Very early implementation for HTMLMediaElement / Web Audio API integration

content/renderer/media/audio_renderer_impl.cc

Index: content/renderer/media/audio_renderer_impl.cc
===================================================================
--- content/renderer/media/audio_renderer_impl.cc	(revision 96995)
+++ content/renderer/media/audio_renderer_impl.cc	(working copy)
@@ -6,8 +6,8 @@
 
 #include <math.h>
 
-#include "content/common/child_process.h"
 #include "base/command_line.h"
+#include "content/common/child_process.h"
 #include "content/common/content_switches.h"
 #include "content/common/media/audio_messages.h"
 #include "content/renderer/render_thread.h"
@@ -15,45 +15,21 @@
 #include "media/audio/audio_output_controller.h"
 #include "media/base/filter_host.h"
 
-// Static variable that says what code path we are using -- low or high
-// latency. Made separate variable so we don't have to go to command line
-// for every DCHECK().
-AudioRendererImpl::LatencyType AudioRendererImpl::latency_type_ =
-    AudioRendererImpl::kUninitializedLatency;
+const size_t kBufferSize = 128; //2048;
 
-AudioRendererImpl::AudioRendererImpl()
+AudioRendererImpl::AudioRendererImpl(MessageLoop* render_loop)
     : AudioRendererBase(),
       bytes_per_second_(0),
-      stream_id_(0),
-      shared_memory_(NULL),
-      shared_memory_size_(0),
       stopped_(false),
       pending_request_(false),
       prerolling_(false),
-      preroll_bytes_(0) {
-  filter_ = RenderThread::current()->audio_message_filter();
-  // Figure out if we are planning to use high or low latency code path.
-  // We are initializing only one variable and double initialization is Ok,
-  // so there would not be any issues caused by CPU memory model.
-  if (latency_type_ == kUninitializedLatency) {
-    if (CommandLine::ForCurrentProcess()->HasSwitch(
-        switches::kLowLatencyAudio)) {
-      latency_type_ = kLowLatency;
-    } else {
-      latency_type_ = kHighLatency;
-    }
-  }
+      preroll_bytes_(0),
+      render_loop_(render_loop) {
 }
 
 AudioRendererImpl::~AudioRendererImpl() {
 }
 
-// static
-void AudioRendererImpl::set_latency_type(LatencyType latency_type) {
-  DCHECK_EQ(kUninitializedLatency, latency_type_);
-  latency_type_ = latency_type;
-}
-
 base::TimeDelta AudioRendererImpl::ConvertToDuration(int bytes) {
   if (bytes_per_second_) {
     return base::TimeDelta::FromMicroseconds(
@@ -64,13 +40,9 @@
 
 bool AudioRendererImpl::OnInitialize(const media::AudioDecoderConfig& config) {
   AudioParameters params(config);
-  params.format = AudioParameters::AUDIO_PCM_LINEAR;
-
-  bytes_per_second_ = params.GetBytesPerSecond();
-
-  ChildProcess::current()->io_message_loop()->PostTask(
+  render_loop_->PostTask(
       FROM_HERE,
-      NewRunnableMethod(this, &AudioRendererImpl::CreateStreamTask, params));
+      NewRunnableMethod(this, &AudioRendererImpl::InitializeTask, params));
   return true;
 }
 
@@ -80,25 +52,12 @@
     return;
   stopped_ = true;
 
-  ChildProcess::current()->io_message_loop()->PostTask(
-      FROM_HERE,
-      NewRunnableMethod(this, &AudioRendererImpl::DestroyTask));
-
-  if (audio_thread_.get()) {
-    socket_->Close();
-    audio_thread_->Join();
+  if (audio_device_.get()) {
+    audio_device_->Stop();
+    audio_device_ = NULL;
   }
 }
 
-void AudioRendererImpl::NotifyDataAvailableIfNecessary() {
-  if (latency_type_ == kHighLatency) {
-    // Post a task to render thread to notify a packet reception.
-    ChildProcess::current()->io_message_loop()->PostTask(
-        FROM_HERE,
-        NewRunnableMethod(this, &AudioRendererImpl::NotifyPacketReadyTask));
-  }
-}
-
 void AudioRendererImpl::ConsumeAudioSamples(
     scoped_refptr<media::Buffer> buffer_in) {
   base::AutoLock auto_lock(lock_);
@@ -110,9 +69,61 @@
   // Use the base class to queue the buffer.
   AudioRendererBase::ConsumeAudioSamples(buffer_in);
 
-  NotifyDataAvailableIfNecessary();
 }
 
+void AudioRendererImpl::Render(const std::vector<float*>& audio_data,

[scherkus (not reviewing), 2011/08/23 15:16:01]

I haven't taken an indepth look at this class but we'll likely have to do some
manual testing to make sure it works

[Chris Rogers, 2011/08/24 00:41:52]

Agreed.  Also, we should loop Aaron into the conversation since he's the one who
provided the majority of the code for this file.

