media/base/audio_converter.cc

   1 // Copyright (c) 2012 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 // AudioConverter implementation.  Uses MultiChannelSincResampler for resampling
   6 // audio, ChannelMixer for channel mixing, and AudioPullFifo for buffering.
   7 //
   8 // Delay estimates are provided to InputCallbacks based on the frame delay
   9 // information reported via the resampler and FIFO units.
  10
  11 #include "media/base/audio_converter.h"
  12
  13 #include <algorithm>
  14
  15 #include "base/bind.h"
  16 #include "base/bind_helpers.h"
  17 #include "media/base/audio_bus.h"
  18 #include "media/base/audio_pull_fifo.h"
  19 #include "media/base/channel_mixer.h"
  20 #include "media/base/multi_channel_resampler.h"
  21 #include "media/base/vector_math.h"
  22
  23 namespace media {
  24
  25 AudioConverter::AudioConverter(const AudioParameters& input_params,
  26                                const AudioParameters& output_params,
  27                                bool disable_fifo)
  28     : chunk_size_(input_params.frames_per_buffer()),
  29       downmix_early_(false),
  30       resampler_frame_delay_(0),
  31       input_channel_count_(input_params.channels()) {
  32   CHECK(input_params.IsValid());
  33   CHECK(output_params.IsValid());
  34
  35   // Handle different input and output channel layouts.
  36   if (input_params.channel_layout() != output_params.channel_layout()) {
  37     DVLOG(1) << "Remixing channel layout from " << input_params.channel_layout()
  38              << " to " << output_params.channel_layout() << "; from "
  39              << input_params.channels() << " channels to "
  40              << output_params.channels() << " channels.";
  41     channel_mixer_.reset(new ChannelMixer(input_params, output_params));
  42
  43     // Pare off data as early as we can for efficiency.
  44     downmix_early_ = input_params.channels() > output_params.channels();
  45   }
  46
  47   // Only resample if necessary since it's expensive.
  48   if (input_params.sample_rate() != output_params.sample_rate()) {
  49     DVLOG(1) << "Resampling from " << input_params.sample_rate() << " to "
  50              << output_params.sample_rate();
  51     const int request_size = disable_fifo ? SincResampler::kDefaultRequestSize :
  52         input_params.frames_per_buffer();
  53     const double io_sample_rate_ratio =
  54         input_params.sample_rate() /
  55         static_cast<double>(output_params.sample_rate());
  56     resampler_.reset(new MultiChannelResampler(
  57         downmix_early_ ? output_params.channels() : input_params.channels(),
  58         io_sample_rate_ratio,
  59         request_size,
  60         base::Bind(&AudioConverter::ProvideInput, base::Unretained(this))));
  61   }
  62
  63   input_frame_duration_ = base::TimeDelta::FromMicroseconds(
  64       base::Time::kMicrosecondsPerSecond /
  65       static_cast<double>(input_params.sample_rate()));
  66   output_frame_duration_ = base::TimeDelta::FromMicroseconds(
  67       base::Time::kMicrosecondsPerSecond /
  68       static_cast<double>(output_params.sample_rate()));
  69
  70   // The resampler can be configured to work with a specific request size, so a
  71   // FIFO is not necessary when resampling.
  72   if (disable_fifo || resampler_)
  73     return;
  74
  75   // Since the output device may want a different buffer size than the caller
  76   // asked for, we need to use a FIFO to ensure that both sides read in chunk
  77   // sizes they're configured for.
  78   if (input_params.frames_per_buffer() != output_params.frames_per_buffer()) {
  79     DVLOG(1) << "Rebuffering from " << input_params.frames_per_buffer()
  80              << " to " << output_params.frames_per_buffer();
  81     chunk_size_ = input_params.frames_per_buffer();
  82     audio_fifo_.reset(new AudioPullFifo(
  83         downmix_early_ ? output_params.channels() : input_params.channels(),
  84         chunk_size_,
  85         base::Bind(&AudioConverter::SourceCallback, base::Unretained(this))));
  86   }
  87 }
  88
  89 AudioConverter::~AudioConverter() {}
  90
  91 void AudioConverter::AddInput(InputCallback* input) {
  92   DCHECK(std::find(transform_inputs_.begin(), transform_inputs_.end(), input) ==
  93          transform_inputs_.end());
  94   transform_inputs_.push_back(input);
  95 }
  96
  97 void AudioConverter::RemoveInput(InputCallback* input) {
  98   DCHECK(std::find(transform_inputs_.begin(), transform_inputs_.end(), input) !=
  99          transform_inputs_.end());
 100   transform_inputs_.remove(input);
 101
 102   if (transform_inputs_.empty())
 103     Reset();
 104 }
 105
 106 void AudioConverter::Reset() {
 107   if (audio_fifo_)
 108     audio_fifo_->Clear();
 109   if (resampler_)
 110     resampler_->Flush();
 111 }
 112
 113 int AudioConverter::ChunkSize() const {
 114   if (!resampler_)
 115     return chunk_size_;
 116   return resampler_->ChunkSize();
 117 }
 118
 119 void AudioConverter::ConvertWithDelay(const base::TimeDelta& initial_delay,
 120                                       AudioBus* dest) {
 121   initial_delay_ = initial_delay;
 122
 123   if (transform_inputs_.empty()) {
 124     dest->Zero();
 125     return;
 126   }
 127
 128   // Determine if channel mixing should be done and if it should be done before
 129   // or after resampling.  If it's possible to reduce the channel count prior to
 130   // resampling we can save a lot of processing time.  Vice versa, we don't want
 131   // to increase the channel count prior to resampling for the same reason.
