Merge AudioRendererImpl and AudioRendererBase; add NullAudioSink

media/filters/audio_renderer_base.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/filters/audio_renderer_base.h"
 
-#include <algorithm>
-#include <string>
+#include <math.h>
 
 #include "base/bind.h"
 #include "base/callback.h"
 #include "base/logging.h"
 #include "media/base/filter_host.h"
+#include "media/audio/audio_util.h"
 
 namespace media {
 
-AudioRendererBase::AudioRendererBase()
+AudioRendererBase::AudioRendererBase(media::AudioRendererSink* sink)
     : state_(kUninitialized),
       pending_read_(false),
       received_end_of_stream_(false),
       rendered_end_of_stream_(false),
       bytes_per_frame_(0),
+      bytes_per_second_(0),
+      stopped_(false),
+      sink_(sink),
+      is_initialized_(false),
       read_cb_(base::Bind(&AudioRendererBase::DecodedAudioReady,
                           base::Unretained(this))) {
 }
 
 AudioRendererBase::~AudioRendererBase() {
   // Stop() should have been called and |algorithm_| should have been destroyed.
   DCHECK(state_ == kUninitialized || state_ == kStopped);
   DCHECK(!algorithm_.get());
 }
 
 void AudioRendererBase::Play(const base::Closure& callback) {
-  base::AutoLock auto_lock(lock_);
-  DCHECK_EQ(kPaused, state_);
-  state_ = kPlaying;
-  callback.Run();
+  {
+    base::AutoLock auto_lock(lock_);
+    DCHECK_EQ(kPaused, state_);
+    state_ = kPlaying;
+    callback.Run();
+  }
+
+  if (stopped_)
+    return;
+
+  if (GetPlaybackRate() != 0.0f) {
+    DoPlay();
+  } else {
+    DoPause();
+  }
+}
+
+void AudioRendererBase::DoPlay() {
+  earliest_end_time_ = base::Time::Now();
+  DCHECK(sink_.get());
+  sink_->Play();
 }
 
 void AudioRendererBase::Pause(const base::Closure& callback) {
-  base::AutoLock auto_lock(lock_);
-  DCHECK(state_ == kPlaying || state_ == kUnderflow || state_ == kRebuffering);
-  pause_cb_ = callback;
-  state_ = kPaused;
+  {
+    base::AutoLock auto_lock(lock_);
+    DCHECK(state_ == kPlaying || state_ == kUnderflow ||
+           state_ == kRebuffering);
+    pause_cb_ = callback;
+    state_ = kPaused;
 
-  // Pause only when we've completed our pending read.
-  if (!pending_read_) {
-    pause_cb_.Run();
-    pause_cb_.Reset();
-  } else {
-    state_ = kPaused;
+    // Pause only when we've completed our pending read.
+    if (!pending_read_) {
+      pause_cb_.Run();
+      pause_cb_.Reset();
+    } else {
+      state_ = kPaused;
+    }
   }
+
+  if (stopped_)
+    return;
+
+  DoPause();
+}
+
+void AudioRendererBase::DoPause() {
+  DCHECK(sink_.get());
+  sink_->Pause(false);
 }
 
 void AudioRendererBase::Flush(const base::Closure& callback) {
   decoder_->Reset(callback);
 }
 
 void AudioRendererBase::Stop(const base::Closure& callback) {
-  OnStop();
+  if (!stopped_) {
+    DCHECK(sink_.get());
+    sink_->Stop();
+
+    stopped_ = true;
+  }
   {
     base::AutoLock auto_lock(lock_);
     state_ = kStopped;
     algorithm_.reset(NULL);
     time_cb_.Reset();
     underflow_cb_.Reset();
   }
   if (!callback.is_null()) {
     callback.Run();
   }
 }
 
 void AudioRendererBase::Seek(base::TimeDelta time, const PipelineStatusCB& cb) {
   base::AutoLock auto_lock(lock_);
   DCHECK_EQ(kPaused, state_);
   DCHECK(!pending_read_) << "Pending read must complete before seeking";
   DCHECK(pause_cb_.is_null());
   DCHECK(seek_cb_.is_null());
   state_ = kSeeking;
   seek_cb_ = cb;
   seek_timestamp_ = time;
 
   // Throw away everything and schedule our reads.
-  last_fill_buffer_time_ = base::TimeDelta();
+  audio_time_buffered_ = base::TimeDelta();
   received_end_of_stream_ = false;
   rendered_end_of_stream_ = false;
 
   // |algorithm_| will request more reads.
   algorithm_->FlushBuffers();
+
+  if (stopped_)
+    return;
+
+  DoSeek();
+}
+
+void AudioRendererBase::DoSeek() {
+  earliest_end_time_ = base::Time::Now();
+
+  // Pause and flush the stream when we seek to a new location.
+  DCHECK(sink_.get());

[scherkus (not reviewing), 2012/03/23 15:24:52]

nit: DCHECK-before-dereferencing aren't very useful

[vrk (LEFT CHROMIUM), 2012/04/02 21:17:54]

Deleted! Yeah I just copy/pasted logic verbatim from AudioRendererImpl, with the
intention to clean these sorts of things up in a follow-up CL. Wanted to keep
things separate to catch bugs more easily if they come up.

