Removed dead code.
Review URL: https://webrtc-codereview.appspot.com/352010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1437 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
parent
aafa5a331c
commit
df9bd9bdbd
@ -249,292 +249,254 @@ RTPSenderAudio::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const
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return false;
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}
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WebRtc_Word32
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RTPSenderAudio::SendAudio(const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 captureTimeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 dataSize,
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const RTPFragmentationHeader* fragmentation)
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{
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WebRtc_UWord16 payloadSize = (WebRtc_UWord16)dataSize;
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WebRtc_UWord16 maxPayloadLength = _rtpSender->MaxPayloadLength();
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bool dtmfToneStarted = false;
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WebRtc_UWord16 dtmfLengthMS = 0;
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WebRtc_UWord8 key = 0;
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WebRtc_Word32 RTPSenderAudio::SendAudio(
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const FrameType frameType,
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 captureTimeStamp,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord32 dataSize,
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const RTPFragmentationHeader* fragmentation) {
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// TODO(pwestin) Breakup function in smaller functions.
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WebRtc_UWord16 payloadSize = static_cast<WebRtc_UWord16>(dataSize);
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WebRtc_UWord16 maxPayloadLength = _rtpSender->MaxPayloadLength();
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bool dtmfToneStarted = false;
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WebRtc_UWord16 dtmfLengthMS = 0;
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WebRtc_UWord8 key = 0;
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// Check if we have pending DTMFs to send
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if ( !_dtmfEventIsOn && PendingDTMF())
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{
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CriticalSectionScoped cs(_sendAudioCritsect);
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// Check if we have pending DTMFs to send
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if (!_dtmfEventIsOn && PendingDTMF()) {
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CriticalSectionScoped cs(_sendAudioCritsect);
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WebRtc_UWord32 delaySinceLastDTMF = (_clock.GetTimeInMS() - _dtmfTimeLastSent);
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WebRtc_UWord32 delaySinceLastDTMF = _clock.GetTimeInMS() -
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_dtmfTimeLastSent;
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if(delaySinceLastDTMF > 100)
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{
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// New tone to play
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_dtmfTimestamp = captureTimeStamp;
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if (NextDTMF(&key, &dtmfLengthMS, &_dtmfLevel) >= 0)
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{
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_dtmfEventFirstPacketSent = false;
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_dtmfKey = key;
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_dtmfLengthSamples = (_frequency/1000)*dtmfLengthMS;
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dtmfToneStarted = true;
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_dtmfEventIsOn = true;
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}
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}
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if (delaySinceLastDTMF > 100) {
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// New tone to play
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_dtmfTimestamp = captureTimeStamp;
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if (NextDTMF(&key, &dtmfLengthMS, &_dtmfLevel) >= 0) {
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_dtmfEventFirstPacketSent = false;
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_dtmfKey = key;
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_dtmfLengthSamples = (_frequency / 1000) * dtmfLengthMS;
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dtmfToneStarted = true;
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_dtmfEventIsOn = true;
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}
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}
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if(dtmfToneStarted)
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{
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CriticalSectionScoped cs(_audioFeedbackCritsect);
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if(_audioFeedback)
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{
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_audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
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}
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if (dtmfToneStarted) {
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CriticalSectionScoped cs(_audioFeedbackCritsect);
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if (_audioFeedback) {
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_audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
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}
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}
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// A source MAY send events and coded audio packets for the same time
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// but we don't support it
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{
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_sendAudioCritsect->Enter();
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if (_dtmfEventIsOn) {
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if (frameType == kFrameEmpty) {
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// kFrameEmpty is used to drive the DTMF when in CN mode
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// it can be triggered more frequently than we want to send the
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// DTMF packets.
