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
Review-Url: https://codereview.webrtc.org/2524923002
Cr-Commit-Position: refs/heads/master@{#15232}
146 lines
5.0 KiB
C++
146 lines
5.0 KiB
C++
/*
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* Copyright (c) 2013 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_PAYLOAD_REGISTRY_H_
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#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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#include <map>
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#include <memory>
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#include <set>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/deprecation.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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namespace webrtc {
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struct CodecInst;
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class VideoCodec;
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// TODO(magjed): Remove once external code is updated.
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class RTPPayloadStrategy {
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public:
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static RTPPayloadStrategy* CreateStrategy(bool handling_audio) {
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return nullptr;
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}
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};
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class RTPPayloadRegistry {
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public:
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RTPPayloadRegistry();
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~RTPPayloadRegistry();
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// TODO(magjed): Remove once external code is updated.
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explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
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: RTPPayloadRegistry() {}
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// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
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// and simplify the code. http://crbug/webrtc/6743.
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int32_t RegisterReceivePayload(const CodecInst& audio_codec,
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bool* created_new_payload_type);
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int32_t RegisterReceivePayload(const VideoCodec& video_codec);
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int32_t DeRegisterReceivePayload(int8_t payload_type);
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int32_t ReceivePayloadType(const CodecInst& audio_codec,
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int8_t* payload_type) const;
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int32_t ReceivePayloadType(const VideoCodec& video_codec,
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int8_t* payload_type) const;
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bool RtxEnabled() const;
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void SetRtxSsrc(uint32_t ssrc);
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bool GetRtxSsrc(uint32_t* ssrc) const;
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void SetRtxPayloadType(int payload_type, int associated_payload_type);
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bool IsRtx(const RTPHeader& header) const;
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bool RestoreOriginalPacket(uint8_t* restored_packet,
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const uint8_t* packet,
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size_t* packet_length,
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uint32_t original_ssrc,
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const RTPHeader& header);
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bool IsRed(const RTPHeader& header) const;
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// Returns true if the media of this RTP packet is encapsulated within an
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// extra header, such as RTX or RED.
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bool IsEncapsulated(const RTPHeader& header) const;
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bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
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int GetPayloadTypeFrequency(uint8_t payload_type) const;
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const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
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void ResetLastReceivedPayloadTypes() {
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rtc::CritScope cs(&crit_sect_);
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last_received_payload_type_ = -1;
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last_received_media_payload_type_ = -1;
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}
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// This sets the payload type of the packets being received from the network
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// on the media SSRC. For instance if packets are encapsulated with RED, this
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// payload type will be the RED payload type.
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void SetIncomingPayloadType(const RTPHeader& header);
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// Returns true if the new media payload type has not changed.
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bool ReportMediaPayloadType(uint8_t media_payload_type);
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int8_t red_payload_type() const { return GetPayloadTypeWithName("red"); }
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int8_t ulpfec_payload_type() const {
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return GetPayloadTypeWithName("ulpfec");
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}
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int8_t last_received_payload_type() const {
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rtc::CritScope cs(&crit_sect_);
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return last_received_payload_type_;
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}
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void set_last_received_payload_type(int8_t last_received_payload_type) {
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rtc::CritScope cs(&crit_sect_);
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last_received_payload_type_ = last_received_payload_type;
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}
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int8_t last_received_media_payload_type() const {
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rtc::CritScope cs(&crit_sect_);
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return last_received_media_payload_type_;
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}
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RTC_DEPRECATED void set_use_rtx_payload_mapping_on_restore(bool val) {}
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private:
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// Prunes the payload type map of the specific payload type, if it exists.
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void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
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const CodecInst& audio_codec);
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bool IsRtxInternal(const RTPHeader& header) const;
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// Returns the payload type for the payload with name |payload_name|, or -1 if
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// no such payload is registered.
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int8_t GetPayloadTypeWithName(const char* payload_name) const;
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rtc::CriticalSection crit_sect_;
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std::map<int, RtpUtility::Payload> payload_type_map_;
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int8_t incoming_payload_type_;
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int8_t last_received_payload_type_;
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int8_t last_received_media_payload_type_;
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bool rtx_;
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// Mapping rtx_payload_type_map_[rtx] = associated.
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std::map<int, int> rtx_payload_type_map_;
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uint32_t ssrc_rtx_;
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// Only warn once per payload type, if an RTX packet is received but
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// no associated payload type found in |rtx_payload_type_map_|.
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std::set<int> payload_types_with_suppressed_warnings_ GUARDED_BY(crit_sect_);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
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