Add the copresence DirectiveHandler.

components/copresence/mediums/audio/audio_recorder.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 "components/copresence/mediums/audio/audio_recorder.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "base/bind.h"
+#include "base/bind_helpers.h"
+#include "base/logging.h"
+#include "base/run_loop.h"
+#include "base/synchronization/waitable_event.h"
+#include "components/copresence/public/copresence_constants.h"
+#include "content/public/browser/browser_thread.h"
+#include "media/audio/audio_manager.h"
+#include "media/audio/audio_manager_base.h"
+#include "media/base/audio_bus.h"
+
+namespace {
+
+const float kProcessIntervalMs = 500.0f;  // milliseconds.
+
+void AudioBusToString(scoped_ptr<media::AudioBus> source, std::string* buffer) {
+  buffer->resize(source->frames() * source->channels() * sizeof(float));
+  float* buffer_view = reinterpret_cast<float*>(string_as_array(buffer));
+
+  const int channels = source->channels();
+  for (int ch = 0; ch < channels; ++ch) {
+    for (int si = 0, di = ch; si < source->frames(); ++si, di += channels)
+      buffer_view[di] = source->channel(ch)[si];
+  }
+}
+
+// Called every kProcessIntervalMs to process the recorded audio. This
+// converts our samples to the required sample rate, interleaves the samples
+// and sends them to the whispernet decoder to process.
+void ProcessSamples(
+    scoped_ptr<media::AudioBus> bus,
+    const copresence::AudioRecorder::DecodeSamplesCallback& callback) {

[Daniel Erat, 2014/07/31 22:31:16]

nit: nest the anon namespace under the copresence namespace so you don't need
the qualifier here

[rkc, 2014/08/01 21:08:57]

Done.

+  std::string samples;
+  AudioBusToString(bus.Pass(), &samples);
+  content::BrowserThread::PostTask(
+      content::BrowserThread::UI, FROM_HERE, base::Bind(callback, samples));
+}
+
+}  // namespace
+
+namespace copresence {
+
+// Public methods.
+
+AudioRecorder::AudioRecorder(const DecodeSamplesCallback& decode_callback)
+    : stream_(NULL),
+      is_recording_(false),
+      decode_callback_(decode_callback),
+      total_buffer_frames_(0),
+      buffer_frame_index_(0),
+      input_stream_for_testing_(NULL) {
+}
+
+void AudioRecorder::Initialize() {
+  media::AudioManager::Get()->GetTaskRunner()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioRecorder::InitializeOnAudioThread,
+                 base::Unretained(this)));
+}
+
+AudioRecorder::~AudioRecorder() {
+}
+
+void AudioRecorder::Record() {
+  media::AudioManager::Get()->GetTaskRunner()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioRecorder::RecordOnAudioThread, base::Unretained(this)));
+}
+
+void AudioRecorder::Stop() {
+  media::AudioManager::Get()->GetTaskRunner()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioRecorder::StopOnAudioThread, base::Unretained(this)));
+}
+
+void AudioRecorder::Finalize() {
+  media::AudioManager::Get()->GetTaskRunner()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioRecorder::FinalizeOnAudioThread,
+                 base::Unretained(this)));
+}
+
+// Private methods.
+
+void AudioRecorder::InitializeOnAudioThread() {
+  DCHECK(media::AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread());
+
+  media::AudioParameters params =
+      params_for_testing_
+          ? *params_for_testing_
+          : media::AudioManager::Get()->GetInputStreamParameters(
+                media::AudioManagerBase::kDefaultDeviceId);
+
+  const media::AudioParameters dest_params(params.format(),
+                                           kDefaultChannelLayout,
+                                           kDefaultChannels,
+                                           params.input_channels(),
+                                           kDefaultSampleRate,
+                                           kDefaultBitsPerSample,
+                                           params.frames_per_buffer(),
+                                           media::AudioParameters::NO_EFFECTS);
+
+  converter_.reset(new media::AudioConverter(
+      params, dest_params, params.sample_rate() == dest_params.sample_rate()));
+  converter_->AddInput(this);
+
+  total_buffer_frames_ = kProcessIntervalMs * dest_params.sample_rate() / 1000;
+  buffer_ =
+      media::AudioBus::Create(dest_params.channels(), total_buffer_frames_);
+  buffer_frame_index_ = 0;
+
+  stream_ = input_stream_for_testing_ ? input_stream_for_testing_ : stream_ =

[Daniel Erat, 2014/07/31 22:31:16]

uh, what's up with the "stream_ =" at the end of this line?

