MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webaudio_media_stream_source.cc

Index: content/renderer/media/webaudio_media_stream_source.cc
diff --git a/content/renderer/media/webaudio_media_stream_source.cc b/content/renderer/media/webaudio_media_stream_source.cc
new file mode 100644
index 0000000000000000000000000000000000000000..efdd7ed8364a269276d318e5f8bf1c9553fd855e
--- /dev/null
+++ b/content/renderer/media/webaudio_media_stream_source.cc
@@ -0,0 +1,109 @@
+// Copyright (c) 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_media_stream_source.h"
+
+#include "base/bind.h"
+#include "base/bind_helpers.h"
+#include "base/logging.h"
+
+namespace content {
+
+WebAudioMediaStreamSource::WebAudioMediaStreamSource(
+    const blink::WebMediaStreamSource& blink_source)
+    : MediaStreamAudioSource(false /* is_remote */),
+      blink_source_(blink_source),
+      is_started_(false),
+      rechunker_(base::TimeDelta::FromMilliseconds(10),
+                 base::Bind(&WebAudioMediaStreamSource::DeliverRechunkedAudio,
+                            base::Unretained(this))) {
+  DVLOG(1) << "WebAudioMediaStreamSource::WebAudioMediaStreamSource()";
+}
+
+WebAudioMediaStreamSource::~WebAudioMediaStreamSource() {
+  DVLOG(1) << "WebAudioMediaStreamSource::~WebAudioMediaStreamSource()";
+  // Superclass will call StopSource() just in case.
+}
+
+void WebAudioMediaStreamSource::setFormat(
+    size_t number_of_channels, float sample_rate) {
+  VLOG(1) << "WebAudio media stream source changed format to: channels="
+          << number_of_channels << ", sample_rate=" << sample_rate;
+
+  rechunker_.SetSampleRate(sample_rate);
+  if (!wrapper_bus_ ||
+      wrapper_bus_->channels() != static_cast<int>(number_of_channels)) {
+    wrapper_bus_ = media::AudioBus::CreateWrapper(number_of_channels);
+  }
+
+  // If the channel count is greater than 8, use discrete layout. However,
+  // anything beyond 8 is ignored by some audio tracks/sinks.
+  const media::ChannelLayout channel_layout =
+      number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
+                             : media::GuessChannelLayout(number_of_channels);
+
+  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).
+      rechunker_.output_frames()));
+}
+
+void WebAudioMediaStreamSource::consumeAudio(
+    const blink::WebVector<const float*>& audio_data,
+    size_t number_of_frames) {
+  // 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
+  base::TimeTicks reference_time = base::TimeTicks::Now();
+
+  wrapper_bus_->set_frames(number_of_frames);
+  DCHECK_EQ(wrapper_bus_->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]));
+
+  rechunker_.Push(*wrapper_bus_, reference_time - base::TimeTicks());
+}
+
+void WebAudioMediaStreamSource::DeliverRechunkedAudio(
+    const media::AudioBus& audio_bus,
+    base::TimeDelta reference_timestamp) {
+  MediaStreamAudioSource::DeliverDataToTracks(
+      audio_bus, reference_timestamp + base::TimeTicks());
+}
+
+void WebAudioMediaStreamSource::DoStopSource() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (is_stopped_)
+    return;
+  if (is_started_) {
+    if (!blink_source_.isNull()) {
+      blink_source_.removeAudioConsumer(this);
+      blink_source_.reset();
+      VLOG(1) << "Stopped WebAudio media stream source.  Final audio "
+                 "parameters={"
+              << GetAudioParameters().AsHumanReadableString() << "}.";
+    }
+  }
+  is_stopped_ = true;
+}
+
+bool WebAudioMediaStreamSource::EnsureSourceIsStarted() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (is_stopped_)
+    return false;
+  if (is_started_)
+    return true;
+  if (blink_source_.isNull() || !blink_source_.requiresAudioConsumer()) {
+    StopSource();
+    return false;
+  }
+  VLOG(1) << "Starting WebAudio media stream source.";
+  blink_source_.addAudioConsumer(this);
+  is_started_ = true;
+  return true;
+}
+
+}  // namespace content