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if (_packetSizeSamples > (captureTimeStamp - _dtmfTimestampLastSent)) {
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// not time to send yet
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_sendAudioCritsect->Leave();
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return 0;
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}
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}
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_dtmfTimestampLastSent = captureTimeStamp;
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WebRtc_UWord32 dtmfDurationSamples = captureTimeStamp - _dtmfTimestamp;
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bool ended = false;
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bool send = true;
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if (_dtmfLengthSamples > dtmfDurationSamples) {
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if (dtmfDurationSamples <= 0) {
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// Skip send packet at start, since we shouldn't use duration 0
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send = false;
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}
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} else {
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ended = true;
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_dtmfEventIsOn = false;
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_dtmfTimeLastSent = _clock.GetTimeInMS();
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}
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// don't hold the critsect while calling SendTelephoneEventPacket
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_sendAudioCritsect->Leave();
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if (send) {
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if (dtmfDurationSamples > 0xffff) {
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// RFC 4733 2.5.2.3 Long-Duration Events
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SendTelephoneEventPacket(ended, _dtmfTimestamp,
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static_cast<WebRtc_UWord16>(0xffff), false);
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// set new timestap for this segment
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_dtmfTimestamp = captureTimeStamp;
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dtmfDurationSamples -= 0xffff;
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_dtmfLengthSamples -= 0xffff;
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return SendTelephoneEventPacket(
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ended,
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_dtmfTimestamp,
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static_cast<WebRtc_UWord16>(dtmfDurationSamples),
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false);
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} else {
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// set markerBit on the first packet in the burst
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_dtmfEventFirstPacketSent = true;
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return SendTelephoneEventPacket(
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ended,
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_dtmfTimestamp,
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static_cast<WebRtc_UWord16>(dtmfDurationSamples),
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!_dtmfEventFirstPacketSent);
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}
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}
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return 0;
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}
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_sendAudioCritsect->Leave();
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}
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if (payloadSize == 0 || payloadData == NULL) {
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if (frameType == kFrameEmpty) {
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// we don't send empty audio RTP packets
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// no error since we use it to drive DTMF when we use VAD
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return 0;
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}
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return -1;
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}
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WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
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bool markerBit = MarkerBit(frameType, payloadType);
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WebRtc_Word32 rtpHeaderLength = 0;
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WebRtc_UWord16 timestampOffset = 0;
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if (_REDPayloadType >= 0 && fragmentation && !markerBit &&
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fragmentation->fragmentationVectorSize > 1) {
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// have we configured RED? use its payload type
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// we need to get the current timestamp to calc the diff
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WebRtc_UWord32 oldTimeStamp = _rtpSender->Timestamp();
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rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, _REDPayloadType,
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markerBit, captureTimeStamp);
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timestampOffset = WebRtc_UWord16(_rtpSender->Timestamp() - oldTimeStamp);
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} else {
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rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, payloadType,
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markerBit, captureTimeStamp);
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}
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if (rtpHeaderLength <= 0) {
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return -1;
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}
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{
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CriticalSectionScoped cs(_sendAudioCritsect);
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// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
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if (_includeAudioLevelIndication) {
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dataBuffer[0] |= 0x10; // set eXtension bit
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/*
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0 1 2 3
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0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| 0xBE | 0xDE | length=1 |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| ID | len=0 |V| level | 0x00 | 0x00 |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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// add our ID (0xBEDE)
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength,
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RTP_AUDIO_LEVEL_UNIQUE_ID);
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rtpHeaderLength += 2;
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// add the length (length=1) in number of word32
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const WebRtc_UWord8 length = 1;
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength,
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length);
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rtpHeaderLength += 2;
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// add ID (defined by the user) and len(=0) byte
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const WebRtc_UWord8 id = _audioLevelIndicationID;
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const WebRtc_UWord8 len = 0;
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dataBuffer[rtpHeaderLength++] = (id << 4) + len;
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// add voice-activity flag (V) bit and the audio level (in dBov)
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const WebRtc_UWord8 V = (frameType == kAudioFrameSpeech);
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WebRtc_UWord8 level = _audioLevel_dBov;
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dataBuffer[rtpHeaderLength++] = (V << 7) + level;
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// add two bytes zero padding
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, 0);
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rtpHeaderLength += 2;
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}
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// A source MAY send events and coded audio packets for the same time
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// but we don't support it
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{
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_sendAudioCritsect->Enter();
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if (_dtmfEventIsOn)
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{
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if(frameType == kFrameEmpty)
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{
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// kFrameEmpty is used to drive the DTMF when in CN mode
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// it can be triggered more frequently than we want to send the DTMF packets
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if(_packetSizeSamples > (captureTimeStamp - _dtmfTimestampLastSent) )
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{
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// not time to send yet
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_sendAudioCritsect->Leave();
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return 0;
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}
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}
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_dtmfTimestampLastSent = captureTimeStamp;
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WebRtc_UWord32 dtmfDurationSamples = (captureTimeStamp - _dtmfTimestamp);
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bool ended = false;
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bool send = true;
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if(_dtmfLengthSamples > dtmfDurationSamples)
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{
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if (dtmfDurationSamples > 0) // Skip send packet at start, since we shouldn't use duration 0
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{
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} else
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{
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send = false;
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}
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}else
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{
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ended = true;
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_dtmfEventIsOn = false;
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_dtmfTimeLastSent = _clock.GetTimeInMS();
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}
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// don't hold the critsect while calling SendTelephoneEventPacket
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_sendAudioCritsect->Leave();
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if(send)
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{
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if(dtmfDurationSamples > 0xffff)
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{
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// RFC 4733 2.5.2.3 Long-Duration Events
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SendTelephoneEventPacket(ended, _dtmfTimestamp, (WebRtc_UWord16)0xffff, false);
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// set new timestap for this segment
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_dtmfTimestamp = captureTimeStamp;
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dtmfDurationSamples -= 0xffff;
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_dtmfLengthSamples -= 0xffff;
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return SendTelephoneEventPacket(ended, _dtmfTimestamp, (WebRtc_UWord16)dtmfDurationSamples, false);
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} else
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{
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// set markerBit on the first packet in the burst
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WebRtc_Word32 retVal = SendTelephoneEventPacket(ended, _dtmfTimestamp, (WebRtc_UWord16)dtmfDurationSamples, !_dtmfEventFirstPacketSent);
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_dtmfEventFirstPacketSent = true;
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return retVal;
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}
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}
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return(0);
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}
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_sendAudioCritsect->Leave();
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}
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if(payloadSize == 0 || payloadData == NULL)
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{
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if(frameType == kFrameEmpty)
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{
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// we don't send empty audio RTP packets
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// no error since we use it to drive DTMF when we use VAD
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return 0;
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}else
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{
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return -1;
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}
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if(maxPayloadLength < rtpHeaderLength + payloadSize ) {
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// too large payload buffer
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return -1;
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}
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WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
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bool markerBit = MarkerBit(frameType, payloadType);
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WebRtc_Word32 rtpHeaderLength = 0;
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WebRtc_UWord16 timestampOffset = 0;
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if( _REDPayloadType >= 0 &&
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if (_REDPayloadType >= 0 && // Have we configured RED?