[rkc, 2014/08/01 21:08:57]

Done.

+                media::AudioManager::Get()->MakeAudioInputStream(
+                    params, media::AudioManagerBase::kDefaultDeviceId);
+
+  if (!stream_ || !stream_->Open()) {
+    LOG(ERROR) << "Failed to open an input stream.";
+    if (stream_) {
+      stream_->Close();

[Daniel Erat, 2014/07/31 22:31:16]

do you actually need to call Close() here after Open() fails? if so, can this
method just call StopAndCloseOnAudioThread()?

[rkc, 2014/08/01 21:08:57]

If the stream open fails, we should close the stream (at least that seems to be
the way the current audio code is recommended. dalecurtis@ reviewed this while
we were still closed source and recommended this method).
StopAndClose tries to stop the stream, which isn't started yet.

+      stream_ = NULL;
+    }
+    return;
+  }
+  stream_->SetVolume(stream_->GetMaxVolume());
+}
+
+void AudioRecorder::RecordOnAudioThread() {
+  DCHECK(media::AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread());
+  if (!stream_ || is_recording_)
+    return;
+
+  DVLOG(2) << "Recording Audio.";
+  converter_->Reset();
+  stream_->Start(this);
+  is_recording_ = true;
+}
+
+void AudioRecorder::StopOnAudioThread() {
+  DCHECK(media::AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread());
+  if (!stream_)
+    return;
+
+  stream_->Stop();
+  is_recording_ = false;
+}
+
+void AudioRecorder::StopAndCloseOnAudioThread() {
+  DCHECK(media::AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread());
+  if (!stream_)
+    return;
+
+  if (is_recording_)

[Daniel Erat, 2014/07/31 22:31:16]

why do you check is_recording_ here but not in StopOnAudioThread()?

[rkc, 2014/08/01 21:08:57]

StopOnAudioThread is called when a user explicitly wants to stop a stream, so we
make the assumption that the user _has started recording before then.
StopAndClose can get called even when the user hasn't started recording (in case
he creates the recorder and just destructs it).

I added a check in Stop also, just in case the user doesn't invoke the method
correctly.

+    stream_->Stop();

[Daniel Erat, 2014/07/31 22:31:16]

can this method just call StopOnAudioThread() and then close the stream and set
it to null?

[rkc, 2014/08/01 21:08:57]

Sure.
Done.

+  stream_->Close();
+  stream_ = NULL;
+
+  is_recording_ = false;
+}
+
+void AudioRecorder::FinalizeOnAudioThread() {
+  DCHECK(media::AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread());
+  StopAndCloseOnAudioThread();
+  if (input_stream_for_testing_)
+    delete input_stream_for_testing_;
+  delete this;
+}
+
+void AudioRecorder::OnData(media::AudioInputStream* stream,
+                           const media::AudioBus* source,
+                           uint32 /* hardware_delay_bytes */,
+                           double /* volume */) {
+  temp_conversion_buffer_ = source;
+  while (temp_conversion_buffer_) {
+    // source->frames() == source_params.frames_per_buffer(), so we only have
+    // one chunk of data in the source; correspondingly set the destination
+    // size to one chunk.
+    // TODO(rkc): Optimize this to directly write into buffer_ so we can avoid
+    // the copy into this buffer and then the copy back into buffer_.
+    scoped_ptr<media::AudioBus> converted_source =
+        media::AudioBus::Create(kDefaultChannels, converter_->ChunkSize());

[Daniel Erat, 2014/07/31 22:31:16]

nit: add blank line after this one so this isn't a huge block of text

[rkc, 2014/08/01 21:08:57]

Done.

+    // Convert accumulated samples into converted_source. Note: One call may not
+    // be enough to consume the samples from |source|.  The converter may have
+    // accumulated samples over time due to a fractional input:output sample
+    // rate ratio.  Since |source| is ephemeral, Convert() must be called until
+    // |source| is at least buffered into the converter.  Once |source| is
+    // consumed during ProvideInput(), |temp_conversion_buffer_| will be set to
+    // NULL, which will break the conversion loop.

