Refactored ViEChannel.
Pointers/references and types will be in a future CL. Review URL: http://webrtc-codereview.appspot.com/322016 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1205 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
parent
c74b2861f3
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@ -8,38 +8,26 @@
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* be found in the AUTHORS file in the root of the source tree.
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*/
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/*
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* vie_channel.h
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*/
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// vie_channel.h
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#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_CHANNEL_H_
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#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_CHANNEL_H_
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#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
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#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
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#include <list>
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// Defines
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#include "vie_defines.h"
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#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "modules/udp_transport/interface/udp_transport.h"
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#include "modules/video_coding/main/interface/video_coding_defines.h"
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#include "system_wrappers/interface/tick_util.h"
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#include "typedefs.h"
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#include "vie_network.h"
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#include "vie_rtp_rtcp.h"
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#include "rtp_rtcp_defines.h"
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#include "udp_transport.h"
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#include "video_coding_defines.h"
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#ifdef WEBRTC_SRTP
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#include "SrtpModule.h"
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#endif
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#include "tick_util.h"
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#include "vie_frame_provider_base.h"
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#include "vie_file_recorder.h"
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#include "video_engine/main/interface/vie_network.h"
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#include "video_engine/main/interface/vie_rtp_rtcp.h"
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#include "video_engine/vie_defines.h"
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#include "video_engine/vie_file_recorder.h"
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#include "video_engine/vie_frame_provider_base.h"
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// Forward declarations
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namespace webrtc {
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class SrtpModule;
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class VideoRenderCallback;
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namespace webrtc
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{
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class CriticalSectionWrapper;
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class Encryption;
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class ProcessThread;
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@ -47,6 +35,7 @@ class RtpRtcp;
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class ThreadWrapper;
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class VideoCodingModule;
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class VideoDecoder;
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class VideoRenderCallback;
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class ViEDecoderObserver;
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class ViEEffectFilter;
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class ViENetworkObserver;
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@ -57,437 +46,353 @@ class ViESender;
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class ViESyncModule;
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class VoEVideoSync;
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class ViEChannel:
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public VCMFrameTypeCallback, // VCM Module
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public VCMReceiveCallback, // VCM Module
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public VCMReceiveStatisticsCallback, // VCM Module
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public VCMPacketRequestCallback, // VCM Module
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public VCMFrameStorageCallback, // VCM Module
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public RtcpFeedback, // RTP/RTCP Module
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public RtpFeedback, // RTP/RTCP Module
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public ViEFrameProviderBase
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{
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public:
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ViEChannel(WebRtc_Word32 channelId, WebRtc_Word32 engineId,
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WebRtc_UWord32 numberOfCores,
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ProcessThread& moduleProcessThread);
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~ViEChannel();
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class ViEChannel
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: public VCMFrameTypeCallback,
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public VCMReceiveCallback,
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public VCMReceiveStatisticsCallback,
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public VCMPacketRequestCallback,
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public VCMFrameStorageCallback,
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public RtcpFeedback,
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public RtpFeedback,
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public ViEFrameProviderBase {
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public:
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ViEChannel(WebRtc_Word32 channel_id,
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WebRtc_Word32 engine_id,
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WebRtc_UWord32 number_of_cores,
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ProcessThread& module_process_thread);
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~ViEChannel();
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WebRtc_Word32 Init();
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// Sets the encoder to use for the channel. |new_stream| indicates the encoder
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// type has changed and we should start a new RTP stream.
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WebRtc_Word32 SetSendCodec(const VideoCodec& video_codec,
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bool new_stream = true);
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WebRtc_Word32 SetReceiveCodec(const VideoCodec& video_codec);
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WebRtc_Word32 GetReceiveCodec(VideoCodec& video_codec);
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WebRtc_Word32 RegisterCodecObserver(ViEDecoderObserver* observer);
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// Registers an external decoder. |decoder_render| is set to true if the
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// decoder will do the rendering. If |decoder_render| is set,|render_delay|
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// indicates the time needed to decode and render a frame.
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WebRtc_Word32 RegisterExternalDecoder(const WebRtc_UWord8 pl_type,
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VideoDecoder* decoder,
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bool decoder_render,
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WebRtc_Word32 render_delay);
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WebRtc_Word32 DeRegisterExternalDecoder(const WebRtc_UWord8 pl_type);
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WebRtc_Word32 ReceiveCodecStatistics(WebRtc_UWord32& num_key_frames,
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WebRtc_UWord32& num_delta_frames);
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WebRtc_UWord32 DiscardedPackets() const;
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// Only affects calls to SetReceiveCodec done after this call.
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WebRtc_Word32 WaitForKeyFrame(bool wait);
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// If enabled, a key frame request will be sent as soon as there are lost
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// packets. If |only_key_frames| are set, requests are only sent for loss in
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// key frames.
