Browser-wide audio mirroring for TabCapture API

media/audio/virtual_audio_input_stream.cc

+// 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 "media/audio/virtual_audio_input_stream.h"
+
+#include <algorithm>
+
+#include "base/bind.h"
+#include "base/message_loop.h"
+#include "media/audio/virtual_audio_output_stream.h"
+
+namespace media {
+
+class MEDIA_EXPORT LoopbackAudioConverter

[DaleCurtis, 2012/11/21 23:45:43]

Class comment.

[miu, 2012/11/22 00:53:50]

I don't think you want to export this.  Or, does the linker disagree?

[justinlin, 2012/11/26 20:19:20]

Done.

[justinlin, 2012/11/26 20:19:20]

Done.

+    : public AudioConverter,
+      public AudioConverter::InputCallback {
+ public:
+  LoopbackAudioConverter(const AudioParameters& input_params,
+                         const AudioParameters& output_params);
+  virtual ~LoopbackAudioConverter() {}
+
+ private:
+  virtual double ProvideInput(
+      AudioBus* audio_bus, base::TimeDelta buffer_delay) OVERRIDE;
+
+  DISALLOW_COPY_AND_ASSIGN(LoopbackAudioConverter);
+};
+
+VirtualAudioInputStream* VirtualAudioInputStream::MakeStream(
+    AudioManagerBase* manager, const AudioParameters& params) {
+  return new VirtualAudioInputStream(manager, params);
+}
+
+VirtualAudioInputStream::VirtualAudioInputStream(AudioManagerBase* manager,
+                                                 const AudioParameters& params)
+    : audio_manager_(manager),
+      callback_(NULL),
+      buffer_(new uint8[params.GetBytesPerBuffer()]),
+      buffer_size_(params.GetBytesPerBuffer()),

[DaleCurtis, 2012/11/21 23:45:43]

Class variable necessary if you're storing params_ ?

[justinlin, 2012/11/26 20:19:20]

Done.

+      params_(params),
+      audio_bus_(AudioBus::Create(params_)),
+      mixer_(params_, params_, false) {
+}
+
+VirtualAudioInputStream::~VirtualAudioInputStream() {
+  for (AudioConvertersMap::iterator it = converters_.begin();
+       it != converters_.end(); ++it) {
+    delete it->second;
+  }
+}
+
+bool VirtualAudioInputStream::Open() {
+  memset(buffer_.get(), 0, buffer_size_);
+  return true;
+}
+
+void VirtualAudioInputStream::Start(AudioInputCallback* callback) {

[DaleCurtis, 2012/11/21 23:45:43]

Methods need DCHECK(manager->GetMessageLoop()->BelongsTOCurrentThread()) for
clarity.

[justinlin, 2012/11/26 20:19:20]

Done.

+  callback_ = callback;
+  on_more_data_cb_.Reset(base::Bind(
+      &VirtualAudioInputStream::DoCallback, base::Unretained(this)));
+  audio_manager_->GetMessageLoop()->PostTask(
+      FROM_HERE, on_more_data_cb_.callback());
+}
+
+void VirtualAudioInputStream::Stop() {
+  on_more_data_cb_.Cancel();
+}
+
+void VirtualAudioInputStream::AddOutputStream(
+    VirtualAudioOutputStream* stream, const AudioParameters& params) {
+  output_params_.insert(
+      std::make_pair(stream, params));

[miu, 2012/11/22 00:53:50]

Fits on one line.

[justinlin, 2012/11/26 20:19:20]

Done.

+
+  LoopbackAudioConverter* converter =

[DaleCurtis, 2012/11/21 23:45:43]

You should check if the converter exists first before creating a new one.  Also
it'd be nice if you used scoped_ptr() instead of raw pointer everywhere.

[miu, 2012/11/22 00:53:50]

This code could be simplified as:

  LoopbackAudioConverter*& converter = converters_[params];
  if (!converter) {
    converter = new LoopbackAudioConverter(params, params_);
    mixer_.AddInput(converter);
  }
  converter->AddInput(stream);

[justinlin, 2012/11/26 20:19:20]

Done. Thanks, that looks good.

[justinlin, 2012/11/26 20:19:20]

Done.

