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Unified Diff: content/renderer/media/external_media_stream_audio_source.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: Created 4 years, 9 months ago
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Index: content/renderer/media/external_media_stream_audio_source.cc
diff --git a/content/renderer/media/webaudio_capturer_source.cc b/content/renderer/media/external_media_stream_audio_source.cc
similarity index 10%
copy from content/renderer/media/webaudio_capturer_source.cc
copy to content/renderer/media/external_media_stream_audio_source.cc
index 3fca41942df83562b16d38c44f0c6af72069713d..28ece803975994cceffcaed5c23e8724ebe69822 100644
--- a/content/renderer/media/webaudio_capturer_source.cc
+++ b/content/renderer/media/external_media_stream_audio_source.cc
@@ -1,136 +1,84 @@
-// Copyright (c) 2012 The Chromium Authors. All rights reserved.
+// Copyright 2016 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/webaudio_capturer_source.h"
-
-#include "base/bind.h"
-#include "base/bind_helpers.h"
-#include "base/logging.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-
-using media::AudioBus;
-using media::AudioParameters;
-using media::ChannelLayout;
-using media::CHANNEL_LAYOUT_MONO;
-using media::CHANNEL_LAYOUT_STEREO;
+#include "content/renderer/media/external_media_stream_audio_source.h"
namespace content {
-WebAudioCapturerSource::WebAudioCapturerSource(
- blink::WebMediaStreamSource* blink_source)
- : track_(NULL),
- audio_format_changed_(false),
- fifo_(base::Bind(&WebAudioCapturerSource::DeliverRebufferedAudio,
- base::Unretained(this))),
- blink_source_(*blink_source) {
- DCHECK(blink_source);
- DCHECK(!blink_source_.isNull());
- DVLOG(1) << "WebAudioCapturerSource::WebAudioCapturerSource()";
- blink_source_.addAudioConsumer(this);
+ExternalMediaStreamAudioSource::ExternalMediaStreamAudioSource(
+ scoped_refptr<media::AudioCapturerSource> source,
+ int sample_rate,
+ media::ChannelLayout channel_layout,
+ int frames_per_buffer,
+ bool is_remote)
+ : MediaStreamAudioSource(!is_remote), source_(std::move(source)),
+ is_started_(false) {
+ DVLOG(1)
+ << "ExternalMediaStreamAudioSource::ExternalMediaStreamAudioSource()";
+ DCHECK(source_.get());
+ MediaStreamAudioSource::SetFormat(media::AudioParameters(
+ media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
+ sample_rate,
+ 16, // Legacy parameter (data is always in 32-bit float format).
+ frames_per_buffer));
}
-WebAudioCapturerSource::~WebAudioCapturerSource() {
- DCHECK(thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebAudioCapturerSource::~WebAudioCapturerSource()";
- DeregisterFromBlinkSource();
+ExternalMediaStreamAudioSource::~ExternalMediaStreamAudioSource() {
+ DVLOG(1)
+ << "ExternalMediaStreamAudioSource::~ExternalMediaStreamAudioSource()";
+ // Ensure the source is stopped.
+ MediaStreamAudioSource::StopSource();
}
-void WebAudioCapturerSource::setFormat(
- size_t number_of_channels, float sample_rate) {
+void ExternalMediaStreamAudioSource::DoStopSource() {
DCHECK(thread_checker_.CalledOnValidThread());
- DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
- << sample_rate << ")";
-
- // If the channel count is greater than 8, use discrete layout. However,
- // anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline.
- ChannelLayout channel_layout =
- number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
- : media::GuessChannelLayout(number_of_channels);
-
- base::AutoLock auto_lock(lock_);
-
- // Set the format used by this WebAudioCapturerSource. We are using 10ms data
- // as buffer size since that is the native buffer size of WebRtc packet
- // running on.
- fifo_.Reset(sample_rate / 100);
- params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
- sample_rate, 16, fifo_.frames_per_buffer());
-
- // Take care of the discrete channel layout case.