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WebRtc_Word32 SetSignalPacketLossStatus(bool enable, bool only_key_frames);
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WebRtc_Word32 SetRTCPMode(const RTCPMethod rtcp_mode);
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WebRtc_Word32 GetRTCPMode(RTCPMethod& rtcp_mode);
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WebRtc_Word32 SetNACKStatus(const bool enable);
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WebRtc_Word32 SetFECStatus(const bool enable,
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const unsigned char payload_typeRED,
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const unsigned char payload_typeFEC);
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WebRtc_Word32 SetHybridNACKFECStatus(const bool enable,
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const unsigned char payload_typeRED,
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const unsigned char payload_typeFEC);
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WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
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WebRtc_Word32 EnableTMMBR(const bool enable);
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WebRtc_Word32 EnableKeyFrameRequestCallback(const bool enable);
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// Sets SSRC for outgoing stream.
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WebRtc_Word32 SetSSRC(const WebRtc_UWord32 SSRC,
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const StreamType usage,
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const unsigned char simulcast_idx);
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// Gets SSRC for outgoing stream.
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WebRtc_Word32 GetLocalSSRC(WebRtc_UWord32& SSRC);
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// Gets SSRC for the incoming stream.
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WebRtc_Word32 GetRemoteSSRC(WebRtc_UWord32& SSRC);
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// Gets the CSRC for the incoming stream.
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WebRtc_Word32 GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]);
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// Sets the starting sequence number, must be called before StartSend.
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WebRtc_Word32 SetStartSequenceNumber(WebRtc_UWord16 sequence_number);
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// Sets the CName for the outgoing stream on the channel.
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WebRtc_Word32 SetRTCPCName(const WebRtc_Word8 rtcp_cname[]);
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// Gets the CName for the outgoing stream on the channel.
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WebRtc_Word32 GetRTCPCName(WebRtc_Word8 rtcp_cname[]);
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// Gets the CName of the incoming stream.
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WebRtc_Word32 GetRemoteRTCPCName(WebRtc_Word8 rtcp_cname[]);
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WebRtc_Word32 RegisterRtpObserver(ViERTPObserver* observer);
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WebRtc_Word32 RegisterRtcpObserver(ViERTCPObserver* observer);
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WebRtc_Word32 SendApplicationDefinedRTCPPacket(
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const WebRtc_UWord8 sub_type,
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WebRtc_UWord32 name,
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const WebRtc_UWord8* data,
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WebRtc_UWord16 data_length_in_bytes);
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// Gets statistics sent in RTCP packets to remote side.
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WebRtc_Word32 GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
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WebRtc_UWord32& cumulative_lost,
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WebRtc_UWord32& extended_max,
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WebRtc_UWord32& jitter_samples,
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WebRtc_Word32& rtt_ms);
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// Gets statistics received in RTCP packets from remote side.
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WebRtc_Word32 GetReceivedRtcpStatistics(WebRtc_UWord16& fraction_lost,
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WebRtc_UWord32& cumulative_lost,
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WebRtc_UWord32& extended_max,
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WebRtc_UWord32& jitter_samples,
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WebRtc_Word32& rtt_ms);
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// Gets sent/received packets statistics.
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WebRtc_Word32 GetRtpStatistics(WebRtc_UWord32& bytes_sent,
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WebRtc_UWord32& packets_sent,
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WebRtc_UWord32& bytes_received,
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WebRtc_UWord32& packets_received) const;
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void GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
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WebRtc_UWord32& video_bitrate_sent,
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WebRtc_UWord32& fec_bitrate_sent,
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WebRtc_UWord32& nackBitrateSent) const;
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WebRtc_Word32 SetKeepAliveStatus(const bool enable,
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const WebRtc_Word8 unknown_payload_type,
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const WebRtc_UWord16 delta_transmit_timeMS);
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WebRtc_Word32 GetKeepAliveStatus(bool& enable,
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WebRtc_Word8& unknown_payload_type,
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WebRtc_UWord16& delta_transmit_timeMS);
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WebRtc_Word32 StartRTPDump(const char file_nameUTF8[1024],
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RTPDirections direction);
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WebRtc_Word32 StopRTPDump(RTPDirections direction);
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// Implements RtcpFeedback.
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virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
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const WebRtc_Word32 audio_video_offset);
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virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
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const WebRtc_UWord8 sub_type,
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const WebRtc_UWord32 name,
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const WebRtc_UWord16 length,
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const WebRtc_UWord8* data);
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// Implements RtpFeedback.
