2011-07-07 08:21:25 +00:00
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/*
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* Copyright (c) 2011 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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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <math.h>
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2018-02-01 14:12:55 +01:00
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#include "modules/audio_coding/codecs/isac/main/util/utility.h"
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2011-07-07 08:21:25 +00:00
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/* function for reading audio data from PCM file */
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2012-01-27 13:44:26 +00:00
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int
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2011-07-07 08:21:25 +00:00
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readframe(
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2012-01-27 13:44:26 +00:00
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short* data,
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2011-07-07 08:21:25 +00:00
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FILE* inp,
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2012-01-27 13:44:26 +00:00
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int length)
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2011-07-07 08:21:25 +00:00
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{
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short k, rlen, status = 0;
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unsigned char* ptrUChar;
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ptrUChar = (unsigned char*)data;
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2012-01-27 13:44:26 +00:00
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2011-07-07 08:21:25 +00:00
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rlen = (short)fread(data, sizeof(short), length, inp);
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if (rlen < length) {
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for (k = rlen; k < length; k++)
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data[k] = 0;
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status = 1;
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2012-01-27 13:44:26 +00:00
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}
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2011-07-07 08:21:25 +00:00
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// Assuming that our PCM files are written in Intel machines
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for(k = 0; k < length; k++)
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{
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data[k] = (short)ptrUChar[k<<1] | ((((short)ptrUChar[(k<<1) + 1]) << 8) & 0xFF00);
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2012-01-27 13:44:26 +00:00
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}
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2011-07-07 08:21:25 +00:00
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return status;
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}
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2012-01-27 13:44:26 +00:00
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short
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2011-07-07 08:21:25 +00:00
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readSwitch(
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2012-01-27 13:44:26 +00:00
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int argc,
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char* argv[],
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2011-07-07 08:21:25 +00:00
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char* strID)
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{
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short n;
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for(n = 0; n < argc; n++)
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{
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if(strcmp(argv[n], strID) == 0)
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{
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return 1;
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}
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}
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return 0;
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}
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2012-01-27 13:44:26 +00:00
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double
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2011-07-07 08:21:25 +00:00
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readParamDouble(
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2012-01-27 13:44:26 +00:00
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int argc,
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char* argv[],
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char* strID,
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2011-07-07 08:21:25 +00:00
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double defaultVal)
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{
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double returnVal = defaultVal;
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short n;
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for(n = 0; n < argc; n++)
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{
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if(strcmp(argv[n], strID) == 0)
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{
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n++;
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if(n < argc)
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{
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returnVal = atof(argv[n]);
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}
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break;
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}
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}
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return returnVal;
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}
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2012-01-27 13:44:26 +00:00
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int
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2011-07-07 08:21:25 +00:00
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readParamInt(
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2012-01-27 13:44:26 +00:00
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int argc,
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char* argv[],
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char* strID,
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2011-07-07 08:21:25 +00:00
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int defaultVal)
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{
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int returnVal = defaultVal;
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short n;
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for(n = 0; n < argc; n++)
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{
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if(strcmp(argv[n], strID) == 0)
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{
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n++;
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if(n < argc)
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{
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returnVal = atoi(argv[n]);
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}
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break;
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}
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}
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return returnVal;
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}
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2012-01-27 13:44:26 +00:00
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int
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2011-07-07 08:21:25 +00:00
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readParamString(
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2012-01-27 13:44:26 +00:00
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int argc,
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char* argv[],
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char* strID,
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char* stringParam,
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2011-07-07 08:21:25 +00:00
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int maxSize)
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{
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int paramLenght = 0;
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short n;
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for(n = 0; n < argc; n++)
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{
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if(strcmp(argv[n], strID) == 0)
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{
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n++;
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if(n < argc)
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{
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strncpy(stringParam, argv[n], maxSize);
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paramLenght = (int)strlen(argv[n]);
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}
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break;
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}
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}
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return paramLenght;
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}
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2012-01-27 13:44:26 +00:00
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void
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2011-07-07 08:21:25 +00:00
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get_arrival_time(
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int current_framesamples, /* samples */
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Update a ton of audio code to use size_t more correctly and in general reduce
use of int16_t/uint16_t.
This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.
This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002
The change is being landed as TBR to all the folks who reviewed the above.
BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher
Review URL: https://codereview.webrtc.org/1230503003 .
Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
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size_t packet_size, /* bytes */
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2011-07-07 08:21:25 +00:00
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int bottleneck, /* excluding headers; bits/s */
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BottleNeckModel* BN_data,
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short senderSampFreqHz,
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short receiverSampFreqHz)
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{
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unsigned int travelTimeMs;
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2012-01-27 13:44:26 +00:00
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const int headerSizeByte = 35;
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2011-07-07 08:21:25 +00:00
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int headerRate;
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BN_data->whenPackGeneratedMs += (current_framesamples / (senderSampFreqHz / 1000));
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headerRate = headerSizeByte * 8 * senderSampFreqHz / current_framesamples; /* bits/s */
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/* everything in samples */
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BN_data->sample_count = BN_data->sample_count + current_framesamples;
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//travelTimeMs = ((packet_size + HeaderSize) * 8 * sampFreqHz) /
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// (bottleneck + HeaderRate)
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2012-01-27 13:44:26 +00:00
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travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000)
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/ (double)(bottleneck + headerRate) + 0.5);
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2011-07-07 08:21:25 +00:00
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if(BN_data->whenPrevPackLeftMs > BN_data->whenPackGeneratedMs)
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{
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BN_data->whenPrevPackLeftMs += travelTimeMs;
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}
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else
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{
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2012-01-27 13:44:26 +00:00
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BN_data->whenPrevPackLeftMs = BN_data->whenPackGeneratedMs +
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2011-07-07 08:21:25 +00:00
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travelTimeMs;
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}
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2012-01-27 13:44:26 +00:00
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BN_data->arrival_time = (BN_data->whenPrevPackLeftMs *
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2011-07-07 08:21:25 +00:00
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(receiverSampFreqHz / 1000));
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// if (BN_data->arrival_time < BN_data->sample_count)
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// BN_data->arrival_time = BN_data->sample_count;
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BN_data->rtp_number++;
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}
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