Make media::AudioClock track frames written to compute time.

media/filters/audio_renderer_impl.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_impl.h"
 
 #include <math.h>
 
 #include <algorithm>
 
@@ skipping 129 matching lines @@
 
 void AudioRendererImpl::SetMediaTime(base::TimeDelta time) {
   DVLOG(1) << __FUNCTION__ << "(" << time.InMicroseconds() << ")";
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   base::AutoLock auto_lock(lock_);
   DCHECK(!rendering_);
   DCHECK_EQ(state_, kFlushed);
 
   start_timestamp_ = time;
+  audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate()));
 }
 
 base::TimeDelta AudioRendererImpl::CurrentMediaTime() {
   DVLOG(1) << __FUNCTION__;
   DCHECK(task_runner_->BelongsToCurrentThread());
 
   // TODO(scherkus): Finish implementing when ready to switch Pipeline to using
   // TimeSource http://crbug.com/370634
   NOTIMPLEMENTED();
 
@@ skipping 34 matching lines @@
 }
 
 void AudioRendererImpl::ResetDecoderDone() {
   DCHECK(task_runner_->BelongsToCurrentThread());
   {
     base::AutoLock auto_lock(lock_);
 
     DCHECK_EQ(state_, kFlushed);
     DCHECK(!flush_cb_.is_null());
 
-    audio_clock_.reset(new AudioClock(audio_parameters_.sample_rate()));
+    audio_clock_.reset();
     received_end_of_stream_ = false;
     rendered_end_of_stream_ = false;
 
     // Flush() may have been called while underflowed/not fully buffered.
     if (buffering_state_ != BUFFERING_HAVE_NOTHING)
       SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
 
     splicer_->Reset();
     if (buffer_converter_)
       buffer_converter_->Reset();
@@ skipping 72 matching lines @@
         // unnecessary upmixing overhead.
         stream->audio_decoder_config().channel_layout(),
         ChannelLayoutToChannelCount(
             stream->audio_decoder_config().channel_layout()),
         hw_params.input_channels(),
         hw_params.sample_rate(),
         hw_params.bits_per_sample(),
         hardware_config_->GetHighLatencyBufferSize());
   }
 
-  audio_clock_.reset(new AudioClock(audio_parameters_.sample_rate()));
+  audio_clock_.reset(
+      new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate()));
 
   audio_buffer_stream_->Initialize(
       stream,
       false,
       statistics_cb,
       base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
                  weak_factory_.GetWeakPtr()));
 }
 
 void AudioRendererImpl::OnAudioBufferStreamInitialized(bool success) {
@@ skipping 234 matching lines @@
       audio_delay_milliseconds);
   const int delay_frames = static_cast<int>(playback_delay.InSecondsF() *
                                             audio_parameters_.sample_rate());
   int frames_written = 0;
   base::Closure time_cb;
   {
     base::AutoLock auto_lock(lock_);
 
     // Ensure Stop() hasn't destroyed our |algorithm_| on the pipeline thread.
     if (!algorithm_) {
-      audio_clock_->WroteSilence(requested_frames, delay_frames);
+      audio_clock_->WroteAudio(
+          0, requested_frames, delay_frames, playback_rate_);
       return 0;
     }
 
     if (playback_rate_ == 0) {
-      audio_clock_->WroteSilence(requested_frames, delay_frames);
+      audio_clock_->WroteAudio(
+          0, requested_frames, delay_frames, playback_rate_);
       return 0;
     }
 
     // Mute audio by returning 0 when not playing.
     if (state_ != kPlaying) {
-      audio_clock_->WroteSilence(requested_frames, delay_frames);
+      audio_clock_->WroteAudio(
+          0, requested_frames, delay_frames, playback_rate_);
       return 0;
     }
 
     // We use the following conditions to determine end of playback:
     //   1) Algorithm can not fill the audio callback buffer
     //   2) We received an end of stream buffer
     //   3) We haven't already signalled that we've ended
     //   4) We've played all known audio data sent to hardware
     //
     // We use the following conditions to determine underflow:
     //   1) Algorithm can not fill the audio callback buffer
     //   2) We have NOT received an end of stream buffer
     //   3) We are in the kPlaying state
     //
     // Otherwise the buffer has data we can send to the device.
-    const base::TimeDelta media_timestamp_before_filling =
-        audio_clock_->CurrentMediaTimestamp(base::TimeDelta());
     if (algorithm_->frames_buffered() > 0) {
       frames_written =
           algorithm_->FillBuffer(audio_bus, requested_frames, playback_rate_);
-      audio_clock_->WroteAudio(
-          frames_written, delay_frames, playback_rate_, algorithm_->GetTime());
     }
-    audio_clock_->WroteSilence(requested_frames - frames_written, delay_frames);
+    audio_clock_->WroteAudio(
+        frames_written, requested_frames, delay_frames, playback_rate_);
 
     if (frames_written == 0) {
       if (received_end_of_stream_ && !rendered_end_of_stream_ &&
-          audio_clock_->CurrentMediaTimestamp(base::TimeDelta()) ==
-              audio_clock_->last_endpoint_timestamp()) {
+          !audio_clock_->AudioDataBuffered()) {
         rendered_end_of_stream_ = true;
         task_runner_->PostTask(FROM_HERE, ended_cb_);
       } else if (!received_end_of_stream_ && state_ == kPlaying) {
         if (buffering_state_ != BUFFERING_HAVE_NOTHING) {
           algorithm_->IncreaseQueueCapacity();
           SetBufferingState_Locked(BUFFERING_HAVE_NOTHING);
         }
       }
     }
 
     if (CanRead_Locked()) {
       task_runner_->PostTask(FROM_HERE,
                              base::Bind(&AudioRendererImpl::AttemptRead,
                                         weak_factory_.GetWeakPtr()));
     }
 
-    // We only want to execute |time_cb_| if time has progressed and we haven't
-    // signaled end of stream yet.
-    if (media_timestamp_before_filling !=
-            audio_clock_->CurrentMediaTimestamp(base::TimeDelta()) &&
-        !rendered_end_of_stream_) {
-      time_cb =
-          base::Bind(time_cb_,
-                     audio_clock_->CurrentMediaTimestamp(base::TimeDelta()),
-                     audio_clock_->last_endpoint_timestamp());
+    // Firing |ended_cb_| means we no longer need to run |time_cb_|.
+    if (!rendered_end_of_stream_) {
+      // Since |max_time| uses linear interpolation, only provide an upper bound
+      // that is for audio data at the same playback rate. Failing to do so can
+      // make time jump backwards when the linear interpolated time advances
+      // past buffered regions of audio at different rates.
+      base::TimeDelta current_timestamp =
+          audio_clock_->CurrentMediaTimestamp(base::TimeDelta());
+      base::TimeDelta max_time =
+          current_timestamp +
+          audio_clock_->ContiguousAudioDataBufferedAtSameRate();
+      time_cb = base::Bind(time_cb_, current_timestamp, max_time);
     }
   }
 
   if (!time_cb.is_null())
     task_runner_->PostTask(FROM_HERE, time_cb);
 
   DCHECK_LE(frames_written, requested_frames);
   return frames_written;
 }
 
@@ skipping 65 matching lines @@
            << buffering_state;
   DCHECK_NE(buffering_state_, buffering_state);
   lock_.AssertAcquired();
   buffering_state_ = buffering_state;
 
   task_runner_->PostTask(FROM_HERE,
                          base::Bind(buffering_state_cb_, buffering_state_));
 }
 
 }  // namespace media