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virtual WebRtc_Word32 OnInitializeDecoder(
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const WebRtc_Word32 id,
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const WebRtc_Word8 payload_type,
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const WebRtc_Word8 payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int frequency,
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const WebRtc_UWord8 channels,
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const WebRtc_UWord32 rate);
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virtual void OnPacketTimeout(const WebRtc_Word32 id);
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virtual void OnReceivedPacket(const WebRtc_Word32 id,
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const RtpRtcpPacketType packet_type);
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virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
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const RTPAliveType alive);
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virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
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const WebRtc_UWord32 SSRC);
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virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
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const WebRtc_UWord32 CSRC,
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const bool added);
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WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtp_port,
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const WebRtc_UWord16 rtcp_port,
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const WebRtc_Word8* ip_address);
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WebRtc_Word32 GetLocalReceiver(WebRtc_UWord16& rtp_port,
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WebRtc_UWord16& rtcp_port,
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WebRtc_Word8* ip_address) const;
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WebRtc_Word32 SetSendDestination(const WebRtc_Word8* ip_address,
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const WebRtc_UWord16 rtp_port,
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const WebRtc_UWord16 rtcp_port,
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const WebRtc_UWord16 source_rtp_port,
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const WebRtc_UWord16 source_rtcp_port);
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WebRtc_Word32 GetSendDestination(WebRtc_Word8* ip_address,
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WebRtc_UWord16& rtp_port,
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WebRtc_UWord16& rtcp_port,
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WebRtc_UWord16& source_rtp_port,
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WebRtc_UWord16& source_rtcp_port) const;
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WebRtc_Word32 GetSourceInfo(WebRtc_UWord16& rtp_port,
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WebRtc_UWord16& rtcp_port,
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WebRtc_Word8* ip_address,
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WebRtc_UWord32 ip_address_length);
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WebRtc_Word32 StartSend();
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WebRtc_Word32 StopSend();
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bool Sending();
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WebRtc_Word32 StartReceive();
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WebRtc_Word32 StopReceive();
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bool Receiving();
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WebRtc_Word32 RegisterSendTransport(Transport& transport);
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WebRtc_Word32 DeregisterSendTransport();
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// Incoming packet from external transport.
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WebRtc_Word32 ReceivedRTPPacket(const void* rtp_packet,
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const WebRtc_Word32 rtp_packet_length);
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// Incoming packet from external transport.
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WebRtc_Word32 ReceivedRTCPPacket(const void* rtcp_packet,
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const WebRtc_Word32 rtcp_packet_length);
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WebRtc_Word32 EnableIPv6();
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bool IsIPv6Enabled();
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WebRtc_Word32 SetSourceFilter(const WebRtc_UWord16 rtp_port,
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const WebRtc_UWord16 rtcp_port,
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const WebRtc_Word8* ip_address);
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WebRtc_Word32 GetSourceFilter(WebRtc_UWord16& rtp_port,
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WebRtc_UWord16& rtcp_port,
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WebRtc_Word8* ip_address) const;
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WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP, const bool use_set_sockOpt);
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WebRtc_Word32 GetToS(WebRtc_Word32& DSCP, bool& use_set_sockOpt) const;
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WebRtc_Word32 SetSendGQoS(const bool enable,
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const WebRtc_Word32 service_type,
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const WebRtc_UWord32 max_bitrate,
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const WebRtc_Word32 overrideDSCP);
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WebRtc_Word32 GetSendGQoS(bool& enabled, WebRtc_Word32& service_type,
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WebRtc_Word32& overrideDSCP) const;
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// Sets the maximum transfer unit size for the network link, i.e. including
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// IP, UDP and RTP headers.
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WebRtc_Word32 SetMTU(WebRtc_UWord16 mtu);
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// Returns maximum allowed payload size, i.e. the maximum allowed size of
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// encoded data in each packet.
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WebRtc_UWord16 MaxDataPayloadLength() const;
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WebRtc_Word32 SetMaxPacketBurstSize(WebRtc_UWord16 max_number_of_packets);
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WebRtc_Word32 SetPacketBurstSpreadState(bool enable,
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const WebRtc_UWord16 frame_periodMS);
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WebRtc_Word32 SetPacketTimeoutNotification(bool enable,
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WebRtc_UWord32 timeout_seconds);
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WebRtc_Word32 RegisterNetworkObserver(ViENetworkObserver* observer);
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bool NetworkObserverRegistered();
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WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
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const bool enable, const WebRtc_UWord32 sample_time_seconds);
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WebRtc_Word32 SendUDPPacket(const WebRtc_Word8* data,
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const WebRtc_UWord32 length,
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WebRtc_Word32& transmitted_bytes,
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bool use_rtcp_socket);
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WebRtc_Word32 EnableColorEnhancement(bool enable);
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// Register send RTP RTCP module, which will deliver encoded frames to the
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// to the channel RTP module.
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WebRtc_Word32 RegisterSendRtpRtcpModule(RtpRtcp& send_rtp_rtcp_module);
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// Deregisters the send RTP RTCP module, which will stop the encoder input to
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// the channel.
