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

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add missing updated_options Created 5 years, 2 months 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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45 #include "talk/media/webrtc/webrtcvoe.h" 45 #include "talk/media/webrtc/webrtcvoe.h"
46 #include "webrtc/base/base64.h" 46 #include "webrtc/base/base64.h"
47 #include "webrtc/base/byteorder.h" 47 #include "webrtc/base/byteorder.h"
48 #include "webrtc/base/common.h" 48 #include "webrtc/base/common.h"
49 #include "webrtc/base/helpers.h" 49 #include "webrtc/base/helpers.h"
50 #include "webrtc/base/logging.h" 50 #include "webrtc/base/logging.h"
51 #include "webrtc/base/stringencode.h" 51 #include "webrtc/base/stringencode.h"
52 #include "webrtc/base/stringutils.h" 52 #include "webrtc/base/stringutils.h"
53 #include "webrtc/common.h" 53 #include "webrtc/common.h"
54 #include "webrtc/modules/audio_processing/include/audio_processing.h" 54 #include "webrtc/modules/audio_processing/include/audio_processing.h"
55 #include "webrtc/system_wrappers/interface/field_trial.h"
55 56
56 namespace cricket { 57 namespace cricket {
57 namespace { 58 namespace {
58 59
59 const int kMaxNumPacketSize = 6; 60 const int kMaxNumPacketSize = 6;
60 struct CodecPref { 61 struct CodecPref {
61 const char* name; 62 const char* name;
62 int clockrate; 63 int clockrate;
63 int channels; 64 int channels;
64 int payload_type; 65 int payload_type;
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424 // Load our audio codec list. 425 // Load our audio codec list.
425 ConstructCodecs(); 426 ConstructCodecs();
426 427
427 // Load our RTP Header extensions. 428 // Load our RTP Header extensions.
428 rtp_header_extensions_.push_back( 429 rtp_header_extensions_.push_back(
429 RtpHeaderExtension(kRtpAudioLevelHeaderExtension, 430 RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
430 kRtpAudioLevelHeaderExtensionDefaultId)); 431 kRtpAudioLevelHeaderExtensionDefaultId));
431 rtp_header_extensions_.push_back( 432 rtp_header_extensions_.push_back(
432 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 433 RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
433 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId)); 434 kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
435 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
436 rtp_header_extensions_.push_back(RtpHeaderExtension(
437 kRtpTransportSequenceNumberHeaderExtension,
438 kRtpTransportSequenceNumberHeaderExtensionDefaultId));
439 }
434 options_ = GetDefaultEngineOptions(); 440 options_ = GetDefaultEngineOptions();
435 } 441 }
436 442
437 void WebRtcVoiceEngine::ConstructCodecs() { 443 void WebRtcVoiceEngine::ConstructCodecs() {
438 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:"; 444 LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
439 int ncodecs = voe_wrapper_->codec()->NumOfCodecs(); 445 int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
440 for (int i = 0; i < ncodecs; ++i) { 446 for (int i = 0; i < ncodecs; ++i) {
441 webrtc::CodecInst voe_codec; 447 webrtc::CodecInst voe_codec;
442 if (GetVoeCodec(i, &voe_codec)) { 448 if (GetVoeCodec(i, &voe_codec)) {
443 // Skip uncompressed formats. 449 // Skip uncompressed formats.
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3037 LOG(LS_WARNING) << "Unknown codec " << ToString(codec); 3043 LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
3038 return false; 3044 return false;
3039 } 3045 }
3040 } 3046 }
3041 return true; 3047 return true;
3042 } 3048 }
3043 3049
3044 } // namespace cricket 3050 } // namespace cricket
3045 3051
3046 #endif // HAVE_WEBRTC_VOICE 3052 #endif // HAVE_WEBRTC_VOICE
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