Arun Raghavan 753eada3aa Update audio_processing module
Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
2015-10-15 16:18:45 +05:30

25 lines
801 B
C

/*
* 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 WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
#include "webrtc/typedefs.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 // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_