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WebRtc_Word32 DeregisterSendRtpRtcpModule();
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// Implements VCMReceiveCallback.
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virtual WebRtc_Word32 FrameToRender(VideoFrame& video_frame);
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// Implements VCMReceiveCallback.
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virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
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const WebRtc_UWord64 picture_id);
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// Implements VCM.
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virtual WebRtc_Word32 StoreReceivedFrame(
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const EncodedVideoData& frame_to_store);
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// Implements VideoReceiveStatisticsCallback.
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virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bit_rate,
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const WebRtc_UWord32 frame_rate);
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// Implements VideoFrameTypeCallback.
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virtual WebRtc_Word32 FrameTypeRequest(const FrameType frame_type);
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// Implements VideoFrameTypeCallback.
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virtual WebRtc_Word32 SliceLossIndicationRequest(
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const WebRtc_UWord64 picture_id);
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// Implements VideoPacketRequestCallback.
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virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequence_numbers,
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WebRtc_UWord16 length);
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WebRtc_Word32 RegisterExternalEncryption(Encryption* encryption);
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WebRtc_Word32 DeRegisterExternalEncryption();
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WebRtc_Word32 SetVoiceChannel(WebRtc_Word32 ve_channel_id,
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VoEVideoSync* ve_sync_interface);
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WebRtc_Word32 VoiceChannel();
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// Implements ViEFrameProviderBase.
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virtual int FrameCallbackChanged() {return -1;}
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WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
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WebRtc_Word32 SetInverseH263Logic(const bool enable);
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ViEFileRecorder& GetIncomingFileRecorder();
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void ReleaseIncomingFileRecorder();
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protected:
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static bool ChannelDecodeThreadFunction(void* obj);
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bool ChannelDecodeProcess();
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private:
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// Assumed to be protected.
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WebRtc_Word32 StartDecodeThread();
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WebRtc_Word32 StopDecodeThread();
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WebRtc_Word32 ProcessNACKRequest(const bool enable);
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WebRtc_Word32 ProcessFECRequest(const bool enable,
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const unsigned char payload_typeRED,
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const unsigned char payload_typeFEC);
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WebRtc_Word32 channel_id_;
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WebRtc_Word32 engine_id_;
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WebRtc_UWord32 number_of_cores_;
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WebRtc_UWord8 num_socket_threads_;
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// Used for all registered callbacks except rendering.
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CriticalSectionWrapper& callbackCritsect_;
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WebRtc_Word32 Init();
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//-----------------------------------------------------------------
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// Codecs
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//-----------------------------------------------------------------
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WebRtc_Word32 SetSendCodec(const VideoCodec& videoCodec,
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bool newStream = true);
