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/*
* 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_frame.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);
}
void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
AudioFrame* output) {
RTC_DCHECK(output);
const size_t samples_to_read = std::min(FutureLength(), requested_len);
output->ResetWithoutMuting();
const size_t tot_samples_read = ReadInterleavedFromIndex(
next_index_, samples_to_read, output->mutable_data());
const size_t samples_read_per_channel = tot_samples_read / Channels();
next_index_ += samples_read_per_channel;
output->num_channels_ = Channels();
output->samples_per_channel_ = samples_read_per_channel;
}
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
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