+  sink_->Pause(true);
 }
 
 void AudioRendererBase::Initialize(const scoped_refptr<AudioDecoder>& decoder,
                                    const PipelineStatusCB& init_cb,
                                    const base::Closure& underflow_cb,
                                    const TimeCB& time_cb) {
   DCHECK(decoder);
   DCHECK(!init_cb.is_null());
   DCHECK(!underflow_cb.is_null());
   DCHECK(!time_cb.is_null());
@@ skipping 12 matching lines @@
   // and a callback to request more reads from the data source.
   ChannelLayout channel_layout = decoder_->channel_layout();
   int channels = ChannelLayoutToChannelCount(channel_layout);
   int bits_per_channel = decoder_->bits_per_channel();
   int sample_rate = decoder_->samples_per_second();
   // TODO(vrk): Add method to AudioDecoder to compute bytes per frame.
   bytes_per_frame_ = channels * bits_per_channel / 8;
 
   bool config_ok = algorithm_->ValidateConfig(channels, sample_rate,
                                               bits_per_channel);
-  if (config_ok)
-    algorithm_->Initialize(channels, sample_rate, bits_per_channel, 0.0f, cb);
-
-  // Give the subclass an opportunity to initialize itself.
-  if (!config_ok || !OnInitialize(bits_per_channel, channel_layout,
-                                  sample_rate)) {
+  if (!config_ok || is_initialized_) {

[scherkus (not reviewing), 2012/03/23 15:24:52]

wouldn't calling Initialize() when is_initialized_ == true programmer error?

it seems that is_initialized_ should be a DCHECK then this if statement can
inline the ValidateConfig() call and we can remove config_ok

[vrk (LEFT CHROMIUM), 2012/04/02 21:17:54]

I agree, and actually I think the DCHECK_EQ(kUninitialized, state_); obviates
the need for the field altogether. I think I will leave the field alone in this
CL and fix it in the CL to clean up the merged AudioRendererBase/Impl class. 

     init_cb.Run(PIPELINE_ERROR_INITIALIZATION_FAILED);
     return;
   }
 
+  if (config_ok)
+    algorithm_->Initialize(channels, sample_rate, bits_per_channel, 0.0f, cb);
+
+  // We use the AUDIO_PCM_LINEAR flag because AUDIO_PCM_LOW_LATENCY
+  // does not currently support all the sample-rates that we require.
+  // Please see: http://code.google.com/p/chromium/issues/detail?id=103627
+  // for more details.
+  audio_parameters_ = AudioParameters(
+      AudioParameters::AUDIO_PCM_LINEAR, channel_layout, sample_rate,
+      bits_per_channel, GetHighLatencyOutputBufferSize(sample_rate));
+
+  bytes_per_second_ = audio_parameters_.GetBytesPerSecond();
+
+  DCHECK(sink_.get());
+  DCHECK(!is_initialized_);
+
+  sink_->Initialize(audio_parameters_, this);
+
+  sink_->Start();
+  is_initialized_ = true;
+
   // Finally, execute the start callback.
   state_ = kPaused;
   init_cb.Run(PIPELINE_OK);
 }
 
 bool AudioRendererBase::HasEnded() {
   base::AutoLock auto_lock(lock_);
   DCHECK(!rendered_end_of_stream_ || algorithm_->NeedsMoreData());
 
   return received_end_of_stream_ && rendered_end_of_stream_;
 }
 
 void AudioRendererBase::ResumeAfterUnderflow(bool buffer_more_audio) {
   base::AutoLock auto_lock(lock_);
   if (state_ == kUnderflow) {
     if (buffer_more_audio)
       algorithm_->IncreaseQueueCapacity();
 
     state_ = kRebuffering;
   }
 }
 
+void AudioRendererBase::SetVolume(float volume) {
+  if (stopped_)
+    return;
+  DCHECK(sink_.get());

[scherkus (not reviewing), 2012/03/23 15:24:52]

ditto

[vrk (LEFT CHROMIUM), 2012/04/02 21:17:54]

Done.

