#include <rtp_receiver_strategy.h>
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virtual | ~RTPReceiverStrategy () |
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virtual int32_t | ParseRtpPacket (WebRtcRTPHeader *rtp_header, const PayloadUnion &specific_payload, bool is_red, const uint8_t *payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet)=0 |
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virtual TelephoneEventHandler * | GetTelephoneEventHandler ()=0 |
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virtual RTPAliveType | ProcessDeadOrAlive (uint16_t last_payload_length) const =0 |
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virtual bool | ShouldReportCsrcChanges (uint8_t payload_type) const =0 |
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virtual int32_t | OnNewPayloadTypeCreated (const CodecInst &audio_codec)=0 |
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virtual int32_t | InvokeOnInitializeDecoder (RtpFeedback *callback, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion &specific_payload) const =0 |
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virtual void | CheckPayloadChanged (int8_t payload_type, PayloadUnion *specific_payload, bool *should_discard_changes) |
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virtual int | Energy (uint8_t array_of_energy[kRtpCsrcSize]) const |
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void | GetLastMediaSpecificPayload (PayloadUnion *payload) const |
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void | SetLastMediaSpecificPayload (const PayloadUnion &payload) |
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virtual | ~RTPReceiverStrategy () |
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virtual int32_t | ParseRtpPacket (WebRtcRTPHeader *rtp_header, const PayloadUnion &specific_payload, bool is_red, const uint8_t *payload, size_t payload_length, int64_t timestamp_ms, bool is_first_packet)=0 |
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virtual TelephoneEventHandler * | GetTelephoneEventHandler ()=0 |
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virtual RTPAliveType | ProcessDeadOrAlive (uint16_t last_payload_length) const =0 |
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virtual bool | ShouldReportCsrcChanges (uint8_t payload_type) const =0 |
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virtual int32_t | OnNewPayloadTypeCreated (const CodecInst &audio_codec)=0 |
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virtual int32_t | InvokeOnInitializeDecoder (RtpFeedback *callback, int8_t payload_type, const char payload_name[RTP_PAYLOAD_NAME_SIZE], const PayloadUnion &specific_payload) const =0 |
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virtual void | CheckPayloadChanged (int8_t payload_type, PayloadUnion *specific_payload, bool *should_discard_changes) |
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virtual int | Energy (uint8_t array_of_energy[kRtpCsrcSize]) const |
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void | GetLastMediaSpecificPayload (PayloadUnion *payload) const |
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void | SetLastMediaSpecificPayload (const PayloadUnion &payload) |
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◆ ~RTPReceiverStrategy() [1/2]
virtual webrtc::RTPReceiverStrategy::~RTPReceiverStrategy |
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inlinevirtual |
◆ RTPReceiverStrategy() [1/2]
webrtc::RTPReceiverStrategy::RTPReceiverStrategy |
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RtpData * |
data_callback | ) |
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explicitprotected |
◆ ~RTPReceiverStrategy() [2/2]
virtual webrtc::RTPReceiverStrategy::~RTPReceiverStrategy |
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inlinevirtual |
◆ RTPReceiverStrategy() [2/2]
webrtc::RTPReceiverStrategy::RTPReceiverStrategy |
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RtpData * |
data_callback | ) |
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explicitprotected |
◆ CheckPayloadChanged() [1/2]
void webrtc::RTPReceiverStrategy::CheckPayloadChanged |
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int8_t |
payload_type, |
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PayloadUnion * |
specific_payload, |
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bool * |
should_discard_changes |
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) |
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virtual |
◆ CheckPayloadChanged() [2/2]
virtual void webrtc::RTPReceiverStrategy::CheckPayloadChanged |
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int8_t |
payload_type, |
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PayloadUnion * |