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WebRtc_Word32 SetReceiveCodec(const VideoCodec& videoCodec);
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WebRtc_Word32 GetReceiveCodec(VideoCodec& videoCodec);
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WebRtc_Word32 RegisterCodecObserver(ViEDecoderObserver* observer);
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WebRtc_Word32 RegisterExternalDecoder(const WebRtc_UWord8 plType,
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VideoDecoder* decoder,
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bool decoderRender,
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WebRtc_Word32 renderDelay);
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WebRtc_Word32 DeRegisterExternalDecoder(const WebRtc_UWord8 plType);
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WebRtc_Word32 ReceiveCodecStatistics(WebRtc_UWord32& numKeyFrames,
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WebRtc_UWord32& numDeltaFrames);
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WebRtc_UWord32 DiscardedPackets() const;
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WebRtc_Word32 WaitForKeyFrame(bool wait);
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WebRtc_Word32 SetSignalPacketLossStatus(bool enable, bool onlyKeyFrames);
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//-----------------------------------------------------------------
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// RTP/RTCP
|
||||
//-----------------------------------------------------------------
|
||||
WebRtc_Word32 SetRTCPMode(const RTCPMethod rtcpMode);
|
||||
|
||||
WebRtc_Word32 GetRTCPMode(RTCPMethod& rtcpMode);
|
||||
|
||||
WebRtc_Word32 SetNACKStatus(const bool enable);
|
||||
|
||||
WebRtc_Word32 SetFECStatus(const bool enable,
|
||||
const unsigned char payloadTypeRED,
|
||||
const unsigned char payloadTypeFEC);
|
||||
WebRtc_Word32 SetHybridNACKFECStatus(const bool enable,
|
||||
const unsigned char payloadTypeRED,
|
||||
const unsigned char payloadTypeFEC);
|
||||
|
||||
WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
|
||||
|
||||
WebRtc_Word32 EnableTMMBR(const bool enable);
|
||||
|
||||
WebRtc_Word32 EnableKeyFrameRequestCallback(const bool enable);
|
||||
|
||||
WebRtc_Word32 SetSSRC(const WebRtc_UWord32 SSRC,
|
||||
const StreamType usage,
|
||||
const unsigned char simulcastIdx);
|
||||
|
||||
WebRtc_Word32 GetLocalSSRC(WebRtc_UWord32& SSRC);
|
||||
|
||||
WebRtc_Word32 GetRemoteSSRC(WebRtc_UWord32& SSRC);
|
||||
|
||||
WebRtc_Word32 GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]);
|
||||
|
||||
WebRtc_Word32 SetStartSequenceNumber(WebRtc_UWord16 sequenceNumber);
|
||||
|
||||
WebRtc_Word32 SetRTCPCName(const WebRtc_Word8 rtcpCName[]);
|
||||
|
||||
WebRtc_Word32 GetRTCPCName(WebRtc_Word8 rtcpCName[]);
|
||||
|
||||
WebRtc_Word32 GetRemoteRTCPCName(WebRtc_Word8 rtcpCName[]);
|
||||
|
||||
WebRtc_Word32 RegisterRtpObserver(ViERTPObserver* observer);
|
||||
|
||||
WebRtc_Word32 RegisterRtcpObserver(ViERTCPObserver* observer);
|
||||
|
||||
WebRtc_Word32 SendApplicationDefinedRTCPPacket(
|
||||
const WebRtc_UWord8 subType,
|
||||
WebRtc_UWord32 name,
|
||||
const WebRtc_UWord8* data,
|
||||
WebRtc_UWord16 dataLengthInBytes);
|
||||
|
||||
WebRtc_Word32 GetSendRtcpStatistics(WebRtc_UWord16& fractionLost,
|
||||
WebRtc_UWord32& cumulativeLost,
|
||||
WebRtc_UWord32& extendedMax,
|
||||
WebRtc_UWord32& jitterSamples,
|
||||
WebRtc_Word32& rttMs);
|
||||
|
||||
WebRtc_Word32 GetReceivedRtcpStatistics(WebRtc_UWord16& fractionLost,
|
||||
WebRtc_UWord32& cumulativeLost,
|
||||
WebRtc_UWord32& extendedMax,
|
||||
WebRtc_UWord32& jitterSamples,
|
||||
WebRtc_Word32& rttMs);
|
||||
|
||||
WebRtc_Word32 GetRtpStatistics(WebRtc_UWord32& bytesSent,
|
||||
WebRtc_UWord32& packetsSent,
|
||||
WebRtc_UWord32& bytesReceived,
|
||||
WebRtc_UWord32& packetsReceived) const;
|
||||
|
||||
void GetBandwidthUsage(WebRtc_UWord32& totalBitrateSent,
|
||||
WebRtc_UWord32& videoBitrateSent,
|
||||
WebRtc_UWord32& fecBitrateSent,
|
||||
WebRtc_UWord32& nackBitrateSent) const;
|
||||
|
||||
WebRtc_Word32 SetKeepAliveStatus(const bool enable,
|
||||
const WebRtc_Word8 unknownPayloadType,
|
||||
const WebRtc_UWord16 deltaTransmitTimeMS);
|
||||
|
||||
WebRtc_Word32 GetKeepAliveStatus(bool& enable,
|
||||
WebRtc_Word8& unknownPayloadType,
|
||||
WebRtc_UWord16& deltaTransmitTimeMS);
|
||||
|
||||
WebRtc_Word32 StartRTPDump(const char fileNameUTF8[1024],
|
||||
RTPDirections direction);
|
||||
|
||||
WebRtc_Word32 StopRTPDump(RTPDirections direction);
|
||||
|
||||
// Implements RtcpFeedback
|
||||
virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
|
||||
const WebRtc_Word32 audioVideoOffset);
|
||||
|
||||
virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
|
||||
const WebRtc_UWord8 subType,
|
||||
const WebRtc_UWord32 name,
|
||||
const WebRtc_UWord16 length,
|
||||
const WebRtc_UWord8* data);
|
||||
|
||||
// Implements RtpFeedback
|
||||
virtual WebRtc_Word32 OnInitializeDecoder(
