In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
131 lines
4.5 KiB
C++
131 lines
4.5 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 MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
|
|
#define MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
|
|
|
|
#include <map>
|
|
#include <set>
|
|
|
|
#include "api/audio_codecs/audio_format.h"
|
|
#include "modules/rtp_rtcp/source/rtp_utility.h"
|
|
#include "rtc_base/criticalsection.h"
|
|
|
|
namespace webrtc {
|
|
|
|
struct CodecInst;
|
|
class VideoCodec;
|
|
|
|
class RTPPayloadRegistry {
|
|
public:
|
|
RTPPayloadRegistry();
|
|
~RTPPayloadRegistry();
|
|
|
|
// TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
|
|
// and simplify the code. http://crbug/webrtc/6743.
|
|
|
|
// Replace all audio receive payload types with the given map.
|
|
void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
|
|
|
|
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 IsRed(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_;
|
|
}
|
|
|
|
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_
|
|
RTC_GUARDED_BY(crit_sect_);
|
|
|
|
// As a first step in splitting this class up in separate cases for audio and
|
|
// video, DCHECK that no instance is used for both audio and video.
|
|
#if RTC_DCHECK_IS_ON
|
|
bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
|
|
bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;
|
|
#endif
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
|