Add support for midstream audio configuration changes.

media/base/audio_buffer_converter.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/base/audio_buffer_converter.h"
+
+#include <cstdlib>
+#include <list>
+
+#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/sinc_resampler.h"
+#include "media/base/vector_math.h"
+
+namespace media {
+
+// Is the config presented by |buffer| a config change from |params|?
+static bool IsConfigChange(const AudioParameters& params,
+                           const scoped_refptr<AudioBuffer>& buffer) {
+  return buffer->sample_rate() != params.sample_rate() ||
+         buffer->channel_count() != params.channels() ||
+         buffer->channel_layout() != params.channel_layout();
+}
+
+AudioBufferConverter::AudioBufferConverter(const AudioParameters& output_params)
+    : output_params_(output_params),
+      input_params_(output_params),
+      offset_into_queue_(0),
+      input_frames_(0),
+      buffered_input_frames_(0.0),
+      sample_rate_ratio_(1.0),
+      timestamp_helper_(output_params_.sample_rate()),
+      is_flushing_(false) {}
+
+AudioBufferConverter::~AudioBufferConverter() {}
+
+void AudioBufferConverter::AddInput(
+    const scoped_refptr<AudioBuffer>& buffer) {
+
+  // On EOS flush any remaining buffered data.
+  if (buffer->end_of_stream()) {
+    Flush();
+    queued_outputs_.push_back(buffer);
+    return;
+  }
+
+  // We'll need a new |audio_converter_| if there was a config change.
+  if (IsConfigChange(input_params_, buffer))
+    ResetConverter(buffer);
+
+  // Pass straight through if there's no work to be done.
+  if (!audio_converter_) {
+    queued_outputs_.push_back(buffer);
+    return;
+  }
+
+  if (timestamp_helper_.base_timestamp() == kNoTimestamp())
+    timestamp_helper_.SetBaseTimestamp(buffer->timestamp());
+
+  queued_inputs_.push_back(buffer);
+  input_frames_ += buffer->frame_count();
+
+  ConvertIfPossible();
+}
+
+bool AudioBufferConverter::HasNextBuffer() {
+  return !queued_outputs_.empty();
+}
+
+scoped_refptr<AudioBuffer> AudioBufferConverter::GetNextBuffer() {
+  DCHECK(!queued_outputs_.empty());
+  scoped_refptr<AudioBuffer> out = queued_outputs_.front();
+  queued_outputs_.pop_front();
+  return out;
+}
+
+void AudioBufferConverter::Reset() {
+  audio_converter_.reset(NULL);

[DaleCurtis, 2014/03/21 23:18:56]

shouldn't need NULL.

[rileya (GONE FROM CHROMIUM), 2014/03/24 18:07:05]

Done.

+  queued_inputs_.clear();
+  queued_outputs_.clear();
+  timestamp_helper_.SetBaseTimestamp(kNoTimestamp());
+  input_params_ = output_params_;
+  input_frames_ = 0;
+  buffered_input_frames_ = 0.0;
+  offset_into_queue_ = 0;
+}
+
+double AudioBufferConverter::ProvideInput(AudioBus* audio_bus,
+                                          base::TimeDelta buffer_delay) {
+  DCHECK(is_flushing_ || input_frames_ >= audio_bus->frames());
+
+  int requested_frames_left = audio_bus->frames();
+  int dest_index = 0;
+
+  while (requested_frames_left > 0 && !queued_inputs_.empty()) {
+    scoped_refptr<AudioBuffer> input_buffer = queued_inputs_.front();
+
+    int frames_to_read =
+        std::min(requested_frames_left,
+                 input_buffer->frame_count() - offset_into_queue_);
+    input_buffer->ReadFrames(
+        frames_to_read, offset_into_queue_, dest_index, audio_bus);
+    offset_into_queue_ += frames_to_read;
+
+    if (offset_into_queue_ == input_buffer->frame_count()) {
+      // We've consumed all the frames in |input_buffer|.
+      queued_inputs_.pop_front();
+      offset_into_queue_ = 0;
+    }
+
+    requested_frames_left -= frames_to_read;
+    dest_index += frames_to_read;
+  }
+
+  // If we're flushing, zero any extra space, otherwise we should always have
+  // enough data to completely fulfill the request.
+  if (is_flushing_ && requested_frames_left > 0)
+    audio_bus->ZeroFramesPartial(audio_bus->frames() - requested_frames_left,
+                                requested_frames_left);

[DaleCurtis, 2014/03/21 23:18:56]

alignment is off and multiline if's require {}

[rileya (GONE FROM CHROMIUM), 2014/03/24 18:07:05]

Done.