+  sink_->SetVolume(volume);
+}
+
 void AudioRendererBase::DecodedAudioReady(scoped_refptr<Buffer> buffer) {
   base::AutoLock auto_lock(lock_);
   DCHECK(state_ == kPaused || state_ == kSeeking || state_ == kPlaying ||
          state_ == kUnderflow || state_ == kRebuffering || state_ == kStopped);
 
   CHECK(pending_read_);
   pending_read_ = false;
 
   if (buffer && buffer->IsEndOfStream()) {
     received_end_of_stream_ = true;
@@ skipping 34 matching lines @@
         algorithm_->EnqueueBuffer(buffer);
       return;
     case kStopped:
       return;
   }
 }
 
 uint32 AudioRendererBase::FillBuffer(uint8* dest,
                                      uint32 requested_frames,
                                      const base::TimeDelta& playback_delay) {
-  // The timestamp of the last buffer written during the last call to
-  // FillBuffer().
-  base::TimeDelta last_fill_buffer_time;
+  // The |audio_time_buffered_| is the ending timestamp of the last frame
+  // buffered at the audio device. |playback_delay| is the amount of time
+  // buffered at the audio device. The current time can be computed by their
+  // difference.
+  base::TimeDelta current_time = audio_time_buffered_ - playback_delay;
+
   size_t frames_written = 0;
   base::Closure underflow_cb;
   {
     base::AutoLock auto_lock(lock_);
 
     if (state_ == kRebuffering && algorithm_->IsQueueFull())
       state_ = kPlaying;
 
     // Mute audio by returning 0 when not playing.
     if (state_ != kPlaying) {
       // TODO(scherkus): To keep the audio hardware busy we write at most 8k of
       // zeros.  This gets around the tricky situation of pausing and resuming
       // the audio IPC layer in Chrome.  Ideally, we should return zero and then
       // the subclass can restart the conversation.
       //
       // This should get handled by the subclass http://crbug.com/106600
       const uint32 kZeroLength = 8192;
       size_t zeros_to_write =
           std::min(kZeroLength, requested_frames * bytes_per_frame_);
       memset(dest, 0, zeros_to_write);
       return zeros_to_write / bytes_per_frame_;
     }
 
-    // Save a local copy of last fill buffer time and reset the member.
-    last_fill_buffer_time = last_fill_buffer_time_;
-    last_fill_buffer_time_ = base::TimeDelta();
-
     // Use three conditions to determine the end of playback:
     // 1. Algorithm needs more audio data.
     // 2. We've received an end of stream buffer.
     //    (received_end_of_stream_ == true)
     // 3. Browser process has no audio data being played.
     //    There is no way to check that condition that would work for all
     //    derived classes, so call virtual method that would either render
     //    end of stream or schedule such rendering.
     //
     // Three conditions determine when an underflow occurs:
     // 1. Algorithm has no audio data.
     // 2. Currently in the kPlaying state.
     // 3. Have not received an end of stream buffer.
     if (algorithm_->NeedsMoreData()) {
       if (received_end_of_stream_) {
-        OnRenderEndOfStream();
+        // TODO(enal): schedule callback instead of polling.
+        if (base::Time::Now() >= earliest_end_time_)
+          SignalEndOfStream();
       } else if (state_ == kPlaying) {
         state_ = kUnderflow;
         underflow_cb = underflow_cb_;
       }
     } else {
       // Otherwise fill the buffer.
       frames_written = algorithm_->FillBuffer(dest, requested_frames);
     }
-
-    // Get the current time.
-    last_fill_buffer_time_ = algorithm_->GetTime();
   }
 
-  // Update the pipeline's time if it was set last time.
-  base::TimeDelta new_current_time = last_fill_buffer_time - playback_delay;
-  if (last_fill_buffer_time.InMicroseconds() > 0 &&
-      (last_fill_buffer_time != last_fill_buffer_time_ ||
-       new_current_time > host()->GetTime())) {
-    time_cb_.Run(new_current_time, last_fill_buffer_time);

[vrk (LEFT CHROMIUM), 2012/03/22 21:28:53]

acolwell: The bug for crbug.com/119549 is here: the 2nd parameter should be
|last_fill_buffer_time_| (the field), not the variable that represents the prev
value. I fixed the bug and reworked this area a bit to hopefully make things
more clear.

