2c0fb05f16
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d
434 lines
15 KiB
C++
434 lines
15 KiB
C++
/*
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* libjingle
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* Copyright 2014 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#ifndef TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
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#define TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
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#include <map>
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#include <vector>
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#include <string>
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#include "webrtc/base/cpumonitor.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "talk/media/base/mediaengine.h"
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#include "talk/media/webrtc/webrtcvideochannelfactory.h"
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#include "webrtc/common_video/interface/i420_video_frame.h"
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#include "webrtc/system_wrappers/interface/thread_annotations.h"
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#include "webrtc/transport.h"
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#include "webrtc/video_renderer.h"
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#include "webrtc/video_send_stream.h"
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#include "webrtc/video_receive_stream.h"
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namespace webrtc {
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class Call;
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class VideoCaptureModule;
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class VideoDecoder;
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class VideoEncoder;
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class VideoRender;
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class VideoSendStreamInput;
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class VideoReceiveStream;
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}
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namespace rtc {
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class CpuMonitor;
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class Thread;
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} // namespace rtc
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namespace cricket {
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class VideoCapturer;
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class VideoFrame;
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class VideoProcessor;
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class VideoRenderer;
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class VoiceMediaChannel;
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class WebRtcVideoChannel2;
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class WebRtcDecoderObserver;
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class WebRtcEncoderObserver;
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class WebRtcLocalStreamInfo;
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class WebRtcRenderAdapter;
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class WebRtcVideoChannelRecvInfo;
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class WebRtcVideoChannelSendInfo;
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class WebRtcVideoDecoderFactory;
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class WebRtcVoiceEngine;
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struct CapturedFrame;
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struct Device;
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class WebRtcVideoEngine2;
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class WebRtcVideoChannel2;
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class WebRtcVideoRenderer;
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class UnsignalledSsrcHandler {
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public:
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enum Action {
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kDropPacket,
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kDeliverPacket,
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};
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virtual Action OnUnsignalledSsrc(VideoMediaChannel* engine,
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uint32_t ssrc) = 0;
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};
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// TODO(pbos): Remove, use external handlers only.
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class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
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public:
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DefaultUnsignalledSsrcHandler();
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virtual Action OnUnsignalledSsrc(VideoMediaChannel* engine,
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uint32_t ssrc) OVERRIDE;
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VideoRenderer* GetDefaultRenderer() const;
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void SetDefaultRenderer(VideoMediaChannel* channel, VideoRenderer* renderer);
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private:
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uint32_t default_recv_ssrc_;
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VideoRenderer* default_renderer_;
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};
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class WebRtcVideoEncoderFactory2 {
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public:
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virtual ~WebRtcVideoEncoderFactory2();
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virtual std::vector<webrtc::VideoStream> CreateVideoStreams(
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const VideoCodec& codec,
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const VideoOptions& options,
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size_t num_streams);
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virtual webrtc::VideoEncoder* CreateVideoEncoder(
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const VideoCodec& codec,
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const VideoOptions& options);
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virtual void* CreateVideoEncoderSettings(
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const VideoCodec& codec,
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const VideoOptions& options);
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virtual void DestroyVideoEncoderSettings(const VideoCodec& codec,
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void* encoder_settings);
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virtual bool SupportsCodec(const cricket::VideoCodec& codec);
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};
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// WebRtcVideoEngine2 is used for the new native WebRTC Video API (webrtc:1667).
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class WebRtcVideoEngine2 : public sigslot::has_slots<> {
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public:
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// Creates the WebRtcVideoEngine2 with internal VideoCaptureModule.
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WebRtcVideoEngine2();
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// Custom WebRtcVideoChannelFactory for testing purposes.
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explicit WebRtcVideoEngine2(WebRtcVideoChannelFactory* channel_factory);
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~WebRtcVideoEngine2();
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// Basic video engine implementation.
