Enhance AudioSplicer to crossfade marked splice frames.

media/base/audio_splicer.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/base/audio_splicer.h"
 
 #include <cstdlib>
+#include <deque>
 
 #include "base/logging.h"
 #include "media/base/audio_buffer.h"
+#include "media/base/audio_bus.h"
 #include "media/base/audio_decoder_config.h"
 #include "media/base/audio_timestamp_helper.h"
 #include "media/base/buffers.h"
+#include "media/base/vector_math.h"
 
 namespace media {
 
 // Largest gap or overlap allowed by this class. Anything
 // larger than this will trigger an error.
 // This is an arbitrary value, but the initial selection of 50ms
 // roughly represents the duration of 2 compressed AAC or MP3 frames.
 static const int kMaxTimeDeltaInMilliseconds = 50;
 
-AudioSplicer::AudioSplicer(int samples_per_second)
-    : output_timestamp_helper_(samples_per_second),
-      min_gap_size_(2),
-      received_end_of_stream_(false) {
+// Minimum gap size needed before the splicer will take action to
+// fill a gap. This avoids periodically inserting and then dropping samples
+// when the buffer timestamps are slightly off because of timestamp rounding
+// in the source content. Unit is frames.
+static const int kMinGapSize = 2;
+
+// The number of milliseconds to crossfade before trimming when buffers overlap.
+static const int kCrossfadeDurationInMilliseconds = 5;
+
+// AudioBuffer::TrimStart() is not as accurate as the timestamp helper, so
+// manually adjust the duration and timestamp after trimming.
+static void AccurateTrimStart(int frames_to_trim,
+                              const scoped_refptr<AudioBuffer> buffer,
+                              const AudioTimestampHelper& timestamp_helper) {
+  buffer->TrimStart(frames_to_trim);
+  buffer->set_timestamp(timestamp_helper.GetTimestamp());
+  buffer->set_duration(
+      timestamp_helper.GetFrameDuration(buffer->frame_count()));
 }
 
-AudioSplicer::~AudioSplicer() {
+// AudioBuffer::TrimEnd() is not as accurate as the timestamp helper, so
+// manually adjust the duration after trimming.
+static void AccurateTrimEnd(int frames_to_trim,
+                            const scoped_refptr<AudioBuffer> buffer,
+                            const AudioTimestampHelper& timestamp_helper) {
+  buffer->TrimEnd(frames_to_trim);

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

nit: DCHECK_EQ(buffer->timestamp(), timestamp_helper.GetTimestamp()) ?

[DaleCurtis, 2014/02/28 00:24:41]

Done.

+  buffer->set_duration(
+      timestamp_helper.GetFrameDuration(buffer->frame_count()));
 }
 
-void AudioSplicer::Reset() {
-  output_timestamp_helper_.SetBaseTimestamp(kNoTimestamp());
+typedef std::deque<scoped_refptr<AudioBuffer> > BufferQueue;

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

nit: I think this should be inside the AudioStreamSanitizer since that appears
to be the only place it is being used right now.

[DaleCurtis, 2014/02/28 00:24:41]

Done.