|
||||
const WebRtc_Word32 id,
|
||||
const WebRtc_Word8 payloadType,
|
||||
const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
|
||||
const int frequency,
|
||||
const WebRtc_UWord8 channels,
|
||||
const WebRtc_UWord32 rate);
|
||||
|
||||
virtual void OnPacketTimeout(const WebRtc_Word32 id);
|
||||
|
||||
virtual void OnReceivedPacket(const WebRtc_Word32 id,
|
||||
const RtpRtcpPacketType packetType);
|
||||
|
||||
virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
|
||||
const RTPAliveType alive);
|
||||
|
||||
virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
|
||||
const WebRtc_UWord32 SSRC);
|
||||
|
||||
virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
|
||||
const WebRtc_UWord32 CSRC,
|
||||
const bool added);
|
||||
|
||||
//-----------------------------------------------------------------
|
||||
// Network
|
||||
//-----------------------------------------------------------------
|
||||
|
||||
// Soure and destination
|
||||
WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtpPort,
|
||||
const WebRtc_UWord16 rtcpPort,
|
||||
const WebRtc_Word8* ipAddress);
|
||||
|
||||
WebRtc_Word32 GetLocalReceiver(WebRtc_UWord16& rtpPort,
|
||||
WebRtc_UWord16& rtcpPort,
|
||||
WebRtc_Word8* ipAddress) const;
|
||||
|
||||
WebRtc_Word32 SetSendDestination(const WebRtc_Word8* ipAddress,
|
||||
const WebRtc_UWord16 rtpPort,
|
||||
const WebRtc_UWord16 rtcpPort,
|
||||
const WebRtc_UWord16 sourceRtpPort,
|
||||
const WebRtc_UWord16 sourceRtcpPort);
|
||||
|
||||
WebRtc_Word32 GetSendDestination(WebRtc_Word8* ipAddress,
|
||||
WebRtc_UWord16& rtpPort,
|
||||
WebRtc_UWord16& rtcpPort,
|
||||
WebRtc_UWord16& sourceRtpPort,
|
||||
WebRtc_UWord16& sourceRtcpPort) const;
|
||||
|
||||
WebRtc_Word32 GetSourceInfo(WebRtc_UWord16& rtpPort,
|
||||
WebRtc_UWord16& rtcpPort,
|
||||
WebRtc_Word8* ipAddress,
|
||||
WebRtc_UWord32 ipAddressLength);
|
||||
|
||||
// Start/Stop Send/Receive
|
||||
WebRtc_Word32 StartSend();
|
||||
WebRtc_Word32 StopSend();
|
||||
bool Sending();
|
||||
WebRtc_Word32 StartReceive();
|
||||
WebRtc_Word32 StopReceive();
|
||||
bool Receiving();
|
||||
|
||||
// External transport
|
||||
WebRtc_Word32 RegisterSendTransport(Transport& transport);
|
||||
|
||||
WebRtc_Word32 DeregisterSendTransport();
|
||||
|
||||
WebRtc_Word32 ReceivedRTPPacket(const void* rtpPacket,
|
||||
const WebRtc_Word32 rtpPacketLength);
|
||||
|
||||
WebRtc_Word32 ReceivedRTCPPacket(const void* rtcpPacket,
|
||||
const WebRtc_Word32 rtcpPacketLength);
|
||||
|
||||
// IPv6
|
||||
WebRtc_Word32 EnableIPv6();
|
||||
|
||||
bool IsIPv6Enabled();
|
||||
|
||||
// Source IP address and port filter
|
||||
WebRtc_Word32 SetSourceFilter(const WebRtc_UWord16 rtpPort,
|
||||
const WebRtc_UWord16 rtcpPort,
|
||||
const WebRtc_Word8* ipAddress);
|
||||
|
||||
WebRtc_Word32 GetSourceFilter(WebRtc_UWord16& rtpPort,
|
||||
WebRtc_UWord16& rtcpPort,
|
||||
WebRtc_Word8* ipAddress) const;
|
||||
|
||||
// ToS
|
||||
WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP, const bool useSetSockOpt);
|
||||
|
||||
WebRtc_Word32 GetToS(WebRtc_Word32& DSCP, bool& useSetSockOpt) const;
|
||||
|
||||
// GQoS
|
||||
WebRtc_Word32 SetSendGQoS(const bool enable,
|
||||
const WebRtc_Word32 serviceType,
|
||||
const WebRtc_UWord32 maxBitrate,
|
||||
const WebRtc_Word32 overrideDSCP);
|
||||
|
||||
WebRtc_Word32 GetSendGQoS(bool& enabled, WebRtc_Word32& serviceType,
|
||||
WebRtc_Word32& overrideDSCP) const;
|
||||
|
||||
// Network settings
|
||||
WebRtc_Word32 SetMTU(WebRtc_UWord16 mtu);
|
||||
|
||||
WebRtc_UWord16 MaxDataPayloadLength() const;
|
||||
|
||||
WebRtc_Word32 SetMaxPacketBurstSize(WebRtc_UWord16 maxNumberOfPackets);
|
||||
|
||||
WebRtc_Word32 SetPacketBurstSpreadState(bool enable,
|
||||
const WebRtc_UWord16 framePeriodMS);
|
||||
|
||||
// Packet timout notification
|
||||
WebRtc_Word32 SetPacketTimeoutNotification(bool enable,
|
||||
WebRtc_UWord32 timeoutSeconds);
|
||||
|
||||
// Periodic dead-or-alive reports
|
||||
WebRtc_Word32 RegisterNetworkObserver(ViENetworkObserver* observer);
|
||||
bool NetworkObserverRegistered();
|
||||
|
||||
WebRtc_Word32
|
||||
SetPeriodicDeadOrAliveStatus(const bool enable,
|
||||
const WebRtc_UWord32 sampleTimeSeconds);
|
||||
|
||||
WebRtc_Word32 SendUDPPacket(const WebRtc_Word8* data,
|
||||
const WebRtc_UWord32 length,
|
||||
WebRtc_Word32& transmittedBytes,
|
||||
bool useRtcpSocket);
|
||||
|
||||
//-----------------------------------------------------------------
|
||||
// Image processing
|
||||
//-----------------------------------------------------------------
|
||||
WebRtc_Word32 EnableColorEnhancement(bool enable);
|
||||
|
||||
//-----------------------------------------------------------------
|
||||
// Register sender
|
||||
//-----------------------------------------------------------------
|
||||
WebRtc_Word32
|
||||
RegisterSendRtpRtcpModule(RtpRtcp& sendRtpRtcpModule);
|
||||
|
||||
WebRtc_Word32 DeregisterSendRtpRtcpModule();
|
||||
|
||||
// Implements VCM::VCMReceiveCallback, getting decoded frames from
|
||||
// VCM.
