Alessio Bazzica fbeb76ab51 Revert "[ACM] iSAC audio codec removed"
This reverts commit b46c4bf27ba5c417fcba7f200d80fa4634e7e1a1.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
2022-11-16 20:40:52 +00:00

180 lines
4.0 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.
*/
#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 */
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++;
}