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

media/filters/audio_clock.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 "media/filters/audio_clock.h"
 
 #include "base/logging.h"
 #include "media/base/buffers.h"
 
 namespace media {
 
-AudioClock::AudioClock(int sample_rate)
-    : sample_rate_(sample_rate), last_endpoint_timestamp_(kNoTimestamp()) {
+AudioClock::AudioClock(base::TimeDelta start_timestamp, int sample_rate)
+    : start_timestamp_(start_timestamp), sample_rate_(sample_rate) {
 }
 
 AudioClock::~AudioClock() {
 }
 
-void AudioClock::WroteAudio(int frames,
+void AudioClock::WroteAudio(int frames_written,
+                            int frames_requested,
                             int delay_frames,
-                            float playback_rate,
-                            base::TimeDelta timestamp) {
-  CHECK_GT(playback_rate, 0);
-  CHECK(timestamp != kNoTimestamp());
-  DCHECK_GE(frames, 0);
+                            float playback_rate) {
+  DCHECK_GE(frames_written, 0);
+  DCHECK_LE(frames_written, frames_requested);
   DCHECK_GE(delay_frames, 0);
+  DCHECK_GE(playback_rate, 0);
 
-  if (last_endpoint_timestamp_ == kNoTimestamp())
-    PushBufferedAudio(delay_frames, 0, kNoTimestamp());
+  // First write: initialize buffer with silence.
+  if (buffered_.empty())
+    PushAudioData(&buffered_, delay_frames, 0.0f);
 
-  TrimBufferedAudioToMatchDelay(delay_frames);
-  PushBufferedAudio(frames, playback_rate, timestamp);
+  // Move frames from |buffered_| to |played_| based on |delay_frames|.
+  int64_t played_frames = std::max(0L, TotalFrames(buffered_) - delay_frames);

[DaleCurtis, 2014/08/02 00:51:32]

Seems like you could simply keep track of TotalFrames(buffered_) instead of
recomputing it every time.

[scherkus (not reviewing), 2014/08/02 01:55:24]

Done.

+  while (played_frames > 0) {
+    int64_t frames_to_move = std::min(buffered_.front().frames, played_frames);
 
-  last_endpoint_timestamp_ = timestamp;
-}
+    // No need to keep track of silent audio.
+    if (buffered_.front().playback_rate > 0.0f)
+      PushAudioData(&played_, frames_to_move, buffered_.front().playback_rate);
 
-void AudioClock::WroteSilence(int frames, int delay_frames) {
-  DCHECK_GE(frames, 0);
-  DCHECK_GE(delay_frames, 0);
+    buffered_.front().frames -= frames_to_move;
+    if (buffered_.front().frames == 0)
+      buffered_.pop_front();
 
-  if (last_endpoint_timestamp_ == kNoTimestamp())
-    PushBufferedAudio(delay_frames, 0, kNoTimestamp());
+    played_frames -= frames_to_move;
+  }
 
-  TrimBufferedAudioToMatchDelay(delay_frames);
-  PushBufferedAudio(frames, 0, kNoTimestamp());
+  // Push in newly buffered data.
+  PushAudioData(&buffered_, frames_written, playback_rate);
+  PushAudioData(&buffered_, frames_requested - frames_written, 0.0f);
 }
 
 base::TimeDelta AudioClock::CurrentMediaTimestamp(
     base::TimeDelta time_since_writing) const {
-  int frames_to_skip =
-      static_cast<int>(time_since_writing.InSecondsF() * sample_rate_);
-  int silence_frames = 0;
-  for (size_t i = 0; i < buffered_audio_.size(); ++i) {
-    int frames = buffered_audio_[i].frames;
-    if (frames_to_skip > 0) {
-      if (frames <= frames_to_skip) {
-        frames_to_skip -= frames;
-        continue;
-      }
-      frames -= frames_to_skip;
-      frames_to_skip = 0;
+  // Count up all |played_| audio since |start_timestamp_|.
+  base::TimeDelta current_timestamp = start_timestamp_;
+  for (size_t i = 0; i < played_.size(); ++i) {

[DaleCurtis, 2014/08/02 00:51:33]

played_ never shrinks, so you should just cache current_timestamp and update it
only based on new frames during WroteAudio().  Otherwise the degenerative
flip-flop playback-rate case is expensive.  Especially so on arm.

