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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
25 lines
771 B
C
25 lines
771 B
C
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
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#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
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#include <stdint.h>
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typedef struct {
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int in_use;
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int32_t send_bw_avg;
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int32_t send_max_delay_avg;
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int16_t bottleneck_idx;
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int16_t jitter_info;
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} IsacBandwidthInfo;
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#endif // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
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