webrtc/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 "webrtc/modules/audio_processing/audio_buffer.h"

#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"

namespace webrtc {
namespace {

enum {
  kSamplesPer8kHzChannel = 80,
  kSamplesPer16kHzChannel = 160,
  kSamplesPer32kHzChannel = 320
};

void StereoToMono(const float* left, const float* right, float* out,
                  int samples_per_channel) {
  for (int i = 0; i < samples_per_channel; ++i) {
    out[i] = (left[i] + right[i]) / 2;
  }
}

void StereoToMono(const int16_t* left, const int16_t* right, int16_t* out,
                  int samples_per_channel) {
  for (int i = 0; i < samples_per_channel; i++)
    out[i] = (left[i] + right[i]) >> 1;
}

}  // namespace

class SplitChannelBuffer {
 public:
  SplitChannelBuffer(int samples_per_split_channel, int num_channels)
      : low_(samples_per_split_channel, num_channels),
        high_(samples_per_split_channel, num_channels) {
  }
  ~SplitChannelBuffer() {}

  int16_t* low_channel(int i) { return low_.channel(i); }
  int16_t* high_channel(int i) { return high_.channel(i); }

 private:
  ChannelBuffer<int16_t> low_;
  ChannelBuffer<int16_t> high_;
};

AudioBuffer::AudioBuffer(int input_samples_per_channel,
                         int num_input_channels,
                         int process_samples_per_channel,
                         int num_process_channels,
                         int output_samples_per_channel)
  : input_samples_per_channel_(input_samples_per_channel),
    num_input_channels_(num_input_channels),
    proc_samples_per_channel_(process_samples_per_channel),
    num_proc_channels_(num_process_channels),
    output_samples_per_channel_(output_samples_per_channel),
    samples_per_split_channel_(proc_samples_per_channel_),
    num_mixed_channels_(0),
    num_mixed_low_pass_channels_(0),
    data_was_mixed_(false),
    reference_copied_(false),
    activity_(AudioFrame::kVadUnknown),
    is_muted_(false),
    data_(NULL),
    channels_(new ChannelBuffer<int16_t>(proc_samples_per_channel_,
                                         num_proc_channels_)) {
  assert(input_samples_per_channel_ > 0);
  assert(proc_samples_per_channel_ > 0);
  assert(output_samples_per_channel_ > 0);
  assert(num_input_channels_ > 0 && num_input_channels_ <= 2);
  assert(num_proc_channels_ <= num_input_channels);

  if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
    input_buffer_.reset(new ChannelBuffer<float>(input_samples_per_channel_,
                                                 num_proc_channels_));
  }

  if (input_samples_per_channel_ != proc_samples_per_channel_ ||
      output_samples_per_channel_ != proc_samples_per_channel_) {
    // Create an intermediate buffer for resampling.
    process_buffer_.reset(new ChannelBuffer<float>(proc_samples_per_channel_,
                                                   num_proc_channels_));
  }

  if (input_samples_per_channel_ != proc_samples_per_channel_) {
    input_resamplers_.reserve(num_proc_channels_);
    for (int i = 0; i < num_proc_channels_; ++i) {
      input_resamplers_.push_back(
          new PushSincResampler(input_samples_per_channel_,
                                proc_samples_per_channel_));
    }
  }

  if (output_samples_per_channel_ != proc_samples_per_channel_) {
    output_resamplers_.reserve(num_proc_channels_);
    for (int i = 0; i < num_proc_channels_; ++i) {
      output_resamplers_.push_back(
          new PushSincResampler(proc_samples_per_channel_,
                                output_samples_per_channel_));
    }
  }

  if (proc_samples_per_channel_ == kSamplesPer32kHzChannel) {
    samples_per_split_channel_ = kSamplesPer16kHzChannel;
    split_channels_.reset(new SplitChannelBuffer(samples_per_split_channel_,
                                                 num_proc_channels_));
    filter_states_.reset(new SplitFilterStates[num_proc_channels_]);
  }
}

void AudioBuffer::CopyFrom(const float* const* data,
                           int samples_per_channel,
                           AudioProcessing::ChannelLayout layout) {
  assert(samples_per_channel == input_samples_per_channel_);
  assert(ChannelsFromLayout(layout) == num_input_channels_);
  InitForNewData();

