Binding media stream audio track to speech recognition [renderer]

content/renderer/media/speech_recognition_audio_source_provider.cc

+// Copyright 2014 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/speech_recognition_audio_source_provider.h"
+
+#include "base/logging.h"
+#include "base/memory/shared_memory.h"
+#include "base/time/time.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_fifo.h"
+
+namespace content {
+
+SpeechRecognitionAudioSourceProvider::SpeechRecognitionAudioSourceProvider(
+    const blink::WebMediaStreamTrack& track,
+    const media::AudioParameters& params, const base::SharedMemoryHandle memory,
+    base::SyncSocket* socket, OnStoppedCB on_stopped_cb)
+    : track_(track),
+      shared_memory_(memory, false),
+      socket_(socket),
+      output_params_(params),
+      track_stopped_(false),
+      buffer_index_(0),
+      on_stopped_cb_(on_stopped_cb) {
+  DCHECK(main_render_thread_checker_.CalledOnValidThread());
+  DCHECK(params.IsValid());
+  DCHECK(IsSupportedTrack(track));
+  const size_t memory_length = media::AudioBus::CalculateMemorySize(params) +
+                               sizeof(media::AudioInputBufferParameters);
+  CHECK(shared_memory_.Map(memory_length));
+
+  // Buffer index for sync with client is |params.size| on the shared memory.
+  uint8* ptr = static_cast<uint8*>(shared_memory_.memory());
+  media::AudioInputBuffer* buffer =
+      reinterpret_cast<media::AudioInputBuffer*>(ptr);
+  peer_buffer_index_ = &(buffer->params.size);
+
+  // Client must manage his own counter and reset it.
+  DCHECK_EQ(0U, *peer_buffer_index_);
+  output_bus_ = media::AudioBus::WrapMemory(params, buffer->audio);
+
+  // Connect the source provider to the track as a sink.
+  MediaStreamAudioSink::AddToAudioTrack(this, track_);
+}
+
+SpeechRecognitionAudioSourceProvider::~SpeechRecognitionAudioSourceProvider() {
+  DCHECK(main_render_thread_checker_.CalledOnValidThread());
+  if (audio_converter_.get()) audio_converter_->RemoveInput(this);

[no longer working on chromium, 2014/09/17 15:55:19]

nit, new line

[burnik, 2014/09/18 09:19:35]

Done.

