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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
// vie_channel.h
#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
#include <list>
#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "modules/udp_transport/interface/udp_transport.h"
#include "modules/video_coding/main/interface/video_coding_defines.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "system_wrappers/interface/tick_util.h"
#include "typedefs.h"
#include "video_engine/include/vie_network.h"
#include "video_engine/include/vie_rtp_rtcp.h"
#include "video_engine/vie_defines.h"
#include "video_engine/vie_file_recorder.h"
#include "video_engine/vie_frame_provider_base.h"
namespace webrtc {
class CriticalSectionWrapper;
class Encryption;
class ProcessThread;
class RtpRtcp;
class ThreadWrapper;
class VideoCodingModule;
class VideoDecoder;
class VideoRenderCallback;
class ViEDecoderObserver;
class ViEEffectFilter;
class ViENetworkObserver;
class ViEReceiver;
class ViERTCPObserver;
class ViERTPObserver;
class ViESender;
class ViESyncModule;
class VoEVideoSync;
class ViEChannel
: public VCMFrameTypeCallback,
public VCMReceiveCallback,
public VCMReceiveStatisticsCallback,
public VCMPacketRequestCallback,
public VCMFrameStorageCallback,
public RtcpFeedback,
public RtpFeedback,
public ViEFrameProviderBase {
public:
ViEChannel(WebRtc_Word32 channel_id,
WebRtc_Word32 engine_id,
WebRtc_UWord32 number_of_cores,
ProcessThread& module_process_thread);
~ViEChannel();
WebRtc_Word32 Init();
// Sets the encoder to use for the channel. |new_stream| indicates the encoder
// type has changed and we should start a new RTP stream.
WebRtc_Word32 SetSendCodec(const VideoCodec& video_codec,
bool new_stream = true);
WebRtc_Word32 SetReceiveCodec(const VideoCodec& video_codec);
WebRtc_Word32 GetReceiveCodec(VideoCodec& video_codec);
WebRtc_Word32 RegisterCodecObserver(ViEDecoderObserver* observer);
// Registers an external decoder. |decoder_render| is set to true if the
// decoder will do the rendering. If |decoder_render| is set,|render_delay|
// indicates the time needed to decode and render a frame.
WebRtc_Word32 RegisterExternalDecoder(const WebRtc_UWord8 pl_type,
VideoDecoder* decoder,
bool decoder_render,
WebRtc_Word32 render_delay);
WebRtc_Word32 DeRegisterExternalDecoder(const WebRtc_UWord8 pl_type);
WebRtc_Word32 ReceiveCodecStatistics(WebRtc_UWord32& num_key_frames,
WebRtc_UWord32& num_delta_frames);
WebRtc_UWord32 DiscardedPackets() const;
// Only affects calls to SetReceiveCodec done after this call.
WebRtc_Word32 WaitForKeyFrame(bool wait);
// If enabled, a key frame request will be sent as soon as there are lost
// packets. If |only_key_frames| are set, requests are only sent for loss in
// key frames.
WebRtc_Word32 SetSignalPacketLossStatus(bool enable, bool only_key_frames);
WebRtc_Word32 SetRTCPMode(const RTCPMethod rtcp_mode);
WebRtc_Word32 GetRTCPMode(RTCPMethod& rtcp_mode);
WebRtc_Word32 SetNACKStatus(const bool enable);
WebRtc_Word32 SetFECStatus(const bool enable,
const unsigned char payload_typeRED,
const unsigned char payload_typeFEC);
WebRtc_Word32 SetHybridNACKFECStatus(const bool enable,
const unsigned char payload_typeRED,
const unsigned char payload_typeFEC);
WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
bool EnableRemb(bool enable);
WebRtc_Word32 EnableTMMBR(const bool enable);
WebRtc_Word32 EnableKeyFrameRequestCallback(const bool enable);
// Sets SSRC for outgoing stream.
WebRtc_Word32 SetSSRC(const WebRtc_UWord32 SSRC,
const StreamType usage,
const unsigned char simulcast_idx);
// Gets SSRC for outgoing stream.
WebRtc_Word32 GetLocalSSRC(WebRtc_UWord32& SSRC);
// Gets SSRC for the incoming stream.
WebRtc_Word32 GetRemoteSSRC(WebRtc_UWord32& SSRC);
// Gets the CSRC for the incoming stream.
WebRtc_Word32 GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]);
// Sets the starting sequence number, must be called before StartSend.
WebRtc_Word32 SetStartSequenceNumber(WebRtc_UWord16 sequence_number);
// Sets the CName for the outgoing stream on the channel.
WebRtc_Word32 SetRTCPCName(const WebRtc_Word8 rtcp_cname[]);
// Gets the CName for the outgoing stream on the channel.
WebRtc_Word32 GetRTCPCName(WebRtc_Word8 rtcp_cname[]);
// Gets the CName of the incoming stream.