[scherkus (not reviewing), 2014/08/02 01:55:24]

have to think about it some more ... but doesn't that put us at risk for
introducing error over time?

do you think it's worth optimizing for such a case?

[DaleCurtis, 2014/08/04 18:55:18]

I'd guess only slightly more error than you're already introducing; if you store
the timestamp as a double, it'd be equivalent to your present calculation.  If
you want to avoid as much error as possible you need to collect frames by
playback_rate.  I don't know how much playback rates flip flop in practice or
how long it would take for this to be noticeable.

I just wanted to point it out since it's a potentially hidden performance
problem.  If you do decide it's worth optimizing, I think it could be done
simply with a float hash_map.

Just note that this is called on the AudioThread at up to 375Hz, though the
common case is now ~47Hz

+    DCHECK_NE(played_[i].playback_rate, 0.0f)
+        << "Silent audio doesn't need to be tracked in |played_|.";
+    current_timestamp += base::TimeDelta::FromMicroseconds(

[DaleCurtis, 2014/08/02 00:51:32]

calculate this as a double or float and only divide by sample rate and convert
to microseconds once.

[scherkus (not reviewing), 2014/08/02 01:55:25]

Done.

+        (played_[i].frames * played_[i].playback_rate) / sample_rate_ *
+        base::Time::kMicrosecondsPerSecond);
+  }
+
+  // Count up all |buffered_| audio based on |time_since_writing|.
+  int64_t frames_played_since_writing =
+      static_cast<int64_t>(time_since_writing.InSecondsF() * sample_rate_);
+  for (size_t i = 0; i < buffered_.size() && frames_played_since_writing > 0;

[DaleCurtis, 2014/08/02 00:51:33]

You could cache this too and subtract off time_since_writing, no?  Then you
remove this loop here and in ContiguousAudioDataBuffered.

[scherkus (not reviewing), 2014/08/02 01:55:24]

Done.

[DaleCurtis, 2014/08/04 18:55:18]

You said done, but didn't do this.  Was that your intent?

+       ++i) {
+    int64_t frames_played =
+        std::min(buffered_[i].frames, frames_played_since_writing);
+    current_timestamp += base::TimeDelta::FromMicroseconds(

[DaleCurtis, 2014/08/02 00:51:32]

Ditto.

[DaleCurtis, 2014/08/04 18:55:18]

I meant break out the TimeDelta conversion like above.

+        (frames_played * buffered_[i].playback_rate) / sample_rate_ *
+        base::Time::kMicrosecondsPerSecond);
+    frames_played_since_writing -= frames_played;
+  }
+
+  return current_timestamp;
+}
+
+base::TimeDelta AudioClock::ContiguousAudioDataBuffered() const {
+  base::TimeDelta buffered;
+  for (size_t i = 0; i < buffered_.size(); ++i) {
+    // Any buffered silence breaks our contiguous stretch of audio data.
+    if (buffered_[i].playback_rate == 0)
+      break;
+
+    // Multiply by playback rate as frames represent time-scaled audio.
+    buffered += base::TimeDelta::FromMicroseconds(
+        (buffered_[i].frames * buffered_[i].playback_rate) / sample_rate_ *
+        base::Time::kMicrosecondsPerSecond);
+  }
+
+  return buffered;
+}
+
+base::TimeDelta AudioClock::ContiguousAudioDataBufferedAtSameRate() const {
+  base::TimeDelta buffered;
+  for (size_t i = 0; i < buffered_.size(); ++i) {
+    // Any buffered silence breaks our contiguous stretch of audio data.
+    if (buffered_[i].playback_rate == 0)
+      break;
+
+    // Multiply by playback rate as frames represent time-scaled audio.
+    buffered = base::TimeDelta::FromMicroseconds(

[DaleCurtis, 2014/08/02 00:51:33]

Should this be += ? I don't understand why you have the loop otherwise.  In
which case I'm not sure how this differs from above?

[scherkus (not reviewing), 2014/08/02 01:55:24]

Nah this was just silly. We always break so this doesn't even need to be a loop.

