From a4a9be041636a08ad45f2917f26159fbf0096ab5 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Fri, 22 Apr 2011 11:41:05 -0400
Subject: [PATCH] [#5564] Fix audio recording resampling for g722
---
sflphone-common/src/audio/audiorecord.cpp | 7 ++---
.../audio/audiortp/AudioRtpRecordHandler.cpp | 1 -
.../src/audio/audiortp/AudioRtpSession.cpp | 2 --
sflphone-common/src/audio/mainbuffer.cpp | 3 +-
.../src/audio/pulseaudio/audiostream.cpp | 2 +-
sflphone-common/src/managerimpl.cpp | 30 ++++++++-----------
6 files changed, 17 insertions(+), 28 deletions(-)
diff --git a/sflphone-common/src/audio/audiorecord.cpp b/sflphone-common/src/audio/audiorecord.cpp
index ab9751d992..343b65aa60 100644
--- a/sflphone-common/src/audio/audiorecord.cpp
+++ b/sflphone-common/src/audio/audiorecord.cpp
@@ -291,7 +291,7 @@ bool AudioRecord::setRawFile()
fp = fopen (savePath_.c_str(), "wb");
if (!fp) {
- _warn ("AudioRecord::setRawFile() : could not create RAW file!");
+ _warn ("AudioRecord: Could not create RAW file!");
return false;
}
@@ -308,7 +308,7 @@ bool AudioRecord::setRawFile()
bool AudioRecord::setWavFile()
{
- _debug ("AudioRecord: Create wave file %s", savePath_.c_str());
+ _debug ("AudioRecord: Create new wave file %s, sampling rate: %d", savePath_.c_str(), sndSmplRate_);
fp = fopen (savePath_.c_str(), "wb");
@@ -322,11 +322,8 @@ bool AudioRecord::setWavFile()
};
hdr.riff[3] = 'F';
-
hdr.wave[3] = 'E';
-
hdr.fmt[3] = ' ';
-
hdr.data[3] = 'a';
hdr.num_chans = channels_;
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index d5514b66c0..2d9167c258 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -158,7 +158,6 @@ void AudioRtpRecordHandler::updateRtpMedia (AudioCodec *audioCodec)
_audioRtpRecord.audioCodecMutex.leave();
- // Manager::instance().getMainBuffer()->setInternalSamplingRate (_audioRtpRecord._codecSampleRate);
Manager::instance().audioSamplingRateChanged(_audioRtpRecord._codecSampleRate);
if (lastSamplingRate != _audioRtpRecord._codecSampleRate)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index 6feaee0628..7ad415b7a1 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -123,7 +123,6 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
}
}
- // Set recording sampling rate
_ca->setRecordingSmplRate (getCodecSampleRate());
}
@@ -164,7 +163,6 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
}
- // Set recording sampling rate
_ca->setRecordingSmplRate (getCodecSampleRate());
_timestamp = getCurrentTimestamp();
diff --git a/sflphone-common/src/audio/mainbuffer.cpp b/sflphone-common/src/audio/mainbuffer.cpp
index 01a5dd87a8..6bdc431d89 100644
--- a/sflphone-common/src/audio/mainbuffer.cpp
+++ b/sflphone-common/src/audio/mainbuffer.cpp
@@ -49,10 +49,11 @@ MainBuffer::~MainBuffer()
void MainBuffer::setInternalSamplingRate (int sr)
{
+ _debug("MainBuffer: ------------------------------------- Set internal sampling rate");
if (sr > _internalSamplingRate) {
- _debug ("MainBuffer: Internal sampling rate changed %d", sr);
+ _debug ("MainBuffer: ---------------------------------- Internal sampling rate changed %d", sr);
// This call takes the mutex
flushAllBuffers();
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index 86f12f3dc7..63e7e166d3 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -198,7 +198,7 @@ AudioStream::createStream (pa_context* c, std::string *deviceName)
assert (pa_sample_spec_valid (&_sample_spec));
assert (pa_channel_map_valid (&channel_map));
- _info ("Audio: Create pulseaudio stream");
+ _info ("Audio: Create pulseaudio stream: %d", _sample_spec.rate);
pa_buffer_attr* attributes = (pa_buffer_attr*) malloc (sizeof (pa_buffer_attr));
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index d7c013bcf6..5b6d84f537 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -3072,22 +3072,25 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
int type, currentSamplerate, framesize, numCardIn, numCardOut, numCardRing;
std::string alsaPlugin;
- _debug ("Manager: Audio sampling rate changed");
-
- if (!_audiodriver)
+ if (!_audiodriver) {
+ _debug("Manager: No Audio driver set");
return;
+ }
// Only modify internal sampling rate if new sampling rate is higher
currentSamplerate = _mainBuffer.getInternalSamplingRate();
- if(currentSamplerate <= samplerate)
+ if(currentSamplerate >= samplerate) {
+ _debug("Manager: No need to update audio layer sampling rate");
return;
+ }
+ else {
+ _debug ("Manager: Audio sampling rate changed");
+ }
type = _audiodriver->getLayerType();
-
- // samplerate = _mainBuffer.getInternalSamplingRate();
framesize = audioPreference.getFramesize();
- _debug ("Manager: new samplerate: %d, new framesize %d", samplerate, framesize);
+ _debug ("Manager: New samplerate: %d, New framesize %d", samplerate, framesize);
alsaPlugin = audioPreference.getPlugin();
@@ -3152,18 +3155,9 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
_dtmfKey = new DTMF (sampleRate);
}
- if (hasCurrentCall())
+ if (hasCurrentCall()) {
_audiodriver->startStream();
-
- // ost::MutexLock unlock (*getAudioLayerMutex());
- // getAudioLayerMutex()->leave();
-
- // need to stop audio streams if there is currently no call
- // if ( (type != PULSEAUDIO) && (!hasCurrentCall())) {
- // _debug("There is currently a call!!");
- // _audiodriver->stopStream();
-
- // }
+ }
}
/**
--
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