media/base/audio_buffer_converter.cc

   1 // Copyright 2014 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #include "media/base/audio_buffer_converter.h"
   6
   7 #include <cmath>
   8
   9 #include "base/logging.h"
  10 #include "media/base/audio_buffer.h"
  11 #include "media/base/audio_bus.h"
  12 #include "media/base/audio_decoder_config.h"
  13 #include "media/base/audio_timestamp_helper.h"
  14 #include "media/base/buffers.h"
  15 #include "media/base/sinc_resampler.h"
  16 #include "media/base/vector_math.h"
  17
  18 namespace media {
  19
  20 // Is the config presented by |buffer| a config change from |params|?
  21 static bool IsConfigChange(const AudioParameters& params,
  22                            const scoped_refptr<AudioBuffer>& buffer) {
  23   return buffer->sample_rate() != params.sample_rate() ||
  24          buffer->channel_count() != params.channels() ||
  25          buffer->channel_layout() != params.channel_layout();
  26 }
  27
  28 AudioBufferConverter::AudioBufferConverter(const AudioParameters& output_params)
  29     : output_params_(output_params),
  30       input_params_(output_params),
  31       last_input_buffer_offset_(0),
  32       input_frames_(0),
  33       buffered_input_frames_(0.0),
  34       io_sample_rate_ratio_(1.0),
  35       timestamp_helper_(output_params_.sample_rate()),
  36       is_flushing_(false) {}
  37
  38 AudioBufferConverter::~AudioBufferConverter() {}
  39
  40 void AudioBufferConverter::AddInput(const scoped_refptr<AudioBuffer>& buffer) {
  41   // On EOS flush any remaining buffered data.
  42   if (buffer->end_of_stream()) {
  43     Flush();
  44     queued_outputs_.push_back(buffer);
  45     return;
  46   }
  47
  48   // We'll need a new |audio_converter_| if there was a config change.
  49   if (IsConfigChange(input_params_, buffer))
  50     ResetConverter(buffer);
  51
  52   // Pass straight through if there's no work to be done.
  53   if (!audio_converter_) {
  54     queued_outputs_.push_back(buffer);
  55     return;
  56   }
  57
  58   if (timestamp_helper_.base_timestamp() == kNoTimestamp())
  59     timestamp_helper_.SetBaseTimestamp(buffer->timestamp());
  60
  61   queued_inputs_.push_back(buffer);
  62   input_frames_ += buffer->frame_count();
  63
  64   ConvertIfPossible();
  65 }
  66
  67 bool AudioBufferConverter::HasNextBuffer() { return !queued_outputs_.empty(); }
  68
  69 scoped_refptr<AudioBuffer> AudioBufferConverter::GetNextBuffer() {
  70   DCHECK(!queued_outputs_.empty());
  71   scoped_refptr<AudioBuffer> out = queued_outputs_.front();
  72   queued_outputs_.pop_front();
  73   return out;
  74 }
  75
  76 void AudioBufferConverter::Reset() {
  77   audio_converter_.reset();
  78   queued_inputs_.clear();
  79   queued_outputs_.clear();
  80   timestamp_helper_.SetBaseTimestamp(kNoTimestamp());
  81   input_params_ = output_params_;
  82   input_frames_ = 0;
  83   buffered_input_frames_ = 0.0;
  84   last_input_buffer_offset_ = 0;
  85 }
  86
  87 void AudioBufferConverter::ResetTimestampState() {
  88   Flush();
  89   timestamp_helper_.SetBaseTimestamp(kNoTimestamp());
  90 }
  91
  92 double AudioBufferConverter::ProvideInput(AudioBus* audio_bus,
  93                                           base::TimeDelta buffer_delay) {
  94   DCHECK(is_flushing_ || input_frames_ >= audio_bus->frames());
  95
  96   int requested_frames_left = audio_bus->frames();
  97   int dest_index = 0;
  98
  99   while (requested_frames_left > 0 && !queued_inputs_.empty()) {
 100     scoped_refptr<AudioBuffer> input_buffer = queued_inputs_.front();
 101
 102     int frames_to_read =
 103         std::min(requested_frames_left,
 104                  input_buffer->frame_count() - last_input_buffer_offset_);
 105     input_buffer->ReadFrames(
 106         frames_to_read, last_input_buffer_offset_, dest_index, audio_bus);
 107     last_input_buffer_offset_ += frames_to_read;
 108
 109     if (last_input_buffer_offset_ == input_buffer->frame_count()) {
 110       // We've consumed all the frames in |input_buffer|.
 111       queued_inputs_.pop_front();
 112       last_input_buffer_offset_ = 0;
 113     }
 114
 115     requested_frames_left -= frames_to_read;
 116     dest_index += frames_to_read;
 117   }
 118
 119   // If we're flushing, zero any extra space, otherwise we should always have
 120   // enough data to completely fulfill the request.
