
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information. This CL was reviewed and approved in pieces in the following CLs: https://webrtc-codereview.appspot.com/24209004/ https://webrtc-codereview.appspot.com/24229004/ https://webrtc-codereview.appspot.com/24259004/ https://webrtc-codereview.appspot.com/25109004/ https://webrtc-codereview.appspot.com/26099004/ https://webrtc-codereview.appspot.com/27069004/ https://webrtc-codereview.appspot.com/27969004/ https://webrtc-codereview.appspot.com/27989004/ https://webrtc-codereview.appspot.com/29009004/ https://webrtc-codereview.appspot.com/30929004/ https://webrtc-codereview.appspot.com/30939004/ https://webrtc-codereview.appspot.com/31999004/ Committing as TBR to the original reviewers. BUG=chromium:81439 TEST=none TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom Review URL: https://webrtc-codereview.appspot.com/23129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
104 lines
3.8 KiB
C++
104 lines
3.8 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(const RTPHeader& rtp_header,
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const uint8_t* payload,
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size_t payload_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 void SetNACKStatus(const NACKMethod method) = 0;
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// Gets the last received timestamp. Returns true if a packet has been
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// received, false otherwise.
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virtual bool Timestamp(uint32_t* timestamp) const = 0;
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// Gets the time in milliseconds when the last timestamp was received.
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// Returns true if a packet has been received, false otherwise.
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virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) 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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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
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