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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
108 lines
3.6 KiB
C++
108 lines
3.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <algorithm> // Access to min.
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#include "modules/audio_coding/neteq/sync_buffer.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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size_t SyncBuffer::FutureLength() const {
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return Size() - next_index_;
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}
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void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
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size_t samples_added = append_this.Size();
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AudioMultiVector::PushBack(append_this);
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AudioMultiVector::PopFront(samples_added);
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if (samples_added <= next_index_) {
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next_index_ -= samples_added;
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} else {
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// This means that we are pushing out future data that was never used.
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// assert(false);
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// TODO(hlundin): This assert must be disabled to support 60 ms frames.
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// This should not happen even for 60 ms frames, but it does. Investigate
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// why.
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next_index_ = 0;
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}
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dtmf_index_ -= std::min(dtmf_index_, samples_added);
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}
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void SyncBuffer::PushFrontZeros(size_t length) {
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InsertZerosAtIndex(length, 0);
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}
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void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
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position = std::min(position, Size());
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length = std::min(length, Size() - position);
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AudioMultiVector::PopBack(length);
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for (size_t channel = 0; channel < Channels(); ++channel) {
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channels_[channel]->InsertZerosAt(length, position);
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}
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if (next_index_ >= position) {
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// We are moving the |next_index_| sample.
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set_next_index(next_index_ + length); // Overflow handled by subfunction.
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}
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if (dtmf_index_ > 0 && dtmf_index_ >= position) {
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// We are moving the |dtmf_index_| sample.
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set_dtmf_index(dtmf_index_ + length); // Overflow handled by subfunction.
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}
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}
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void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
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size_t length,
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size_t position) {
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position = std::min(position, Size()); // Cap |position| in the valid range.
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length = std::min(length, Size() - position);
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AudioMultiVector::OverwriteAt(insert_this, length, position);
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}
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void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
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size_t position) {
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ReplaceAtIndex(insert_this, insert_this.Size(), position);
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}
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void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
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AudioFrame* output) {
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RTC_DCHECK(output);
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const size_t samples_to_read = std::min(FutureLength(), requested_len);
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output->ResetWithoutMuting();
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const size_t tot_samples_read =
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ReadInterleavedFromIndex(next_index_, samples_to_read,
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output->mutable_data());
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const size_t samples_read_per_channel = tot_samples_read / Channels();
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next_index_ += samples_read_per_channel;
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output->num_channels_ = Channels();
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output->samples_per_channel_ = samples_read_per_channel;
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}
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void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
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end_timestamp_ += increment;
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}
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void SyncBuffer::Flush() {
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Zeros(Size());
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next_index_ = Size();
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end_timestamp_ = 0;
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dtmf_index_ = 0;
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}
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void SyncBuffer::set_next_index(size_t value) {
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// Cannot set |next_index_| larger than the size of the buffer.
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next_index_ = std::min(value, Size());
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}
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void SyncBuffer::set_dtmf_index(size_t value) {
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// Cannot set |dtmf_index_| larger than the size of the buffer.
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dtmf_index_ = std::min(value, Size());
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}
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} // namespace webrtc
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