From 608ed45f1c2175a7b0e10c5c70da0fc22319d380 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.net>
Date: Thu, 22 Oct 2009 13:39:16 -0400
Subject: [PATCH] [#2333] Change (again) Pulseaudio settings to fit logiteck
usb hdw requirement
---
sflphone-client-gnome/doc/C/sflphone.xml | 52 +++++++++++--------
.../src/audio/pulseaudio/audiostream.cpp | 22 ++++----
.../src/audio/pulseaudio/pulselayer.cpp | 43 +++++++++++++--
.../src/audio/pulseaudio/pulselayer.h | 2 +
sflphone-common/src/managerimpl.cpp | 9 ++++
5 files changed, 90 insertions(+), 38 deletions(-)
diff --git a/sflphone-client-gnome/doc/C/sflphone.xml b/sflphone-client-gnome/doc/C/sflphone.xml
index 00cb6a6d18..07f31faa79 100644
--- a/sflphone-client-gnome/doc/C/sflphone.xml
+++ b/sflphone-client-gnome/doc/C/sflphone.xml
@@ -99,8 +99,11 @@
<application>SFLphone</application> is a free software and is distributed under the GNU General Public License version 3. It is developed by Savoir-Faire Linux, a Canadian Linux consulting company, in partnership with the global community.
</para>
<para>
- Among the many features we developed for you, we could highlight the high definition sound (wide-band audio codecs - speex, G722, Celt), audio recording, voicemail notification and call history.
- More than a simple softphone, <application>SFLphone</application> supports advanced enterprise-class call features: unlimited number of calls, call transfer and on/off hold option.
+ Among the many features we developed for you, we could
+ highlight the high definition sound (wide-band audio
+ codecs - speex, G722, Celt), multiple conference call,
+ audio recording, voicemail notification, and call history.
+ More than a simple softphone, <application>SFLphone</application> supports advanced enterprise-class call features: unlimited number of calls, call transfer, and on/off hold option.
</para>
<para>
This release also provides advanced security features (secure RTP and TLS).
@@ -137,7 +140,8 @@
<sect1 id="accounts">
<title>Managing your accounts</title>
<para>
- <application>SFLphone</application> supports both IAX2 and SIP accounts.
+ <application>SFLphone</application> supports both IAX2
+ and SIP accounts.
</para>
<sect2 id="accounts_basic">
@@ -507,36 +511,34 @@
</sect2>
<sect2 id="conferencecall" label="Conference calls">
- <para>SFLphone now supports conference calling
- integrating server like features in a simple
- GUI.</para>
+ <para>SFLphone now supports conference call hosting
+ integrating server like features in a simple GUI.
+ </para>
<para>
<itemizedlist>
<listitem>
<guilabel>Create a conference</guilabel>
- <para>To create a conference, drag and drop one call
+ <para>To host a conference, simply drag and drop one call
on another. Additional participants are
added the same way, dragging a call on the
- conference icon. SFLphone automatically
- includes into the newly created conference.
+ conference icon.
</para>
</listitem>
<listitem>
<guilabel>Leave a conference</guilabel>
<para>
- The Main Participant may leave at any time to
- answer any other incoming communication or initiate
- new ones. The conference is not interupted if Main
- Participant left it. Double clicking the conference
- let the Main Participant reintroduce
- the conference.
+ SFLphone conference model let you leave a conference that
+ you are currently hosting to answer any other incoming
+ communication or even initiate new ones. The conference is
+ not interupted Double clicking the conference icon
+ let you reintroduce the conference.
</para>
</listitem>
<listitem>
<guilabel>Multiple conference</guilabel>
<para>
SFLphone supports multiple conferences running
- simultaneously. Conferences may be joined the same way
+ simultaneously. Two conferences can be joined the same way
they are created, dragging one on the
other.
</para>
@@ -582,33 +584,39 @@
<para>
<itemizedlist>
<listitem>
- <guilabel>PCMU</guilabel>
+ <guilabel>PCMU/PCMA</guilabel>
<para>
+ ITU-T telefony standard PCM formats, 8kHz, 64
+ kbit/s, using logarithmic byte compression algorithm.
</para>
</listitem>
- <listitem>
- <guilabel>PCMA</guilabel>
- <para>
- </para>
- </listitem>
<listitem>
<guilabel>GSM</guilabel>
<para>
+ Global System for Mobile communications (GSM)
+ narrowband 8kHz standard based on linear prediction encoding.
</para>
</listitem>
<listitem>
<guilabel>G722</guilabel>
<para>
+ ITU-T standard wideband 16kHz standard based on linear prediction.
</para>
</listitem>
<listitem>
<guilabel>SPEEX</guilabel>
<para>
+ High quality voice encoding/decoding available
+ in narrowband 8Khz, wideband 16khz (HD Voice),
+ and ultra-wideband 32 kHz.
+ Integrate additional features such as Variable Bit
+ Rate (VBR) and noise reduction.