+
+class AudioStreamSanitizer {
+ public:
+  explicit AudioStreamSanitizer(int samples_per_second);
+  ~AudioStreamSanitizer();
+
+  // Resets the sanitizer state by clearing the output buffers queue, and
+  // resetting the timestamp helper.
+  void Reset();
+
+  // Similar to Reset(), but initializes the timestamp helper with the given
+  // parameters.
+  void ResetTimestampState(int64 frame_count, base::TimeDelta base_timestamp);
+
+  // Adds a new buffer full of samples or end of stream buffer to the splicer.
+  // Returns true if the buffer was accepted. False is returned if an error
+  // occurred.
+  bool AddInput(const scoped_refptr<AudioBuffer>& input);
+
+  // Returns true if the sanitizer has a buffer to return.
+  bool HasNextBuffer() const;
+
+  // Removes the next buffer from the output buffer queue and returns it; should
+  // only be called if HasNextBuffer() returns true.
+  scoped_refptr<AudioBuffer> GetNextBuffer();
+
+  // Returns the total frame count of all buffers available for output.
+  int GetFrameCount() const;
+
+  // Returns the duration of all buffers added to the output queue thus far.
+  base::TimeDelta GetDuration() const;
+
+  // Sets the base timestamp of the AudioTimestampHelper.  Must only be called
+  // if no base timestamp has been set.
+  void SetBaseTimestamp(base::TimeDelta base_timestamp);
+
+  const AudioTimestampHelper& timestamp_helper() {
+    return output_timestamp_helper_;
+  }
+
+ private:
+  void AddOutputBuffer(const scoped_refptr<AudioBuffer>& buffer);
+
+  AudioTimestampHelper output_timestamp_helper_;
+  BufferQueue output_buffers_;
+  bool received_end_of_stream_;
+
+  DISALLOW_ASSIGN(AudioStreamSanitizer);
+};
+
+AudioStreamSanitizer::AudioStreamSanitizer(int samples_per_second)
+    : output_timestamp_helper_(samples_per_second),
+      received_end_of_stream_(false) {}
+
+AudioStreamSanitizer::~AudioStreamSanitizer() {}
+
+void AudioStreamSanitizer::Reset() {
+  ResetTimestampState(0, kNoTimestamp());
+}
+
+void AudioStreamSanitizer::ResetTimestampState(int64 frame_count,
+                                               base::TimeDelta base_timestamp) {
   output_buffers_.clear();
   received_end_of_stream_ = false;
+  output_timestamp_helper_.SetBaseTimestamp(base_timestamp);
+  if (frame_count > 0)
+    output_timestamp_helper_.AddFrames(frame_count);
 }
 
-bool AudioSplicer::AddInput(const scoped_refptr<AudioBuffer>& input) {
+bool AudioStreamSanitizer::AddInput(const scoped_refptr<AudioBuffer>& input) {
   DCHECK(!received_end_of_stream_ || input->end_of_stream());
 
   if (input->end_of_stream()) {
     output_buffers_.push_back(input);
     received_end_of_stream_ = true;
     return true;
   }
 
   DCHECK(input->timestamp() != kNoTimestamp());
   DCHECK(input->duration() > base::TimeDelta());
   DCHECK_GT(input->frame_count(), 0);
 
   if (output_timestamp_helper_.base_timestamp() == kNoTimestamp())
     output_timestamp_helper_.SetBaseTimestamp(input->timestamp());
 
   if (output_timestamp_helper_.base_timestamp() > input->timestamp()) {
     DVLOG(1) << "Input timestamp is before the base timestamp.";
     return false;
   }
 
-  base::TimeDelta timestamp = input->timestamp();
-  base::TimeDelta expected_timestamp = output_timestamp_helper_.GetTimestamp();
-  base::TimeDelta delta = timestamp - expected_timestamp;
+  const base::TimeDelta timestamp = input->timestamp();
+  const base::TimeDelta expected_timestamp =
+      output_timestamp_helper_.GetTimestamp();
+  const base::TimeDelta delta = timestamp - expected_timestamp;
 
   if (std::abs(delta.InMilliseconds()) > kMaxTimeDeltaInMilliseconds) {
     DVLOG(1) << "Timestamp delta too large: " << delta.InMicroseconds() << "us";
     return false;
   }
 
   int frames_to_fill = 0;
   if (delta != base::TimeDelta())
     frames_to_fill = output_timestamp_helper_.GetFramesToTarget(timestamp);
 
-  if (frames_to_fill == 0 || std::abs(frames_to_fill) < min_gap_size_) {
+  if (frames_to_fill == 0 || std::abs(frames_to_fill) < kMinGapSize) {
     AddOutputBuffer(input);
     return true;
   }
 
   if (frames_to_fill > 0) {
     DVLOG(1) << "Gap detected @ " << expected_timestamp.InMicroseconds()
              << " us: " << delta.InMicroseconds() << " us";
 
     // Create a buffer with enough silence samples to fill the gap and
     // add it to the output buffer.
     scoped_refptr<AudioBuffer> gap = AudioBuffer::CreateEmptyBuffer(
         input->channel_count(),
         frames_to_fill,
         expected_timestamp,
         output_timestamp_helper_.GetFrameDuration(frames_to_fill));
     AddOutputBuffer(gap);
 
     // Add the input buffer now that the gap has been filled.
     AddOutputBuffer(input);
     return true;
   }
 