+  else
+    DCHECK_EQ(requested_frames_left, 0);
+
+  input_frames_ -= (audio_bus->frames() - requested_frames_left);
+  DCHECK_GE(input_frames_, 0);
+
+  buffered_input_frames_ += audio_bus->frames() - requested_frames_left;
+
+  // Full volume.
+  return 1.0;
+}
+
+void AudioBufferConverter::ResetConverter(
+    const scoped_refptr<AudioBuffer>& buffer) {
+  Flush();
+  audio_converter_.reset(NULL);
+  input_params_.Reset(
+      input_params_.format(),
+      buffer->channel_layout(),
+      buffer->channel_count(),
+      0,
+      buffer->sample_rate(),
+      input_params_.bits_per_sample(),
+      // This is arbitrary, but small buffer sizes result in a lot of tiny
+      // ProvideInput calls, so we'll use at least the SincResampler's default
+      // request size.
+      std::max(buffer->frame_count(),

[DaleCurtis, 2014/03/21 23:18:56]

Probably this should round to the nearest multiple of kKernelSize, but is
probably fine.

+               static_cast<int>(SincResampler::kDefaultRequestSize)));
+
+  sample_rate_ratio_ = static_cast<double>(input_params_.sample_rate()) /

[DaleCurtis, 2014/03/21 23:18:56]

alignment is off. git cl format is your friend :)

[rileya (GONE FROM CHROMIUM), 2014/03/24 18:07:05]

Fixed.

+                      output_params_.sample_rate();
+
+  // If |buffer| matches |output_params_| we don't need an AudioConverter at
+  // all, and can early-out here.
+  if (!IsConfigChange(output_params_, buffer))
+    return;
+
+  audio_converter_.reset(
+      new AudioConverter(input_params_, output_params_, false));
+  audio_converter_->AddInput(this);
+}
+
+void AudioBufferConverter::ConvertIfPossible() {
+  DCHECK(audio_converter_);
+
+  int request_frames = 0;
+
+  if (is_flushing_) {
+    // If we're flushing we want to convert *everything* even if this means
+    // we'll have to pad some silence in ProvideInput().
+    request_frames =
+        ceil((buffered_input_frames_ + input_frames_) / sample_rate_ratio_);
+  } else {
+    // How many calls to ProvideInput() we can satisfy completely.
+    int chunks = input_frames_ / input_params_.frames_per_buffer();
+
+    // How many output frames that corresponds to:
+    request_frames = chunks * audio_converter_->ChunkSize();
+  }
+
+  if (!request_frames)
+    return;
+
+  scoped_refptr<AudioBuffer> output_buffer =
+      AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
+                                output_params_.channel_layout(),
+                                output_params_.sample_rate(),
+                                request_frames);
+  scoped_ptr<AudioBus> output_bus =
+      AudioBus::CreateWrapper(output_buffer->channel_count());
+
+  int frames_remaining = request_frames;
+
+  // The AudioConverter wants requests of a fixed size, so we'll slide an
+  // AudioBus of that size across the |output_buffer|.
+  while (frames_remaining != 0) {
+    int frames_this_iteration =
+        std::min(output_params_.frames_per_buffer(), frames_remaining);
+
+    int offset_into_buffer = output_buffer->frame_count() - frames_remaining;
+
+    // Wrap the portion of the AudioBuffer in an AudioBus so the AudioConverter
+    // can fill it.
+    output_bus->set_frames(frames_this_iteration);
+    for (int ch = 0; ch < output_buffer->channel_count(); ++ch) {
+      output_bus->SetChannelData(
+          ch,
+          reinterpret_cast<float*>(output_buffer->channel_data()[ch]) +
+              offset_into_buffer);
+    }
+
+    // Do the actual conversion.
+    audio_converter_->Convert(output_bus.get());
+    frames_remaining -= frames_this_iteration;
+    buffered_input_frames_ -= frames_this_iteration * sample_rate_ratio_;
+  }
+
+  // Compute the timestamp.
+  output_buffer->set_timestamp(timestamp_helper_.GetTimestamp());
+  output_buffer->set_duration(
+      timestamp_helper_.GetFrameDuration(request_frames));
+  timestamp_helper_.AddFrames(request_frames);
+
+  queued_outputs_.push_back(output_buffer);
+}
+
+void AudioBufferConverter::Flush() {
+  if (!audio_converter_)
+    return;
+  is_flushing_ = true;
+  ConvertIfPossible();
+  is_flushing_ = false;
+  DCHECK_EQ(input_frames_, 0);
+  DCHECK_EQ(offset_into_queue_, 0);
+  DCHECK_LT(buffered_input_frames_, 1.0);
+  DCHECK(queued_inputs_.empty());
+  buffered_input_frames_ = 0.0;
+}
+
+} // namespace media