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}
180 lines
4.0 KiB
C
180 lines
4.0 KiB
C
/*
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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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#include "modules/audio_coding/codecs/isac/main/util/utility.h"
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/* function for reading audio data from PCM file */
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int
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readframe(
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short* data,
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FILE* inp,
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int length)
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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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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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}
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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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}
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return status;
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}
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short
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readSwitch(
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int argc,
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char* argv[],
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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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double
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readParamDouble(
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int argc,
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char* argv[],
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char* strID,
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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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int
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readParamInt(
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int argc,
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char* argv[],
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char* strID,
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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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int
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readParamString(
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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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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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void
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get_arrival_time(
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int current_framesamples, /* samples */
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size_t packet_size, /* bytes */
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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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const int headerSizeByte = 35;
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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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travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000)
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/ (double)(bottleneck + headerRate) + 0.5);
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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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BN_data->whenPrevPackLeftMs = BN_data->whenPackGeneratedMs +
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travelTimeMs;
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}
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BN_data->arrival_time = (BN_data->whenPrevPackLeftMs *
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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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