diff options
Diffstat (limited to 'chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.h')
| -rw-r--r-- | chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.h | 52 |
1 files changed, 33 insertions, 19 deletions
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.h index 289903a0722..775f4e46de1 100644 --- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.h +++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.h @@ -45,6 +45,7 @@ #error "Bogus include." #endif + namespace webrtc { class VideoCaptureModule; class VideoDecoder; @@ -60,12 +61,13 @@ class CpuMonitor; namespace cricket { +class CoordinatedVideoAdapter; +class ViETraceWrapper; +class ViEWrapper; class VideoCapturer; class VideoFrame; class VideoProcessor; class VideoRenderer; -class ViETraceWrapper; -class ViEWrapper; class VoiceMediaChannel; class WebRtcDecoderObserver; class WebRtcEncoderObserver; @@ -199,6 +201,7 @@ class WebRtcVideoEngine : public sigslot::has_slots<>, void SetTraceFilter(int filter); void SetTraceOptions(const std::string& options); bool InitVideoEngine(); + bool VerifyApt(const VideoCodec& in, int expected_apt) const; // webrtc::TraceCallback implementation. virtual void Print(webrtc::TraceLevel level, const char* trace, int length); @@ -227,10 +230,6 @@ class WebRtcVideoEngine : public sigslot::has_slots<>, 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_; - talk_base::scoped_ptr<talk_base::CpuMonitor> cpu_monitor_; }; @@ -248,6 +247,9 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, int video_channel() const { return vie_channel_; } bool sending() const { return sending_; } + // Public for testing purpose. + uint32 GetDefaultChannelSsrc(); + // VideoMediaChannel implementation virtual bool SetRecvCodecs(const std::vector<VideoCodec> &codecs); virtual bool SetSendCodecs(const std::vector<VideoCodec> &codecs); @@ -261,7 +263,7 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, virtual bool AddRecvStream(const StreamParams& sp); virtual bool RemoveRecvStream(uint32 ssrc); virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer); - virtual bool GetStats(VideoMediaInfo* info); + virtual bool GetStats(const StatsOptions& options, VideoMediaInfo* info); virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer); virtual bool SendIntraFrame(); virtual bool RequestIntraFrame(); @@ -276,7 +278,9 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, const std::vector<RtpHeaderExtension>& extensions); virtual bool SetSendRtpHeaderExtensions( const std::vector<RtpHeaderExtension>& extensions); - virtual bool SetSendBandwidth(bool autobw, int bps); + virtual int GetRtpSendTimeExtnId() const; + virtual bool SetStartSendBandwidth(int bps); + virtual bool SetMaxSendBandwidth(int bps); virtual bool SetOptions(const VideoOptions &options); virtual bool GetOptions(VideoOptions *options) const { *options = options_; @@ -288,6 +292,7 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, // Public functions for use by tests and other specialized code. uint32 send_ssrc() const { return 0; } bool GetRenderer(uint32 ssrc, VideoRenderer** renderer); + bool GetVideoAdapter(uint32 ssrc, CoordinatedVideoAdapter** video_adapter); void SendFrame(VideoCapturer* capturer, const VideoFrame* frame); bool SendFrame(WebRtcVideoChannelSendInfo* channel_info, const VideoFrame* frame, bool is_screencast); @@ -309,6 +314,7 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, private: typedef std::map<uint32, WebRtcVideoChannelRecvInfo*> RecvChannelMap; typedef std::map<uint32, WebRtcVideoChannelSendInfo*> SendChannelMap; + typedef std::map<uint32, uint32> SsrcMap; typedef int (webrtc::ViERTP_RTCP::* ExtensionSetterFunction)(int, bool, int); enum MediaDirection { MD_RECV, MD_SEND, MD_SENDRECV }; @@ -323,32 +329,32 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, // returning false. bool CreateChannel(uint32 