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Reformat the WebRTC code base

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Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This commit is contained in:
Yves Gerey
2018-06-19 15:03:05 +02:00
parent b602123a5a
commit 665174fdbb
1569 changed files with 30495 additions and 30309 deletions
Download Patch File Download Diff File Expand all files Collapse all files

46
modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -11,11 +11,11 @@
// ReleaseTest-API.cpp : Defines the entry point for the console application.
//
#include <ctype.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include <ctype.h>
#include <iostream>
/* include API */
@ -24,10 +24,13 @@
#include "rtc_base/format_macros.h"
/* Defines */
#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
#define MAX_FRAMESAMPLES 960 /* max number of samples per frame
(= 60 ms frame & 16 kHz) or
(= 30 ms frame & 32 kHz) */
#define SEED_FILE \
"randseed.txt" /* Used when running decoder on garbage data \
*/
#define MAX_FRAMESAMPLES \
960 /* max number of samples per frame \
(= 60 ms frame & 16 kHz) or \
(= 30 ms frame & 32 kHz) */
#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
#define SWBFRAMESAMPLES_10ms 320
//#define FS 16000 /* sampling frequency (Hz) */
@ -42,7 +45,7 @@ using namespace std;
int main(int argc, char* argv[]) {
char inname[100], outname[100], bottleneck_file[100], vadfile[100];
FILE* inp, *outp, * f_bn = NULL, * vadp = NULL, *bandwidthp;
FILE *inp, *outp, *f_bn = NULL, *vadp = NULL, *bandwidthp;
int framecnt, endfile;
size_t i;
@ -230,8 +233,10 @@ int main(int argc, char* argv[]) {
rateBPS = atoi(argv[i + 1]);
setControlBWE = 1;
if ((rateBPS < 10000) || (rateBPS > 32000)) {
printf("\n%d is not a initial rate. Valid values are in the range "
"10000 to 32000.\n", rateBPS);
printf(
"\n%d is not a initial rate. Valid values are in the range "
"10000 to 32000.\n",
rateBPS);
exit(0);
}
printf("New initial rate: %d\n", rateBPS);
@ -242,8 +247,10 @@ int main(int argc, char* argv[]) {
if (!strcmp("-FL", argv[i])) {
framesize = atoi(argv[i + 1]);
if ((framesize != 30) && (framesize != 60)) {
printf("\n%d is not a valid frame length. Valid length are 30 and 60 "
"msec.\n", framesize);
printf(
"\n%d is not a valid frame length. Valid length are 30 and 60 "
"msec.\n",
framesize);
exit(0);
}
setControlBWE = 1;
@ -277,8 +284,10 @@ int main(int argc, char* argv[]) {
testNum = atoi(argv[i + 1]);
printf("Fault test: %d\n", testNum);
if (testNum < 1 || testNum > 10) {
printf("\n%d is not a valid Fault Scenario number. Valid Fault "
"Scenarios are numbered 1-10.\n", testNum);
printf(
"\n%d is not a valid Fault Scenario number. Valid Fault "
"Scenarios are numbered 1-10.\n",
testNum);
exit(0);
}
i++;
@ -336,8 +345,10 @@ int main(int argc, char* argv[]) {
sscanf(argv[i], "%s", bottleneck_file);
f_bn = fopen(bottleneck_file, "rb");
if (f_bn == NULL) {
printf("Error No value provided for BottleNeck and cannot read file "
"%s.\n", bottleneck_file);
printf(
"Error No value provided for BottleNeck and cannot read file "
"%s.\n",
bottleneck_file);
exit(0);
} else {
printf("reading bottleneck rates from file %s\n\n", bottleneck_file);
@ -637,8 +648,8 @@ int main(int argc, char* argv[]) {
}
if (fwrite(streamDataTransCoding, sizeof(uint8_t),
streamLenTransCoding, transcodingBitstream) !=
streamLenTransCoding) {
streamLenTransCoding,
transcodingBitstream) != streamLenTransCoding) {
return -1;
}
@ -718,8 +729,7 @@ int main(int argc, char* argv[]) {
fprintf(stderr, "Error in RED trans-coding\n");
exit(0);
}
streamLenTransCoding =
static_cast<size_t>(streamLenTransCoding_int);
streamLenTransCoding = static_cast<size_t>(streamLenTransCoding_int);
}
}