Files
platform-external-webrtc/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
magjed f3feeffe03 Reland of move RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #1 id:1 of https://codereview.webrtc.org/2528993002/ )
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}
2016-11-25 14:40:30 +00:00

146 lines
5.0 KiB
C++

/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
#include <map>
#include <memory>
#include <set>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/deprecation.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc {
struct CodecInst;
class VideoCodec;
// TODO(magjed): Remove once external code is updated.
class RTPPayloadStrategy {
public:
static RTPPayloadStrategy* CreateStrategy(bool handling_audio) {
return nullptr;
}
};
class RTPPayloadRegistry {
public:
RTPPayloadRegistry();
~RTPPayloadRegistry();
// TODO(magjed): Remove once external code is updated.
explicit RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy)
: RTPPayloadRegistry() {}
// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
// and simplify the code. http://crbug/webrtc/6743.
int32_t RegisterReceivePayload(const CodecInst& audio_codec,
bool* created_new_payload_type);
int32_t RegisterReceivePayload(const VideoCodec& video_codec);
int32_t DeRegisterReceivePayload(int8_t payload_type);
int32_t ReceivePayloadType(const CodecInst& audio_codec,
int8_t* payload_type) const;
int32_t ReceivePayloadType(const VideoCodec& video_codec,
int8_t* payload_type) const;
bool RtxEnabled() const;
void SetRtxSsrc(uint32_t ssrc);
bool GetRtxSsrc(uint32_t* ssrc) const;
void SetRtxPayloadType(int payload_type, int associated_payload_type);
bool IsRtx(const RTPHeader& header) const;
bool RestoreOriginalPacket(uint8_t* restored_packet,
const uint8_t* packet,
size_t* packet_length,
uint32_t original_ssrc,
const RTPHeader& header);
bool IsRed(const RTPHeader& header) const;
// Returns true if the media of this RTP packet is encapsulated within an
// extra header, such as RTX or RED.
bool IsEncapsulated(const RTPHeader& header) const;
bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
int GetPayloadTypeFrequency(uint8_t payload_type) const;
const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
void ResetLastReceivedPayloadTypes() {
rtc::CritScope cs(&crit_sect_);
last_received_payload_type_ = -1;
last_received_media_payload_type_ = -1;
}
// This sets the payload type of the packets being received from the network
// on the media SSRC. For instance if packets are encapsulated with RED, this
// payload type will be the RED payload type.
void SetIncomingPayloadType(const RTPHeader& header);
// Returns true if the new media payload type has not changed.
bool ReportMediaPayloadType(uint8_t media_payload_type);
int8_t red_payload_type() const { return GetPayloadTypeWithName("red"); }
int8_t ulpfec_payload_type() const {
return GetPayloadTypeWithName("ulpfec");
}
int8_t last_received_payload_type() const {
rtc::CritScope cs(&crit_sect_);
return last_received_payload_type_;
}
void set_last_received_payload_type(int8_t last_received_payload_type) {
rtc::CritScope cs(&crit_sect_);
last_received_payload_type_ = last_received_payload_type;
}
int8_t last_received_media_payload_type() const {
rtc::CritScope cs(&crit_sect_);
return last_received_media_payload_type_;
}
RTC_DEPRECATED void set_use_rtx_payload_mapping_on_restore(bool val) {}
private:
// Prunes the payload type map of the specific payload type, if it exists.
void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
const CodecInst& audio_codec);
bool IsRtxInternal(const RTPHeader& header) const;
// Returns the payload type for the payload with name |payload_name|, or -1 if
// no such payload is registered.
int8_t GetPayloadTypeWithName(const char* payload_name) const;
rtc::CriticalSection crit_sect_;
std::map<int, RtpUtility::Payload> payload_type_map_;
int8_t incoming_payload_type_;
int8_t last_received_payload_type_;
int8_t last_received_media_payload_type_;
bool rtx_;
// Mapping rtx_payload_type_map_[rtx] = associated.
std::map<int, int> rtx_payload_type_map_;
uint32_t ssrc_rtx_;
// Only warn once per payload type, if an RTX packet is received but
// no associated payload type found in |rtx_payload_type_map_|.
std::set<int> payload_types_with_suppressed_warnings_ GUARDED_BY(crit_sect_);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_