From b4390a3f803df6932dc1b5e1271d0a2224eafb02 Mon Sep 17 00:00:00 2001
From: asavard <asavard@asavard-KT378AA-A2L-a6552f.(none)>
Date: Wed, 27 Oct 2010 11:52:10 -0400
Subject: [PATCH] [#4367] Refactor mic data encoding in rtp session
---
.../audio/audiortp/AudioRtpRecordHandler.cpp | 18 ++++++------------
.../src/audio/audiortp/AudioRtpSession.cpp | 11 ++++-------
2 files changed, 10 insertions(+), 19 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index 7995247659..bdab551501 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -405,26 +405,19 @@ int AudioRtpRecordHandler::processDataEncode(void)
float fixedCodecFramesize = computeCodecFrameSize(getCodecFrameSize(), getCodecSampleRate());
// compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
- int maxBytesToGet = computeNbByteAudioLayer(fixedCodecFramesize);
+ int bytesToGet = computeNbByteAudioLayer(fixedCodecFramesize);
+
+ _debug(" byte to get %d", bytesToGet);
// available bytes inside ringbuffer
int availBytesFromMic = audioLayer->getMainBuffer()->availForGet(_ca->getCallId());
_debug(" avail byte from mic %d", availBytesFromMic);
- if(!(availBytesFromMic > 320))
+ if(availBytesFromMic < bytesToGet)
return 0;
- // set available byte to maxByteToGet
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
-
- if(bytesAvail == 0){
- memset(micDataEncoded, 0, sizeof (SFLDataFormat));
- return getCodecFrameSize();
- }
-
// Get bytes from micRingBuffer to data_from_mic
- // int nbSample = audioLayer->getMainBuffer()->getData(micData, bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
- int nbSample = audioLayer->getMainBuffer()->getData(micData, 320, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
+ int nbSample = audioLayer->getMainBuffer()->getData(micData, bytesToGet, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
// process mic fade in
if(!_audioRtpRecord.getMicFadeInComplete())
@@ -453,6 +446,7 @@ int AudioRtpRecordHandler::processDataEncode(void)
// no resampling required
compSize = audioCodec->codecEncode(micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
}
+
return compSize;
}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index d80dde79f0..ea0a6529a8 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -235,7 +235,7 @@ bool AudioRtpSession::onRTPPacketRecv (ost::IncomingRTPPkt&)
void AudioRtpSession::sendMicData()
{
- // _debug("============== sendMicData ===============");
+ _debug("============== sendMicData ===============");
int compSize = processDataEncode();
@@ -246,8 +246,8 @@ void AudioRtpSession::sendMicData()
// Increment timestamp for outgoing packet
_timestamp += _timestampIncrement;
- // _debug(" compSize: %d", compSize);
- // _debug(" timestamp: %d", _timestamp);
+ _debug(" compSize: %d", compSize);
+ _debug(" timestamp: %d", _timestamp);
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
@@ -357,10 +357,7 @@ void AudioRtpSession::run ()
sendMicData ();
}
- // Recv session
- // TODO should not be called here anymore
-// receiveSpeakerData ();
-
+ // This also should be moved
notifyIncomingCall();
_manager->getAudioLayerMutex()->leave();
--
GitLab