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bool Init(rtc::Thread* worker_thread);
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void Terminate();
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int GetCapabilities();
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bool SetOptions(const VideoOptions& options);
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bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
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VideoEncoderConfig GetDefaultEncoderConfig() const;
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WebRtcVideoChannel2* CreateChannel(VoiceMediaChannel* voice_channel);
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const std::vector<VideoCodec>& codecs() const;
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const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
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void SetLogging(int min_sev, const char* filter);
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bool EnableTimedRender();
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// No-op, never used.
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bool SetLocalRenderer(VideoRenderer* renderer);
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// This is currently ignored.
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sigslot::repeater2<VideoCapturer*, CaptureState> SignalCaptureStateChange;
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// Set the VoiceEngine for A/V sync. This can only be called before Init.
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bool SetVoiceEngine(WebRtcVoiceEngine* voice_engine);
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// Functions called by WebRtcVideoChannel2.
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const VideoFormat& default_codec_format() const {
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return default_codec_format_;
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}
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bool FindCodec(const VideoCodec& in);
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bool CanSendCodec(const VideoCodec& in,
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const VideoCodec& current,
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VideoCodec* out);
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// Check whether the supplied trace should be ignored.
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bool ShouldIgnoreTrace(const std::string& trace);
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VideoFormat GetStartCaptureFormat() const { return default_codec_format_; }
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rtc::CpuMonitor* cpu_monitor() { return cpu_monitor_.get(); }
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virtual WebRtcVideoEncoderFactory2* GetVideoEncoderFactory();
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private:
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void Construct(WebRtcVideoChannelFactory* channel_factory,
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WebRtcVoiceEngine* voice_engine,
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rtc::CpuMonitor* cpu_monitor);
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rtc::Thread* worker_thread_;
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WebRtcVoiceEngine* voice_engine_;
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std::vector<VideoCodec> video_codecs_;
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std::vector<RtpHeaderExtension> rtp_header_extensions_;
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VideoFormat default_codec_format_;
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bool initialized_;
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bool capture_started_;
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// Critical section to protect the media processor register/unregister
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// while processing a frame
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rtc::CriticalSection signal_media_critical_;
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rtc::scoped_ptr<rtc::CpuMonitor> cpu_monitor_;
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WebRtcVideoChannelFactory* channel_factory_;
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WebRtcVideoEncoderFactory2 default_video_encoder_factory_;
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};
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class WebRtcVideoChannel2 : public rtc::MessageHandler,
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public VideoMediaChannel,
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public webrtc::newapi::Transport {
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public:
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WebRtcVideoChannel2(WebRtcVideoEngine2* engine,
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VoiceMediaChannel* voice_channel,
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WebRtcVideoEncoderFactory2* encoder_factory);
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// For testing purposes insert a pre-constructed call to verify that
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// WebRtcVideoChannel2 calls the correct corresponding methods.
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WebRtcVideoChannel2(webrtc::Call* call,
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WebRtcVideoEngine2* engine,
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WebRtcVideoEncoderFactory2* encoder_factory);
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~WebRtcVideoChannel2();
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bool Init();
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// VideoMediaChannel implementation
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virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) OVERRIDE;
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virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs) OVERRIDE;
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virtual bool GetSendCodec(VideoCodec* send_codec) OVERRIDE;
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virtual bool SetSendStreamFormat(uint32 ssrc,
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const VideoFormat& format) OVERRIDE;
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virtual bool SetRender(bool render) OVERRIDE;
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virtual bool SetSend(bool send) OVERRIDE;
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virtual bool AddSendStream(const StreamParams& sp) OVERRIDE;
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virtual bool RemoveSendStream(uint32 ssrc) OVERRIDE;
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virtual bool AddRecvStream(const StreamParams& sp) OVERRIDE;
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virtual bool RemoveRecvStream(uint32 ssrc) OVERRIDE;
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virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) OVERRIDE;
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virtual bool GetStats(const StatsOptions& options,
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VideoMediaInfo* info) OVERRIDE;
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virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) OVERRIDE;
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virtual bool SendIntraFrame() OVERRIDE;
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virtual bool RequestIntraFrame() OVERRIDE;
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virtual void OnPacketReceived(rtc::Buffer* packet,
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const rtc::PacketTime& packet_time)
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OVERRIDE;
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virtual void OnRtcpReceived(rtc::Buffer* packet,
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const rtc::PacketTime& packet_time)
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OVERRIDE;
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virtual void OnReadyToSend(bool ready) OVERRIDE;
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virtual bool MuteStream(uint32 ssrc, bool mute) OVERRIDE;
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// Set send/receive RTP header extensions. This must be done before creating
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// streams as it only has effect on future streams.
