diff options
Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc | 107 |
1 files changed, 107 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc new file mode 100644 index 00000000000..d1802e174fc --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/sync_buffer.cc @@ -0,0 +1,107 @@ +/* + * 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 <assert.h> + +#include <algorithm> // Access to min. + +#include "webrtc/modules/audio_coding/neteq/sync_buffer.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. +// assert(false); + // 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::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); + AudioMultiVector::OverwriteAt(insert_this, length, position); +} + +void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this, + size_t position) { + ReplaceAtIndex(insert_this, insert_this.Size(), position); +} + +size_t SyncBuffer::GetNextAudioInterleaved(size_t requested_len, + int16_t* output) { + if (!output) { + assert(false); + return 0; + } + size_t samples_to_read = std::min(FutureLength(), requested_len); + ReadInterleavedFromIndex(next_index_, samples_to_read, output); + next_index_ += samples_to_read; + return samples_to_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 |