|
||||
virtual WebRtc_Word32 FrameToRender(VideoFrame& videoFrame);
|
||||
|
||||
// Implements VCM::VCMReceiveCallback, getting info about decoded
|
||||
// frames from VCM.
|
||||
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
|
||||
const WebRtc_UWord64 pictureId);
|
||||
|
||||
//Implements VCM::VideoFrameStorageCallback
|
||||
virtual WebRtc_Word32 StoreReceivedFrame(
|
||||
const EncodedVideoData& frameToStore);
|
||||
|
||||
// Implements VCM::VideoReceiveStatisticsCallback
|
||||
virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bitRate,
|
||||
const WebRtc_UWord32 frameRate);
|
||||
|
||||
// Implements VCM::VideoFrameTypeCallback
|
||||
virtual WebRtc_Word32 FrameTypeRequest(const FrameType frameType);
|
||||
|
||||
// Implements VCM::VideoFrameTypeCallback
|
||||
virtual WebRtc_Word32 SliceLossIndicationRequest(
|
||||
const WebRtc_UWord64 pictureId);
|
||||
|
||||
// Implements VCM::VideoPacketRequestCallback
|
||||
virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
|
||||
WebRtc_UWord16 length);
|
||||
|
||||
#ifdef WEBRTC_SRTP
|
||||
//SRTP
|
||||
WebRtc_Word32 EnableSRTPSend(
|
||||
const SrtpModule::CipherTypes cipherType,
|
||||
const unsigned int cipherKeyLength,
|
||||
const SrtpModule::AuthenticationTypes authType,
|
||||
const unsigned int authKeyLength,
|
||||
const unsigned int authTagLength,
|
||||
const SrtpModule::SecurityLevels level,
|
||||
const WebRtc_UWord8* key,
|
||||
const bool useForRTCP);
|
||||
|
||||
WebRtc_Word32 DisableSRTPSend();
|
||||
|
||||
WebRtc_Word32 EnableSRTPReceive(
|
||||
const SrtpModule::CipherTypes cipherType,
|
||||
const unsigned int cipherKeyLength,
|
||||
const SrtpModule::AuthenticationTypes authType,
|
||||
const unsigned int authKeyLength,
|
||||
const unsigned int authTagLength,
|
||||
const SrtpModule::SecurityLevels level,
|
||||
const WebRtc_UWord8* key,
|
||||
const bool useForRTCP);
|
||||
WebRtc_Word32 DisableSRTPReceive();
|
||||
#endif
|
||||
|
||||
WebRtc_Word32 RegisterExternalEncryption(Encryption* encryption);
|
||||
WebRtc_Word32 DeRegisterExternalEncryption();
|
||||
|
||||
//Voice Engine
|
||||
WebRtc_Word32 SetVoiceChannel(WebRtc_Word32 veChannelId,
|
||||
VoEVideoSync* veSyncInterface);
|
||||
WebRtc_Word32 VoiceChannel();
|
||||
|
||||
//ViEFrameProviderBase
|
||||
virtual int FrameCallbackChanged(){return -1;}
|
||||
|
||||
// Effect filter
|
||||
WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effectFilter);
|
||||
|
||||
WebRtc_Word32 SetInverseH263Logic(const bool enable);
|
||||
|
||||
// File recording
|
||||
ViEFileRecorder& GetIncomingFileRecorder();
|
||||
void ReleaseIncomingFileRecorder();
|
||||
|
||||
protected:
|
||||
// Thread function according to ThreadWrapper
|
||||
static bool ChannelDecodeThreadFunction(void* obj);
|
||||
bool ChannelDecodeProcess();
|
||||
|
||||
private:
|
||||
|
||||
WebRtc_Word32 StartDecodeThread();
|
||||
WebRtc_Word32 StopDecodeThread();
|
||||
|
||||
// Protection
|
||||
WebRtc_Word32 ProcessNACKRequest(const bool enable);
|
||||
|
||||
WebRtc_Word32 ProcessFECRequest(const bool enable,
|
||||
const unsigned char payloadTypeRED,
|
||||
const unsigned char payloadTypeFEC);
|
||||
|
||||
// General members
|
||||
WebRtc_Word32 _channelId;
|
||||
WebRtc_Word32 _engineId;
|
||||
WebRtc_UWord32 _numberOfCores;
|
||||
WebRtc_UWord8 _numSocketThreads;
|
||||
|
||||
// Critical sections
|
||||
// Used for all registered callbacks except rendering.
