modules/audio_processing/audio_buffer.cc

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "modules/audio_processing/audio_buffer.h"

#include <string.h>

#include <cstdint>

#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "modules/audio_processing/splitting_filter.h"
#include "rtc_base/checks.h"

namespace webrtc {
namespace {

const size_t kSamplesPer16kHzChannel = 160;
const size_t kSamplesPer32kHzChannel = 320;
const size_t kSamplesPer48kHzChannel = 480;

size_t NumBandsFromSamplesPerChannel(size_t num_frames) {
  size_t num_bands = 1;
  if (num_frames == kSamplesPer32kHzChannel ||
      num_frames == kSamplesPer48kHzChannel) {
    num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel);
  }
  return num_bands;
}

}  // namespace

AudioBuffer::AudioBuffer(size_t input_num_frames,
                         size_t num_input_channels,
                         size_t process_num_frames,
                         size_t num_process_channels,
                         size_t output_num_frames)
    : input_num_frames_(input_num_frames),
      num_input_channels_(num_input_channels),
      proc_num_frames_(process_num_frames),
      num_proc_channels_(num_process_channels),
      output_num_frames_(output_num_frames),
      num_channels_(num_process_channels),
      num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
      num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
      data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)),
      output_buffer_(new IFChannelBuffer(output_num_frames_, num_channels_)) {
  RTC_DCHECK_GT(input_num_frames_, 0);
  RTC_DCHECK_GT(proc_num_frames_, 0);
  RTC_DCHECK_GT(output_num_frames_, 0);
  RTC_DCHECK_GT(num_input_channels_, 0);
  RTC_DCHECK_GT(num_proc_channels_, 0);
  RTC_DCHECK_LE(num_proc_channels_, num_input_channels_);

  if (input_num_frames_ != proc_num_frames_ ||
      output_num_frames_ != proc_num_frames_) {
    // Create an intermediate buffer for resampling.
    process_buffer_.reset(
        new ChannelBuffer<float>(proc_num_frames_, num_proc_channels_));

    if (input_num_frames_ != proc_num_frames_) {
      for (size_t i = 0; i < num_proc_channels_; ++i) {
        input_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
            new PushSincResampler(input_num_frames_, proc_num_frames_)));
      }
    }

    if (output_num_frames_ != proc_num_frames_) {
      for (size_t i = 0; i < num_proc_channels_; ++i) {
        output_resamplers_.push_back(std::unique_ptr<PushSincResampler>(
            new PushSincResampler(proc_num_frames_, output_num_frames_)));
      }
    }
  }

  if (num_bands_ > 1) {
    split_data_.reset(
        new IFChannelBuffer(proc_num_frames_, num_proc_channels_, num_bands_));
    splitting_filter_.reset(
        new SplittingFilter(num_proc_channels_, num_bands_, proc_num_frames_));
  }
}

AudioBuffer::~AudioBuffer() {}

void AudioBuffer::CopyFrom(const float* const* data,
                           const StreamConfig& stream_config) {
  RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_);
  RTC_DCHECK_EQ(stream_config.num_channels(), num_input_channels_);
  InitForNewData();
  // Initialized lazily because there's a different condition in
  // DeinterleaveFrom.
  const bool need_to_downmix =
      num_input_channels_ > 1 && num_proc_channels_ == 1;
  if (need_to_downmix && !input_buffer_) {
    input_buffer_.reset(
        new IFChannelBuffer(input_num_frames_, num_proc_channels_));
  }

  // Downmix.
  const float* const* data_ptr = data;
  if (need_to_downmix) {
    DownmixToMono<float, float>(data, input_num_frames_, num_input_channels_,
                                input_buffer_->fbuf()->channels()[0]);
    data_ptr = input_buffer_->fbuf_const()->channels();
  }

  // Resample.
  if (input_num_frames_ != proc_num_frames_) {
    for (size_t i = 0; i < num_proc_channels_; ++i) {
      input_resamplers_[i]->Resample(data_ptr[i], input_num_frames_,
                                     process_buffer_->channels()[i],
                                     proc_num_frames_);
    }
    data_ptr = process_buffer_->channels();
  }

  // Convert to the S16 range.
  for (size_t i = 0; i < num_proc_channels_; ++i) {
    FloatToFloatS16(data_ptr[i], proc_num_frames_,
                    data_->fbuf()->channels()[i]);
  }
}

void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                         float* const* data) {
  RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_);
  RTC_DCHECK(stream_config.num_channels() == num_channels_ ||
             num_channels_ == 1);

  // Convert to the float range.
  float* const* data_ptr = data;
  if (output_num_frames_ != proc_num_frames_) {
    // Convert to an intermediate buffer for subsequent resampling.
    data_ptr = process_buffer_->channels();
  }
  for (size_t i = 0; i < num_channels_; ++i) {
    FloatS16ToFloat(data_->fbuf()->channels()[i], proc_num_frames_,
                    data_ptr[i]);
  }

  // Resample.
  if (output_num_frames_ != proc_num_frames_) {
    for (size_t i = 0; i < num_channels_; ++i) {
      output_resamplers_[i]->Resample(data_ptr[i], proc_num_frames_, data[i],
                                      output_num_frames_);
    }
  }

