blob: 99676504cdd74319ab368cd443d7313e64bfcd76 [file] [log] [blame]
/*
* Copyright (c) 2011 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.
*/
/*
* decode_bwe.c
*
* This C file contains the internal decode bandwidth estimate function.
*
*/
#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
Bitstr_dec *streamdata,
size_t packet_size,
uint16_t rtp_seq_number,
uint32_t send_ts,
uint32_t arr_ts)
{
int16_t index;
size_t frame_samples;
int err;
/* decode framelength */
err = WebRtcIsacfix_DecodeFrameLen(streamdata, &frame_samples);
/* error check */
if (err<0) {
return err;
}
/* decode BW estimation */
err = WebRtcIsacfix_DecodeSendBandwidth(streamdata, &index);
/* error check */
if (err<0) {
return err;
}
/* Update BWE with received data */
err = WebRtcIsacfix_UpdateUplinkBwImpl(
bwest_str,
rtp_seq_number,
(int16_t)(frame_samples * 1000 / FS),
send_ts,
arr_ts,
packet_size, /* in bytes */
index);
/* error check */
if (err<0) {
return err;
}
/* Succesful */
return 0;
}