WebRtc_Word32 GetRemoteRTCPCName(WebRtc_Word8 rtcp_cname[]);
WebRtc_Word32 RegisterRtpObserver(ViERTPObserver* observer);
WebRtc_Word32 RegisterRtcpObserver(ViERTCPObserver* observer);
WebRtc_Word32 SendApplicationDefinedRTCPPacket(
const WebRtc_UWord8 sub_type,
WebRtc_UWord32 name,
const WebRtc_UWord8* data,
WebRtc_UWord16 data_length_in_bytes);
// Returns statistics reported by the remote client in an RTCP packet.
WebRtc_Word32 GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
WebRtc_UWord32& cumulative_lost,
WebRtc_UWord32& extended_max,
WebRtc_UWord32& jitter_samples,
WebRtc_Word32& rtt_ms);
// Returns our localy created statistics of the received RTP stream.
WebRtc_Word32 GetReceivedRtcpStatistics(WebRtc_UWord16& fraction_lost,
WebRtc_UWord32& cumulative_lost,
WebRtc_UWord32& extended_max,
WebRtc_UWord32& jitter_samples,
WebRtc_Word32& rtt_ms);
// Gets sent/received packets statistics.
WebRtc_Word32 GetRtpStatistics(WebRtc_UWord32& bytes_sent,
WebRtc_UWord32& packets_sent,
WebRtc_UWord32& bytes_received,
WebRtc_UWord32& packets_received) const;
void GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
WebRtc_UWord32& video_bitrate_sent,
WebRtc_UWord32& fec_bitrate_sent,
WebRtc_UWord32& nackBitrateSent) const;
WebRtc_Word32 SetKeepAliveStatus(const bool enable,
const WebRtc_Word8 unknown_payload_type,
const WebRtc_UWord16 delta_transmit_timeMS);
WebRtc_Word32 GetKeepAliveStatus(bool& enable,
WebRtc_Word8& unknown_payload_type,
WebRtc_UWord16& delta_transmit_timeMS);
WebRtc_Word32 StartRTPDump(const char file_nameUTF8[1024],
RTPDirections direction);
WebRtc_Word32 StopRTPDump(RTPDirections direction);
// Implements RtcpFeedback.
virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
const WebRtc_Word32 audio_video_offset);
virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
const WebRtc_UWord8 sub_type,
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 payload_type,
const WebRtc_Word8 payload_name[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 packet_type);
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);
WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtp_port,
const WebRtc_UWord16 rtcp_port,
const WebRtc_Word8* ip_address);
WebRtc_Word32 GetLocalReceiver(WebRtc_UWord16& rtp_port,
WebRtc_UWord16& rtcp_port,
WebRtc_Word8* ip_address) const;
WebRtc_Word32 SetSendDestination(const WebRtc_Word8* ip_address,
const WebRtc_UWord16 rtp_port,
const WebRtc_UWord16 rtcp_port,
const WebRtc_UWord16 source_rtp_port,
const WebRtc_UWord16 source_rtcp_port);
WebRtc_Word32 GetSendDestination(WebRtc_Word8* ip_address,
WebRtc_UWord16& rtp_port,
WebRtc_UWord16& rtcp_port,
WebRtc_UWord16& source_rtp_port,
WebRtc_UWord16& source_rtcp_port) const;
WebRtc_Word32 GetSourceInfo(WebRtc_UWord16& rtp_port,
WebRtc_UWord16& rtcp_port,
WebRtc_Word8* ip_address,
WebRtc_UWord32 ip_address_length);
WebRtc_Word32 SetRemoteSSRCType(const StreamType usage,
const uint32_t SSRC) const;
WebRtc_Word32 StartSend();
WebRtc_Word32 StopSend();
bool Sending();
WebRtc_Word32 StartReceive();
WebRtc_Word32 StopReceive();
bool Receiving();
WebRtc_Word32 RegisterSendTransport(Transport& transport);
WebRtc_Word32 DeregisterSendTransport();
// Incoming packet from external transport.
WebRtc_Word32 ReceivedRTPPacket(const void* rtp_packet,
const WebRtc_Word32 rtp_packet_length);
// Incoming packet from external transport.
WebRtc_Word32 ReceivedRTCPPacket(const void* rtcp_packet,
const WebRtc_Word32 rtcp_packet_length);
WebRtc_Word32 EnableIPv6();
bool IsIPv6Enabled();
WebRtc_Word32 SetSourceFilter(const WebRtc_UWord16 rtp_port,
const WebRtc_UWord16 rtcp_port,
const WebRtc_Word8* ip_address);
WebRtc_Word32 GetSourceFilter(WebRtc_UWord16& rtp_port,
WebRtc_UWord16& rtcp_port,
WebRtc_Word8* ip_address) const;
WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP, const bool use_set_sockOpt);
WebRtc_Word32 GetToS(WebRtc_Word32& DSCP, bool& use_set_sockOpt) const;
WebRtc_Word32 SetSendGQoS(const bool enable,
const WebRtc_Word32 service_type,
const WebRtc_UWord32 max_bitrate,
const WebRtc_Word32 overrideDSCP);
WebRtc_Word32 GetSendGQoS(bool& enabled, WebRtc_Word32& service_type,
WebRtc_Word32& overrideDSCP) const;
// Sets the maximum transfer unit size for the network link, i.e. including
// IP, UDP and RTP headers.
