822fbd8b68
Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
121 lines
4.5 KiB
C++
121 lines
4.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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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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#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
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#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class RTPPayloadRegistry;
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class TelephoneEventHandler {
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public:
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virtual ~TelephoneEventHandler() {}
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// The following three methods implement the TelephoneEventHandler interface.
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// Forward DTMFs to decoder for playout.
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virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
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// Is forwarding of outband telephone events turned on/off?
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virtual bool TelephoneEventForwardToDecoder() const = 0;
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// Is TelephoneEvent configured with payload type payload_type
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virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
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};
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class RtpReceiver {
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public:
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// Creates a video-enabled RTP receiver.
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static RtpReceiver* CreateVideoReceiver(
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int id, Clock* clock,
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RtpData* incoming_payload_callback,
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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry);
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// Creates an audio-enabled RTP receiver.
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static RtpReceiver* CreateAudioReceiver(
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int id, Clock* clock,
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RtpAudioFeedback* incoming_audio_feedback,
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RtpData* incoming_payload_callback,
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RtpFeedback* incoming_messages_callback,
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RTPPayloadRegistry* rtp_payload_registry);
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virtual ~RtpReceiver() {}
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// Returns a TelephoneEventHandler if available.
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virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
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// Registers a receive payload in the payload registry and notifies the media
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// receiver strategy.
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virtual int32_t RegisterReceivePayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payload_type,
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const uint32_t frequency,
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const uint8_t channels,
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const uint32_t rate) = 0;
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// De-registers |payload_type| from the payload registry.
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virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
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// Parses the media specific parts of an RTP packet and updates the receiver
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// state. This for instance means that any changes in SSRC and payload type is
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// detected and acted upon.
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virtual bool IncomingRtpPacket(RTPHeader* rtp_header,
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const uint8_t* incoming_rtp_packet,
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int incoming_rtp_packet_length,
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PayloadUnion payload_specific,
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bool in_order) = 0;
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// Returns the currently configured NACK method.
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virtual NACKMethod NACK() const = 0;
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// Turn negative acknowledgement (NACK) requests on/off.
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virtual int32_t SetNACKStatus(const NACKMethod method,
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int max_reordering_threshold) = 0;
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// Returns the last received timestamp.
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virtual uint32_t Timestamp() const = 0;
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// Returns the time in milliseconds when the last timestamp was received.
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virtual int32_t LastReceivedTimeMs() const = 0;
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// Returns the remote SSRC of the currently received RTP stream.
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virtual uint32_t SSRC() const = 0;
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// Returns the current remote CSRCs.
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virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
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// Returns the current energy of the RTP stream received.
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virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
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// Enable/disable RTX and set the SSRC to be used.
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virtual void SetRTXStatus(bool enable, uint32_t ssrc) = 0;
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// Returns the current RTX status and the SSRC and payload type used.
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virtual void RTXStatus(bool* enable, uint32_t* ssrc,
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int* payload_type) const = 0;
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// Sets the RTX payload type.
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virtual void SetRtxPayloadType(int payload_type) = 0;
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// Returns true if the packet with RTP header |header| is likely to be a
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// retransmitted packet, false otherwise.
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virtual bool RetransmitOfOldPacket(const RTPHeader& header, int jitter,
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int min_rtt) const = 0;
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// Returns true if |sequence_number| is received in order, false otherwise.
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virtual bool InOrderPacket(const uint16_t sequence_number) const = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
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