[TO 54] AudioRendererImpl: Don't advance time when rendering stops.

media/renderers/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/renderers/audio_renderer_impl.h"
 
 #include <math.h>
 #include <stddef.h>
 #include <algorithm>
 #include <utility>
@@ skipping 84 matching lines @@
   // After this call, the |sink_| will not call back into |this| anymore.
   sink_->Stop();
 
   if (!init_cb_.is_null())
     base::ResetAndReturn(&init_cb_).Run(PIPELINE_ERROR_ABORT);
 }
 
 void AudioRendererImpl::StartTicking() {
   DVLOG(1) << __func__;
   DCHECK(task_runner_->BelongsToCurrentThread());
+
+  base::AutoLock auto_lock(lock_);
+
   DCHECK(!rendering_);
   rendering_ = true;
 
-  base::AutoLock auto_lock(lock_);
   // Wait for an eventual call to SetPlaybackRate() to start rendering.
   if (playback_rate_ == 0) {
     DCHECK(!sink_playing_);
     return;
   }
 
   StartRendering_Locked();
 }
 
 void AudioRendererImpl::StartRendering_Locked() {
   DVLOG(1) << __func__;
   DCHECK(task_runner_->BelongsToCurrentThread());
   DCHECK_EQ(state_, kPlaying);
   DCHECK(!sink_playing_);
   DCHECK_NE(playback_rate_, 0.0);
   lock_.AssertAcquired();
 
   sink_playing_ = true;
 
   base::AutoUnlock auto_unlock(lock_);
   sink_->Play();
 }
 
 void AudioRendererImpl::StopTicking() {
   DVLOG(1) << __func__;
   DCHECK(task_runner_->BelongsToCurrentThread());
+
+  base::AutoLock auto_lock(lock_);
+
   DCHECK(rendering_);
   rendering_ = false;
 
-  base::AutoLock auto_lock(lock_);
   // Rendering should have already been stopped with a zero playback rate.
   if (playback_rate_ == 0) {
     DCHECK(!sink_playing_);
     return;
   }
 
   StopRendering_Locked();
 }
 
 void AudioRendererImpl::StopRendering_Locked() {
@@ skipping 16 matching lines @@
   base::AutoLock auto_lock(lock_);
   DCHECK(!rendering_);
   DCHECK_EQ(state_, kFlushed);
 
   start_timestamp_ = time;
   ended_timestamp_ = kInfiniteDuration;
   last_render_time_ = stop_rendering_time_ = base::TimeTicks();
   first_packet_timestamp_ = kNoTimestamp;
   last_media_timestamp_ = base::TimeDelta();
   audio_clock_.reset(new AudioClock(time, audio_parameters_.sample_rate()));
+  last_reported_media_time_ = kNoTimestamp;
 }
 
 base::TimeDelta AudioRendererImpl::CurrentMediaTime() {
   base::AutoLock auto_lock(lock_);
 
+  // Re-use last reported time if rendering has stopped to avoid confusing blink
+  // layer (avoid currentTime advancing after signaling buffer underflow).
+  // TODO(chcunningham): Do this in blink instead. Patching it here for now
+  // because HTMLMediaElement's currentTime is a mess.
+  if (!rendering_ && last_reported_media_time_ != kNoTimestamp) {
+    // If rendering stops, be sure to at least report the front time of the most
+    // recently rendered audio buffer.
+    if (last_reported_media_time_ < audio_clock_->front_timestamp())
+      last_reported_media_time_ = audio_clock_->front_timestamp();
+
+    DVLOG(3) << __func__ << " Returning cached time while rendering stopped:"
+             << last_reported_media_time_.InMicroseconds();
+    return last_reported_media_time_;
+  }
+
   // Return the current time based on the known extents of the rendered audio
   // data plus an estimate based on the last time those values were calculated.
   base::TimeDelta current_media_time = audio_clock_->front_timestamp();
   if (!last_render_time_.is_null()) {
     current_media_time +=
         (tick_clock_->NowTicks() - last_render_time_) * playback_rate_;
     if (current_media_time > audio_clock_->back_timestamp())
       current_media_time = audio_clock_->back_timestamp();
   }
 
   // Clamp current media time to the last reported value, this prevents higher
   // level clients from seeing time go backwards based on inaccurate or spurious
   // delay values reported to the AudioClock.
   //
   // It is expected that such events are transient and will be recovered as
   // rendering continues over time.
   if (current_media_time < last_media_timestamp_) {
     DVLOG(2) << __func__ << ": " << last_media_timestamp_
              << " (clamped), actual: " << current_media_time;
     return last_media_timestamp_;
   }
 
   DVLOG(2) << __func__ << ": " << current_media_time;
   last_media_timestamp_ = current_media_time;
+  last_reported_media_time_ = current_media_time;
   return current_media_time;
 }
 
 bool AudioRendererImpl::GetWallClockTimes(
     const std::vector<base::TimeDelta>& media_timestamps,
     std::vector<base::TimeTicks>* wall_clock_times) {
   base::AutoLock auto_lock(lock_);
   DCHECK(wall_clock_times->empty());
 
   // When playback is paused (rate is zero), assume a rate of 1.0.
@@ skipping 224 matching lines @@
             : stream->audio_decoder_config().channel_layout();
 
     audio_parameters_.Reset(hw_params.format(), renderer_channel_layout,
                             sample_rate, hw_params.bits_per_sample(),
                             media::AudioLatency::GetHighLatencyBufferSize(
                                 sample_rate, preferred_buffer_size));
   }
 
   audio_clock_.reset(
       new AudioClock(base::TimeDelta(), audio_parameters_.sample_rate()));
+  last_reported_media_time_ = kNoTimestamp;
 
   audio_buffer_stream_->Initialize(
       stream, base::Bind(&AudioRendererImpl::OnAudioBufferStreamInitialized,
                          weak_factory_.GetWeakPtr()),
       cdm_context, base::Bind(&AudioRendererImpl::OnStatisticsUpdate,
                               weak_factory_.GetWeakPtr()),
       base::Bind(&AudioRendererImpl::OnWaitingForDecryptionKey,
                  weak_factory_.GetWeakPtr()));
 }
 
@@ skipping 510 matching lines @@
   DCHECK_NE(buffering_state_, buffering_state);
   lock_.AssertAcquired();
   buffering_state_ = buffering_state;
 
   task_runner_->PostTask(
       FROM_HERE, base::Bind(&AudioRendererImpl::OnBufferingStateChange,
                             weak_factory_.GetWeakPtr(), buffering_state_));
 }
 
 }  // namespace media