  // Downmix.
  const float* const* data_ptr = data;
  if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
    StereoToMono(data[0],
                 data[1],
                 input_buffer_->channel(0),
                 input_samples_per_channel_);
    data_ptr = input_buffer_->channels();
  }

  // Resample.
  if (input_samples_per_channel_ != proc_samples_per_channel_) {
    for (int i = 0; i < num_proc_channels_; ++i) {
      input_resamplers_[i]->Resample(data_ptr[i],
                                     input_samples_per_channel_,
                                     process_buffer_->channel(i),
                                     proc_samples_per_channel_);
    }
    data_ptr = process_buffer_->channels();
  }

  // Convert to int16.
  for (int i = 0; i < num_proc_channels_; ++i) {
    ScaleAndRoundToInt16(data_ptr[i], proc_samples_per_channel_,
                         channels_->channel(i));
  }
}

void AudioBuffer::CopyTo(int samples_per_channel,
                         AudioProcessing::ChannelLayout layout,
                         float* const* data) {
  assert(samples_per_channel == output_samples_per_channel_);
  assert(ChannelsFromLayout(layout) == num_proc_channels_);

  // Convert to float.
  float* const* data_ptr = data;
  if (output_samples_per_channel_ != proc_samples_per_channel_) {
    // Convert to an intermediate buffer for subsequent resampling.
    data_ptr = process_buffer_->channels();
  }
  for (int i = 0; i < num_proc_channels_; ++i) {
    ScaleToFloat(channels_->channel(i), proc_samples_per_channel_, data_ptr[i]);
  }

  // Resample.
  if (output_samples_per_channel_ != proc_samples_per_channel_) {
    for (int i = 0; i < num_proc_channels_; ++i) {
      output_resamplers_[i]->Resample(data_ptr[i],
                                      proc_samples_per_channel_,
                                      data[i],
                                      output_samples_per_channel_);
    }
  }
}

AudioBuffer::~AudioBuffer() {}

void AudioBuffer::InitForNewData() {
  data_ = NULL;
  data_was_mixed_ = false;
  num_mixed_channels_ = 0;
  num_mixed_low_pass_channels_ = 0;
  reference_copied_ = false;
  activity_ = AudioFrame::kVadUnknown;
  is_muted_ = false;
}

int16_t* AudioBuffer::data(int channel) const {
  assert(channel >= 0 && channel < num_proc_channels_);
  if (data_ != NULL) {
    return data_;
  }

  return channels_->channel(channel);
}

int16_t* AudioBuffer::low_pass_split_data(int channel) const {
  assert(channel >= 0 && channel < num_proc_channels_);
  if (split_channels_.get() == NULL) {
    return data(channel);
  }

  return split_channels_->low_channel(channel);
}

int16_t* AudioBuffer::high_pass_split_data(int channel) const {
  assert(channel >= 0 && channel < num_proc_channels_);
  if (split_channels_.get() == NULL) {
    return NULL;
  }

  return split_channels_->high_channel(channel);
}

int16_t* AudioBuffer::mixed_data(int channel) const {
  assert(channel >= 0 && channel < num_mixed_channels_);

  return mixed_channels_->channel(channel);
}

int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
  assert(channel >= 0 && channel < num_mixed_low_pass_channels_);

  return mixed_low_pass_channels_->channel(channel);
}

int16_t* AudioBuffer::low_pass_reference(int channel) const {
  assert(channel >= 0 && channel < num_proc_channels_);
  if (!reference_copied_) {
    return NULL;
  }

  return low_pass_reference_channels_->channel(channel);
}

SplitFilterStates* AudioBuffer::filter_states(int channel) const {
  assert(channel >= 0 && channel < num_proc_channels_);
  return &filter_states_[channel];
}

void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
  activity_ = activity;
}

