From adda9e4b4921366adaf4fb4cd42e815062838515 Mon Sep 17 00:00:00 2001 From: yanmorin <yanmorin> Date: Wed, 3 May 2006 21:04:36 +0000 Subject: [PATCH] Fix onhold/getcallstatus issues --- src/audio/audiortp.cpp | 3 ++- src/audio/tone.cpp | 3 ++- src/managerimpl.cpp | 2 +- src/sipvoiplink.cpp | 1 + 4 files changed, 6 insertions(+), 3 deletions(-) diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp index 98d0fbc59d..aacc61bd3a 100644 --- a/src/audio/audiortp.cpp +++ b/src/audio/audiortp.cpp @@ -97,7 +97,8 @@ AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, AudioLayer* driver, bool sym) { _sym = sym; _audioDevice = driver; if (_audioDevice!=0) { - _nbFrames = 20 * _audioDevice->getSampleRate()/1000; // 20 ms +// _nbFrames = 20 * _audioDevice->getSampleRate()/1000; // 20 ms + _nbFrames = _audioDevice->getSampleRate()/50; // 20 ms / 1000 } else { _nbFrames = RTP_FRAMES2SEND; } diff --git a/src/audio/tone.cpp b/src/audio/tone.cpp index 8a84e140b2..ab329dfd26 100644 --- a/src/audio/tone.cpp +++ b/src/audio/tone.cpp @@ -116,7 +116,8 @@ Tone::genSin(int16 *buffer, int frequency1, int frequency2, int nb) double var1 = (double)2 * (double)M_PI * (double)frequency1 / (double)_sampleRate; double var2 = (double)2 * (double)M_PI * (double)frequency2 / (double)_sampleRate; - double amp = (double)(INT16_AMPLITUDE >> 2); + // softer + double amp = (double)(INT16_AMPLITUDE >> 3); if (_nbChannel == 2) { // stereo int k = 0; for(int t = 0; t < nb; t++) { diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp index 49cbd91582..6ccde181ce 100644 --- a/src/managerimpl.cpp +++ b/src/managerimpl.cpp @@ -1237,7 +1237,7 @@ ManagerImpl::getCallStatus(const std::string& sequenceId) std::string destination; std::string number; - while (iter != _callAccountMap.begin()) + while (iter != _callAccountMap.end()) { Call* call = getAccountLink(iter->second)->getCall(iter->first); Call::ConnectionState state = call->getConnectionState(); diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp index d92a0704b9..8137b61b18 100644 --- a/src/sipvoiplink.cpp +++ b/src/sipvoiplink.cpp @@ -663,6 +663,7 @@ SIPVoIPLink::onhold(const CallID& id) // Stop sound call->setAudioStart(false); + call->setState(Call::Hold); _debug("SIP: Stopping AudioRTP when onhold\n"); _audiortp.closeRtpSession(); -- GitLab