Use LocalMediaStreamAudioSource for screen-casting use cases.

content/renderer/media/local_media_stream_audio_source.cc

Index: content/renderer/media/local_media_stream_audio_source.cc
diff --git a/content/renderer/media/local_media_stream_audio_source.cc b/content/renderer/media/local_media_stream_audio_source.cc
new file mode 100644
index 0000000000000000000000000000000000000000..3678b4740f05acc67f4bd770ccbbd837ecad33ca
--- /dev/null
+++ b/content/renderer/media/local_media_stream_audio_source.cc
@@ -0,0 +1,96 @@
+// 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/local_media_stream_audio_source.h"
+
+#include "content/common/media/media_stream_options.h"
+#include "content/renderer/media/audio_device_factory.h"
+#include "content/renderer/render_frame_impl.h"
+
+namespace content {
+
+LocalMediaStreamAudioSource::LocalMediaStreamAudioSource(
+    int consumer_render_frame_id,
+    const StreamDeviceInfo& device_info)
+    : MediaStreamAudioSource(true /* is_local_source */),
+      consumer_render_frame_id_(consumer_render_frame_id) {
+  DVLOG(1) << "LocalMediaStreamAudioSource::LocalMediaStreamAudioSource()";
+  MediaStreamSource::SetDeviceInfo(device_info);
+  MediaStreamAudioSource::SetFormat(media::AudioParameters(
+      media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      static_cast<media::ChannelLayout>(
+          device_info.device.input.channel_layout),
+      device_info.device.input.sample_rate,
+      16,  // Legacy parameter (data is always in 32-bit float format).
+      // If specified, use frames_per_buffer; else, default to 10 ms.
+      device_info.device.input.frames_per_buffer > 0 ?
+          device_info.device.input.frames_per_buffer :
+          device_info.device.input.sample_rate / 100));

[o1ka, 2016/08/15 15:15:00]

Should we probably unify ProcessedLocalAudioSource::GetBufferSize()
(https://cs.chromium.org/chromium/src/content/renderer/media/webrtc/processed_...)
and use it here? Do we probably want to keep Android behavior unchanged in this
CL (2 * sample_rate / 100)?

[miu, 2016/08/16 04:30:02]

Done. Opened a crbug for tracking (http://crbug.com/638081) with TODO comment.

+}
+
+LocalMediaStreamAudioSource::~LocalMediaStreamAudioSource() {
+  DVLOG(1) << "LocalMediaStreamAudioSource::~LocalMediaStreamAudioSource()";
+  EnsureSourceIsStopped();
+}
+
+bool LocalMediaStreamAudioSource::EnsureSourceIsStarted() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  if (source_)
+    return true;
+
+  // Sanity-check that the consuming RenderFrame still exists. This is required
+  // by AudioDeviceFactory.
+  if (!RenderFrameImpl::FromRoutingID(consumer_render_frame_id_))
+    return false;
+
+  VLOG(1) << "Starting local audio input device (session_id="
+          << device_info().session_id << ") for render frame "
+          << consumer_render_frame_id_ << " with audio parameters={"
+          << GetAudioParameters().AsHumanReadableString() << "}.";
+
+  source_ =
+      AudioDeviceFactory::NewAudioCapturerSource(consumer_render_frame_id_);
+  source_->Initialize(GetAudioParameters(), this, device_info().session_id);
+  source_->Start();
+  return true;
+}
+
+void LocalMediaStreamAudioSource::EnsureSourceIsStopped() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+
+  if (!source_)
+    return;
+
+  source_->Stop();
+  source_ = nullptr;
+
+  VLOG(1) << "Stopped local audio input device (session_id="
+          << device_info().session_id << ") for render frame "
+          << consumer_render_frame_id_ << " with audio parameters={"
+          << GetAudioParameters().AsHumanReadableString() << "}.";
+}
+
+void LocalMediaStreamAudioSource::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
+  MediaStreamAudioSource::DeliverDataToTracks(
+      *audio_bus,
+      base::TimeTicks::Now() -
+      base::TimeDelta::FromMilliseconds(audio_delay_milliseconds));
+}
+
+void LocalMediaStreamAudioSource::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