |sinks_|.
149 SinkList::ItemList sinks;
150 {
151 base::AutoLock auto_lock(lock_);
152 sinks = sinks_.Items();
153 sinks_.Clear();
154 }
155
156 for (SinkList::ItemList::const_iterator it = sinks.begin();
157 it != sinks.end();
158 ++it){
159 (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
160 (*it)->Reset();
161 }
162 }
163
164 webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
165 DCHECK(main_render_thread_checker_.CalledOnValidThread());
166 return adapter_.get();
167 }
168
169 } // namespace content
OLDNEW
« no previous file with comments | « content/renderer/media/webrtc_local_audio_track.h ('k') | content/renderer/media/webrtc_local_audio_track_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698