-  int frames_to_skip = -frames_to_fill;
+  // Overlapping buffers marked as splice frames are handled by AudioSplicer,
+  // but decoder and demuxer quirks may sometimes produce overlapping samples
+  // which need to be sanitized.
+  //
+  // A crossfade can't be done here because only the current buffer is available
+  // at this point, not previous buffers.
+  DVLOG(1) << "Overlap detected @ " << expected_timestamp.InMicroseconds()
+           << " us: " << -delta.InMicroseconds() << " us";
 
-  DVLOG(1) << "Overlap detected @ " << expected_timestamp.InMicroseconds()
-           << " us: "  << -delta.InMicroseconds() << " us";
-
+  const int frames_to_skip = -frames_to_fill;
   if (input->frame_count() <= frames_to_skip) {
     DVLOG(1) << "Dropping whole buffer";
     return true;
   }
 
   // Copy the trailing samples that do not overlap samples already output
   // into a new buffer. Add this new buffer to the output queue.
-  //
-  // TODO(acolwell): Implement a cross-fade here so the transition is less
-  // jarring.
-  input->TrimStart(frames_to_skip);
+  AccurateTrimStart(frames_to_skip, input, output_timestamp_helper_);
   AddOutputBuffer(input);
   return true;
 }
 
-bool AudioSplicer::HasNextBuffer() const {
+bool AudioStreamSanitizer::HasNextBuffer() const {
   return !output_buffers_.empty();
 }
 
-scoped_refptr<AudioBuffer> AudioSplicer::GetNextBuffer() {
+scoped_refptr<AudioBuffer> AudioStreamSanitizer::GetNextBuffer() {
   scoped_refptr<AudioBuffer> ret = output_buffers_.front();
   output_buffers_.pop_front();
   return ret;
 }
 
-void AudioSplicer::AddOutputBuffer(const scoped_refptr<AudioBuffer>& buffer) {
+void AudioStreamSanitizer::AddOutputBuffer(
+    const scoped_refptr<AudioBuffer>& buffer) {
   output_timestamp_helper_.AddFrames(buffer->frame_count());
   output_buffers_.push_back(buffer);
 }
 
+int AudioStreamSanitizer::GetFrameCount() const {
+  int frame_count = 0;
+  for(BufferQueue::const_iterator it = output_buffers_.begin();

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

nit: space after for

[DaleCurtis, 2014/02/28 00:24:41]

Done.

+      it != output_buffers_.end(); ++it) {
+    frame_count += (*it)->frame_count();
+  }
+  return frame_count;
+}
+
+base::TimeDelta AudioStreamSanitizer::GetDuration() const {
+  DCHECK(output_timestamp_helper_.base_timestamp() != kNoTimestamp());
+  return output_timestamp_helper_.GetTimestamp() -
+         output_timestamp_helper_.base_timestamp();
+}
+
+void AudioStreamSanitizer::SetBaseTimestamp(base::TimeDelta base_timestamp) {
+  DCHECK(output_timestamp_helper_.base_timestamp() == kNoTimestamp());
+  output_timestamp_helper_.SetBaseTimestamp(base_timestamp);
+}
+
+AudioSplicer::AudioSplicer(int samples_per_second)
+    : max_crossfade_duration_(
+          base::TimeDelta::FromMilliseconds(kCrossfadeDurationInMilliseconds)),
+      splice_timestamp_(kNoTimestamp()),
+      output_sanitizer_(new AudioStreamSanitizer(samples_per_second)),
+      pre_splice_sanitizer_(new AudioStreamSanitizer(samples_per_second)),
+      post_splice_sanitizer_(new AudioStreamSanitizer(samples_per_second)) {}
+
+AudioSplicer::~AudioSplicer() {}
+
+void AudioSplicer::Reset() {
+  output_sanitizer_->Reset();
+  pre_splice_sanitizer_->Reset();
+  post_splice_sanitizer_->Reset();
+  splice_timestamp_ = kNoTimestamp();
+}
+
+bool AudioSplicer::AddInput(const scoped_refptr<AudioBuffer>& input) {
+  // If we're not processing a splice, add the input to the output queue.
+  if (splice_timestamp_ == kNoTimestamp())
+    return output_sanitizer_->AddInput(input);

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

DCHECK(!pre_splice_sanitizer_->HasNextBuffer());
DCHECK(!post_splice_sanitizer_->HasNextBuffer());

in this block just to make sure our assumptions below hold?

[DaleCurtis, 2014/02/28 00:24:41]

Done.

