webrtc/modules/audio_coding/codecs/opus/opus_inst.h
Yves Gerey 665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00

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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 "opus.h"
RTC_POP_IGNORING_WUNDEF()
struct WebRtcOpusEncInst {
OpusEncoder* encoder;
size_t channels;
int in_dtx_mode;
};
struct WebRtcOpusDecInst {
OpusDecoder* decoder;
int prev_decoded_samples;
size_t channels;
int in_dtx_mode;
};
#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_