webrtc/modules/audio_coding/codecs/opus/opus_inst.h
Danil Chapovalov c2160b14b1 Delete expired field trial Audio-OpusAvoidNoisePumpingDuringDtx
Bug: webrtc:42222522, chromium:40174928
Change-Id: I2391b3078e5fff93edca3c3e6e568560b2a1c1cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357742
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42691}
2024-07-30 09:43:52 +00:00

41 lines
1.1 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()
#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;
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_