diff options
Diffstat (limited to 'webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h')
| -rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h | 171 |
1 files changed, 85 insertions, 86 deletions
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h index dd16fe51b4..1e96d17a67 100644 --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h @@ -19,92 +19,91 @@ #include "webrtc/typedefs.h" namespace webrtc { -class RTPSenderAudio: public DTMFqueue -{ -public: - RTPSenderAudio(Clock* clock, - RTPSender* rtpSender, - RtpAudioFeedback* audio_feedback); - virtual ~RTPSenderAudio(); - - int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int8_t payloadType, - const uint32_t frequency, - const uint8_t channels, - const uint32_t rate, - RtpUtility::Payload*& payload); - - int32_t SendAudio(const FrameType frameType, - const int8_t payloadType, - const uint32_t captureTimeStamp, - const uint8_t* payloadData, - const size_t payloadSize, - const RTPFragmentationHeader* fragmentation); - - // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG) - int32_t SetAudioPacketSize(const uint16_t packetSizeSamples); - - // Store the audio level in dBov for header-extension-for-audio-level-indication. - // Valid range is [0,100]. Actual value is negative. - int32_t SetAudioLevel(const uint8_t level_dBov); - - // Send a DTMF tone using RFC 2833 (4733) - int32_t SendTelephoneEvent(const uint8_t key, - const uint16_t time_ms, - const uint8_t level); - - int AudioFrequency() const; - - // Set payload type for Redundant Audio Data RFC 2198 - int32_t SetRED(const int8_t payloadType); - - // Get payload type for Redundant Audio Data RFC 2198 - int32_t RED(int8_t& payloadType) const; - -protected: - int32_t SendTelephoneEventPacket(bool ended, - int8_t dtmf_payload_type, - uint32_t dtmfTimeStamp, - uint16_t duration, - bool markerBit); // set on first packet in talk burst - - bool MarkerBit(const FrameType frameType, - const int8_t payloadType); - -private: - Clock* const _clock; - RTPSender* const _rtpSender; - RtpAudioFeedback* const _audioFeedback; - - rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect; - - uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect); - - // DTMF - bool _dtmfEventIsOn; - bool _dtmfEventFirstPacketSent; - int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect); - uint32_t _dtmfTimestamp; - uint8_t _dtmfKey; - uint32_t _dtmfLengthSamples; - uint8_t _dtmfLevel; - int64_t _dtmfTimeLastSent; - uint32_t _dtmfTimestampLastSent; - - int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect); - - // VAD detection, used for markerbit - bool _inbandVADactive GUARDED_BY(_sendAudioCritsect); - int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect); - int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect); - int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect); - int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect); - int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect); - - // Audio level indication - // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) - uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect); +class RTPSenderAudio : public DTMFqueue { + public: + RTPSenderAudio(Clock* clock, + RTPSender* rtpSender, + RtpAudioFeedback* audio_feedback); + virtual ~RTPSenderAudio(); + + int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], + int8_t payloadType, + uint32_t frequency, + size_t channels, + uint32_t rate, + RtpUtility::Payload** payload); + + int32_t SendAudio(FrameType frameType, + int8_t payloadType, + uint32_t captureTimeStamp, + const uint8_t* payloadData, + size_t payloadSize, + const RTPFragmentationHeader* fragmentation); + + // set audio packet size, used to determine when it's time to send a DTMF + // packet in silence (CNG) + int32_t SetAudioPacketSize(uint16_t packetSizeSamples); + + // Store the audio level in dBov for + // header-extension-for-audio-level-indication. + // Valid range is [0,100]. Actual value is negative. + int32_t SetAudioLevel(uint8_t level_dBov); + + // Send a DTMF tone using RFC 2833 (4733) + int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); + + int AudioFrequency() const; + + // Set payload type for Redundant Audio Data RFC 2198 + int32_t SetRED(int8_t payloadType); + + // Get payload type for Redundant Audio Data RFC 2198 + int32_t RED(int8_t* payloadType) const; + + protected: + int32_t SendTelephoneEventPacket( + bool ended, + int8_t dtmf_payload_type, + uint32_t dtmfTimeStamp, + uint16_t duration, + bool markerBit); // set on first packet in talk burst + + bool MarkerBit(const FrameType frameType, const int8_t payloadType); + + private: + Clock* const _clock; + RTPSender* const _rtpSender; + RtpAudioFeedback* const _audioFeedback; + + rtc::scoped_ptr<CriticalSectionWrapper> _sendAudioCritsect; + + uint16_t _packetSizeSamples GUARDED_BY(_sendAudioCritsect); + + // DTMF + bool _dtmfEventIsOn; + bool _dtmfEventFirstPacketSent; + int8_t _dtmfPayloadType GUARDED_BY(_sendAudioCritsect); + uint32_t _dtmfTimestamp; + uint8_t _dtmfKey; + uint32_t _dtmfLengthSamples; + uint8_t _dtmfLevel; + int64_t _dtmfTimeLastSent; + uint32_t _dtmfTimestampLastSent; + + int8_t _REDPayloadType GUARDED_BY(_sendAudioCritsect); + + // VAD detection, used for markerbit + bool _inbandVADactive GUARDED_BY(_sendAudioCritsect); + int8_t _cngNBPayloadType GUARDED_BY(_sendAudioCritsect); + int8_t _cngWBPayloadType GUARDED_BY(_sendAudioCritsect); + int8_t _cngSWBPayloadType GUARDED_BY(_sendAudioCritsect); + int8_t _cngFBPayloadType GUARDED_BY(_sendAudioCritsect); + int8_t _lastPayloadType GUARDED_BY(_sendAudioCritsect); + + // Audio level indication + // (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/) + uint8_t _audioLevel_dBov GUARDED_BY(_sendAudioCritsect); }; } // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ +#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_ |