ssrc_key, MediaDirection direction, int* channel_id); + bool CreateUnsignalledRecvChannel(uint32 ssrc_key, int* channel_id); bool ConfigureChannel(int channel_id, MediaDirection direction, uint32 ssrc_key); bool ConfigureReceiving(int channel_id, uint32 remote_ssrc_key); bool ConfigureSending(int channel_id, uint32 local_ssrc_key); bool SetNackFec(int channel_id, int red_payload_type, int fec_payload_type, bool nack_enabled); - bool SetSendCodec(const webrtc::VideoCodec& codec, int min_bitrate, - int start_bitrate, int max_bitrate); + bool SetSendCodec(const webrtc::VideoCodec& codec); bool SetSendCodec(WebRtcVideoChannelSendInfo* send_channel, - const webrtc::VideoCodec& codec, int min_bitrate, - int start_bitrate, int max_bitrate); + const webrtc::VideoCodec& codec); void LogSendCodecChange(const std::string& reason); // Prepares the channel with channel id |info->channel_id()| to receive all // codecs in |receive_codecs_| and start receive packets. bool SetReceiveCodecs(WebRtcVideoChannelRecvInfo* info); // Returns the channel number that receives the stream with SSRC |ssrc|. int GetRecvChannelNum(uint32 ssrc); + bool MaybeSetRtxSsrc(const StreamParams& sp, int channel_id); // Given captured video frame size, checks if we need to reset vie send codec. // |reset| is set to whether resetting has happened on vie or not. // Returns false on error. bool MaybeResetVieSendCodec(WebRtcVideoChannelSendInfo* send_channel, int new_width, int new_height, bool is_screencast, bool* reset); - // Checks the current bitrate estimate and modifies the start bitrate - // accordingly. - void MaybeChangeStartBitrate(int channel_id, webrtc::VideoCodec* video_codec); + // Checks the current bitrate estimate and modifies the bitrates + // accordingly, including converting kAutoBandwidth to the correct defaults. + void MaybeChangeBitrates(int channel_id, webrtc::VideoCodec* video_codec); // Helper function for starting the sending of media on all channels or // |channel_id|. Note that these two function do not change |sending_|. bool StartSend(); @@ -376,7 +382,6 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, bool IsDefaultChannel(int channel_id) const { return channel_id == vie_channel_; } - uint32 GetDefaultChannelSsrc(); bool DeleteSendChannel(uint32 ssrc_key); @@ -412,6 +417,8 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, // to one send channel, i.e. the last send channel. void MaybeDisconnectCapturer(VideoCapturer* capturer); + bool RemoveRecvStreamInternal(uint32 ssrc); + // Global state. WebRtcVideoEngine* engine_; VoiceMediaChannel* voice_channel_; @@ -427,10 +434,20 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, // work properly), resides in both recv_channels_ and send_channels_ with the // ssrc key 0. RecvChannelMap recv_channels_; // Contains all receive channels. + // A map from the SSRCs on which RTX packets are received to the media SSRCs + // the RTX packets are associated with. RTX packets will be delivered to the + // streams matching the primary SSRC. + SsrcMap rtx_to_primary_ssrc_; std::vector<webrtc::VideoCodec> receive_codecs_; + // A map from codec payload types to their associated payload types, if any. + // TODO(holmer): This is a temporary solution until webrtc::VideoCodec has + // an associated payload type member, when it does we can rely on + // receive_codecs_. + std::map<int, int> associated_payload_types_; bool render_started_; uint32 first_receive_ssrc_; std::vector<RtpHeaderExtension> receive_extensions_; + int num_unsignalled_recv_channels_; // Global send side state. SendChannelMap send_channels_; @@ -438,9 +455,6 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler, int send_rtx_type_; int send_red_type_; int send_fec_type_; - int send_min_bitrate_; - int send_start_bitrate_; - int send_max_bitrate_; bool sending_; std::vector<RtpHeaderExtension> send_extensions_; |