  // Upmix.
  for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
    memcpy(data[i], data[0], output_num_frames_ * sizeof(**data));
  }
}

void AudioBuffer::InitForNewData() {
  num_channels_ = num_proc_channels_;
  data_->set_num_channels(num_proc_channels_);
  if (split_data_.get()) {
    split_data_->set_num_channels(num_proc_channels_);
  }
}

const int16_t* const* AudioBuffer::channels_const() const {
  return data_->ibuf_const()->channels();
}

int16_t* const* AudioBuffer::channels() {
  return data_->ibuf()->channels();
}

const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
  return split_data_.get() ? split_data_->ibuf_const()->bands(channel)
                           : data_->ibuf_const()->bands(channel);
}

int16_t* const* AudioBuffer::split_bands(size_t channel) {
  return split_data_.get() ? split_data_->ibuf()->bands(channel)
                           : data_->ibuf()->bands(channel);
}

const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
  if (split_data_.get()) {
    return split_data_->ibuf_const()->channels(band);
  } else {
    return band == kBand0To8kHz ? data_->ibuf_const()->channels() : nullptr;
  }
}

const float* const* AudioBuffer::channels_const_f() const {
  return data_->fbuf_const()->channels();
}

float* const* AudioBuffer::channels_f() {
  return data_->fbuf()->channels();
}

const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
  return split_data_.get() ? split_data_->fbuf_const()->bands(channel)
                           : data_->fbuf_const()->bands(channel);
}

float* const* AudioBuffer::split_bands_f(size_t channel) {
  return split_data_.get() ? split_data_->fbuf()->bands(channel)
                           : data_->fbuf()->bands(channel);
}

size_t AudioBuffer::num_channels() const {
  return num_channels_;
}

void AudioBuffer::set_num_channels(size_t num_channels) {
  num_channels_ = num_channels;
  data_->set_num_channels(num_channels);
  if (split_data_.get()) {
    split_data_->set_num_channels(num_channels);
  }
}

size_t AudioBuffer::num_frames() const {
  return proc_num_frames_;
}

size_t AudioBuffer::num_frames_per_band() const {
  return num_split_frames_;
}

size_t AudioBuffer::num_bands() const {
  return num_bands_;
}

// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
void AudioBuffer::DeinterleaveFrom(const AudioFrame* frame) {
  RTC_DCHECK_EQ(frame->num_channels_, num_input_channels_);
  RTC_DCHECK_EQ(frame->samples_per_channel_, input_num_frames_);
  InitForNewData();
  // Initialized lazily because there's a different condition in CopyFrom.
  if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
    input_buffer_.reset(
        new IFChannelBuffer(input_num_frames_, num_proc_channels_));
  }

  int16_t* const* deinterleaved;
  if (input_num_frames_ == proc_num_frames_) {
    deinterleaved = data_->ibuf()->channels();
  } else {
    deinterleaved = input_buffer_->ibuf()->channels();
  }
  // TODO(yujo): handle muted frames more efficiently.
  if (num_proc_channels_ == 1) {
    // Downmix and deinterleave simultaneously.
    DownmixInterleavedToMono(frame->data(), input_num_frames_,
                             num_input_channels_, deinterleaved[0]);
  } else {
    RTC_DCHECK_EQ(num_proc_channels_, num_input_channels_);
    Deinterleave(frame->data(), input_num_frames_, num_proc_channels_,
                 deinterleaved);
  }

  // Resample.
  if (input_num_frames_ != proc_num_frames_) {
    for (size_t i = 0; i < num_proc_channels_; ++i) {
      input_resamplers_[i]->Resample(
          input_buffer_->fbuf_const()->channels()[i], input_num_frames_,
          data_->fbuf()->channels()[i], proc_num_frames_);
    }
  }
}

void AudioBuffer::InterleaveTo(AudioFrame* frame) const {
  RTC_DCHECK(frame->num_channels_ == num_channels_ || num_channels_ == 1);
  RTC_DCHECK_EQ(frame->samples_per_channel_, output_num_frames_);

  // Resample if necessary.
  IFChannelBuffer* data_ptr = data_.get();
  if (proc_num_frames_ != output_num_frames_) {
    for (size_t i = 0; i < num_channels_; ++i) {
      output_resamplers_[i]->Resample(
          data_->fbuf()->channels()[i], proc_num_frames_,
          output_buffer_->fbuf()->channels()[i], output_num_frames_);
    }
    data_ptr = output_buffer_.get();
  }

  // TODO(yujo): handle muted frames more efficiently.
  if (frame->num_channels_ == num_channels_) {
    Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
               frame->mutable_data());
  } else {
    UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
                           frame->num_channels_, frame->mutable_data());
  }
}

void AudioBuffer::SplitIntoFrequencyBands() {
  splitting_filter_->Analysis(data_.get(), split_data_.get());
}

void AudioBuffer::MergeFrequencyBands() {
  splitting_filter_->Synthesis(split_data_.get(), data_.get());
}

}  // namespace webrtc