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/*
* 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 "api/optional.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;
int GetPayloadTypeFrequency(uint8_t payload_type) const;
rtc::Optional<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_