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}
109 lines
4.4 KiB
C
109 lines
4.4 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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#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
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#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
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#include <stdio.h>
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#include <stdlib.h>
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#if defined(__cplusplus)
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extern "C" {
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#endif
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#define OPEN_FILE_WB(filePtr, fullPath) \
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do { \
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if (fullPath != NULL) { \
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filePtr = fopen(fullPath, "wb"); \
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if (filePtr == NULL) { \
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printf("could not open %s to write to.", fullPath); \
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return -1; \
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} \
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} else { \
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filePtr = NULL; \
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} \
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} while (0)
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#define OPEN_FILE_AB(filePtr, fullPath) \
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do { \
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if (fullPath != NULL) { \
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filePtr = fopen(fullPath, "ab"); \
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if (filePtr == NULL) { \
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printf("could not open %s to write to.", fullPath); \
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return -1; \
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} \
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} else { \
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filePtr = NULL; \
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} \
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} while (0)
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#define OPEN_FILE_RB(filePtr, fullPath) \
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do { \
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if (fullPath != NULL) { \
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filePtr = fopen(fullPath, "rb"); \
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if (filePtr == NULL) { \
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printf("could not open %s to read from.", fullPath); \
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return -1; \
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} \
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} else { \
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filePtr = NULL; \
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} \
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} while (0)
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#define WRITE_FILE_D(bufferPtr, len, filePtr) \
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do { \
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if (filePtr != NULL) { \
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double dummy[1000]; \
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int cntr; \
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for (cntr = 0; cntr < (len); cntr++) { \
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dummy[cntr] = (double)bufferPtr[cntr]; \
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} \
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fwrite(dummy, sizeof(double), len, filePtr); \
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fflush(filePtr); \
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} \
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} while (0)
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typedef struct {
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unsigned int whenPackGeneratedMs;
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unsigned int whenPrevPackLeftMs;
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unsigned int sendTimeMs; /* milisecond */
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unsigned int arrival_time; /* samples */
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unsigned int sample_count; /* samples, also used as "send time stamp" */
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unsigned int rtp_number;
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} BottleNeckModel;
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void get_arrival_time(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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/* function for reading audio data from PCM file */
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int readframe(short* data, FILE* inp, int length);
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short readSwitch(int argc, char* argv[], char* strID);
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double readParamDouble(int argc, char* argv[], char* strID, double defaultVal);
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int readParamInt(int argc, char* argv[], char* strID, int defaultVal);
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int readParamString(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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#if defined(__cplusplus)
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}
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#endif
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#endif
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