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/*
* libjingle
* Copyright 2004--2011, 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_SESSION_PHONE_WEBRTCVIDEOENGINE_H_
#define TALK_SESSION_PHONE_WEBRTCVIDEOENGINE_H_
#include <vector>
#include "talk/base/scoped_ptr.h"
#include "talk/session/phone/videocommon.h"
#include "talk/session/phone/codec.h"
#include "talk/session/phone/channel.h"
#include "talk/session/phone/webrtccommon.h"
#ifdef WEBRTC_RELATIVE_PATH
#include "video_engine/main/interface/vie_base.h"
#else
#include "third_party/webrtc/files/include/vie_base.h"
#endif // WEBRTC_RELATIVE_PATH
namespace webrtc {
class VideoCaptureModule;
class VideoRender;
class ViEExternalCapture;
}
namespace cricket {
struct CapturedFrame;
struct Device;
class LocalStreamInfo;
class VideoCapturer;
class VideoFrame;
class VideoProcessor;
class VideoRenderer;
class ViETraceWrapper;
class ViEWrapper;
class VoiceMediaChannel;
class WebRtcRenderAdapter;
class WebRtcVideoMediaChannel;
class WebRtcVoiceEngine;
class WebRtcDecoderObserver;
class WebRtcEncoderObserver;
class WebRtcVideoEngine : public sigslot::has_slots<>,
public webrtc::ViEBaseObserver,
public webrtc::TraceCallback {
public:
// Creates the WebRtcVideoEngine with internal VideoCaptureModule.
WebRtcVideoEngine();
// For testing purposes. Allows the WebRtcVoiceEngine and
// ViEWrapper to be mocks.
// TODO: Remove the 2-arg ctor once fake tracing is implemented.
WebRtcVideoEngine(WebRtcVoiceEngine* voice_engine,
ViEWrapper* vie_wrapper);
WebRtcVideoEngine(WebRtcVoiceEngine* voice_engine,
ViEWrapper* vie_wrapper,
ViETraceWrapper* tracing);
~WebRtcVideoEngine();
// Basic video engine implementation.
bool Init();
void Terminate();
int GetCapabilities();
bool SetOptions(int options);
bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
WebRtcVideoMediaChannel* CreateChannel(VoiceMediaChannel* voice_channel);
const std::vector<VideoCodec>& codecs() const;
void SetLogging(int min_sev, const char* filter);
// Capture-related stuff. Will be removed with capture refactor.
bool SetCaptureDevice(const Device* device);
bool SetCaptureModule(webrtc::VideoCaptureModule* vcm);
// If capturer is NULL, unregisters the capturer and stops capturing.
// Otherwise sets the capturer and starts capturing.
bool SetVideoCapturer(VideoCapturer* capturer, uint32 /*ssrc*/);
bool SetLocalRenderer(VideoRenderer* renderer);
CaptureResult SetCapture(bool capture);
sigslot::repeater2<VideoCapturer*, CaptureResult> SignalCaptureResult;
virtual VideoCapturer* CreateVideoCapturer(const Device& device);
CaptureResult UpdateCapturingState();
bool IsCapturing() const;
void OnFrameCaptured(VideoCapturer* capturer, const CapturedFrame* frame);
// Set the VoiceEngine for A/V sync. This can only be called before Init.
bool SetVoiceEngine(WebRtcVoiceEngine* voice_engine);
// Enable the render module with timing control.
bool EnableTimedRender();
bool RegisterProcessor(VideoProcessor* video_processor);
bool UnregisterProcessor(VideoProcessor* video_processor);
// Functions called by WebRtcVideoMediaChannel.
ViEWrapper* vie() { return vie_wrapper_.get(); }
const VideoFormat& default_codec_format() const {
return default_codec_format_;
}
int GetLastEngineError();
bool FindCodec(const VideoCodec& in);
bool CanSendCodec(const VideoCodec& in, const VideoCodec& current,
VideoCodec* out);
void RegisterChannel(WebRtcVideoMediaChannel* channel);
void UnregisterChannel(WebRtcVideoMediaChannel* channel);
void ConvertToCricketVideoCodec(const webrtc::VideoCodec& in_codec,
VideoCodec& out_codec);
bool ConvertFromCricketVideoCodec(const VideoCodec& in_codec,
webrtc::VideoCodec& out_codec);
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
int GetNumOfChannels();
protected:
// When a video processor registers with the engine.
