webrtc/modules/audio_coding/codecs/opus/opus_inst.h
Jim Gustafson 6e5158df93 m120 merge fixes
- Use worker_thread TaskQueue for channel operations
- Fix use of deprecated DNS resolver
- Restore quantization of audio levels
- Simplify crypto options change
- Move channel blocking operations to pc
- Sync opus for merge
2024-01-24 09:14:46 -08:00

44 lines
1.3 KiB
C

/*
* 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()
// RingRTC change to use a fork of opus
#include "ringrtc/opus/src/include/opus.h"
#include "ringrtc/opus/src/include/opus_multistream.h"
RTC_POP_IGNORING_WUNDEF()
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
OpusMSEncoder* multistream_encoder;
size_t channels;
int in_dtx_mode;
bool avoid_noise_pumping_during_dtx;
int sample_rate_hz;
float smooth_energy_non_active_frames;
};
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
OpusMSDecoder* multistream_decoder;
int prev_decoded_samples;
bool plc_use_prev_decoded_samples;
size_t channels;
int in_dtx_mode;
int sample_rate_hz;
};
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_