AudioFrame::VADActivity AudioBuffer::activity() const {
  return activity_;
}

bool AudioBuffer::is_muted() const {
  return is_muted_;
}

int AudioBuffer::num_channels() const {
  return num_proc_channels_;
}

int AudioBuffer::samples_per_channel() const {
  return proc_samples_per_channel_;
}

int AudioBuffer::samples_per_split_channel() const {
  return samples_per_split_channel_;
}

// TODO(andrew): Do deinterleaving and mixing in one step?
void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
  assert(proc_samples_per_channel_ == input_samples_per_channel_);
  assert(num_proc_channels_ == num_input_channels_);
  assert(frame->num_channels_ == num_proc_channels_);
  assert(frame->samples_per_channel_ ==  proc_samples_per_channel_);
  InitForNewData();
  activity_ = frame->vad_activity_;
  if (frame->energy_ == 0) {
    is_muted_ = true;
  }

  if (num_proc_channels_ == 1) {
    // We can get away with a pointer assignment in this case.
    data_ = frame->data_;
    return;
  }

  int16_t* interleaved = frame->data_;
  for (int i = 0; i < num_proc_channels_; i++) {
    int16_t* deinterleaved = channels_->channel(i);
    int interleaved_idx = i;
    for (int j = 0; j < proc_samples_per_channel_; j++) {
      deinterleaved[j] = interleaved[interleaved_idx];
      interleaved_idx += num_proc_channels_;
    }
  }
}

void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
  assert(proc_samples_per_channel_ == output_samples_per_channel_);
  assert(num_proc_channels_ == num_input_channels_);
  assert(frame->num_channels_ == num_proc_channels_);
  assert(frame->samples_per_channel_ == proc_samples_per_channel_);
  frame->vad_activity_ = activity_;

  if (!data_changed) {
    return;
  }

  if (num_proc_channels_ == 1) {
    if (data_was_mixed_) {
      memcpy(frame->data_,
             channels_->channel(0),
             sizeof(int16_t) * proc_samples_per_channel_);
    } else {
      // These should point to the same buffer in this case.
      assert(data_ == frame->data_);
    }

    return;
  }

  int16_t* interleaved = frame->data_;
  for (int i = 0; i < num_proc_channels_; i++) {
    int16_t* deinterleaved = channels_->channel(i);
    int interleaved_idx = i;
    for (int j = 0; j < proc_samples_per_channel_; j++) {
      interleaved[interleaved_idx] = deinterleaved[j];
      interleaved_idx += num_proc_channels_;
    }
  }
}

void AudioBuffer::CopyAndMix(int num_mixed_channels) {
  // We currently only support the stereo to mono case.
  assert(num_proc_channels_ == 2);
  assert(num_mixed_channels == 1);
  if (!mixed_channels_.get()) {
    mixed_channels_.reset(
        new ChannelBuffer<int16_t>(proc_samples_per_channel_,
                                   num_mixed_channels));
  }

  StereoToMono(channels_->channel(0),
               channels_->channel(1),
               mixed_channels_->channel(0),
               proc_samples_per_channel_);

  num_mixed_channels_ = num_mixed_channels;
}

void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) {
  // We currently only support the stereo to mono case.
  assert(num_proc_channels_ == 2);
  assert(num_mixed_channels == 1);
  if (!mixed_low_pass_channels_.get()) {
    mixed_low_pass_channels_.reset(
        new ChannelBuffer<int16_t>(samples_per_split_channel_,
                                   num_mixed_channels));
  }

  StereoToMono(low_pass_split_data(0),
               low_pass_split_data(1),
               mixed_low_pass_channels_->channel(0),
               samples_per_split_channel_);

  num_mixed_low_pass_channels_ = num_mixed_channels;
}

void AudioBuffer::CopyLowPassToReference() {
  reference_copied_ = true;
  if (!low_pass_reference_channels_.get()) {
    low_pass_reference_channels_.reset(
        new ChannelBuffer<int16_t>(samples_per_split_channel_,
                                   num_proc_channels_));
  }
  for (int i = 0; i < num_proc_channels_; i++) {
    low_pass_reference_channels_->CopyFrom(low_pass_split_data(i), i);
  }
}

}  // namespace webrtc