specific_payload, |
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bool * |
should_discard_changes |
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) |
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virtual |
◆ CreateAudioStrategy() [1/2]
◆ CreateAudioStrategy() [2/2]
◆ CreateVideoStrategy() [1/2]
◆ CreateVideoStrategy() [2/2]
◆ Energy() [1/2]
int webrtc::RTPReceiverStrategy::Energy |
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uint8_t |
array_of_energy[kRtpCsrcSize] | ) |
const |
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virtual |
◆ Energy() [2/2]
virtual int webrtc::RTPReceiverStrategy::Energy |
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uint8_t |
array_of_energy[kRtpCsrcSize] | ) |
const |
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virtual |
◆ GetLastMediaSpecificPayload() [1/2]
void webrtc::RTPReceiverStrategy::GetLastMediaSpecificPayload |
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PayloadUnion * |
payload | ) |
const |
◆ GetLastMediaSpecificPayload() [2/2]
void webrtc::RTPReceiverStrategy::GetLastMediaSpecificPayload |
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PayloadUnion * |
payload | ) |
const |
◆ GetTelephoneEventHandler() [1/2]
◆ GetTelephoneEventHandler() [2/2]
◆ InvokeOnInitializeDecoder() [1/2]
virtual int32_t webrtc::RTPReceiverStrategy::InvokeOnInitializeDecoder |
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RtpFeedback * |
callback, |
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int8_t |
payload_type, |
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const char |
payload_name[RTP_PAYLOAD_NAME_SIZE], |
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const PayloadUnion & |
specific_payload |
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) |
| const |
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pure virtual |
◆ InvokeOnInitializeDecoder() [2/2]
virtual int32_t webrtc::RTPReceiverStrategy::InvokeOnInitializeDecoder |
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RtpFeedback * |
callback, |
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int8_t |
payload_type, |
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const char |
payload_name[RTP_PAYLOAD_NAME_SIZE], |
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const PayloadUnion & |
specific_payload |
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) |
| const |
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pure virtual |
◆ OnNewPayloadTypeCreated() [1/2]
virtual int32_t webrtc::RTPReceiverStrategy::OnNewPayloadTypeCreated |
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const CodecInst & |
audio_codec | ) |
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pure virtual |
◆ OnNewPayloadTypeCreated() [2/2]
virtual int32_t webrtc::RTPReceiverStrategy::OnNewPayloadTypeCreated |
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const CodecInst & |
audio_codec | ) |
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pure virtual |
◆ ParseRtpPacket() [1/2]
◆ ParseRtpPacket() [2/2]
◆ ProcessDeadOrAlive() [1/2]
virtual RTPAliveType webrtc::RTPReceiverStrategy::ProcessDeadOrAlive |
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uint16_t |
last_payload_length | ) |
const |
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pure virtual |
◆ ProcessDeadOrAlive() [2/2]
virtual RTPAliveType webrtc::RTPReceiverStrategy::ProcessDeadOrAlive |
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uint16_t |
last_payload_length | ) |
const |
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pure virtual |
◆ SetLastMediaSpecificPayload() [1/2]
void webrtc::RTPReceiverStrategy::SetLastMediaSpecificPayload |
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const PayloadUnion & |
payload | ) |
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◆ SetLastMediaSpecificPayload() [2/2]
void webrtc::RTPReceiverStrategy::SetLastMediaSpecificPayload |
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const PayloadUnion & |
payload | ) |
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◆ ShouldReportCsrcChanges() [1/2]
virtual bool webrtc::RTPReceiverStrategy::ShouldReportCsrcChanges |
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uint8_t |
payload_type | ) |
const |
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pure virtual |
◆ ShouldReportCsrcChanges() [2/2]
virtual bool webrtc::RTPReceiverStrategy::ShouldReportCsrcChanges |
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uint8_t |
payload_type | ) |
const |
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pure virtual |
◆ crit_sect_
◆ data_callback_
RtpData * webrtc::RTPReceiverStrategy::data_callback_ |
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protected |
◆ last_payload_
The documentation for this class was generated from the following files: