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/*
* 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.
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "modules/audio_coding/codecs/isac/main/util/utility.h"
/* function for reading audio data from PCM file */
int
readframe(
short* data,
FILE* inp,
int length)
{
short k, rlen, status = 0;
unsigned char* ptrUChar;
ptrUChar = (unsigned char*)data;
rlen = (short)fread(data, sizeof(short), length, inp);
if (rlen < length) {
for (k = rlen; k < length; k++)
data[k] = 0;
status = 1;
}
// Assuming that our PCM files are written in Intel machines
for(k = 0; k < length; k++)
{
data[k] = (short)ptrUChar[k<<1] | ((((short)ptrUChar[(k<<1) + 1]) << 8) & 0xFF00);
}
return status;
}
short
readSwitch(
int argc,
char* argv[],
char* strID)
{
short n;
for(n = 0; n < argc; n++)
{
if(strcmp(argv[n], strID) == 0)
{
return 1;
}
}
return 0;
}
double
readParamDouble(
int argc,
char* argv[],
char* strID,
double defaultVal)
{
double returnVal = defaultVal;
short n;
for(n = 0; n < argc; n++)
{
if(strcmp(argv[n], strID) == 0)
{
n++;
if(n < argc)
{
returnVal = atof(argv[n]);
}
break;
}
}
return returnVal;
}
int
readParamInt(
int argc,
char* argv[],
char* strID,
int defaultVal)
{
int returnVal = defaultVal;
short n;
for(n = 0; n < argc; n++)
{
if(strcmp(argv[n], strID) == 0)
{
n++;
if(n < argc)
{
returnVal = atoi(argv[n]);
}
break;
}
}
return returnVal;
}
int
readParamString(
int argc,
char* argv[],
char* strID,
char* stringParam,
int maxSize)
{
int paramLenght = 0;
short n;
for(n = 0; n < argc; n++)
{
if(strcmp(argv[n], strID) == 0)
{
n++;
if(n < argc)
{
strncpy(stringParam, argv[n], maxSize);
paramLenght = (int)strlen(argv[n]);
}
break;
}
}
return paramLenght;
}
void
get_arrival_time(
int current_framesamples, /* samples */
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
size_t packet_size, /* bytes */
int bottleneck, /* excluding headers; bits/s */
BottleNeckModel* BN_data,
short senderSampFreqHz,
short receiverSampFreqHz)
{
unsigned int travelTimeMs;
const int headerSizeByte = 35;
int headerRate;
BN_data->whenPackGeneratedMs += (current_framesamples / (senderSampFreqHz / 1000));
headerRate = headerSizeByte * 8 * senderSampFreqHz / current_framesamples; /* bits/s */
/* everything in samples */
BN_data->sample_count = BN_data->sample_count + current_framesamples;
//travelTimeMs = ((packet_size + HeaderSize) * 8 * sampFreqHz) /
// (bottleneck + HeaderRate)
travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000)
/ (double)(bottleneck + headerRate) + 0.5);
if(BN_data->whenPrevPackLeftMs > BN_data->whenPackGeneratedMs)
{
BN_data->whenPrevPackLeftMs += travelTimeMs;
}
else
{
BN_data->whenPrevPackLeftMs = BN_data->whenPackGeneratedMs +
travelTimeMs;
}
BN_data->arrival_time = (BN_data->whenPrevPackLeftMs *
(receiverSampFreqHz / 1000));
// if (BN_data->arrival_time < BN_data->sample_count)
// BN_data->arrival_time = BN_data->sample_count;
BN_data->rtp_number++;
}