* #10797: send DTMF over RTP as per RFC2833

They must be sent repeatedly and eventually with an end bit set.
Fixes regression from commit 9a8bc388

daemon/src/audio/audiortp/audio_rtp_record_handler.cpp

@@ -47,6 +47,14 @@ std::ofstream rtpResampled ("testRtpOutputResampled.raw", std::ifstream::binary)
 std::ofstream rtpNotResampled("testRtpOutput.raw", std::ifstream::binary);
 #endif
 
+DTMFEvent::DTMFEvent(int digit) : payload(), newevent(true), length(1000)
+{
+    payload.event = digit;
+    payload.ebit = false; // end of event bit
+    payload.rbit = false; // reserved bit
+    payload.duration = 1; // duration for this event
+}
+
 AudioRtpRecord::AudioRtpRecord() :
     audioCodec_(0)
     , audioCodecMutex_()
@@ -131,7 +139,8 @@ void AudioRtpRecordHandler::initNoiseSuppress()
 
 void AudioRtpRecordHandler::putDtmfEvent(int digit)
 {
-    audioRtpRecord_.dtmfQueue_.push_back(digit);
+    DTMFEvent dtmf(digit);
+    audioRtpRecord_.dtmfQueue_.push_back(dtmf);
 }
 
 int AudioRtpRecordHandler::processDataEncode()

daemon/src/audio/audiortp/audio_rtp_record_handler.h

@@ -67,6 +67,13 @@ timeval2microtimeout(const timeval& t)
     return ((t.tv_sec * 1000000ul) + t.tv_usec);
 }
 
+struct DTMFEvent {
+    DTMFEvent(int digit);
+    ost::RTPPacket::RFC2833Payload payload;
+    bool newevent;
+    int length;
+};
+
 /**
  * Class meant to store internal data in order to encode/decode,
  * resample, process, and packetize audio streams. This class should not be
@@ -90,7 +97,7 @@ class AudioRtpRecord {
         int codecSampleRate_;
         int codecFrameSize_;
         int converterSamplingRate_;
-        std::list<int> dtmfQueue_;
+        std::list<DTMFEvent> dtmfQueue_;
         SFLDataFormat fadeFactor_;
         NoiseSuppress *noiseSuppressEncode_;
         NoiseSuppress *noiseSuppressDecode_;
@@ -138,7 +145,7 @@ class AudioRtpRecordHandler {
             return audioRtpRecord_.hasDynamicPayloadType_;
         }
 
-        int DtmfPending() const {
+        bool hasDTMFPending() const {
             return not audioRtpRecord_.dtmfQueue_.empty();
         }
 

daemon/src/audio/audiortp/audio_rtp_session.cpp

@@ -110,25 +110,13 @@ void AudioRtpSession::setSessionMedia(AudioCodec &audioCodec)
     }
 }
 
-void AudioRtpSession::incrementTimestampForDTMF()
-{
-    timestamp_ += timestampIncrement_;
-}
-
 void AudioRtpSession::sendDtmfEvent()
 {
-    ost::RTPPacket::RFC2833Payload payload;
-
-    payload.event = audioRtpRecord_.dtmfQueue_.front();
-    payload.ebit = false; // end of event bit
-    payload.rbit = false; // reserved bit
-    payload.duration = 1; // duration for this event
-
-    audioRtpRecord_.dtmfQueue_.pop_front();
-
-    DEBUG("Send RTP Dtmf (%d)", payload.event);
+    DTMFEvent &dtmf(audioRtpRecord_.dtmfQueue_.front());
+    DEBUG("Send RTP Dtmf (%d)", dtmf.payload.event);
 
-    incrementTimestampForDTMF();
+    const int increment = getIncrementForDTMF();
+    timestamp_ += increment;
 
     // discard equivalent size of audio
     processDataEncode();
@@ -136,13 +124,29 @@ void AudioRtpSession::sendDtmfEvent()
     // change Payload type for DTMF payload
     queue_.setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) getDtmfPayloadType(), 8000));
 