+        (buffered_[i].frames * buffered_[i].playback_rate) / sample_rate_ *
+        base::Time::kMicrosecondsPerSecond);
+
+    if ((i + 1) < buffered_.size()) {
+      DCHECK_NE(buffered_[i].playback_rate, buffered_[i + 1].playback_rate)
+          << "Adjacent AudioData elements shouldn't have same playback rate";
     }
 
-    // Account for silence ahead of the buffer closest to being played.
-    if (buffered_audio_[i].playback_rate == 0) {
-      silence_frames += frames;
-      continue;
-    }
-
-    // Multiply by playback rate as frames represent time-scaled audio.
-    return buffered_audio_[i].endpoint_timestamp -
-           base::TimeDelta::FromMicroseconds(
-               ((frames * buffered_audio_[i].playback_rate) + silence_frames) /
-               sample_rate_ * base::Time::kMicrosecondsPerSecond);
-  }
-
-  // Either:
-  //   1) AudioClock is uninitialziated and we'll return kNoTimestamp()
-  //   2) All previously buffered audio has been replaced by silence,
-  //      meaning media time is now at the last endpoint
-  return last_endpoint_timestamp_;
-}
-
-void AudioClock::TrimBufferedAudioToMatchDelay(int delay_frames) {
-  if (buffered_audio_.empty())
-    return;
-
-  size_t i = buffered_audio_.size() - 1;
-  while (true) {
-    if (buffered_audio_[i].frames <= delay_frames) {
-      // Reached the end before accounting for all of |delay_frames|. This
-      // means we haven't written enough audio data yet to account for hardware
-      // delay. In this case, do nothing.
-      if (i == 0)
-        return;
-
-      // Keep accounting for |delay_frames|.
-      delay_frames -= buffered_audio_[i].frames;
-      --i;
-      continue;
-    }
-
-    // All of |delay_frames| has been accounted for: adjust amount of frames
-    // left in current buffer. All preceeding elements with index < |i| should
-    // be considered played out and hence discarded.
-    buffered_audio_[i].frames = delay_frames;
     break;
   }
 
-  // At this point |i| points at what will be the new head of |buffered_audio_|
-  // however if it contains no audio it should be removed as well.
-  if (buffered_audio_[i].frames == 0)
-    ++i;
-
-  buffered_audio_.erase(buffered_audio_.begin(), buffered_audio_.begin() + i);
+  return buffered;
 }
 
-void AudioClock::PushBufferedAudio(int frames,
-                                   float playback_rate,
-                                   base::TimeDelta endpoint_timestamp) {
-  if (playback_rate == 0)
-    DCHECK(endpoint_timestamp == kNoTimestamp());
+bool AudioClock::AudioDataBuffered() const {
+  for (size_t i = 0; i < buffered_.size(); ++i) {
+    if (buffered_[i].playback_rate != 0) {
+      DCHECK_NE(buffered_[i].frames, 0)
+          << "AudioData elements with zero frames shouldn't exist";
+      return true;
+    }
+  }
+  return false;
+}
 
+AudioClock::AudioData::AudioData(int64_t frames, float playback_rate)

[DaleCurtis, 2014/08/02 00:51:33]

Up to you, but you can remove this and use a POD-type with = {} initialization. 

+    : frames(frames), playback_rate(playback_rate) {
+}
+
+// static
+void AudioClock::PushAudioData(std::deque<AudioData>* audio_data,
+                               int64_t frames,
+                               float playback_rate) {
   if (frames == 0)
     return;
 
   // Avoid creating extra elements where possible.
-  if (!buffered_audio_.empty() &&
-      buffered_audio_.back().playback_rate == playback_rate) {
-    buffered_audio_.back().frames += frames;
-    buffered_audio_.back().endpoint_timestamp = endpoint_timestamp;
+  if (!audio_data->empty() &&
+      audio_data->back().playback_rate == playback_rate) {
+    audio_data->back().frames += frames;
     return;
   }
 
-  buffered_audio_.push_back(
-      BufferedAudio(frames, playback_rate, endpoint_timestamp));
+  audio_data->push_back(AudioData(frames, playback_rate));
 }
 
-AudioClock::BufferedAudio::BufferedAudio(int frames,
-                                         float playback_rate,
-                                         base::TimeDelta endpoint_timestamp)
-    : frames(frames),
-      playback_rate(playback_rate),
-      endpoint_timestamp(endpoint_timestamp) {
+// static
+int64_t AudioClock::TotalFrames(const std::deque<AudioData>& audio_data) {
+  int64_t total_frames = 0;
+  for (size_t i = 0; i < audio_data.size(); ++i) {
+    total_frames += audio_data[i].frames;
+  }
+  return total_frames;
 }
 
 }  // namespace media