 132   bool needs_mixing = channel_mixer_ && !downmix_early_;
 133
 134   if (needs_mixing)
 135     CreateUnmixedAudioIfNecessary(dest->frames());
 136
 137   AudioBus* temp_dest = needs_mixing ? unmixed_audio_.get() : dest;
 138   DCHECK(temp_dest);
 139
 140   // Figure out which method to call based on whether we're resampling and
 141   // rebuffering, just resampling, or just mixing.  We want to avoid any extra
 142   // steps when possible since we may be converting audio data in real time.
 143   if (!resampler_ && !audio_fifo_) {
 144     SourceCallback(0, temp_dest);
 145   } else {
 146     if (resampler_)
 147       resampler_->Resample(temp_dest->frames(), temp_dest);
 148     else
 149       ProvideInput(0, temp_dest);
 150   }
 151
 152   // Finally upmix the channels if we didn't do so earlier.
 153   if (needs_mixing) {
 154     DCHECK_EQ(temp_dest->frames(), dest->frames());
 155     channel_mixer_->Transform(temp_dest, dest);
 156   }
 157 }
 158
 159 void AudioConverter::Convert(AudioBus* dest) {
 160   ConvertWithDelay(base::TimeDelta::FromMilliseconds(0), dest);
 161 }
 162
 163 void AudioConverter::SourceCallback(int fifo_frame_delay, AudioBus* dest) {
 164   const bool needs_downmix = channel_mixer_ && downmix_early_;
 165
 166   if (!mixer_input_audio_bus_ ||
 167       mixer_input_audio_bus_->frames() != dest->frames()) {
 168     mixer_input_audio_bus_ =
 169         AudioBus::Create(input_channel_count_, dest->frames());
 170   }
 171
 172   // If we're downmixing early we need a temporary AudioBus which matches
 173   // the the input channel count and input frame size since we're passing
 174   // |unmixed_audio_| directly to the |source_callback_|.
 175   if (needs_downmix)
 176     CreateUnmixedAudioIfNecessary(dest->frames());
 177
 178   AudioBus* const temp_dest = needs_downmix ? unmixed_audio_.get() : dest;
 179
 180   // Sanity check our inputs.
 181   DCHECK_EQ(temp_dest->frames(), mixer_input_audio_bus_->frames());
 182   DCHECK_EQ(temp_dest->channels(), mixer_input_audio_bus_->channels());
 183
 184   // Calculate the buffer delay for this callback.
 185   base::TimeDelta buffer_delay = initial_delay_;
 186   if (resampler_) {
 187     buffer_delay += base::TimeDelta::FromMicroseconds(
 188         resampler_frame_delay_ * output_frame_duration_.InMicroseconds());
 189   }
 190   if (audio_fifo_) {
 191     buffer_delay += base::TimeDelta::FromMicroseconds(
 192         fifo_frame_delay * input_frame_duration_.InMicroseconds());
 193   }
 194
 195   // If we only have a single input, avoid an extra copy.
 196   AudioBus* const provide_input_dest =
 197       transform_inputs_.size() == 1 ? temp_dest : mixer_input_audio_bus_.get();
 198
 199   // Have each mixer render its data into an output buffer then mix the result.
 200   for (InputCallbackSet::iterator it = transform_inputs_.begin();
 201        it != transform_inputs_.end(); ++it) {
 202     InputCallback* input = *it;
 203
 204     const float volume = input->ProvideInput(provide_input_dest, buffer_delay);
 205
 206     // Optimize the most common single input, full volume case.
 207     if (it == transform_inputs_.begin()) {
 208       if (volume == 1.0f) {
 209         if (temp_dest != provide_input_dest)
 210           provide_input_dest->CopyTo(temp_dest);
 211       } else if (volume > 0) {
 212         for (int i = 0; i < provide_input_dest->channels(); ++i) {
 213           vector_math::FMUL(
 214               provide_input_dest->channel(i), volume,
 215               provide_input_dest->frames(), temp_dest->channel(i));
 216         }
 217       } else {
 218         // Zero |temp_dest| otherwise, so we're mixing into a clean buffer.
 219         temp_dest->Zero();
 220       }
 221
 222       continue;
 223     }
 224
 225     // Volume adjust and mix each mixer input into |temp_dest| after rendering.
 226     if (volume > 0) {
 227       for (int i = 0; i < mixer_input_audio_bus_->channels(); ++i) {
 228         vector_math::FMAC(
 229             mixer_input_audio_bus_->channel(i), volume,
 230             mixer_input_audio_bus_->frames(), temp_dest->channel(i));
 231       }
 232     }
 233   }
 234
 235   if (needs_downmix) {
 236     DCHECK_EQ(temp_dest->frames(), dest->frames());
 237     channel_mixer_->Transform(temp_dest, dest);
 238   }
 239 }
 240
 241 void AudioConverter::ProvideInput(int resampler_frame_delay, AudioBus* dest) {
 242   resampler_frame_delay_ = resampler_frame_delay;
 243   if (audio_fifo_)
 244     audio_fifo_->Consume(dest, dest->frames());
 245   else
 246     SourceCallback(0, dest);
 247 }
 248
 249 void AudioConverter::CreateUnmixedAudioIfNecessary(int frames) {
 250   if (!unmixed_audio_ || unmixed_audio_->frames() != frames)
 251     unmixed_audio_ = AudioBus::Create(input_channel_count_, frames);
 252 }
 253
 254 }  // namespace media