modules/audio_coding/neteq/sync_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_coding/neteq/sync_buffer.h"

#include <algorithm>  // Access to min.
#include <cstddef>
#include <cstdint>

#include "api/audio/audio_view.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"

namespace webrtc {

size_t SyncBuffer::FutureLength() const {
  return Size() - next_index_;
}

void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
  size_t samples_added = append_this.Size();
  AudioMultiVector::PushBack(append_this);
  AudioMultiVector::PopFront(samples_added);
  if (samples_added <= next_index_) {
    next_index_ -= samples_added;
  } else {
    // This means that we are pushing out future data that was never used.
    //    RTC_DCHECK_NOTREACHED();
    // TODO(hlundin): This assert must be disabled to support 60 ms frames.
    // This should not happen even for 60 ms frames, but it does. Investigate
    // why.
    next_index_ = 0;
  }
  dtmf_index_ -= std::min(dtmf_index_, samples_added);
}

void SyncBuffer::PushBackInterleaved(const BufferT<int16_t>& append_this) {
  const size_t size_before_adding = Size();
  AudioMultiVector::PushBackInterleaved(append_this);
  const size_t samples_added_per_channel = Size() - size_before_adding;
  RTC_DCHECK_EQ(samples_added_per_channel * Channels(), append_this.size());
  AudioMultiVector::PopFront(samples_added_per_channel);
  next_index_ -= std::min(next_index_, samples_added_per_channel);
  dtmf_index_ -= std::min(dtmf_index_, samples_added_per_channel);
}

void SyncBuffer::PushFrontZeros(size_t length) {
  InsertZerosAtIndex(length, 0);
}

void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
  position = std::min(position, Size());
  length = std::min(length, Size() - position);
  AudioMultiVector::PopBack(length);
  for (size_t channel = 0; channel < Channels(); ++channel) {
    channels_[channel]->InsertZerosAt(length, position);
  }
  if (next_index_ >= position) {
    // We are moving the `next_index_` sample.
    set_next_index(next_index_ + length);  // Overflow handled by subfunction.
  }
  if (dtmf_index_ > 0 && dtmf_index_ >= position) {
    // We are moving the `dtmf_index_` sample.
    set_dtmf_index(dtmf_index_ + length);  // Overflow handled by subfunction.
  }
}

void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
                                size_t length,
                                size_t position) {
  position = std::min(position, Size());  // Cap `position` in the valid range.
  length = std::min(length, Size() - position);
  OverwriteAt(insert_this, length, position);
}

void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
                                size_t position) {
  ReplaceAtIndex(insert_this, insert_this.Size(), position);
}

bool SyncBuffer::GetNextAudioInterleaved(InterleavedView<int16_t> audio) {
  RTC_DCHECK_EQ(audio.num_channels(), Channels());
  bool read = ReadInterleavedFromIndex(next_index_, audio);
  if (read) {
    next_index_ += audio.samples_per_channel();
  }
  return read;
}

void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
  end_timestamp_ += increment;
}

void SyncBuffer::Flush() {
  Zeros(Size());
  next_index_ = Size();
  end_timestamp_ = 0;
  dtmf_index_ = 0;
}

void SyncBuffer::set_next_index(size_t value) {
  // Cannot set `next_index_` larger than the size of the buffer.
  next_index_ = std::min(value, Size());
}

void SyncBuffer::set_dtmf_index(size_t value) {
  // Cannot set `dtmf_index_` larger than the size of the buffer.
  dtmf_index_ = std::min(value, Size());
}

}  // namespace webrtc