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22ec952829
Bug: webrtc:7135 Change-Id: I35fbc76a5ca8d50caff918bbfd2cb13dce4cbd21 Reviewed-on: https://webrtc-review.googlesource.com/4141 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20154}
113 lines
4.2 KiB
C++
113 lines
4.2 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 MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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#define MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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#include <vector>
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#include "api/rtpreceiverinterface.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace webrtc {
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class RTPPayloadRegistry;
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class VideoCodec;
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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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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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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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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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int payload_type,
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const SdpAudioFormat& audio_format) = 0;
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// Deprecated version of the above.
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int32_t RegisterReceivePayload(const CodecInst& audio_codec);
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// Registers a receive payload in the payload registry.
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virtual int32_t RegisterReceivePayload(const VideoCodec& video_codec) = 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) = 0;
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// TODO(nisse): Deprecated version, delete as soon as downstream
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// applications are updated.
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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 /* Ignored */) {
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return IncomingRtpPacket(rtp_header, payload, payload_length,
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payload_specific);
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}
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// Gets the RTP timestamp and the corresponding monotonic system
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// time for the most recent in-order packet. Returns true on
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// success, false if no packet has been received.
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virtual bool GetLatestTimestamps(uint32_t* timestamp,
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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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virtual std::vector<RtpSource> GetSources() const = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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