webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
Mirko Bonadei bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00

69 lines
1.6 KiB
C

/*
* 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 "bandwidth_estimator.h"
#include "codec.h"
#include "entropy_coding.h"
#include "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;
}