Move talk/media to webrtc/media

talk/media/base/mediaengine.h

-/*
- * libjingle
- * Copyright 2004 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.
- */
-
-#ifndef TALK_MEDIA_BASE_MEDIAENGINE_H_
-#define TALK_MEDIA_BASE_MEDIAENGINE_H_
-
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
-#include <CoreAudio/CoreAudio.h>
-#endif
-
-#include <string>
-#include <vector>
-
-#include "talk/media/base/codec.h"
-#include "talk/media/base/mediachannel.h"
-#include "talk/media/base/mediacommon.h"
-#include "talk/media/base/videocapturer.h"
-#include "talk/media/base/videocommon.h"
-#include "talk/media/devices/devicemanager.h"
-#include "webrtc/audio_state.h"
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/sigslotrepeater.h"
-
-#if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
-#define DISABLE_MEDIA_ENGINE_FACTORY
-#endif
-
-namespace webrtc {
-class Call;
-}
-
-namespace cricket {
-
-class VideoCapturer;
-
-struct RtpCapabilities {
-  std::vector<RtpHeaderExtension> header_extensions;
-};
-
-// MediaEngineInterface is an abstraction of a media engine which can be
-// subclassed to support different media componentry backends.
-// It supports voice and video operations in the same class to facilitate
-// proper synchronization between both media types.
-class MediaEngineInterface {
- public:
-  virtual ~MediaEngineInterface() {}
-
-  // Initialization
-  // Starts the engine.
-  virtual bool Init(rtc::Thread* worker_thread) = 0;
-  // Shuts down the engine.
-  virtual void Terminate() = 0;
-  // TODO(solenberg): Remove once VoE API refactoring is done.
-  virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
-
-  // MediaChannel creation
-  // Creates a voice media channel. Returns NULL on failure.
-  virtual VoiceMediaChannel* CreateChannel(
-      webrtc::Call* call,
-      const AudioOptions& options) = 0;
-  // Creates a video media channel, paired with the specified voice channel.
-  // Returns NULL on failure.
-  virtual VideoMediaChannel* CreateVideoChannel(
-      webrtc::Call* call,
-      const VideoOptions& options) = 0;
-
-  // Device configuration
-  // Gets the current speaker volume, as a value between 0 and 255.
-  virtual bool GetOutputVolume(int* level) = 0;
-  // Sets the current speaker volume, as a value between 0 and 255.
-  virtual bool SetOutputVolume(int level) = 0;
-
-  // Gets the current microphone level, as a value between 0 and 10.
-  virtual int GetInputLevel() = 0;
-
-  virtual const std::vector<AudioCodec>& audio_codecs() = 0;
-  virtual RtpCapabilities GetAudioCapabilities() = 0;
-  virtual const std::vector<VideoCodec>& video_codecs() = 0;
-  virtual RtpCapabilities GetVideoCapabilities() = 0;
-
-  // Starts AEC dump using existing file, a maximum file size in bytes can be
-  // specified. Logging is stopped just before the size limit is exceeded.
-  // If max_size_bytes is set to a value <= 0, no limit will be used.
-  virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
-
-  // Stops recording AEC dump.
-  virtual void StopAecDump() = 0;
-
-  // Starts RtcEventLog using existing file.
-  virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
-
-  // Stops recording an RtcEventLog.
-  virtual void StopRtcEventLog() = 0;
-};
-
-
-#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
-class MediaEngineFactory {
- public:
-  typedef cricket::MediaEngineInterface* (*MediaEngineCreateFunction)();
-  // Creates a media engine, using either the compiled system default or the
-  // creation function specified in SetCreateFunction, if specified.
-  static MediaEngineInterface* Create();
-  // Sets the function used when calling Create. If unset, the compiled system
-  // default will be used. Returns the old create function, or NULL if one
-  // wasn't set. Likewise, NULL can be used as the |function| parameter to
-  // reset to the default behavior.
-  static MediaEngineCreateFunction SetCreateFunction(
-      MediaEngineCreateFunction function);
- private:
-  static MediaEngineCreateFunction create_function_;
-};
-#endif
-
-// CompositeMediaEngine constructs a MediaEngine from separate
-// voice and video engine classes.
-template<class VOICE, class VIDEO>
-class CompositeMediaEngine : public MediaEngineInterface {
- public:
-  virtual ~CompositeMediaEngine() {}
-  virtual bool Init(rtc::Thread* worker_thread) {
-    if (!voice_.Init(worker_thread))
-      return false;
-    video_.Init();
-    return true;
-  }
-  virtual void Terminate() {
-    voice_.Terminate();
-  }
-
-  virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
-    return voice_.GetAudioState();
-  }
-  virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
-                                           const AudioOptions& options) {
-    return voice_.CreateChannel(call, options);
-  }
-  virtual VideoMediaChannel* CreateVideoChannel(webrtc::Call* call,
-                                                const VideoOptions& options) {
-    return video_.CreateChannel(call, options);
-  }
-
-  virtual bool GetOutputVolume(int* level) {
-    return voice_.GetOutputVolume(level);
-  }
-  virtual bool SetOutputVolume(int level) {
-    return voice_.SetOutputVolume(level);
-  }
-
-  virtual int GetInputLevel() {
-    return voice_.GetInputLevel();
-  }
-  virtual const std::vector<AudioCodec>& audio_codecs() {
-    return voice_.codecs();
-  }
-  virtual RtpCapabilities GetAudioCapabilities() {
-    return voice_.GetCapabilities();
-  }
-  virtual const std::vector<VideoCodec>& video_codecs() {
-    return video_.codecs();
-  }
-  virtual RtpCapabilities GetVideoCapabilities() {
-    return video_.GetCapabilities();
-  }
-
-  virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
-    return voice_.StartAecDump(file, max_size_bytes);
-  }
-
-  virtual void StopAecDump() {
-    voice_.StopAecDump();
-  }
-
-  virtual bool StartRtcEventLog(rtc::PlatformFile file) {
-    return voice_.StartRtcEventLog(file);
-  }
-
-  virtual void StopRtcEventLog() { voice_.StopRtcEventLog(); }
-
- protected:
-  VOICE voice_;
-  VIDEO video_;
-};
-
-enum DataChannelType {
-  DCT_NONE = 0,
-  DCT_RTP = 1,
-  DCT_SCTP = 2
-};
-
-class DataEngineInterface {
- public:
-  virtual ~DataEngineInterface() {}
-  virtual DataMediaChannel* CreateChannel(DataChannelType type) = 0;
-  virtual const std::vector<DataCodec>& data_codecs() = 0;
-};
-
-}  // namespace cricket
-
-#endif  // TALK_MEDIA_BASE_MEDIAENGINE_H_