On 2011/08/23 15:16:01, scherkus wrote:

+                               size_t number_of_frames,
+                               size_t audio_delay_milliseconds) {
+  // LOG(ERROR) << "Render("<< audio_data.size() << ", "
+  //            << number_of_frames << ", "
+  //            << audio_delay_milliseconds << ")";
+
+  base::AutoLock auto_lock(lock_);
+  if (stopped_)
+    return;
+
+  if (GetPlaybackRate() > 0.0f) {
+    // Adjust the playback delay.
+    base::Time current_time = base::Time::Now();
+
+    base::TimeDelta request_delay =
+        base::TimeDelta::FromMilliseconds(audio_delay_milliseconds);
+
+    // Finally we need to adjust the delay according to playback rate.
+    if (GetPlaybackRate() != 1.0f) {
+      request_delay = base::TimeDelta::FromMicroseconds(
+          static_cast<int64>(ceil(request_delay.InMicroseconds() *
+                                  GetPlaybackRate())));
+    }
+
+    int buf_size = number_of_frames * bytes_per_frame_;
+    int bytes_per_sample = bytes_per_frame_ / audio_data.size();
+    DCHECK_EQ(bytes_per_sample, 2);
+    scoped_array<uint8> buf(new uint8[buf_size]);
+    uint32 filled = FillBuffer(buf.get(), buf_size, request_delay, false);
+    uint32 filled_frames = filled  / bytes_per_frame_;
+    int stride = audio_data.size();
+    for (size_t i = 0; i < audio_data.size(); ++i) {
+      short* pSrc = reinterpret_cast<short*>(buf.get()) + i;
+      float* pDst = audio_data[i];
+
+      for (size_t j = 0; j < filled_frames; ++j) {
+        *pDst++ = *pSrc / 32768.0f;
+        pSrc += stride;
+      }
+    }
+  }
+}
+
+void AudioRendererImpl::SetAudioSink(AudioSink* audio_sink)
+{
+    if (audio_device_.get()) {
+      audio_device_->Stop();
+    }
+    
+    audio_device_ = audio_sink;
+}
+
 void AudioRendererImpl::SetPlaybackRate(float rate) {
   DCHECK_LE(0.0f, rate);
 
@@ -127,22 +138,16 @@
   // Play: GetPlaybackRate() == 0.0 && rate != 0.0
   // Pause: GetPlaybackRate() != 0.0 && rate == 0.0
   if (GetPlaybackRate() == 0.0f && rate != 0.0f) {
-    ChildProcess::current()->io_message_loop()->PostTask(
+    render_loop_->PostTask(
         FROM_HERE,
         NewRunnableMethod(this, &AudioRendererImpl::PlayTask));
   } else if (GetPlaybackRate() != 0.0f && rate == 0.0f) {
     // Pause is easy, we can always pause.
-    ChildProcess::current()->io_message_loop()->PostTask(
+    render_loop_->PostTask(
         FROM_HERE,
         NewRunnableMethod(this, &AudioRendererImpl::PauseTask));
   }
   AudioRendererBase::SetPlaybackRate(rate);
-
-  // If we are playing, give a kick to try fulfilling the packet request as
-  // the previous packet request may be stalled by a pause.
-  if (rate > 0.0f) {
-    NotifyDataAvailableIfNecessary();
-  }
 }
 
 void AudioRendererImpl::Pause(media::FilterCallback* callback) {
@@ -151,7 +156,7 @@
   if (stopped_)
     return;
 
-  ChildProcess::current()->io_message_loop()->PostTask(
+  render_loop_->PostTask(
       FROM_HERE,
       NewRunnableMethod(this, &AudioRendererImpl::PauseTask));
 }
@@ -163,7 +168,7 @@
   if (stopped_)
     return;
 
-  ChildProcess::current()->io_message_loop()->PostTask(
+  render_loop_->PostTask(
       FROM_HERE,
       NewRunnableMethod(this, &AudioRendererImpl::SeekTask));
 }
@@ -176,11 +181,11 @@
     return;
 
   if (GetPlaybackRate() != 0.0f) {
-    ChildProcess::current()->io_message_loop()->PostTask(
+    render_loop_->PostTask(
         FROM_HERE,
         NewRunnableMethod(this, &AudioRendererImpl::PlayTask));
   } else {
-    ChildProcess::current()->io_message_loop()->PostTask(
+    render_loop_->PostTask(
         FROM_HERE,
         NewRunnableMethod(this, &AudioRendererImpl::PauseTask));
   }
@@ -190,218 +195,50 @@
   base::AutoLock auto_lock(lock_);
   if (stopped_)
     return;
-  ChildProcess::current()->io_message_loop()->PostTask(
-      FROM_HERE,
-      NewRunnableMethod(this, &AudioRendererImpl::SetVolumeTask, volume));
-}
 
