src/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       buffer->sample_rate(),
 146       input_params_.bits_per_sample(),
 147       // If resampling is needed and the FIFO disabled, the AudioConverter will
 148       // always request SincResampler::kDefaultRequestSize frames.  Otherwise it
 149       // will use the output frame size.
 150       buffer->sample_rate() == output_params_.sample_rate()
 151           ? output_params_.frames_per_buffer()
 152           : SincResampler::kDefaultRequestSize);
 153
 154   io_sample_rate_ratio_ = static_cast<double>(input_params_.sample_rate()) /
 155                           output_params_.sample_rate();
 156
 157   // If |buffer| matches |output_params_| we don't need an AudioConverter at
 158   // all, and can early-out here.
 159   if (!IsConfigChange(output_params_, buffer))
 160     return;
 161
 162   // Note: The FIFO is disabled to avoid extraneous memcpy().
 163   audio_converter_.reset(
 164       new AudioConverter(input_params_, output_params_, true));
 165   audio_converter_->AddInput(this);
 166 }
 167
 168 void AudioBufferConverter::ConvertIfPossible() {
 169   DCHECK(audio_converter_);
 170
 171   int request_frames = 0;
 172
 173   if (is_flushing_) {
 174     // If we're flushing we want to convert *everything* even if this means
 175     // we'll have to pad some silence in ProvideInput().
 176     request_frames =
 177         ceil((buffered_input_frames_ + input_frames_) / io_sample_rate_ratio_);
 178   } else {
 179     // How many calls to ProvideInput() we can satisfy completely.
 180     int chunks = input_frames_ / input_params_.frames_per_buffer();
 181
 182     // How many output frames that corresponds to:
 183     request_frames = chunks * audio_converter_->ChunkSize();
 184   }
 185
 186   if (!request_frames)
 187     return;
 188
 189   scoped_refptr<AudioBuffer> output_buffer =
 190       AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
 191                                 output_params_.channel_layout(),
 192                                 output_params_.channels(),
 193                                 output_params_.sample_rate(),
 194                                 request_frames);
 195   scoped_ptr<AudioBus> output_bus =
 196       AudioBus::CreateWrapper(output_buffer->channel_count());
 197
 198   int frames_remaining = request_frames;
 199
 200   // The AudioConverter wants requests of a fixed size, so we'll slide an
 201   // AudioBus of that size across the |output_buffer|.
 202   while (frames_remaining != 0) {
 203     // It's important that this is a multiple of AudioBus::kChannelAlignment in
 204     // all requests except for the last, otherwise downstream SIMD optimizations
 205     // will crash on unaligned data.
 206     const int frames_this_iteration = std::min(
 207         static_cast<int>(SincResampler::kDefaultRequestSize), frames_remaining);
 208     const int offset_into_buffer =
 209         output_buffer->frame_count() - frames_remaining;
 210
 211     // Wrap the portion of the AudioBuffer in an AudioBus so the AudioConverter
 212     // can fill it.
 213     output_bus->set_frames(frames_this_iteration);
 214     for (int ch = 0; ch < output_buffer->channel_count(); ++ch) {
 215       output_bus->SetChannelData(
 216           ch,
 217           reinterpret_cast<float*>(output_buffer->channel_data()[ch]) +
 218               offset_into_buffer);
 219     }
 220
 221     // Do the actual conversion.
 222     audio_converter_->Convert(output_bus.get());
 223     frames_remaining -= frames_this_iteration;
 224     buffered_input_frames_ -= frames_this_iteration * io_sample_rate_ratio_;
 225   }
 226
 227   // Compute the timestamp.
 228   output_buffer->set_timestamp(timestamp_helper_.GetTimestamp());
 229   timestamp_helper_.AddFrames(request_frames);
 230
 231   queued_outputs_.push_back(output_buffer);
 232 }
 233
 234 void AudioBufferConverter::Flush() {
 235   if (!audio_converter_)
 236     return;
 237   is_flushing_ = true;
 238   ConvertIfPossible();
 239   is_flushing_ = false;
 240   audio_converter_->Reset();
 241   DCHECK_EQ(input_frames_, 0);
 242   DCHECK_EQ(last_input_buffer_offset_, 0);
 243   DCHECK_LT(buffered_input_frames_, 1.0);
 244   DCHECK(queued_inputs_.empty());
 245   buffered_input_frames_ = 0.0;
 246 }
 247
 248 }  // namespace media