+  base::TimeDelta previous_time_buffered = audio_time_buffered_;
+  // The call to FillBuffer() on |algorithm_| has increased the amount of
+  // buffered audio data. Update the new amount of time buffered.
+  audio_time_buffered_ = algorithm_->GetTime();
+
+  if (previous_time_buffered.InMicroseconds() > 0 &&
+      (previous_time_buffered != audio_time_buffered_ ||
+       current_time > host()->GetTime())) {
+    time_cb_.Run(current_time, audio_time_buffered_);
   }
 
   if (!underflow_cb.is_null())
     underflow_cb.Run();
 
   return frames_written;
 }
 
 void AudioRendererBase::SignalEndOfStream() {
   DCHECK(received_end_of_stream_);
   if (!rendered_end_of_stream_) {
     rendered_end_of_stream_ = true;
     host()->NotifyEnded();
   }
 }
 
 void AudioRendererBase::ScheduleRead_Locked() {
   lock_.AssertAcquired();
   if (pending_read_ || state_ == kPaused)
     return;
   pending_read_ = true;
   decoder_->Read(read_cb_);
 }
 
 void AudioRendererBase::SetPlaybackRate(float playback_rate) {
+  DCHECK_LE(0.0f, playback_rate);
+
+  if (!stopped_) {
+    // Notify sink of new playback rate.
+    sink_->SetPlaybackRate(playback_rate);
+
+    // We have two cases here:
+    // Play: GetPlaybackRate() == 0.0 && playback_rate != 0.0
+    // Pause: GetPlaybackRate() != 0.0 && playback_rate == 0.0
+    if (GetPlaybackRate() == 0.0f && playback_rate != 0.0f) {
+      DoPlay();
+    } else if (GetPlaybackRate() != 0.0f && playback_rate == 0.0f) {
+      // Pause is easy, we can always pause.
+      DoPause();
+    }
+  }
+
   base::AutoLock auto_lock(lock_);
   algorithm_->SetPlaybackRate(playback_rate);
 }
 
 float AudioRendererBase::GetPlaybackRate() {
   base::AutoLock auto_lock(lock_);
   return algorithm_->playback_rate();
 }
 
 bool AudioRendererBase::IsBeforeSeekTime(const scoped_refptr<Buffer>& buffer) {
   return (state_ == kSeeking) && buffer && !buffer->IsEndOfStream() &&
       (buffer->GetTimestamp() + buffer->GetDuration()) < seek_timestamp_;
 }
 
+int AudioRendererBase::Render(const std::vector<float*>& audio_data,
+                              int number_of_frames,
+                              int audio_delay_milliseconds) {
+  if (stopped_ || GetPlaybackRate() == 0.0f) {
+    // Output silence if stopped.
+    for (size_t i = 0; i < audio_data.size(); ++i)
+      memset(audio_data[i], 0, sizeof(float) * number_of_frames);
+    return 0;
+  }
+
+  // Adjust the playback delay.
+  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 bytes_per_frame = audio_parameters_.GetBytesPerFrame();
+
+  const int buf_size = number_of_frames * bytes_per_frame;
+  scoped_array<uint8> buf(new uint8[buf_size]);
+
+  int frames_filled = FillBuffer(buf.get(), number_of_frames, request_delay);
+  int bytes_filled = frames_filled * bytes_per_frame;
+  DCHECK_LE(bytes_filled, buf_size);
+  UpdateEarliestEndTime(bytes_filled, request_delay, base::Time::Now());
+
+  // Deinterleave each audio channel.
+  int channels = audio_data.size();
+  for (int channel_index = 0; channel_index < channels; ++channel_index) {
+    media::DeinterleaveAudioChannel(buf.get(),
+                                    audio_data[channel_index],
+                                    channels,
+                                    channel_index,
+                                    bytes_per_frame / channels,
+                                    frames_filled);
+
+    // If FillBuffer() didn't give us enough data then zero out the remainder.
+    if (frames_filled < number_of_frames) {
+      int frames_to_zero = number_of_frames - frames_filled;
+      memset(audio_data[channel_index] + frames_filled,
+             0,
+             sizeof(float) * frames_to_zero);
+    }
+  }
+  return frames_filled;
+}
+
+void AudioRendererBase::UpdateEarliestEndTime(int bytes_filled,
+                                              base::TimeDelta request_delay,
+                                              base::Time time_now) {
+  if (bytes_filled != 0) {
+    base::TimeDelta predicted_play_time = ConvertToDuration(bytes_filled);
+    float playback_rate = GetPlaybackRate();
+    if (playback_rate != 1.0f) {
+      predicted_play_time = base::TimeDelta::FromMicroseconds(
+          static_cast<int64>(ceil(predicted_play_time.InMicroseconds() *
+                                  playback_rate)));
+    }
+    earliest_end_time_ =
+        std::max(earliest_end_time_,
+                 time_now + request_delay + predicted_play_time);
+  }
+}
+
+base::TimeDelta AudioRendererBase::ConvertToDuration(int bytes) {
+  if (bytes_per_second_) {
+    return base::TimeDelta::FromMicroseconds(
+        base::Time::kMicrosecondsPerSecond * bytes / bytes_per_second_);
+  }
+  return base::TimeDelta();
+}
+
+void AudioRendererBase::OnRenderError() {
+  host()->DisableAudioRenderer();
+}
+
 }  // namespace media