+
+  // If we're still receiving buffers before the splice point figure out which
+  // sanitizer (if any) to put them in.
+  if (!post_splice_sanitizer_->HasNextBuffer()) {
+    DCHECK(!input->end_of_stream());
+
+    // If the provided buffer is entirely before the splice point it can also be
+    // added to the output queue.
+    if (input->timestamp() + input->duration() < splice_timestamp_)
+      return output_sanitizer_->AddInput(input);

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

DCHECK(!pre_splice_sanitizer_->HasNextBuffer()); here just to make sure we don't
allow anything to sneak into the output_sanitizer_?

[DaleCurtis, 2014/02/28 00:24:41]

Done.

+
+    // If we've encountered the first pre splice buffer, reset the pre splice
+    // sanitizer based on |output_sanitizer_|.  This is done so that gaps and
+    // overlaps between buffers across the sanitizers are accounted for prior
+    // to calculating crossfade.
+    if (!pre_splice_sanitizer_->HasNextBuffer()) {
+      pre_splice_sanitizer_->ResetTimestampState(
+          output_sanitizer_->timestamp_helper().frame_count(),
+          output_sanitizer_->timestamp_helper().base_timestamp());
+    }
+
+    // If we're processing a splice and the input buffer does not overlap any of
+    // the existing buffers, append it to the splice queue for processing.
+    if (!pre_splice_sanitizer_->HasNextBuffer() ||
+        input->timestamp() != splice_timestamp_) {

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

We can rely on the "input->timestamp() != splice_timestamp_" condition because
we know the splice always happens in the middle of a pre-splice frame? This is a
little subtle and doesn't really reflect what the comment above is saying. I
wonder if something like
!pre_splice_sanitizer_->IsOverlapping(input->timestamp())

where 
bool IsOverlapping(TimeDelta timestamp) {
  return timestamp >= output_timestamp_helper_.GetTimestamp()
}

[DaleCurtis, 2014/02/27 19:45:01]

It's only strong enough when included with the
!pre_splice_sanitizer_->HasNextBuffer() check, since we could have a perfect
overlap.

We can rely on it because we expect the first overlapping block to have a
timestamp of exactly splice_timestamp (config change, first base_timestamp ==
splice timestamp).

This is definitely subtle... but it's not sufficient to check timestamp >=
GetTimestamp() since there may be decoder fuzz in the timestamps.

[acolwell GONE FROM CHROMIUM, 2014/02/28 18:50:27]

Ok then some of this should be in the comment then because I'm sure we'll both
forget it. :)

[DaleCurtis, 2014/02/28 21:14:26]

Done.  I also added a CHECK() for the unlikely case that the decoder fuzzes the
n>1'th pre splice buffer such that splice_timestamp is seen before it's ready.

+      return pre_splice_sanitizer_->AddInput(input);
+    }
+
+    // We've received the first overlapping buffer.
+  }
+
+  // At this point we have all the fade out preroll buffers from the decoder.
+  // We now need to wait until we have enough data to perform the crossfade (or
+  // we receive an end of stream).
+  if (!post_splice_sanitizer_->AddInput(input))
+    return false;
+
+  if (!input->end_of_stream() &&
+      post_splice_sanitizer_->GetDuration() < max_crossfade_duration_) {
+    return true;
+  }
+
+  // Transfer out preroll buffers involved in the splice, drop those not.  Since
+  // we don't want to care what format the AudioBuffers are in, we need to use
+  // an intermediary AudioBus to convert the data to float.
+  scoped_ptr<AudioBus> pre_splice_bus = ExtractCrossfadeFromPreSplice();
+
+  // Allocate output buffer for crossfade.
+  scoped_refptr<AudioBuffer> crossfade_buffer =
+      AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
+                                pre_splice_bus->channels(),
+                                pre_splice_bus->frames());
+
+  // Use the calculated timestamp and duration to ensure there's no extra gaps
+  // or overlaps to process when adding the buffer to |output_sanitizer_|.
+  const AudioTimestampHelper& output_ts_helper =
+      output_sanitizer_->timestamp_helper();
+  crossfade_buffer->set_timestamp(output_ts_helper.GetTimestamp());
+  crossfade_buffer->set_duration(
+      output_ts_helper.GetFrameDuration(pre_splice_bus->frames()));
+
+  // AudioBuffer::ReadFrames() only allows output into an AudioBus, so wrap
+  // our AudioBuffer in one so we can avoid extra data copies.
+  scoped_ptr<AudioBus> crossfade_bus_wrapper =
+      AudioBus::CreateWrapper(crossfade_buffer->channel_count());
+  crossfade_bus_wrapper->set_frames(crossfade_buffer->frame_count());
+  for (int ch = 0; ch < crossfade_buffer->channel_count(); ++ch) {
+    crossfade_bus_wrapper->SetChannelData(
+        ch, reinterpret_cast<float*>(crossfade_buffer->channel_data()[ch]));
+  }
+
+  // Insert the crossfade buffer into the output queue now so post splice
+  // buffers can be added in processing order.  We will still modify the buffer
+  // during the crossfade step.
+  CHECK(output_sanitizer_->AddInput(crossfade_buffer));

