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Side by Side Diff: talk/media/webrtc/webrtcvoiceengine.cc

Issue 1405023016: Convert usage of ARRAY_SIZE to arraysize. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: static_cast<int> Created 5 years, 1 month ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2004 Google Inc. 3 * Copyright 2004 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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36 #include <algorithm> 36 #include <algorithm>
37 #include <cstdio> 37 #include <cstdio>
38 #include <string> 38 #include <string>
39 #include <vector> 39 #include <vector>
40 40
41 #include "talk/media/base/audioframe.h" 41 #include "talk/media/base/audioframe.h"
42 #include "talk/media/base/audiorenderer.h" 42 #include "talk/media/base/audiorenderer.h"
43 #include "talk/media/base/constants.h" 43 #include "talk/media/base/constants.h"
44 #include "talk/media/base/streamparams.h" 44 #include "talk/media/base/streamparams.h"
45 #include "talk/media/webrtc/webrtcvoe.h" 45 #include "talk/media/webrtc/webrtcvoe.h"
46 #include "webrtc/base/arraysize.h"
46 #include "webrtc/base/base64.h" 47 #include "webrtc/base/base64.h"
47 #include "webrtc/base/byteorder.h" 48 #include "webrtc/base/byteorder.h"
48 #include "webrtc/base/common.h" 49 #include "webrtc/base/common.h"
49 #include "webrtc/base/helpers.h" 50 #include "webrtc/base/helpers.h"
50 #include "webrtc/base/logging.h" 51 #include "webrtc/base/logging.h"
51 #include "webrtc/base/stringencode.h" 52 #include "webrtc/base/stringencode.h"
52 #include "webrtc/base/stringutils.h" 53 #include "webrtc/base/stringutils.h"
53 #include "webrtc/call/rtc_event_log.h" 54 #include "webrtc/call/rtc_event_log.h"
54 #include "webrtc/common.h" 55 #include "webrtc/common.h"
55 #include "webrtc/modules/audio_processing/include/audio_processing.h" 56 #include "webrtc/modules/audio_processing/include/audio_processing.h"
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205 206
206 bool IsCodec(const AudioCodec& codec, const char* ref_name) { 207 bool IsCodec(const AudioCodec& codec, const char* ref_name) {
207 return (_stricmp(codec.name.c_str(), ref_name) == 0); 208 return (_stricmp(codec.name.c_str(), ref_name) == 0);
208 } 209 }
209 210
210 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) { 211 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
211 return (_stricmp(codec.plname, ref_name) == 0); 212 return (_stricmp(codec.plname, ref_name) == 0);
212 } 213 }
213 214
214 bool IsCodecMultiRate(const webrtc::CodecInst& codec) { 215 bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
215 for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) { 216 for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
216 if (IsCodec(codec, kCodecPrefs[i].name) && 217 if (IsCodec(codec, kCodecPrefs[i].name) &&
217 kCodecPrefs[i].clockrate == codec.plfreq) { 218 kCodecPrefs[i].clockrate == codec.plfreq) {
218 return kCodecPrefs[i].is_multi_rate; 219 return kCodecPrefs[i].is_multi_rate;
219 } 220 }
220 } 221 }
221 return false; 222 return false;
222 } 223 }
223 224
224 bool FindCodec(const std::vector<AudioCodec>& codecs, 225 bool FindCodec(const std::vector<AudioCodec>& codecs,
225 const AudioCodec& codec, 226 const AudioCodec& codec,
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448 int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); 449 int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
449 for (int i = 0; i < ncodecs; ++i) { 450 for (int i = 0; i < ncodecs; ++i) {
450 webrtc::CodecInst voe_codec; 451 webrtc::CodecInst voe_codec;
451 if (GetVoeCodec(i, &voe_codec)) { 452 if (GetVoeCodec(i, &voe_codec)) {
452 // Skip uncompressed formats. 453 // Skip uncompressed formats.
453 if (IsCodec(voe_codec, kL16CodecName)) { 454 if (IsCodec(voe_codec, kL16CodecName)) {
454 continue; 455 continue;
455 } 456 }
456 457
457 const CodecPref* pref = NULL; 458 const CodecPref* pref = NULL;
458 for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) { 459 for (size_t j = 0; j < arraysize(kCodecPrefs); ++j) {
459 if (IsCodec(voe_codec, kCodecPrefs[j].name) && 460 if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
460 kCodecPrefs[j].clockrate == voe_codec.plfreq && 461 kCodecPrefs[j].clockrate == voe_codec.plfreq &&
461 kCodecPrefs[j].channels == voe_codec.channels) { 462 kCodecPrefs[j].channels == voe_codec.channels) {
462 pref = &kCodecPrefs[j]; 463 pref = &kCodecPrefs[j];
463 break; 464 break;
464 } 465 }
465 } 466 }
466 467
467 if (pref) { 468 if (pref) {
468 // Use the payload type that we've configured in our pref table; 469 // Use the payload type that we've configured in our pref table;
469 // use the offset in our pref table to determine the sort order. 470 // use the offset in our pref table to determine the sort order.
470 AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq, 471 AudioCodec codec(
471 voe_codec.rate, voe_codec.channels, 472 pref->payload_type, voe_codec.plname, voe_codec.plfreq,
472 ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs)); 473 voe_codec.rate, voe_codec.channels,
474 static_cast<int>(arraysize(kCodecPrefs)) - (pref - kCodecPrefs));
473 LOG(LS_INFO) << ToString(codec); 475 LOG(LS_INFO) << ToString(codec);
474 if (IsCodec(codec, kIsacCodecName)) { 476 if (IsCodec(codec, kIsacCodecName)) {
475 // Indicate auto-bitrate in signaling. 477 // Indicate auto-bitrate in signaling.
476 codec.bitrate = 0; 478 codec.bitrate = 0;
477 } 479 }
478 if (IsCodec(codec, kOpusCodecName)) { 480 if (IsCodec(codec, kOpusCodecName)) {
479 // Only add fmtp parameters that differ from the spec. 481 // Only add fmtp parameters that differ from the spec.
480 if (kPreferredMinPTime != kOpusDefaultMinPTime) { 482 if (kPreferredMinPTime != kOpusDefaultMinPTime) {
481 codec.params[kCodecParamMinPTime] = 483 codec.params[kCodecParamMinPTime] =
482 rtc::ToString(kPreferredMinPTime); 484 rtc::ToString(kPreferredMinPTime);
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2856 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); 2858 LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
2857 return false; 2859 return false;
2858 } 2860 }
2859 } 2861 }
2860 return true; 2862 return true;
2861 } 2863 }
2862 2864
2863 } // namespace cricket 2865 } // namespace cricket
2864 2866
2865 #endif // HAVE_WEBRTC_VOICE 2867 #endif // HAVE_WEBRTC_VOICE
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