Searched refs:dtmfbuffer (Results 1 – 9 of 9) sorted by relevance
/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 438 uint8_t dtmfbuffer[IP_PACKET_SIZE]; in SendTelephoneEventPacket() local 450 _rtpSender->BuildRTPheader(dtmfbuffer, dtmf_payload_type, markerBit, in SendTelephoneEventPacket() 454 dtmfbuffer[0] &= 0xe0; in SendTelephoneEventPacket() 473 dtmfbuffer[12] = _dtmfKey; in SendTelephoneEventPacket() 474 dtmfbuffer[13] = E|R|volume; in SendTelephoneEventPacket() 475 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration); in SendTelephoneEventPacket() 481 retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1, in SendTelephoneEventPacket()
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 308 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 309 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 325 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 326 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 327 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 308 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 309 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 325 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 326 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 327 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 308 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 309 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 325 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 326 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 327 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 323 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 324 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 340 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 341 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 342 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 369 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 370 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 386 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 387 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 388 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/net-im/tg_owt/tg_owt-d578c76/src/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 377 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 378 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 394 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 395 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 396 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 377 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 378 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 394 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 395 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 396 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 380 uint8_t* dtmfbuffer = packet->AllocatePayload(kDtmfSize); in SendTelephoneEventPacket() local 381 RTC_DCHECK(dtmfbuffer); in SendTelephoneEventPacket() 397 dtmfbuffer[0] = dtmf_current_event_.key; in SendTelephoneEventPacket() 398 dtmfbuffer[1] = E | R | volume; in SendTelephoneEventPacket() 399 ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration); in SendTelephoneEventPacket()
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