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libtgvoip/webrtc_dsp/modules/audio_coding/codecs/isac/bandwidth_info.h

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/*
* Copyright (c) 2015 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_ISAC_BANDWIDTH_INFO_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
#include <stdint.h>
typedef struct {
int in_use;
int32_t send_bw_avg;
int32_t send_max_delay_avg;
int16_t bottleneck_idx;
int16_t jitter_info;
} IsacBandwidthInfo;
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_