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virtual bool SetRecvRtpHeaderExtensions(
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const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
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virtual bool SetSendRtpHeaderExtensions(
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const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
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virtual bool SetStartSendBandwidth(int bps) OVERRIDE;
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virtual bool SetMaxSendBandwidth(int bps) OVERRIDE;
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virtual bool SetOptions(const VideoOptions& options) OVERRIDE;
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virtual bool GetOptions(VideoOptions* options) const OVERRIDE {
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*options = options_;
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return true;
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}
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virtual void SetInterface(NetworkInterface* iface) OVERRIDE;
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virtual void UpdateAspectRatio(int ratio_w, int ratio_h) OVERRIDE;
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virtual void OnMessage(rtc::Message* msg) OVERRIDE;
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// Implemented for VideoMediaChannelTest.
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bool sending() const { return sending_; }
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uint32 GetDefaultSendChannelSsrc() { return default_send_ssrc_; }
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bool GetRenderer(uint32 ssrc, VideoRenderer** renderer);
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private:
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void ConfigureReceiverRtp(webrtc::VideoReceiveStream::Config* config,
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const StreamParams& sp) const;
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struct VideoCodecSettings {
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VideoCodecSettings();
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VideoCodec codec;
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webrtc::FecConfig fec;
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int rtx_payload_type;
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};
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// Wrapper for the sender part, this is where the capturer is connected and
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// frames are then converted from cricket frames to webrtc frames.
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class WebRtcVideoSendStream : public sigslot::has_slots<> {
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public:
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WebRtcVideoSendStream(
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webrtc::Call* call,
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WebRtcVideoEncoderFactory2* encoder_factory,
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const VideoOptions& options,
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const Settable<VideoCodecSettings>& codec_settings,
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const StreamParams& sp,
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const std::vector<webrtc::RtpExtension>& rtp_extensions);
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~WebRtcVideoSendStream();
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void SetOptions(const VideoOptions& options);
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void SetCodec(const VideoCodecSettings& codec);
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void SetRtpExtensions(
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const std::vector<webrtc::RtpExtension>& rtp_extensions);
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void InputFrame(VideoCapturer* capturer, const VideoFrame* frame);
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bool SetCapturer(VideoCapturer* capturer);
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bool SetVideoFormat(const VideoFormat& format);
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bool MuteStream(bool mute);
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bool DisconnectCapturer();
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void Start();
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void Stop();
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VideoSenderInfo GetVideoSenderInfo();
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private:
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// Parameters needed to reconstruct the underlying stream.
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// webrtc::VideoSendStream doesn't support setting a lot of options on the
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// fly, so when those need to be changed we tear down and reconstruct with
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// similar parameters depending on which options changed etc.
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struct VideoSendStreamParameters {
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VideoSendStreamParameters(
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const webrtc::VideoSendStream::Config& config,
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const VideoOptions& options,
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const Settable<VideoCodecSettings>& codec_settings);
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webrtc::VideoSendStream::Config config;
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VideoOptions options;
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Settable<VideoCodecSettings> codec_settings;
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// Sent resolutions + bitrates etc. by the underlying VideoSendStream,
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// typically changes when setting a new resolution or reconfiguring
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// bitrates.