WebRtc_Word32 SetMTU(WebRtc_UWord16 mtu);
// Returns maximum allowed payload size, i.e. the maximum allowed size of
// encoded data in each packet.
WebRtc_UWord16 MaxDataPayloadLength() const;
WebRtc_Word32 SetMaxPacketBurstSize(WebRtc_UWord16 max_number_of_packets);
WebRtc_Word32 SetPacketBurstSpreadState(bool enable,
const WebRtc_UWord16 frame_periodMS);
WebRtc_Word32 SetPacketTimeoutNotification(bool enable,
WebRtc_UWord32 timeout_seconds);
WebRtc_Word32 RegisterNetworkObserver(ViENetworkObserver* observer);
bool NetworkObserverRegistered();
WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
const bool enable, const WebRtc_UWord32 sample_time_seconds);
WebRtc_Word32 SendUDPPacket(const WebRtc_Word8* data,
const WebRtc_UWord32 length,
WebRtc_Word32& transmitted_bytes,
bool use_rtcp_socket);
WebRtc_Word32 EnableColorEnhancement(bool enable);
// Register send RTP RTCP module, which will deliver encoded frames to the
// to the channel RTP module.
WebRtc_Word32 RegisterSendRtpRtcpModule(RtpRtcp& send_rtp_rtcp_module);
// Deregisters the send RTP RTCP module, which will stop the encoder input to
// the channel.
WebRtc_Word32 DeregisterSendRtpRtcpModule();
// Gets the modules used by the channel.
RtpRtcp* rtp_rtcp();
// Implements VCMReceiveCallback.
virtual WebRtc_Word32 FrameToRender(VideoFrame& video_frame);
// Implements VCMReceiveCallback.
virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
const WebRtc_UWord64 picture_id);
// Implements VCM.
virtual WebRtc_Word32 StoreReceivedFrame(
const EncodedVideoData& frame_to_store);
// Implements VideoReceiveStatisticsCallback.
virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bit_rate,
const WebRtc_UWord32 frame_rate);
// Implements VideoFrameTypeCallback.
virtual WebRtc_Word32 FrameTypeRequest(const FrameType frame_type);
// Implements VideoFrameTypeCallback.
virtual WebRtc_Word32 SliceLossIndicationRequest(
const WebRtc_UWord64 picture_id);
// Implements VideoPacketRequestCallback.
virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequence_numbers,
WebRtc_UWord16 length);
WebRtc_Word32 RegisterExternalEncryption(Encryption* encryption);
WebRtc_Word32 DeRegisterExternalEncryption();
WebRtc_Word32 SetVoiceChannel(WebRtc_Word32 ve_channel_id,
VoEVideoSync* ve_sync_interface);
WebRtc_Word32 VoiceChannel();
// Implements ViEFrameProviderBase.
virtual int FrameCallbackChanged() {return -1;}
WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
ViEFileRecorder& GetIncomingFileRecorder();
void ReleaseIncomingFileRecorder();
protected:
static bool ChannelDecodeThreadFunction(void* obj);
bool ChannelDecodeProcess();
private:
// Assumed to be protected.
WebRtc_Word32 StartDecodeThread();
WebRtc_Word32 StopDecodeThread();
WebRtc_Word32 ProcessNACKRequest(const bool enable);
WebRtc_Word32 ProcessFECRequest(const bool enable,
const unsigned char payload_typeRED,
const unsigned char payload_typeFEC);
WebRtc_Word32 channel_id_;
WebRtc_Word32 engine_id_;
WebRtc_UWord32 number_of_cores_;
WebRtc_UWord8 num_socket_threads_;
// Used for all registered callbacks except rendering.
scoped_ptr<CriticalSectionWrapper> callback_cs_;
// Owned modules/classes.
RtpRtcp& rtp_rtcp_;
RtpRtcp* default_rtp_rtcp_;
std::list<RtpRtcp*> simulcast_rtp_rtcp_;
#ifndef WEBRTC_EXTERNAL_TRANSPORT
UdpTransport& socket_transport_;
#endif
VideoCodingModule& vcm_;
ViEReceiver& vie_receiver_;
ViESender& vie_sender_;
ViESyncModule& vie_sync_;
// 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_;
Transport* external_transport_;
bool decoder_reset_;
bool wait_for_key_frame_;
ThreadWrapper* decode_thread_;
Encryption* external_encryption_;
ViEEffectFilter* effect_filter_;
bool color_enhancement_;
// Time when RTT time was last reported to VCM JB.
TickTime vcm_rttreported_;
ViEFileRecorder file_recorder_;
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_