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fragmentation &&
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fragmentation->fragmentationVectorSize > 1 &&
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!markerBit)
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{
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// have we configured RED? use its payload type
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// we need to get the current timestamp to calc the diff
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WebRtc_UWord32 oldTimeStamp = _rtpSender->Timestamp();
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rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, _REDPayloadType, markerBit, captureTimeStamp);
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timestampOffset = WebRtc_UWord16(_rtpSender->Timestamp() - oldTimeStamp);
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} else
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{
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rtpHeaderLength= _rtpSender->BuildRTPheader(dataBuffer, payloadType, markerBit, captureTimeStamp);
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!markerBit) {
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if (timestampOffset <= 0x3fff) {
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if(fragmentation->fragmentationVectorSize != 2) {
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// we only support 2 codecs when using RED
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return -1;
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}
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// only 0x80 if we have multiple blocks
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dataBuffer[rtpHeaderLength++] = 0x80 +
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fragmentation->fragmentationPlType[1];
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WebRtc_UWord32 blockLength = fragmentation->fragmentationLength[1];
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// sanity blockLength
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if(blockLength > 0x3ff) { // block length 10 bits 1023 bytes
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return -1;
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}
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WebRtc_UWord32 REDheader = (timestampOffset << 10) + blockLength;
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ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength,
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REDheader);
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rtpHeaderLength += 3;
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dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
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// copy the RED data
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memcpy(dataBuffer+rtpHeaderLength,
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payloadData + fragmentation->fragmentationOffset[1],
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fragmentation->fragmentationLength[1]);
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// copy the normal data
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memcpy(dataBuffer+rtpHeaderLength +
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fragmentation->fragmentationLength[1],
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payloadData + fragmentation->fragmentationOffset[0],
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fragmentation->fragmentationLength[0]);
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payloadSize = static_cast<WebRtc_UWord16>(
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fragmentation->fragmentationLength[0] +
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fragmentation->fragmentationLength[1]);
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} else {
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// silence for too long send only new data
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dataBuffer[rtpHeaderLength++] = static_cast<WebRtc_UWord8>(payloadType);
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memcpy(dataBuffer+rtpHeaderLength,
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payloadData + fragmentation->fragmentationOffset[0],
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fragmentation->fragmentationLength[0]);
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payloadSize = static_cast<WebRtc_UWord16>(
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fragmentation->fragmentationLength[0]);
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}
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} else {
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if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
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// use the fragment info if we have one
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memcpy( dataBuffer+rtpHeaderLength,
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payloadData + fragmentation->fragmentationOffset[0],
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fragmentation->fragmentationLength[0]);
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payloadSize = static_cast<WebRtc_UWord16>(
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fragmentation->fragmentationLength[0]);
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} else {
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memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
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}
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}
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if(rtpHeaderLength == -1)
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{
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return -1;
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}
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{
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CriticalSectionScoped cs(_sendAudioCritsect);
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if (_includeAudioLevelIndication)
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{
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dataBuffer[0] |= 0x10; // set eXtension bit
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// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
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/*
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0 1 2 3
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0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| 0xBE | 0xDE | length=1 |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| ID | len=0 |V| level | 0x00 | 0x00 |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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// add the extension
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// add our ID (0xBEDE)
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, RTP_AUDIO_LEVEL_UNIQUE_ID);
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rtpHeaderLength += 2;
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// add the length (length=1) in number of word32
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const WebRtc_UWord8 length = 1;
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, length);
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rtpHeaderLength += 2;
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// add ID (defined by the user) and len(=0) byte
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const WebRtc_UWord8 id = _audioLevelIndicationID;
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const WebRtc_UWord8 len = 0;
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dataBuffer[rtpHeaderLength++] = (id << 4) + len;
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// add voice-activity flag (V) bit and the audio level (in dBov)
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const WebRtc_UWord8 V = (frameType == kAudioFrameSpeech);
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WebRtc_UWord8 level = _audioLevel_dBov;
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dataBuffer[rtpHeaderLength++] = (V << 7) + level;
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// add two bytes zero padding
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ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, 0);
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rtpHeaderLength += 2;
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}
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if(maxPayloadLength < rtpHeaderLength + payloadSize )
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{
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// too large payload buffer
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return -1;
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}
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if( _REDPayloadType >= 0 && // have we configured RED?