[Daniel Erat, 2014/07/31 22:31:16]

i don't understand this comment. is it saying that some other thread will run
ProvideInput() and set temp_conversion_buffer_ to NULL while this loop is
running? that doesn't seem safe.

[rkc, 2014/08/01 21:08:57]

No, this is all single threaded. ->Convert is what will call ProvideInput (on
the same thread), which will set temp_conversion_buffer_ to NULL once its done.

+    converter_->Convert(converted_source.get());
+
+    int remaining_buffer_frames = buffer_->frames() - buffer_frame_index_;
+    int frames_to_copy =
+        std::min(remaining_buffer_frames, converted_source->frames());
+    converted_source->CopyPartialFramesTo(
+        0, frames_to_copy, buffer_frame_index_, buffer_.get());
+    buffer_frame_index_ += frames_to_copy;
+
+    // Buffer full, send it for processing.
+    if (buffer_->frames() == buffer_frame_index_) {
+      ProcessSamples(buffer_.Pass(), decode_callback_);
+      buffer_ = media::AudioBus::Create(kDefaultChannels, total_buffer_frames_);
+      buffer_frame_index_ = 0;
+
+      // Copy any remaining frames in the source to our buffer.
+      int remaining_source_frames = converted_source->frames() - frames_to_copy;

[Daniel Erat, 2014/07/31 22:31:16]

is converted_source->frames() guaranteed to be <= total_buffer_frames_? if not,
don't you need to check that you're not going to overflow buffer_ here? it seems
like it would be wise to use std::min() to ensure that this doesn't happen
anyway.

[rkc, 2014/08/01 21:08:57]

total_buffer_frames_ is orders of magnitude higher than
converted_source->frames() can get to.

+      converted_source->CopyPartialFramesTo(frames_to_copy,
+                                            remaining_source_frames,
+                                            buffer_frame_index_,
+                                            buffer_.get());

[Daniel Erat, 2014/07/31 22:31:16]

it seems like you'll never process the last chunk of samples that's left over
from the final call to OnData(). is that a problem?

[rkc, 2014/08/01 21:08:57]

Whatever samples are leftover after the last ProcessSamples call will be picked
up by the next OnData call and processed with the next. The only case that won't
happen is if the user stops recording, in which case he doesn't care about
processing any more samples. Depending on when the user stops recording, he will
lose between 0 to 499ms of samples anyway (unless we decide to send out another
ProcessSamples on the user stopping recording - there are pros and cons to this,
we decided on letting the last half a second go).

+      buffer_frame_index_ += remaining_source_frames;
+    }
+  }
+}
+
+void AudioRecorder::OnError(media::AudioInputStream* /* stream */) {
+  LOG(ERROR) << "Error during sound recording.";
+  media::AudioManager::Get()->GetTaskRunner()->PostTask(
+      FROM_HERE,
+      base::Bind(&AudioRecorder::StopAndCloseOnAudioThread,
+                 base::Unretained(this)));
+}
+
+double AudioRecorder::ProvideInput(media::AudioBus* dest,
+                                   base::TimeDelta /* buffer_delay */) {
+  DCHECK(temp_conversion_buffer_);
+  DCHECK_LE(temp_conversion_buffer_->frames(), dest->frames());
+  temp_conversion_buffer_->CopyTo(dest);
+  temp_conversion_buffer_ = NULL;
+  return 1.0;
+}
+
+void AudioRecorder::FlushAudioLoopForTesting() {
+  if (media::AudioManager::Get()->GetTaskRunner()->BelongsToCurrentThread())
+    return;
+
+  // Queue task on the audio thread, when it is executed, that means we've
+  // successfully executed all the tasks before us.
+  base::RunLoop rl;
+  media::AudioManager::Get()->GetTaskRunner()->PostTaskAndReply(
+      FROM_HERE,
+      base::Bind(base::IgnoreResult(&AudioRecorder::FlushAudioLoopForTesting),
+                 base::Unretained(this)),
+      rl.QuitClosure());
+  rl.Run();
+}
+
+}  // namespace copresence