webrtc_m130/talk/app/webrtc/mediacontroller.cc
Fredrik Solenberg 709ed67c38 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1269863005 .

Cr-Commit-Position: refs/heads/master@{#9939}
2015-09-15 10:26:45 +00:00

88 lines
3.1 KiB
C++

/*
* libjingle
* Copyright 2015 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/mediacontroller.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/call.h"
namespace {
const int kMinBandwidthBps = 30000;
const int kStartBandwidthBps = 300000;
const int kMaxBandwidthBps = 2000000;
class MediaController : public webrtc::MediaControllerInterface {
public:
MediaController(rtc::Thread* worker_thread,
webrtc::VoiceEngine* voice_engine)
: worker_thread_(worker_thread) {
DCHECK(nullptr != worker_thread);
worker_thread_->Invoke<void>(
rtc::Bind(&MediaController::Construct_w, this, voice_engine));
}
~MediaController() override {
worker_thread_->Invoke<void>(
rtc::Bind(&MediaController::Destruct_w, this));
}
webrtc::Call* call_w() override {
DCHECK(worker_thread_->IsCurrent());
return call_.get();
}
private:
void Construct_w(webrtc::VoiceEngine* voice_engine) {
DCHECK(worker_thread_->IsCurrent());
webrtc::Call::Config config;
config.voice_engine = voice_engine;
config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
call_.reset(webrtc::Call::Create(config));
}
void Destruct_w() {
DCHECK(worker_thread_->IsCurrent());
call_.reset(nullptr);
}
rtc::Thread* worker_thread_;
rtc::scoped_ptr<webrtc::Call> call_;
DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
};
} // namespace {
namespace webrtc {
MediaControllerInterface* MediaControllerInterface::Create(
rtc::Thread* worker_thread, webrtc::VoiceEngine* voice_engine) {
return new MediaController(worker_thread, voice_engine);
}
} // namespace webrtc