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