AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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8
webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
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@ -8,10 +8,10 @@
* 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_
#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
#include "webrtc/typedefs.h"
#include <stdint.h>
typedef struct {
int in_use;
@ -21,4 +21,4 @@ typedef struct {
int16_t jitter_info;
} IsacBandwidthInfo;
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
#endif // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_