 121   if (is_flushing_ && requested_frames_left > 0) {
 122     audio_bus->ZeroFramesPartial(audio_bus->frames() - requested_frames_left,
 123                                  requested_frames_left);
 124   } else {
 125     DCHECK_EQ(requested_frames_left, 0);
 126   }
 127
 128   input_frames_ -= audio_bus->frames() - requested_frames_left;
 129   DCHECK_GE(input_frames_, 0);
 130
 131   buffered_input_frames_ += audio_bus->frames() - requested_frames_left;
 132
 133   // Full volume.
 134   return 1.0;
 135 }
 136
 137 void AudioBufferConverter::ResetConverter(
 138     const scoped_refptr<AudioBuffer>& buffer) {
 139   Flush();
 140   audio_converter_.reset();
 141   input_params_.Reset(
 142       input_params_.format(),
 143       buffer->channel_layout(),
 144       buffer->channel_count(),
 145       0,
 146       buffer->sample_rate(),
 147       input_params_.bits_per_sample(),
 148       // If resampling is needed and the FIFO disabled, the AudioConverter will
 149       // always request SincResampler::kDefaultRequestSize frames.  Otherwise it
 150       // will use the output frame size.
 151       buffer->sample_rate() == output_params_.sample_rate()
 152           ? output_params_.frames_per_buffer()
 153           : SincResampler::kDefaultRequestSize);
 154
 155   io_sample_rate_ratio_ = static_cast<double>(input_params_.sample_rate()) /
 156                           output_params_.sample_rate();
 157
 158   // If |buffer| matches |output_params_| we don't need an AudioConverter at
 159   // all, and can early-out here.
 160   if (!IsConfigChange(output_params_, buffer))
 161     return;
 162
 163   // Note: The FIFO is disabled to avoid extraneous memcpy().
 164   audio_converter_.reset(
 165       new AudioConverter(input_params_, output_params_, true));
 166   audio_converter_->AddInput(this);
 167 }
 168
 169 void AudioBufferConverter::ConvertIfPossible() {
 170   DCHECK(audio_converter_);
 171
 172   int request_frames = 0;
 173
 174   if (is_flushing_) {
 175     // If we're flushing we want to convert *everything* even if this means
 176     // we'll have to pad some silence in ProvideInput().
 177     request_frames =
 178         ceil((buffered_input_frames_ + input_frames_) / io_sample_rate_ratio_);
 179   } else {
 180     // How many calls to ProvideInput() we can satisfy completely.
 181     int chunks = input_frames_ / input_params_.frames_per_buffer();
 182
 183     // How many output frames that corresponds to:
 184     request_frames = chunks * audio_converter_->ChunkSize();
 185   }
 186
 187   if (!request_frames)
 188     return;
 189
 190   scoped_refptr<AudioBuffer> output_buffer =
 191       AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
 192                                 output_params_.channel_layout(),
 193                                 output_params_.channels(),
 194                                 output_params_.sample_rate(),
 195                                 request_frames);
 196   scoped_ptr<AudioBus> output_bus =
 197       AudioBus::CreateWrapper(output_buffer->channel_count());
 198
 199   int frames_remaining = request_frames;
 200
 201   // The AudioConverter wants requests of a fixed size, so we'll slide an
 202   // AudioBus of that size across the |output_buffer|.
 203   while (frames_remaining != 0) {
 204     // It's important that this is a multiple of AudioBus::kChannelAlignment in
 205     // all requests except for the last, otherwise downstream SIMD optimizations
 206     // will crash on unaligned data.
 207     const int frames_this_iteration = std::min(
 208         static_cast<int>(SincResampler::kDefaultRequestSize), frames_remaining);
 209     const int offset_into_buffer =
 210         output_buffer->frame_count() - frames_remaining;
 211
 212     // Wrap the portion of the AudioBuffer in an AudioBus so the AudioConverter
 213     // can fill it.
 214     output_bus->set_frames(frames_this_iteration);
 215     for (int ch = 0; ch < output_buffer->channel_count(); ++ch) {
 216       output_bus->SetChannelData(
 217           ch,
 218           reinterpret_cast<float*>(output_buffer->channel_data()[ch]) +
 219               offset_into_buffer);
 220     }
 221
 222     // Do the actual conversion.
 223     audio_converter_->Convert(output_bus.get());
 224     frames_remaining -= frames_this_iteration;
 225     buffered_input_frames_ -= frames_this_iteration * io_sample_rate_ratio_;
 226   }
 227
 228   // Compute the timestamp.
 229   output_buffer->set_timestamp(timestamp_helper_.GetTimestamp());
 230   timestamp_helper_.AddFrames(request_frames);
 231
 232   queued_outputs_.push_back(output_buffer);
 233 }
 234
 235 void AudioBufferConverter::Flush() {
 236   if (!audio_converter_)
 237     return;
 238   is_flushing_ = true;
 239   ConvertIfPossible();
 240   is_flushing_ = false;
 241   audio_converter_->Reset();
 242   DCHECK_EQ(input_frames_, 0);
 243   DCHECK_EQ(last_input_buffer_offset_, 0);
 244   DCHECK_LT(buffered_input_frames_, 1.0);
 245   DCHECK(queued_inputs_.empty());
 246   buffered_input_frames_ = 0.0;
 247 }
 248
 249 }  // namespace media