Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1638)

Unified 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 side-by-side diff with in-line comments
Download patch
Index: content/renderer/media/webrtc_local_audio_track.cc
diff --git a/content/renderer/media/webrtc_local_audio_track.cc b/content/renderer/media/webrtc_local_audio_track.cc
deleted file mode 100644
index 53c55b2d0d4c45a7c50c331b547c616dd4f717e2..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_track.cc
+++ /dev/null
@@ -1,169 +0,0 @@
-// Copyright (c) 2012 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.
-
-#include "content/renderer/media/webrtc_local_audio_track.h"
-
-#include <stdint.h>
-
-#include <limits>
-
-#include "content/public/renderer/media_stream_audio_sink.h"
-#include "content/renderer/media/media_stream_audio_level_calculator.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/media_stream_audio_sink_owner.h"
-#include "content/renderer/media/media_stream_audio_track_sink.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-
-namespace content {
-
-WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter)
- : MediaStreamAudioTrack(true), adapter_(std::move(adapter)) {
- signal_thread_checker_.DetachFromThread();
- DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
-
- adapter_->Initialize(this);
-}
-
-WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
- // Ensure the track is stopped.
- MediaStreamAudioTrack::Stop();
-}
-
-media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- base::AutoLock auto_lock(lock_);
- return audio_parameters_;
-}
-
-void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
- base::TimeTicks estimated_capture_time) {
- DCHECK(capture_thread_checker_.CalledOnValidThread());
- DCHECK(!estimated_capture_time.is_null());
-
- SinkList::ItemList sinks;
- SinkList::ItemList sinks_to_notify_format;
- {
- base::AutoLock auto_lock(lock_);
- sinks = sinks_.Items();
- sinks_.RetrieveAndClearTags(&sinks_to_notify_format);
- }
-
- // Notify the tracks on when the format changes. This will do nothing if
- // |sinks_to_notify_format| is empty. Note that accessing |audio_parameters_|
- // without holding the |lock_| is valid since |audio_parameters_| is only
- // changed on the current thread.
- for (const auto& sink : sinks_to_notify_format)
- sink->OnSetFormat(audio_parameters_);
-
- // Feed the data to the sinks.
- // TODO(jiayl): we should not pass the real audio data down if the track is
- // disabled. This is currently done so to feed input to WebRTC typing
- // detection and should be changed when audio processing is moved from
- // WebRTC to the track.
- for (const auto& sink : sinks)
- sink->OnData(audio_bus, estimated_capture_time);
-}
-
-void WebRtcLocalAudioTrack::OnSetFormat(
- const media::AudioParameters& params) {
- DVLOG(1) << "WebRtcLocalAudioTrack::OnSetFormat()";
- // If the source is restarted, we might have changed to another capture
- // thread.
- capture_thread_checker_.DetachFromThread();
- DCHECK(capture_thread_checker_.CalledOnValidThread());
-
- base::AutoLock auto_lock(lock_);
- audio_parameters_ = params;
- // Remember to notify all sinks of the new format.
- sinks_.TagAll();
-}
-
-void WebRtcLocalAudioTrack::SetLevel(
- scoped_refptr<MediaStreamAudioLevelCalculator::Level> level) {
- adapter_->SetLevel(std::move(level));
-}
-
-void WebRtcLocalAudioTrack::SetAudioProcessor(
- scoped_refptr<MediaStreamAudioProcessor> processor) {
- adapter_->SetAudioProcessor(std::move(processor));
-}
-
-void WebRtcLocalAudioTrack::AddSink(MediaStreamAudioSink* sink) {
- // This method is called from webrtc, on the signaling thread, when the local
- // description is set and from the main thread from WebMediaPlayerMS::load
- // (via WebRtcLocalAudioRenderer::Start).
- DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
- signal_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::AddSink()";
- base::AutoLock auto_lock(lock_);
-
- // Verify that |sink| is not already added to the list.
- DCHECK(!sinks_.Contains(
- MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink)));
-
- // Create (and add to the list) a new MediaStreamAudioTrackSink
- // which owns the |sink| and delagates all calls to the
- // MediaStreamAudioSink interface. It will be tagged in the list, so
- // we remember to call OnSetFormat() on the new sink.
- scoped_refptr<MediaStreamAudioTrackSink> sink_owner(
- new MediaStreamAudioSinkOwner(sink));
- sinks_.AddAndTag(sink_owner.get());
-}
-
-void WebRtcLocalAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
- // See AddSink for additional context. When local audio is stopped from
- // webrtc, we'll be called here on the signaling thread.
- DCHECK(main_render_thread_checker_.CalledOnValidThread() ||
- signal_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::RemoveSink()";
-
- scoped_refptr<MediaStreamAudioTrackSink> removed_item;
- {
- base::AutoLock auto_lock(lock_);
- removed_item = sinks_.Remove(
- MediaStreamAudioTrackSink::WrapsMediaStreamSink(sink));
- }
-
- // Clear the delegate to ensure that no more capture callbacks will
- // be sent to this sink. Also avoids a possible crash which can happen
- // if this method is called while capturing is active.
- if (removed_item.get())
- removed_item->Reset();
-}
-
-void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- if (adapter_.get())
- adapter_->set_enabled(enabled);
-}
-
-void WebRtcLocalAudioTrack::OnStop() {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::OnStop()";
-
- // Protect the pointers using the lock when accessing |sinks_|.
- SinkList::ItemList sinks;
- {
- base::AutoLock auto_lock(lock_);
- sinks = sinks_.Items();
- sinks_.Clear();
- }
-
- for (SinkList::ItemList::const_iterator it = sinks.begin();
- it != sinks.end();
- ++it){
- (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateEnded);
- (*it)->Reset();
- }
-}
-
-webrtc::AudioTrackInterface* WebRtcLocalAudioTrack::GetAudioAdapter() {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- return adapter_.get();
-}
-
-} // namespace content
« 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