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Unified Diff: content/renderer/media/webrtc_local_audio_track.cc

Issue 1647773002: MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: NOT FOR REVIEW -- This will be broken-up across multiple CLs. Created 4 years, 10 months ago
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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 cb48668eaeccc09a670c0a150b5d439e8d515778..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc_local_audio_track.cc
+++ /dev/null
@@ -1,241 +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/webaudio_capturer_source.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
-
-namespace content {
-
-WebRtcLocalAudioTrack::WebRtcLocalAudioTrack(
- WebRtcLocalAudioTrackAdapter* adapter,
- const scoped_refptr<WebRtcAudioCapturer>& capturer,
- WebAudioCapturerSource* webaudio_source)
- : MediaStreamAudioTrack(true),
- adapter_(adapter),
- capturer_(capturer),
- webaudio_source_(webaudio_source) {
- DCHECK(capturer.get() || webaudio_source);
- signal_thread_checker_.DetachFromThread();
-
- adapter_->Initialize(this);
-
- DVLOG(1) << "WebRtcLocalAudioTrack::WebRtcLocalAudioTrack()";
-}
-
-WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack() {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::~WebRtcLocalAudioTrack()";
- // Users might not call Stop() on the track.
- Stop();
-}
-
-media::AudioParameters WebRtcLocalAudioTrack::GetOutputFormat() const {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- if (webaudio_source_.get()) {
- return media::AudioParameters();
- } else {
- return capturer_->GetOutputFormat();
- }
-}
-
-void WebRtcLocalAudioTrack::Capture(const media::AudioBus& audio_bus,
- base::TimeTicks estimated_capture_time,
- bool force_report_nonzero_energy) {
- DCHECK(capture_thread_checker_.CalledOnValidThread());
- DCHECK(!estimated_capture_time.is_null());
-
- // Calculate the signal level regardless of whether the track is disabled or
- // enabled. If |force_report_nonzero_energy| is true, |audio_bus| contains
- // post-processed data that may be all zeros even though the signal contained
- // energy before the processing. In this case, report nonzero energy even if
- // the energy of the data in |audio_bus| is zero.
- const float minimum_signal_level =
- force_report_nonzero_energy ? 1.0f / std::numeric_limits<int16_t>::max()
- : 0.0f;
- const float signal_level = std::max(
- minimum_signal_level,
- std::min(1.0f, level_calculator_->Calculate(audio_bus)));
- const int signal_level_as_pcm16 =
- static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
- 0.5f /* rounding to nearest int */);
- adapter_->SetSignalLevel(signal_level_as_pcm16);
-
- scoped_refptr<WebRtcAudioCapturer> capturer;
- SinkList::ItemList sinks;
- SinkList::ItemList sinks_to_notify_format;
- {
- base::AutoLock auto_lock(lock_);
- capturer = capturer_;
- 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.
- 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());
-
- audio_parameters_ = params;
- level_calculator_.reset(new MediaStreamAudioLevelCalculator());
-
- base::AutoLock auto_lock(lock_);
- // Remember to notify all sinks of the new format.
- sinks_.TagAll();
-}
-
-void WebRtcLocalAudioTrack::SetAudioProcessor(
- const scoped_refptr<MediaStreamAudioProcessor>& processor) {
- // if the |processor| does not have audio processing, which can happen if
- // kDisableAudioTrackProcessing is set set or all the constraints in
- // the |processor| are turned off. In such case, we pass NULL to the
- // adapter to indicate that no stats can be gotten from the processor.
- adapter_->SetAudioProcessor(processor->has_audio_processing() ?
- processor : NULL);
-}
-
-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::Start() {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::Start()";
- if (webaudio_source_.get()) {
- // If the track is hooking up with WebAudio, do NOT add the track to the
- // capturer as its sink otherwise two streams in different clock will be
- // pushed through the same track.
- webaudio_source_->Start(this);
- } else if (capturer_.get()) {
- capturer_->AddTrack(this);
- }
-
- SinkList::ItemList sinks;
- {
- base::AutoLock auto_lock(lock_);
- sinks = sinks_.Items();
- }
- for (SinkList::ItemList::const_iterator it = sinks.begin();
- it != sinks.end();
- ++it) {
- (*it)->OnReadyStateChanged(blink::WebMediaStreamSource::ReadyStateLive);
- }
-}
-
-void WebRtcLocalAudioTrack::SetEnabled(bool enabled) {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- if (adapter_.get())
- adapter_->set_enabled(enabled);
-}
-
-void WebRtcLocalAudioTrack::Stop() {
- DCHECK(main_render_thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebRtcLocalAudioTrack::Stop()";
- if (!capturer_.get() && !webaudio_source_.get())
- return;
-
- if (webaudio_source_.get()) {
- // Called Stop() on the |webaudio_source_| explicitly so that
- // |webaudio_source_| won't push more data to the track anymore.
- // Also note that the track is not registered as a sink to the |capturer_|
- // in such case and no need to call RemoveTrack().
- webaudio_source_->Stop();
- } else {
- // It is necessary to call RemoveTrack on the |capturer_| to avoid getting
- // audio callback after Stop().
- capturer_->RemoveTrack(this);
- }
-
- // Protect the pointers using the lock when accessing |sinks_| and
- // setting the |capturer_| to NULL.
- SinkList::ItemList sinks;
- {
- base::AutoLock auto_lock(lock_);
- sinks = sinks_.Items();
- sinks_.Clear();
- webaudio_source_ = NULL;
- capturer_ = NULL;
- }
-
- 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
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