-void AudioRendererImpl::OnCreated(base::SharedMemoryHandle handle,
-                                  uint32 length) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kHighLatency, latency_type_);
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  shared_memory_.reset(new base::SharedMemory(handle, false));
-  shared_memory_->Map(length);
-  shared_memory_size_ = length;
+  if (audio_device_.get())
+    audio_device_->SetVolume(volume);
 }
 
-void AudioRendererImpl::CreateSocket(base::SyncSocket::Handle socket_handle) {
-  DCHECK_EQ(kLowLatency, latency_type_);
-#if defined(OS_WIN)
-  DCHECK(socket_handle);
-#else
-  DCHECK_GE(socket_handle, 0);
-#endif
-  socket_.reset(new base::SyncSocket(socket_handle));
-}
+void AudioRendererImpl::InitializeTask(const AudioParameters& params) {
+  bytes_per_second_ = params.GetBytesPerSecond();
+  bytes_per_frame_ = params.bits_per_sample * params.channels / 8;
 
-void AudioRendererImpl::CreateAudioThread() {
-  DCHECK_EQ(kLowLatency, latency_type_);
-  audio_thread_.reset(
-      new base::DelegateSimpleThread(this, "renderer_audio_thread"));
-  audio_thread_->Start();
+  audio_device_ = new AudioDevice(
+      kBufferSize,
+      params.channels,
+      params.sample_rate,
+      this);
 }
 
-void AudioRendererImpl::OnLowLatencyCreated(
-    base::SharedMemoryHandle handle,
-    base::SyncSocket::Handle socket_handle,
-    uint32 length) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kLowLatency, latency_type_);
-#if defined(OS_WIN)
-  DCHECK(handle);
-#else
-  DCHECK_GE(handle.fd, 0);
-#endif
-  DCHECK_NE(0u, length);
+void AudioRendererImpl::PlayTask() {
+  LOG(ERROR) << "PlayTask()";
 
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  shared_memory_.reset(new base::SharedMemory(handle, false));
-  shared_memory_->Map(length);
-  shared_memory_size_ = length;
-
-  CreateSocket(socket_handle);
-  CreateAudioThread();
-}
-
-void AudioRendererImpl::OnRequestPacket(AudioBuffersState buffers_state) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kHighLatency, latency_type_);
-  {
-    base::AutoLock auto_lock(lock_);
-    DCHECK(!pending_request_);
-    pending_request_ = true;
-    request_buffers_state_ = buffers_state;
+  //Send(new AudioHostMsg_PlayStream(stream_id_));
+  if (audio_device_.get()) {
+    audio_device_->Start();
   }
-
-  // Try to fill in the fulfill the packet request.
-  NotifyPacketReadyTask();
 }
 
-void AudioRendererImpl::OnStateChanged(AudioStreamState state) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
+void AudioRendererImpl::PauseTask() {
+  LOG(ERROR) << "PauseTask()";
 
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  switch (state) {
-    case kAudioStreamError:
-      // We receive this error if we counter an hardware error on the browser
-      // side. We can proceed with ignoring the audio stream.
-      // TODO(hclam): We need more handling of these kind of error. For example
-      // re-try creating the audio output stream on the browser side or fail
-      // nicely and report to demuxer that the whole audio stream is discarded.
-      host()->DisableAudioRenderer();
-      break;
-    // TODO(hclam): handle these events.
-    case kAudioStreamPlaying:
-    case kAudioStreamPaused:
-      break;
-    default:
-      NOTREACHED();
-      break;
+  //Send(new AudioHostMsg_PauseStream(stream_id_));
+  if (audio_device_.get()) {
+    audio_device_->Stop();
   }
 }
 
-void AudioRendererImpl::OnVolume(double volume) {
-  // TODO(hclam): decide whether we need to report the current volume to
-  // pipeline.
-}
-
-void AudioRendererImpl::CreateStreamTask(const AudioParameters& audio_params) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-
-  // Make sure we don't call create more than once.
-  DCHECK_EQ(0, stream_id_);
-  stream_id_ = filter_->AddDelegate(this);
-  ChildProcess::current()->io_message_loop()->AddDestructionObserver(this);
-
-  AudioParameters params_to_send(audio_params);
-  // Let the browser choose packet size.
-  params_to_send.samples_per_packet = 0;
-
-  Send(new AudioHostMsg_CreateStream(stream_id_,
-                                     params_to_send,
-                                     latency_type_ == kLowLatency));
-}
-
-void AudioRendererImpl::PlayTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  Send(new AudioHostMsg_PlayStream(stream_id_));
-}
-
-void AudioRendererImpl::PauseTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  Send(new AudioHostMsg_PauseStream(stream_id_));
-}
-
 void AudioRendererImpl::SeekTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
+  LOG(ERROR) << "SeekTask()";
+  if (audio_device_.get()) {
+    audio_device_->Stop();
+  }
   // We have to pause the audio stream before we can flush.
-  Send(new AudioHostMsg_PauseStream(stream_id_));
-  Send(new AudioHostMsg_FlushStream(stream_id_));
+  //Send(new AudioHostMsg_PauseStream(stream_id_));
+  //Send(new AudioHostMsg_FlushStream(stream_id_));
 }
 