- params_.set_channels_for_discrete(number_of_channels);
-
- audio_format_changed_ = true;
-
- if (!wrapper_bus_ ||
- wrapper_bus_->channels() != static_cast<int>(number_of_channels)) {
- wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
+ if (is_stopped_)
+ return;
+ if (is_started_) {
+ source_->Stop();
+ VLOG(1) << "Stopped externally-provided "
+ << (is_local_source() ? "local" : "remote")
+ << " source with audio parameters={"
+ << GetAudioParameters().AsHumanReadableString() << "}.";
}
+ is_stopped_ = true;
}
-void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
+bool ExternalMediaStreamAudioSource::EnsureSourceIsStarted() {
DCHECK(thread_checker_.CalledOnValidThread());
- DCHECK(track);
- base::AutoLock auto_lock(lock_);
- track_ = track;
+ if (is_stopped_)
+ return false;
+ if (is_started_)
+ return true;
+ VLOG(1) << "Starting externally-provided "
+ << (is_local_source() ? "local" : "remote")
+ << " source with audio parameters={"
+ << GetAudioParameters().AsHumanReadableString() << "}.";
+ source_->Initialize(GetAudioParameters(), this, -1);
+ source_->Start();
+ is_started_ = true;
+ return true;
}
-void WebAudioCapturerSource::Stop() {
- DCHECK(thread_checker_.CalledOnValidThread());
- {
- base::AutoLock auto_lock(lock_);
- track_ = NULL;
- }
- // DeregisterFromBlinkSource() should not be called while |lock_| is acquired,
- // as it could result in a deadlock.
- DeregisterFromBlinkSource();
-}
-
-void WebAudioCapturerSource::consumeAudio(
- const blink::WebVector<const float*>& audio_data,
- size_t number_of_frames) {
+void ExternalMediaStreamAudioSource::Capture(const media::AudioBus* audio_bus,
+ int audio_delay_milliseconds,
+ double volume,
+ bool key_pressed) {
+ DCHECK(audio_bus);
// TODO(miu): Plumbing is needed to determine the actual capture timestamp
// of the audio, instead of just snapshotting TimeTicks::Now(), for proper
// audio/video sync. http://crbug.com/335335
- current_reference_time_ = base::TimeTicks::Now();
-
- base::AutoLock auto_lock(lock_);
- if (!track_)
- return;
-
- // Update the downstream client if the audio format has been changed.
- if (audio_format_changed_) {
- track_->OnSetFormat(params_);
- audio_format_changed_ = false;
- }
-
- wrapper_bus_->set_frames(number_of_frames);
- DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
- for (size_t i = 0; i < audio_data.size(); ++i)
- wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
-
- // The following will result in zero, one, or multiple synchronous calls to
- // DeliverRebufferedAudio().
- fifo_.Push(*wrapper_bus_);
+ MediaStreamAudioSource::DeliverDataToTracks(
+ *audio_bus, base::TimeTicks::Now() - base::TimeDelta::FromMilliseconds(
+ audio_delay_milliseconds));
}
-void WebAudioCapturerSource::DeliverRebufferedAudio(
- const media::AudioBus& audio_bus,
- int frame_delay) {
- lock_.AssertAcquired();
- const base::TimeTicks reference_time =
- current_reference_time_ +
- base::TimeDelta::FromMicroseconds(frame_delay *
- base::Time::kMicrosecondsPerSecond /
- params_.sample_rate());
- track_->Capture(audio_bus, reference_time);
-}
-
-void WebAudioCapturerSource::DeregisterFromBlinkSource() {
- if (!blink_source_.isNull()) {
- blink_source_.removeAudioConsumer(this);
- blink_source_.reset();
- }
+void ExternalMediaStreamAudioSource::OnCaptureError(const std::string& why) {
+ // As of this writing, this method doesn't get called for anything useful,
+ // and all other implementors just log the message, but don't disconnect sinks
+ // or take any other action. So, just log the error.
+ LOG(ERROR) << why;
}
} // namespace content

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