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Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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1 /*
2 * libjingle
3 * Copyright 2015 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28 #include "talk/app/webrtc/mediacontroller.h"
29
30 #include "talk/session/media/channelmanager.h"
31 #include "webrtc/base/bind.h"
32 #include "webrtc/base/checks.h"
33 #include "webrtc/call.h"
34
35 namespace {
36
37 const int kMinBandwidthBps = 30000;
38 const int kStartBandwidthBps = 300000;
39 const int kMaxBandwidthBps = 2000000;
40
41 class MediaController : public webrtc::MediaControllerInterface,
42 public sigslot::has_slots<> {
43 public:
44 MediaController(rtc::Thread* worker_thread,
45 cricket::ChannelManager* channel_manager)
46 : worker_thread_(worker_thread), channel_manager_(channel_manager) {
47 RTC_DCHECK(nullptr != worker_thread);
48 worker_thread_->Invoke<void>(
49 rtc::Bind(&MediaController::Construct_w, this,
50 channel_manager_->media_engine()));
51 }
52 ~MediaController() override {
53 worker_thread_->Invoke<void>(rtc::Bind(&MediaController::Destruct_w, this));
54 }
55
56 webrtc::Call* call_w() override {
57 RTC_DCHECK(worker_thread_->IsCurrent());
58 return call_.get();
59 }
60
61 cricket::ChannelManager* channel_manager() const override {
62 return channel_manager_;
63 }
64
65 private:
66 void Construct_w(cricket::MediaEngineInterface* media_engine) {
67 RTC_DCHECK(worker_thread_->IsCurrent());
68 RTC_DCHECK(media_engine);
69 webrtc::Call::Config config;
70 config.audio_state = media_engine->GetAudioState();
71 config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
72 config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
73 config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
74 call_.reset(webrtc::Call::Create(config));
75 }
76 void Destruct_w() {
77 RTC_DCHECK(worker_thread_->IsCurrent());
78 call_.reset();
79 }
80
81 rtc::Thread* const worker_thread_;
82 cricket::ChannelManager* const channel_manager_;
83 rtc::scoped_ptr<webrtc::Call> call_;
84
85 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
86 };
87 } // namespace {
88
89 namespace webrtc {
90
91 MediaControllerInterface* MediaControllerInterface::Create(
92 rtc::Thread* worker_thread,
93 cricket::ChannelManager* channel_manager) {
94 return new MediaController(worker_thread, channel_manager);
95 }
96 } // namespace webrtc
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