+      new LoopbackAudioConverter(params, params_);
+  std::pair<AudioConvertersMap::iterator, bool> result =
+      converters_.insert(std::make_pair(params, converter));
+
+  // Add to main mixer if we just added a new AudioTransform, otherwise just
+  // use the existing one for these audio params.
+  if (result.second) {
+    mixer_.AddInput(result.first->second);
+  } else {
+    delete converter;
+    converter = result.first->second;
+  }
+
+  converter->AddInput(stream);
+}
+
+void VirtualAudioInputStream::RemoveOutputStream(
+    VirtualAudioOutputStream* stream) {
+  if (output_params_.find(stream) == output_params_.end())

[DaleCurtis, 2012/11/21 23:45:43]

DCHECK() this instead for sanity, seems crazy that RemoveOutputStream is called
with wild values.

[justinlin, 2012/11/26 20:19:20]

Done.

+    return;
+  const AudioParameters& params = output_params_[stream];
+
+  DCHECK(converters_.find(params) != converters_.end());
+  converters_[params]->RemoveInput(stream);
+  // TODO(justinlin): Do we need to delete the AudioConverter and remove it from

[DaleCurtis, 2012/11/21 23:45:43]

Probably okay since virtual input is a temporary task and destroys everything
when it goes away.

[justinlin, 2012/11/26 20:19:20]

OK, maybe I'll just leave the TODO For now? Doing this properly isn't too nice
because we'd need additional book-keeping in this class or we would need to add
a method to query the number of input streams to an AudioConverter.

+  // the mixer when there's no more input streams?
+
+  output_params_.erase(stream);
+}
+
+void VirtualAudioInputStream::DoCallback() {
+  DCHECK(callback_);
+  DCHECK(audio_manager_->GetMessageLoop()->BelongsToCurrentThread());
+
+  int frames_received = params_.frames_per_buffer();

[DaleCurtis, 2012/11/21 23:45:43]

Seems like this is a constant that could be calculated at construction time.

[justinlin, 2012/11/26 20:19:20]

Done.

+  float frames_per_millisecond = params_.sample_rate() / static_cast<float>(
+      base::Time::kMillisecondsPerSecond);
+
+  mixer_.Convert(audio_bus_.get());
+  audio_bus_->ToInterleaved(params_.frames_per_buffer(),
+                            params_.bits_per_sample() / 8,
+                            buffer_.get());
+
+  callback_->OnData(this, buffer_.get(), buffer_size_, buffer_size_, 1.0);
+
+  MessageLoop::current()->PostDelayedTask(

[DaleCurtis, 2012/11/21 23:45:43]

Hmm this seems to indicate that you're driving Audio output by the input clock. 
Should be a good stress test of the delay information present in AudioConverter.
I'm curious what your sync tests look like for playback -> capture -> playback.

[justinlin, 2012/11/26 20:19:20]

Yup, I either use an extension that uses the TabCapture API, and turn that
ON/OFF, or I comment out some stuff in AudioManagerBase so that it always
creates the virtual input stream, and use apprtc.appspot.com (reloading the page
will cancel audio mirroring).

+      FROM_HERE, on_more_data_cb_.callback(),
+      base::TimeDelta::FromMilliseconds(

[miu, 2012/11/22 00:53:50]

Don't you have to subtract the amount of time that was spent in this method? 
Otherwise, the callback is being executed less often than it needs to be to
provide audio fast enough.

[justinlin, 2012/11/26 20:19:20]

Added a DO_NOT_SUBMIT, will look at this in a bit.

+          frames_received / frames_per_millisecond));
+}
+
+void VirtualAudioInputStream::Close() {
+  if (callback_) {
+    callback_->OnClose(this);

[DaleCurtis, 2012/11/21 23:45:43]

DCHECK(on_more_data_cb_.Cancelled()) ?

[justinlin, 2012/11/26 20:19:20]

Done.

+    callback_ = NULL;
+  }
+  audio_manager_->ReleaseInputStream(this);
+}
+
+double VirtualAudioInputStream::GetMaxVolume() {
+  return 1.0;
+}
+
+void VirtualAudioInputStream::SetVolume(double volume) {}
+
+double VirtualAudioInputStream::GetVolume() {
+  return 1.0;
+}
+
+void VirtualAudioInputStream::SetAutomaticGainControl(bool enabled) {}
+
+bool VirtualAudioInputStream::GetAutomaticGainControl() {
+  return false;
+}
+
+LoopbackAudioConverter::LoopbackAudioConverter(
+    const AudioParameters& input_params, const AudioParameters& output_params)
+    : AudioConverter(input_params, output_params, false) {
+}
+
+double LoopbackAudioConverter::ProvideInput(
+    AudioBus* audio_bus, base::TimeDelta buffer_delay) {
+  Convert(audio_bus);
+  return 1.0;
+}
+
+}  // namespace media