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std::vector<webrtc::VideoStream> video_streams;
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};
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void SetCodecAndOptions(const VideoCodecSettings& codec,
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const VideoOptions& options);
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void RecreateWebRtcStream();
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void SetDimensions(int width, int height);
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webrtc::Call* const call_;
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WebRtcVideoEncoderFactory2* const encoder_factory_;
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rtc::CriticalSection lock_;
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webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
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VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
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VideoCapturer* capturer_ GUARDED_BY(lock_);
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bool sending_ GUARDED_BY(lock_);
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bool muted_ GUARDED_BY(lock_);
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VideoFormat format_ GUARDED_BY(lock_);
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rtc::CriticalSection frame_lock_;
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webrtc::I420VideoFrame video_frame_ GUARDED_BY(frame_lock_);
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};
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// Wrapper for the receiver part, contains configs etc. that are needed to
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// reconstruct the underlying VideoReceiveStream. Also serves as a wrapper
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// between webrtc::VideoRenderer and cricket::VideoRenderer.
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class WebRtcVideoReceiveStream : public webrtc::VideoRenderer {
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public:
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WebRtcVideoReceiveStream(
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webrtc::Call*,
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const webrtc::VideoReceiveStream::Config& config,
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const std::vector<VideoCodecSettings>& recv_codecs);
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~WebRtcVideoReceiveStream();
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void SetRecvCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
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void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
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virtual void RenderFrame(const webrtc::I420VideoFrame& frame,
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int time_to_render_ms) OVERRIDE;
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void SetRenderer(cricket::VideoRenderer* renderer);
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cricket::VideoRenderer* GetRenderer();
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VideoReceiverInfo GetVideoReceiverInfo();
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private:
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void SetSize(int width, int height);
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void RecreateWebRtcStream();
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webrtc::Call* const call_;
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webrtc::VideoReceiveStream* stream_;
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webrtc::VideoReceiveStream::Config config_;
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rtc::CriticalSection renderer_lock_;
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cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_);
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int last_width_ GUARDED_BY(renderer_lock_);
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int last_height_ GUARDED_BY(renderer_lock_);
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};
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void Construct(webrtc::Call* call, WebRtcVideoEngine2* engine);
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void SetDefaultOptions();
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virtual bool SendRtp(const uint8_t* data, size_t len) OVERRIDE;
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virtual bool SendRtcp(const uint8_t* data, size_t len) OVERRIDE;
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void StartAllSendStreams();
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void StopAllSendStreams();
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static std::vector<VideoCodecSettings> MapCodecs(
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const std::vector<VideoCodec>& codecs);
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std::vector<VideoCodecSettings> FilterSupportedCodecs(
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const std::vector<VideoCodecSettings>& mapped_codecs);
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void FillSenderStats(VideoMediaInfo* info);
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void FillReceiverStats(VideoMediaInfo* info);
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void FillBandwidthEstimationStats(VideoMediaInfo* info);
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uint32_t rtcp_receiver_report_ssrc_;
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bool sending_;
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rtc::scoped_ptr<webrtc::Call> call_;
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uint32_t default_send_ssrc_;
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DefaultUnsignalledSsrcHandler default_unsignalled_ssrc_handler_;
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UnsignalledSsrcHandler* const unsignalled_ssrc_handler_;
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// Using primary-ssrc (first ssrc) as key.
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std::map<uint32, WebRtcVideoSendStream*> send_streams_;
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std::map<uint32, WebRtcVideoReceiveStream*> receive_streams_;
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Settable<VideoCodecSettings> send_codec_;
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std::vector<webrtc::RtpExtension> send_rtp_extensions_;
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WebRtcVideoEncoderFactory2* const encoder_factory_;
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std::vector<VideoCodecSettings> recv_codecs_;
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std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
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VideoOptions options_;
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};
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} // namespace cricket
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#endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_H_
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