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

This seems risky since AddInput() can modify crossfade_buffer's state. I realize
that we shouldn't trigger the trim logic in there, but I think in general there
is an implict contract that the buffer isn't going to be modified by the caller
once it is passed into the sanitizer.

I think ExtractCrossfadeFromPostSplice() should just pass the one potential
partial buffer out through an out param if necessary and then do all the
appending to the output_sanitizer_ after the crossfade is computed.

[DaleCurtis, 2014/02/28 00:24:41]

This is messy since then trimming and the accuracy update are split across the
main loop and the extraction method.

I much prefer the current setup since it the ExtractXXX methods have symmetry
which makes them easy to understand.

Instead I've added a DCHECK() to ensure trim is not invoked.  I can promote that
to a check if necessary.  But in either case it wouldn't be a use after free
since AudioBuffer isn't actually shrunk during trimming.

[acolwell GONE FROM CHROMIUM, 2014/02/28 18:50:27]

I still think this is a bad idea. You could move the creation of
crossfade_buffer and the actual crossfade down into
ExtractCrossfadeFromPostSplice(). I don't see any reason this code needs to be
up here. That would a be more natural setup. 

While I am sympathetic to the symmetry argument, I don't think it is a good idea
to rely on internal details of AudioBuffer & AudioStreamSanitizer like this.
AddInput() is essentially an ownership transfer so I don't think you should be
modifying the contents of the buffer once you've passed it into this method. 

[DaleCurtis, 2014/02/28 21:14:26]

Switching to a CrossfadePostSplice() method looks good.  Done.

+
+  ExtractCrossfadeFromPostSplice(crossfade_bus_wrapper.get());
+
+  // Crossfade the audio into |crossfade_buffer|.
+  for (int ch = 0; ch < crossfade_bus_wrapper->channels(); ++ch) {
+    vector_math::Crossfade(pre_splice_bus->channel(ch),
+                           pre_splice_bus->frames(),
+                           crossfade_bus_wrapper->channel(ch));
+  }
+
+  // Clear the splice timestamp so new splices can be accepted.
+  splice_timestamp_ = kNoTimestamp();
+  return true;
+}
+
+bool AudioSplicer::HasNextBuffer() const {
+  return output_sanitizer_->HasNextBuffer();
+}
+
+scoped_refptr<AudioBuffer> AudioSplicer::GetNextBuffer() {
+  return output_sanitizer_->GetNextBuffer();
+}
+
+void AudioSplicer::SetSpliceTimestamp(base::TimeDelta splice_timestamp) {
+  DCHECK(splice_timestamp != kNoTimestamp());
+  if (splice_timestamp_ == splice_timestamp)
+    return;
+
+  // TODO(dalecurtis): We may need the concept of a future_splice_timestamp_ to
+  // handle cases where another splice comes in before we've received 5ms of
+  // data from the last one.  Leave this as a CHECK for now to figure out if
+  // this case is possible.
+  CHECK(splice_timestamp_ == kNoTimestamp());
+  splice_timestamp_ = splice_timestamp;
+}
+
+scoped_ptr<AudioBus> AudioSplicer::ExtractCrossfadeFromPreSplice() {
+  const AudioTimestampHelper& output_ts_helper =
+      output_sanitizer_->timestamp_helper();
+
+  // Ensure |output_sanitizer_| has a valid base timestamp so we can use it for
+  // timestamp calculations.
+  if (output_ts_helper.base_timestamp() == kNoTimestamp()) {
+    output_sanitizer_->SetBaseTimestamp(

[acolwell GONE FROM CHROMIUM, 2014/02/27 17:10:14]

Could you use ResetTimestampState() here instead? I think then you won't need
the SetBaseTimestamp() method.

