MediaController/Call instantiation.

talk/app/webrtc/mediacontroller.cc

 /*
  * 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.
  */
 
-// Place holder file to be able to update Chrome's libjingle.gyp before the real
-// implementation goes in.
+#include "talk/app/webrtc/mediacontroller.h"
 
-#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