-void AudioRendererImpl::DestroyTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  // Make sure we don't call destroy more than once.
-  DCHECK_NE(0, stream_id_);
-  filter_->RemoveDelegate(stream_id_);
-  Send(new AudioHostMsg_CloseStream(stream_id_));
-  ChildProcess::current()->io_message_loop()->RemoveDestructionObserver(this);
-  stream_id_ = 0;
-}
-
-void AudioRendererImpl::SetVolumeTask(double volume) {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-  Send(new AudioHostMsg_SetVolume(stream_id_, volume));
-}
-
-void AudioRendererImpl::NotifyPacketReadyTask() {
-  DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
-  DCHECK_EQ(kHighLatency, latency_type_);
-
-  base::AutoLock auto_lock(lock_);
-  if (stopped_)
-    return;
-  if (pending_request_ && GetPlaybackRate() > 0.0f) {
-    DCHECK(shared_memory_.get());
-
-    // Adjust the playback delay.
-    base::Time current_time = base::Time::Now();
-
-    base::TimeDelta request_delay =
-        ConvertToDuration(request_buffers_state_.total_bytes());
-
-    // Add message delivery delay.
-    if (current_time > request_buffers_state_.timestamp) {
-      base::TimeDelta receive_latency =
-          current_time - request_buffers_state_.timestamp;
-
-      // If the receive latency is too much it may offset all the delay.
-      if (receive_latency >= request_delay) {
-        request_delay = base::TimeDelta();
-      } else {
-        request_delay -= receive_latency;
-      }
-    }
-
-    // Finally we need to adjust the delay according to playback rate.
-    if (GetPlaybackRate() != 1.0f) {
-      request_delay = base::TimeDelta::FromMicroseconds(
-          static_cast<int64>(ceil(request_delay.InMicroseconds() *
-                                  GetPlaybackRate())));
-    }
-
-    uint32 filled = FillBuffer(static_cast<uint8*>(shared_memory_->memory()),
-                               shared_memory_size_, request_delay,
-                               request_buffers_state_.pending_bytes == 0);
-    pending_request_ = false;
-    // Then tell browser process we are done filling into the buffer.
-    Send(new AudioHostMsg_NotifyPacketReady(stream_id_, filled));
-  }
-}
-
 void AudioRendererImpl::WillDestroyCurrentMessageLoop() {
   DCHECK(MessageLoop::current() == ChildProcess::current()->io_message_loop());
 
@@ -411,42 +248,8 @@
     return;
 
   stopped_ = true;
-  DestroyTask();
-}
-
-// Our audio thread runs here. We receive requests for more data and send it
-// on this thread.
-void AudioRendererImpl::Run() {
-  audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
-
-  int bytes;
-  while (sizeof(bytes) == socket_->Receive(&bytes, sizeof(bytes))) {
-    LOG(ERROR) << "+++ bytes: " << bytes;
-    if (bytes == media::AudioOutputController::kPauseMark)
-      continue;
-    else if (bytes < 0)
-      break;
-    base::AutoLock auto_lock(lock_);
-    if (stopped_)
-      break;
-    float playback_rate = GetPlaybackRate();
-    if (playback_rate <= 0.0f)
-      continue;
-    DCHECK(shared_memory_.get());
-    base::TimeDelta request_delay = ConvertToDuration(bytes);
-    // We need to adjust the delay according to playback rate.
-    if (playback_rate != 1.0f) {
-      request_delay = base::TimeDelta::FromMicroseconds(
-          static_cast<int64>(ceil(request_delay.InMicroseconds() *
-                                  playback_rate)));
-    }
-    FillBuffer(static_cast<uint8*>(shared_memory_->memory()),
-               shared_memory_size_,
-               request_delay,
-               true  /* buffers empty */);
+  if (audio_device_.get()) {
+    audio_device_->Stop();
+    audio_device_ = NULL;
   }
 }
-
-void AudioRendererImpl::Send(IPC::Message* message) {
-  filter_->Send(message);
-}