f3feeffe03
Reason for revert: Downstream code has been updated. Original issue's description: > Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ ) > > Reason for revert: > Breaks downstream projects. > > Original issue's description: > > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry > > > > This CL removes RTPPayloadStrategy that is currently used to handle > > audio/video specific aspects of payload handling. Instead, the audio and > > video specific aspects will now have different functions, with linear > > code flow. > > > > This CL does not contain any functional changes, and is just a > > preparation for future CL:s. > > > > The main purpose with this CL is to add this function: > > bool PayloadIsCompatible(const RtpUtility::Payload& payload, > > const webrtc::VideoCodec& video_codec); > > that can easily be extended in a future CL to look at video codec > > specific information. > > > > BUG=webrtc:6743 > > > > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166 > > Cr-Commit-Position: refs/heads/master@{#15232} > > TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6743 > > Committed: https://crrev.com/33c81d05613f45f65ee17224ed381c6cdd1c6c6f > Cr-Commit-Position: refs/heads/master@{#15234} TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6743 Review-Url: https://codereview.webrtc.org/2531043002 Cr-Commit-Position: refs/heads/master@{#15245}
96 lines
3.6 KiB
C++
96 lines
3.6 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_INCLUDE_RTP_RECEIVER_H_
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#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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struct CodecInst;
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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(const CodecInst& audio_codec) = 0;
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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,
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bool in_order) = 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_INCLUDE_RTP_RECEIVER_H_
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