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/*
* Copyright (c) 2014 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
#include <stddef.h>
#include "rtc_base/ignore_wundef.h"
RTC_PUSH_IGNORING_WUNDEF()
#include "third_party/opus/src/include/opus.h"
#include "third_party/opus/src/include/opus_multistream.h"
RTC_POP_IGNORING_WUNDEF()
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
OpusMSEncoder* multistream_encoder;
size_t channels;
int in_dtx_mode;
int sample_rate_hz;
};
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
OpusMSDecoder* multistream_decoder;
size_t channels;
int in_dtx_mode;
int sample_rate_hz;
// TODO: https://issues.webrtc.org/376493209 - Remove when libopus gets fixed.
int last_packet_num_channels;
};
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
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