-    queue_.setMark(true);
-    queue_.sendImmediate(timestamp_, (const unsigned char *)(&payload), sizeof(payload));
-    queue_.setMark(false);
+    // Set marker in case this is a new Event
+    if (dtmf.newevent)
+        queue_.setMark(true);
+    queue_.sendImmediate(timestamp_, (const unsigned char *) (& (dtmf.payload)), sizeof (ost::RTPPacket::RFC2833Payload));
 
-    // get back the payload to audio
+    // This is no longer a new event
+    if (dtmf.newevent) {
+        dtmf.newevent = false;
+        queue_.setMark(false);
+    }
+
+    // restore the payload to audio
     const ost::StaticPayloadFormat pf(static_cast<ost::StaticPayloadType>(getCodecPayloadType()));
     queue_.setPayloadFormat(pf);
+
+    // decrease length remaining to process for this event
+    dtmf.length -= increment;
+    dtmf.payload.duration++;
+    // next packet is going to be the last one
+    if ((dtmf.length - increment) < increment)
+        dtmf.payload.ebit = true;
+    if (dtmf.length < increment)
+        audioRtpRecord_.dtmfQueue_.pop_front();
 }
 
 
@@ -253,4 +257,9 @@ bool AudioRtpSession::onRTPPacketRecv(ost::IncomingRTPPkt&)
     return true;
 }
 
+int AudioRtpSession::getIncrementForDTMF() const
+{
+    return timestampIncrement_;
+}
+
 }

daemon/src/audio/audiortp/audio_rtp_session.h

@@ -66,6 +66,8 @@ class AudioRtpSession : public AudioRtpRecordHandler {
          */
         void updateDestinationIpAddress();
 
+        virtual int getIncrementForDTMF() const;
+
     protected:
         /**
          * Set the audio codec for this RTP session
@@ -122,11 +124,6 @@ class AudioRtpSession : public AudioRtpRecordHandler {
          */
         void receiveSpeakerData();
 
-        /**
-         * Increment timestamp for DTMF event
-         */
-        virtual void incrementTimestampForDTMF();
-
         // Main destination address for this rtp session.
         // Stored in case or reINVITE, which may require to forget
         // this destination and update a new one.

daemon/src/audio/audiortp/audio_symmetric_rtp_session.cpp

@@ -70,7 +70,7 @@ void AudioSymmetricRtpSession::AudioRtpThread::run()
 
     while (running_) {
         // Send session
-        if (rtpSession_.DtmfPending())
+        if (rtpSession_.hasDTMFPending())
             rtpSession_.sendDtmfEvent();
         else
             rtpSession_.sendMicData();

daemon/src/audio/audiortp/audio_zrtp_session.cpp

@@ -133,7 +133,7 @@ void AudioZrtpSession::run()
             timeout = getSchedulingTimeout();
 
         // Send session
-        if (DtmfPending())
+        if (hasDTMFPending())
             sendDtmfEvent();
         else
             sendMicData();
@@ -162,9 +162,9 @@ void AudioZrtpSession::run()
     DEBUG("Left main loop for call %s", call_.getCallId().c_str());
 }
 
-void AudioZrtpSession::incrementTimestampForDTMF()
+int AudioZrtpSession::getIncrementForDTMF() const
 {
-    timestamp_ += 160;
+    return 160;
 }
 
 void AudioZrtpSession::setSessionMedia(AudioCodec &audioCodec)

daemon/src/audio/audiortp/audio_zrtp_session.h

@@ -71,8 +71,8 @@ class AudioZrtpSession :
         void sendMicData();
         void initializeZid();
         std::string zidFilename_;
-        void incrementTimestampForDTMF();
         void setSessionMedia(AudioCodec &codec);
+        virtual int getIncrementForDTMF() const;
 };
 
 }