+
+  // Notify the track before this sink goes away.
+  if (!track_stopped_)
+    MediaStreamAudioSink::RemoveFromAudioTrack(this, track_);
+}
+
+// static
+bool SpeechRecognitionAudioSourceProvider::IsSupportedTrack(
+    const blink::WebMediaStreamTrack& track) {
+  if (track.source().type() != blink::WebMediaStreamSource::TypeAudio)
+    return false;
+
+  MediaStreamAudioSource* native_source =
+      static_cast<MediaStreamAudioSource*>(track.source().extraData());
+  if (!native_source)
+    return false;
+
+  const StreamDeviceInfo& device_info = native_source->device_info();
+  // Purposely only support tracks from an audio device. Dissallow WebAudio.
+  return (device_info.device.type == content::MEDIA_DEVICE_AUDIO_CAPTURE);
+}
+
+void SpeechRecognitionAudioSourceProvider::OnSetFormat(
+    const media::AudioParameters& input_params) {
+  // We need detach the thread here because it will be a new capture thread
+  // calling OnSetFormat() and OnData() if the source is restarted.
+  capture_thread_checker_.DetachFromThread();
+  DCHECK(capture_thread_checker_.CalledOnValidThread());
+  DCHECK(input_params.IsValid());
+
+  input_params_ = input_params;
+  fifo_buffer_size_ = output_params_.frames_per_buffer() *
+                      input_params_.sample_rate() /
+                      output_params_.sample_rate();
+  DCHECK_GE(fifo_buffer_size_, input_params_.frames_per_buffer());
+  DCHECK_GE(fifo_buffer_size_, output_params_.frames_per_buffer());
+
+  // Allows for some delays on the endpoint client.
+  static const int kNumberOfBuffersInFifo = 2;
+  int frames_in_fifo = kNumberOfBuffersInFifo * fifo_buffer_size_;
+  fifo_.reset(new media::AudioFifo(input_params.channels(), frames_in_fifo));
+  input_bus_ = media::AudioBus::Create(input_params.channels(),
+                                       input_params.frames_per_buffer());
+
+  // Create the audio converter with |disable_fifo| as false so that the
+  // converter will request input_params.frames_per_buffer() each time.
+  // This will not increase the complexity as there is only one client to
+  // the converter.
+  audio_converter_.reset(
+      new media::AudioConverter(input_params, output_params_, false));
+  audio_converter_->AddInput(this);
+}
+
+void SpeechRecognitionAudioSourceProvider::OnReadyStateChanged(
+    blink::WebMediaStreamSource::ReadyState state) {
+  DCHECK(main_render_thread_checker_.CalledOnValidThread());
+  DCHECK(!track_stopped_);
+
+  if (state == blink::WebMediaStreamSource::ReadyStateEnded) {
+    track_stopped_ = true;
+    if (!on_stopped_cb_.is_null())
+      on_stopped_cb_.Run();
+  }
+}
+
+void SpeechRecognitionAudioSourceProvider::OnData(const int16* audio_data,
+                                                  int sample_rate,
+                                                  int number_of_channels,
+                                                  int number_of_frames) {
+  DCHECK(capture_thread_checker_.CalledOnValidThread());
+  DCHECK(peer_buffer_index_);
+  DCHECK_EQ(input_bus_->frames(), number_of_frames);
+  DCHECK_EQ(input_bus_->channels(), number_of_channels);
+  if (fifo_->frames() + number_of_frames > fifo_->max_frames()) {
+    NOTREACHED() << "Audio FIFO overflow";
+    return;
+  }
+  // TODO(xians): A better way to handle the interleaved and deinterleaved
+  // format switching, see issue/317710.
+  input_bus_->FromInterleaved(audio_data, number_of_frames,
+                              sizeof(audio_data[0]));
+
+  fifo_->Push(input_bus_.get());
+  // Wait for FIFO to have at least |fifo_buffer_size_| frames ready.
+  if (fifo_->frames() < fifo_buffer_size_)
+    return;
+
+  // Make sure the previous output buffer was consumed by client before we send
+  // the next buffer. |peer_buffer_index_| is pointing to shared memory.
+  // The client must write to it (incrementing by 1) once the the buffer was
+  // consumed. This is intentional not to block this audio capturing thread.
+  if (buffer_index_ != (*peer_buffer_index_)) {
+    DLOG(WARNING) << "Buffer synchronization lag";

[no longer working on chromium, 2014/09/17 15:55:19]

hmm, thinking about these code a bit, it seems risky to me. Is it possible that
something wrong with the socket, and the indexes gets out of sync and they will
never get recovered because of this check?

[burnik, 2014/09/18 09:19:35]

If indices are out of sync, that would indicate that the |peer_buffer_index_| is
less than |buffer_index_|.
The return statement here gives a chance to the peer process to catch up. If in
turn it doesn't, we will end up in the NOTREACHED() and in production just
ignore any incoming data since the FIFO is already full.

On 2014/09/17 15:55:19, xians1 wrote:

+    return;
+  }
+
+  audio_converter_->Convert(output_bus_.get());
+
+  // Notify client to consume buffer |buffer_index_| on |output_bus_|.
+  const size_t bytes_sent =
+      socket_->Send(&buffer_index_, sizeof(buffer_index_));
+  if (bytes_sent != sizeof(buffer_index_)) {
+    // The send usually fails if the user changes his input audio device.
+    DVLOG(1) << "Failed sending buffer index to peer";
+    // We have discarded this buffer, but could still recover on the next one.
+    return;
+  }
+
+  // Count the sent buffer. We expect the client to do the same on his end.
+  ++buffer_index_;
+}
+
+double SpeechRecognitionAudioSourceProvider::ProvideInput(
+    media::AudioBus* audio_bus, base::TimeDelta buffer_delay) {
+  DCHECK(capture_thread_checker_.CalledOnValidThread());
+  if (fifo_->frames() >= audio_bus->frames())
+    fifo_->Consume(audio_bus, 0, audio_bus->frames());
+  else
+    audio_bus->Zero();
+
+  return 1.0;
+}
+
+}  // namespace content