// SignalMediaFrame will be invoked for every video frame.
sigslot::signal2<uint32, VideoFrame*> SignalMediaFrame;
private:
typedef std::vector<WebRtcVideoMediaChannel*> VideoChannels;
struct VideoCodecPref {
const char* name;
int payload_type;
int pref;
};
static const VideoCodecPref kVideoCodecPrefs[];
static const VideoFormatPod kVideoFormats[];
static const VideoFormatPod kDefaultVideoFormat;
void Construct(ViEWrapper* vie_wrapper,
ViETraceWrapper* tracing,
WebRtcVoiceEngine* voice_engine);
bool SetDefaultCodec(const VideoCodec& codec);
bool RebuildCodecList(const VideoCodec& max_codec);
void ApplyLogging();
bool InitVideoEngine();
bool SetCapturer(VideoCapturer* capturer, bool own_capturer);
// webrtc::ViEBaseObserver implementation.
virtual void PerformanceAlarm(const unsigned int cpu_load);
// webrtc::TraceCallback implementation.
virtual void Print(const webrtc::TraceLevel level, const char* trace_string,
const int length);
void ClearCapturer();
talk_base::scoped_ptr<ViEWrapper> vie_wrapper_;
bool vie_wrapper_base_initialized_;
talk_base::scoped_ptr<ViETraceWrapper> tracing_;
WebRtcVoiceEngine* voice_engine_;
int log_level_;
talk_base::scoped_ptr<webrtc::VideoRender> render_module_;
std::vector<VideoCodec> video_codecs_;
VideoFormat default_codec_format_;
bool initialized_;
talk_base::CriticalSection channels_crit_;
VideoChannels channels_;
bool owns_capturer_;
VideoCapturer* video_capturer_;
bool capture_started_;
int local_renderer_w_;
int local_renderer_h_;
VideoRenderer* local_renderer_;
// Critical section to protect the media processor register/unregister
// while processing a frame
talk_base::CriticalSection signal_media_critical_;
};
class WebRtcVideoMediaChannel : public VideoMediaChannel,
public webrtc::Transport {
public:
WebRtcVideoMediaChannel(
WebRtcVideoEngine* engine, VoiceMediaChannel* voice_channel);
~WebRtcVideoMediaChannel();
bool Init();
WebRtcVideoEngine* engine() { return engine_; }
VoiceMediaChannel* voice_channel() { return voice_channel_; }
int video_channel() const { return vie_channel_; }
bool sending() const { return sending_; }
// VideoMediaChannel implementation
virtual bool SetRecvCodecs(const std::vector<VideoCodec> &codecs);
virtual bool SetSendCodecs(const std::vector<VideoCodec> &codecs);
virtual bool SetRender(bool render);
virtual bool SetSend(bool send);
virtual bool AddStream(uint32 ssrc, uint32 voice_ssrc);
virtual bool RemoveStream(uint32 ssrc);
virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
virtual bool GetStats(VideoMediaInfo* info);
virtual bool AddScreencast(uint32 ssrc, talk_base::WindowId id) {
return false;
}
virtual bool RemoveScreencast(uint32 ssrc) {
return false;
}
virtual bool SendIntraFrame();
virtual bool RequestIntraFrame();
virtual void OnPacketReceived(talk_base::Buffer* packet);
virtual void OnRtcpReceived(talk_base::Buffer* packet);
virtual void SetSendSsrc(uint32 id);
virtual bool SetRtcpCName(const std::string& cname);
virtual bool Mute(bool on);
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
return false;
}
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
return false;
}
virtual bool SetSendBandwidth(bool autobw, int bps);
virtual bool SetOptions(int options);
virtual void SetInterface(NetworkInterface* iface);
// Public functions for use by tests and other specialized code.
uint32 send_ssrc() const { return 0; }
bool GetRenderer(uint32 ssrc, VideoRenderer** renderer) {
*renderer = NULL;
return false;
}
bool SendFrame(uint32 ssrc, const VideoFrame* frame);
// Thunk functions for use with HybridVideoEngine
void OnLocalFrame(VideoCapturer* capturer, const VideoFrame* frame) {
SendFrame(0, frame);
}
void OnLocalFrameFormat(VideoCapturer* capturer, const VideoFormat* format) {
}
protected:
int GetLastEngineError() { return engine()->GetLastEngineError(); }
virtual int SendPacket(int channel, const void* data, int len);
virtual int SendRTCPPacket(int channel, const void* data, int len);
private:
bool EnableRtcp();
bool EnablePli();
bool EnableTmmbr();
bool EnableNack();
bool SetNackFec(int red_payload_type, int fec_payload_type);
bool SetSendCodec(const webrtc::VideoCodec& codec,
int min_bitrate,
int start_bitrate,
int max_bitrate);
bool ResetRecvCodecs(int channel);
WebRtcVideoEngine* engine_;
VoiceMediaChannel* voice_channel_;
int vie_channel_;
int vie_capture_;
webrtc::ViEExternalCapture* external_capture_;
bool sending_;
bool render_started_;
bool muted_; // Flag to tell if we need to mute video.
int send_min_bitrate_;
int send_start_bitrate_;
int send_max_bitrate_;
talk_base::scoped_ptr<webrtc::VideoCodec> send_codec_;
talk_base::scoped_ptr<WebRtcRenderAdapter> remote_renderer_;
talk_base::scoped_ptr<WebRtcDecoderObserver> decoder_observer_;
talk_base::scoped_ptr<WebRtcEncoderObserver> encoder_observer_;
talk_base::scoped_ptr<LocalStreamInfo> local_stream_info_;
};
} // namespace cricket
#endif // TALK_SESSION_PHONE_WEBRTCVIDEOENGINE_H_