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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: No changes to const parameters. Created 5 years, 2 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2015 Google Inc. 3 * Copyright 2015 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,
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
49 ~MediaController() override { 49 ~MediaController() override {
50 worker_thread_->Invoke<void>( 50 worker_thread_->Invoke<void>(
51 rtc::Bind(&MediaController::Destruct_w, this)); 51 rtc::Bind(&MediaController::Destruct_w, this));
52 } 52 }
53 53
54 webrtc::Call* call_w() override { 54 webrtc::Call* call_w() override {
55 RTC_DCHECK(worker_thread_->IsCurrent()); 55 RTC_DCHECK(worker_thread_->IsCurrent());
56 return call_.get(); 56 return call_.get();
57 } 57 }
58 58
59 void OnPacketSent(const webrtc::SentPacket& sent_packet) override {
60 call_->OnPacketSent(sent_packet);
61 }
62
59 private: 63 private:
60 void Construct_w(webrtc::VoiceEngine* voice_engine) { 64 void Construct_w(webrtc::VoiceEngine* voice_engine) {
61 RTC_DCHECK(worker_thread_->IsCurrent()); 65 RTC_DCHECK(worker_thread_->IsCurrent());
62 webrtc::Call::Config config; 66 webrtc::Call::Config config;
63 config.voice_engine = voice_engine; 67 config.voice_engine = voice_engine;
64 config.bitrate_config.min_bitrate_bps = kMinBandwidthBps; 68 config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
65 config.bitrate_config.start_bitrate_bps = kStartBandwidthBps; 69 config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
66 config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps; 70 config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
67 call_.reset(webrtc::Call::Create(config)); 71 call_.reset(webrtc::Call::Create(config));
68 } 72 }
69 void Destruct_w() { 73 void Destruct_w() {
70 RTC_DCHECK(worker_thread_->IsCurrent()); 74 RTC_DCHECK(worker_thread_->IsCurrent());
71 call_.reset(nullptr); 75 call_.reset(nullptr);
72 } 76 }
73 77
74 rtc::Thread* worker_thread_; 78 rtc::Thread* worker_thread_;
75 rtc::scoped_ptr<webrtc::Call> call_; 79 rtc::scoped_ptr<webrtc::Call> call_;
76 80
77 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController); 81 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
78 }; 82 };
79 } // namespace { 83 } // namespace {
80 84
81 namespace webrtc { 85 namespace webrtc {
82 86
83 MediaControllerInterface* MediaControllerInterface::Create( 87 MediaControllerInterface* MediaControllerInterface::Create(
84 rtc::Thread* worker_thread, webrtc::VoiceEngine* voice_engine) { 88 rtc::Thread* worker_thread, webrtc::VoiceEngine* voice_engine) {
85 return new MediaController(worker_thread, voice_engine); 89 return new MediaController(worker_thread, voice_engine);
86 } 90 }
87 } // namespace webrtc 91 } // namespace webrtc
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