|
||||
CriticalSectionWrapper& _callbackCritsect;
|
||||
|
||||
// Owned modules/classes
|
||||
RtpRtcp& _rtpRtcp;
|
||||
RtpRtcp* _defaultRtpRtcp;
|
||||
std::list<RtpRtcp*> _simulcastRtpRtcp;
|
||||
// Owned modules/classes.
|
||||
RtpRtcp& rtp_rtcp_;
|
||||
RtpRtcp* default_rtp_rtcp_;
|
||||
std::list<RtpRtcp*> simulcast_rtp_rtcp_;
|
||||
#ifndef WEBRTC_EXTERNAL_TRANSPORT
|
||||
UdpTransport& _socketTransport;
|
||||
UdpTransport& socket_transport_;
|
||||
#endif
|
||||
VideoCodingModule& _vcm;
|
||||
ViEReceiver& _vieReceiver;
|
||||
ViESender& _vieSender;
|
||||
ViESyncModule& _vieSync;//Lip syncronization
|
||||
VideoCodingModule& vcm_;
|
||||
ViEReceiver& vie_receiver_;
|
||||
ViESender& vie_sender_;
|
||||
ViESyncModule& vie_sync_;
|
||||
|
||||
//Uses
|
||||
ProcessThread& _moduleProcessThread;
|
||||
ViEDecoderObserver* _codecObserver;
|
||||
bool _doKeyFrameCallbackRequest;
|
||||
ViERTPObserver* _rtpObserver;
|
||||
ViERTCPObserver* _rtcpObserver;
|
||||
ViENetworkObserver* _networkObserver;
|
||||
bool _rtpPacketTimeout;
|
||||
bool _usingPacketSpread;
|
||||
// Not owned.
|
||||
ProcessThread& module_process_thread_;
|
||||
ViEDecoderObserver* codec_observer_;
|
||||
bool do_key_frame_callbackRequest_;
|
||||
ViERTPObserver* rtp_observer_;
|
||||
ViERTCPObserver* rtcp_observer_;
|
||||
ViENetworkObserver* networkObserver_;
|
||||
bool rtp_packet_timeout_;
|
||||
bool using_packet_spread_;
|
||||
|
||||
// Registered members
|
||||
Transport* _ptrExternalTransport;
|
||||
Transport* external_transport_;
|
||||
|
||||
// Codec
|
||||
bool _decoderReset;
|
||||
bool _waitForKeyFrame;
|
||||
bool decoder_reset_;
|
||||
bool wait_for_key_frame_;
|
||||
ThreadWrapper* decode_thread_;
|
||||
|
||||
// Decoder
|
||||
ThreadWrapper* _ptrDecodeThread;
|
||||
Encryption* external_encryption_;
|
||||
|
||||
//SRTP - using seperate pointers for encryption and decryption to support
|
||||
// simultaneous operations.
|
||||
SrtpModule* _ptrSrtpModuleEncryption;
|
||||
SrtpModule* _ptrSrtpModuleDecryption;
|
||||
Encryption* _ptrExternalEncryption;
|
||||
ViEEffectFilter* effect_filter_;
|
||||
bool color_enhancement_;
|
||||
|
||||
// Effect filter and color enhancement
|
||||
ViEEffectFilter* _effectFilter;
|
||||
bool _colorEnhancement;
|
||||
// Time when RTT time was last reported to VCM JB.
|
||||
TickTime vcm_rttreported_;
|
||||
|
||||
// Time when RTT time was last reported to VCM JB.
|
||||
TickTime _vcmRTTReported;
|
||||
|
||||
//Recording
|
||||
ViEFileRecorder _fileRecorder;
|
||||
ViEFileRecorder file_recorder_;
|
||||
};
|
||||
} // namespace webrtc
|
||||
#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_CHANNEL_H_
|
||||
|
||||
} // namespace webrtc
|
||||
|
||||
#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
|
||||
|
||||
Loading…
Reference in New Issue
Block a user