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fragmentation &&
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fragmentation->fragmentationVectorSize > 1 &&
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!markerBit)
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{
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if(fragmentation == NULL)
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{
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// this can't happen any more but save the code incase we want to use it later again
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// we don't send this type of packet due to old NetEq issue
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dataBuffer[rtpHeaderLength++] = (WebRtc_UWord8)payloadType;
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memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
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}else
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{
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if( fragmentation->fragmentationVectorSize > 1 &&
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!markerBit && // markerBit == first packet
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timestampOffset <= 0x3fff) // silence for too long send only new data
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{
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if(fragmentation->fragmentationVectorSize != 2)
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{
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// we only support 2 codecs when using RED
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return -1;
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}
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// only 0x80 if we have multiple blocks
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dataBuffer[rtpHeaderLength++] = 0x80 + fragmentation->fragmentationPlType[1];
|
||||
WebRtc_UWord32 blockLength = fragmentation->fragmentationLength[1];
|
||||
|
||||
// sanity blockLength
|
||||
if(blockLength > 0x3ff) // block length 10 bits 1023 bytes
|
||||
{
|
||||
return -1;
|
||||
}
|
||||
WebRtc_UWord32 REDheader = (timestampOffset << 10) + blockLength;
|
||||
ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer+rtpHeaderLength, REDheader);
|
||||
rtpHeaderLength += 3;
|
||||
|
||||
dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
|
||||
// copy the RED data
|
||||
memcpy(dataBuffer+rtpHeaderLength,
|
||||
payloadData + fragmentation->fragmentationOffset[1],
|
||||
fragmentation->fragmentationLength[1]);
|
||||
|
||||
// copy the normal data
|
||||
memcpy( dataBuffer+rtpHeaderLength + fragmentation->fragmentationLength[1],
|
||||
payloadData + fragmentation->fragmentationOffset[0],
|
||||
fragmentation->fragmentationLength[0]);
|
||||
|
||||
payloadSize = WebRtc_UWord16(fragmentation->fragmentationLength[0] + fragmentation->fragmentationLength[1]);
|
||||
|
||||
} else
|
||||
{
|
||||
dataBuffer[rtpHeaderLength++] = (WebRtc_UWord8)payloadType;
|
||||
|
||||
memcpy( dataBuffer+rtpHeaderLength,
|
||||
payloadData + fragmentation->fragmentationOffset[0],
|
||||
fragmentation->fragmentationLength[0]);
|
||||
|
||||
payloadSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
|
||||
}
|
||||
}
|
||||
}else
|
||||
{
|
||||
if( fragmentation &&
|
||||
fragmentation->fragmentationVectorSize > 0)
|
||||
{
|
||||
// use the fragment info if we have one
|
||||
memcpy( dataBuffer+rtpHeaderLength,
|
||||
payloadData + fragmentation->fragmentationOffset[0],
|
||||
fragmentation->fragmentationLength[0]);
|
||||
|
||||
payloadSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
|
||||
|
||||
}else
|
||||
{
|
||||
memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
|
||||
}
|
||||
}
|
||||
_lastPayloadType = payloadType;
|
||||
|
||||
} // end critical section
|
||||
|
||||
return _rtpSender->SendToNetwork(
|
||||
dataBuffer,
|
||||
payloadSize,
|
||||
static_cast<WebRtc_UWord16>(rtpHeaderLength),
|
||||
kAllowRetransmission);
|
||||
_lastPayloadType = payloadType;
|
||||
} // end critical section
|
||||
return _rtpSender->SendToNetwork(dataBuffer,
|
||||
payloadSize,
|
||||
static_cast<WebRtc_UWord16>(rtpHeaderLength),
|
||||
kAllowRetransmission);
|
||||
}
|
||||
|
||||
|
||||
WebRtc_Word32
|
||||
RTPSenderAudio::SetAudioLevelIndicationStatus(const bool enable,
|
||||
const WebRtc_UWord8 ID)
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user