IIUC this can only happen if the splice is the first thing that happens and no
buffers have been output yet. Right?

[DaleCurtis, 2014/02/27 19:45:01]

Correct and a good idea, but I I was thinking of changing ResetTimestampState()
to take only a timestamp_helper() and extract the necessary information itself. 
Which do you prefer?

+        pre_splice_sanitizer_->timestamp_helper().base_timestamp());
+  }
+
+  int frames_before_splice =
+      output_ts_helper.GetFramesToTarget(splice_timestamp_);
+
+  // Determine crossfade frame count based on available frames in each splicer
+  // and capping to the maximum crossfade duration.
+  const int max_crossfade_frame_count =
+      output_ts_helper.GetFramesToTarget(splice_timestamp_ +
+                                         max_crossfade_duration_) -
+      frames_before_splice;
+  const int frames_to_crossfade = std::min(
+      max_crossfade_frame_count,
+      std::min(pre_splice_sanitizer_->GetFrameCount() - frames_before_splice,
+               post_splice_sanitizer_->GetFrameCount()));
+
+  int frames_read = 0;
+  scoped_ptr<AudioBus> output_bus;
+  while (pre_splice_sanitizer_->HasNextBuffer() &&
+         frames_read < frames_to_crossfade) {
+    scoped_refptr<AudioBuffer> preroll = pre_splice_sanitizer_->GetNextBuffer();
+
+    // We don't know the channel count until we see the first buffer, so wait
+    // until the first buffer to allocate the output AudioBus.
+    if (!output_bus) {
+      output_bus =
+          AudioBus::Create(preroll->channel_count(), frames_to_crossfade);
+    }
+
+    // There may be enough of a gap introduced during decoding such that an
+    // entire buffer exists before the splice point.
+    if (frames_before_splice >= preroll->frame_count()) {
+      frames_before_splice -= preroll->frame_count();
+      CHECK(output_sanitizer_->AddInput(preroll));
+      continue;
+    }
+
+    const int frames_to_read =
+        std::min(preroll->frame_count() - frames_before_splice,
+                 output_bus->frames() - frames_read);
+    preroll->ReadFrames(
+        frames_to_read, frames_before_splice, frames_read, output_bus.get());
+    frames_read += frames_to_read;
+
+    // If only part of the buffer was consumed, trim it appropriately and stick
+    // it into the output queue.
+    if (frames_before_splice) {
+      AccurateTrimEnd(preroll->frame_count() - frames_before_splice,
+                      preroll,
+                      output_ts_helper);
+      CHECK(output_sanitizer_->AddInput(preroll));
+      frames_before_splice = 0;
+    }
+  }
+
+  // All necessary buffers have been processed, it's safe to reset.
+  pre_splice_sanitizer_->Reset();
+  DCHECK_EQ(output_bus->frames(), frames_read);
+  DCHECK_EQ(output_ts_helper.GetFramesToTarget(splice_timestamp_), 0);
+  return output_bus.Pass();
+}
+
+void AudioSplicer::ExtractCrossfadeFromPostSplice(AudioBus* output_bus) {
+  int frames_read = 0;
+  while (post_splice_sanitizer_->HasNextBuffer() &&
+         frames_read < output_bus->frames()) {
+    scoped_refptr<AudioBuffer> postroll =
+        post_splice_sanitizer_->GetNextBuffer();
+    const int frames_to_read =
+        std::min(postroll->frame_count(), output_bus->frames() - frames_read);
+    postroll->ReadFrames(frames_to_read, 0, frames_read, output_bus);
+    frames_read += frames_to_read;
+
+    // If only part of the buffer was consumed, trim it appropriately and stick
+    // it into the output queue.
+    if (frames_to_read < postroll->frame_count()) {
+      AccurateTrimStart(
+          frames_to_read, postroll, output_sanitizer_->timestamp_helper());
+      CHECK(output_sanitizer_->AddInput(postroll));
+    }
+  }
+
+  DCHECK_EQ(output_bus->frames(), frames_read);
+
+  // Transfer all remaining buffers out and reset once empty.
+  while (post_splice_sanitizer_->HasNextBuffer())
+    CHECK(output_sanitizer_->AddInput(post_splice_sanitizer_->GetNextBuffer()));
+  post_splice_sanitizer_->Reset();
+}
+
 }  // namespace media