From 71d6b7bf210a6043a8c778318139ff228b1d685f Mon Sep 17 00:00:00 2001
From: asavard <asavard@asavard-KT378AA-A2L-a6552f.(none)>
Date: Tue, 26 Oct 2010 16:27:41 -0400
Subject: [PATCH] [#4367] Fix AudioRtpSession putDtmfEvent shadowing
---
.../audio/audiortp/AudioRtpRecordHandler.cpp | 32 +++++++++----------
.../src/audio/audiortp/AudioRtpSession.h | 2 --
.../src/audio/audiortp/AudioZrtpSession.h | 2 --
3 files changed, 16 insertions(+), 20 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index e7f104fe78..e5a59b0911 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -475,22 +475,22 @@ void AudioRtpRecordHandler::processDataDecode(unsigned char *spkrData, unsigned
}
}
- bool AudioRtpRecordHandler::fadeIn(SFLDataFormat *audio, int size, SFLDataFormat *factor)
- {
- // apply amplitude factor;
- while(size){
- size--;
- audio[size] /= *factor;
- }
- // decrease factor
- *factor /= FADEIN_STEP_SIZE
- ;
- // if factor reach 0, thsi function should no be called anymore
- if(*factor == 0)
- return true;
-
- return false;
- }
+bool AudioRtpRecordHandler::fadeIn(SFLDataFormat *audio, int size, SFLDataFormat *factor)
+{
+ // apply amplitude factor;
+ while(size){
+ size--;
+ audio[size] /= *factor;
+ }
+ // decrease factor
+ *factor /= FADEIN_STEP_SIZE;
+
+ // if factor reach 0, thsi function should no be called anymore
+ if(*factor == 0)
+ return true;
+
+ return false;
+}
}
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 0c4eec82d3..272d2b68ca 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -78,8 +78,6 @@ class AudioRtpSession : public ost::TimerPort, public ost::SymmetricRTPSession,
*/
void updateDestinationIpAddress (void);
- void putDtmfEvent (int digit);
-
/**
* Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
* incremented as if it was microphone audio. This function change the payload type of the rtp session,
diff --git a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
index 190aba9060..8cc5531223 100644
--- a/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioZrtpSession.h
@@ -64,8 +64,6 @@ class AudioZrtpSession : public ost::TimerPort, public ost::SymmetricZRTPSession
*/
void updateDestinationIpAddress (void);
- void putDtmfEvent (int digit);
-
/**
* Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
* incremented as if it was microphone audio. This function change the payload type of the rtp session,
--
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