</para>
</listitem>
<listitem>
<guilabel>CELT</guilabel>
<para>
+
</para>
</listitem>
</itemizedlist>
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index 54309276ec..5e10a6b91f 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -184,27 +184,27 @@ AudioStream::createStream (pa_context* c)
// parameters are defined as number of bytes
// 2048 bytes (1024 int16) is 20 ms at 44100 Hz
if (_streamType == PLAYBACK_STREAM) {
- attributes->maxlength = 16000;
- attributes->tlength = 8192;
+ attributes->maxlength = 88200;
+ attributes->tlength = 22050;
attributes->prebuf = 4096; // Pulseaudio will not start if prebuffering is not reached
- attributes->minreq = 2048; // The server side playback framesize
- attributes->fragsize = 4096; // Fragment size at wich we receive an interupt
- pa_stream_connect_playback( s , NULL , attributes, PA_STREAM_INTERPOLATE_TIMING, &_volume, NULL);
+ attributes->minreq = 2048; // The server side playback framesize
+ attributes->fragsize = 11025; // Fragment size at wich we receive an interupt
+ pa_stream_connect_playback( s , NULL , attributes, PA_STREAM_NOFLAGS, &_volume, NULL);
// pa_stream_connect_playback (s , NULL , attributes, PA_STREAM_START_CORKED, &_volume, NULL);
} else if (_streamType == CAPTURE_STREAM) {
// attributes->maxlength = 66500;
// attributes->fragsize = (uint32_t)-1;
- attributes->maxlength = 32000;
- attributes->tlength = (uint32_t)-1;
- attributes->prebuf = (uint32_t)-1;
- attributes->minreq = (uint32_t)-1;
- attributes->fragsize = 4096;
+ attributes->maxlength = 88200;
+ attributes->tlength = 22050;
+ attributes->prebuf = 4096;
+ attributes->minreq = 2048;
+ attributes->fragsize = 11025;
// pa_stream_connect_record (s , NULL , attributes , PA_STREAM_START_CORKED);
// pa_stream_connect_record( s , NULL , attributes , PA_STREAM_INTERPOLATE_TIMING );
- pa_stream_connect_record( s, NULL, NULL, PA_STREAM_INTERPOLATE_TIMING );
+ pa_stream_connect_record( s, NULL, NULL, PA_STREAM_NOFLAGS);
} else if (_streamType == UPLOAD_STREAM) {
pa_stream_connect_upload (s , 1024);
} else {
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index e46e1e9524..7801582cb7 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -30,6 +30,7 @@ static void playback_callback (pa_stream* s, size_t bytes, void* userdata)
assert (s && bytes);
assert (bytes > 0);
static_cast<PulseLayer*> (userdata)->processPlaybackData();
+ // static_cast<PulseLayer*> (userdata)->processData();
}
static void capture_callback (pa_stream* s, size_t bytes, void* userdata)
@@ -39,6 +40,7 @@ static void capture_callback (pa_stream* s, size_t bytes, void* userdata)
assert(s && bytes);
assert(bytes > 0);
static_cast<PulseLayer*> (userdata)->processCaptureData();
+ // static_cast<PulseLayer*> (userdata)->processData();
}
@@ -357,12 +359,15 @@ void PulseLayer::startStream (void)
// Create Streams
connectPulseAudioServer();
- _urgentRingBuffer.flushAll();
- _mainBuffer.flushAllBuffers();
+ // _urgentRingBuffer.flushAll();
+ // _mainBuffer.flushAllBuffers();
is_started = true;
}
+ _urgentRingBuffer.flushAll();
+ _mainBuffer.flushAllBuffers();
+
}
void
@@ -451,6 +456,30 @@ void PulseLayer::processCaptureData(void)
}
+
+void PulseLayer::processData(void)
+{
+
+ // Handle the data for the speakers
+ if ( playback &&(playback->pulseStream()) && (pa_stream_get_state (playback->pulseStream()) == PA_STREAM_READY)) {
+
+ // _debug("PulseLayer::processPlaybackData()\n");
+
+ // If the playback buffer is full, we don't overflow it; wait for it to have free space
+ if (pa_stream_writable_size (playback->pulseStream()) == 0)
+ return;
+
+ writeToSpeaker();
+ }
+
+ // Handle the mic
+ // We check if the stream is ready
+ if ( record &&(record->pulseStream()) && (pa_stream_get_state (record->pulseStream()) == PA_STREAM_READY))
+ readFromMic();
+
+}
+
+
void PulseLayer::writeToSpeaker (void)
{
/** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
@@ -471,9 +500,9 @@ void PulseLayer::writeToSpeaker (void)
urgentAvailBytes = _urgentRingBuffer.AvailForGet();
- if (urgentAvailBytes > 0) {
+ if (urgentAvailBytes > (framesPerBuffer*sizeof(SFLDataFormat))) {
- // _debug("urgentAvailBytes: %i\n", urgentAvailBytes);
+ _debug("urgentAvailBytes: %i\n", urgentAvailBytes);
toGet = (urgentAvailBytes < (int) (framesPerBuffer * sizeof (SFLDataFormat))) ? urgentAvailBytes : framesPerBuffer * sizeof (SFLDataFormat);
out = (SFLDataFormat*) pa_xmalloc (toGet * sizeof (SFLDataFormat));
@@ -490,6 +519,8 @@ void PulseLayer::writeToSpeaker (void)
AudioLoop* tone = _manager->getTelephoneTone();
AudioLoop* file_tone = _manager->getTelephoneFile();
+ _urgentRingBuffer.flushAll();
+
if (tone != 0) {
if (playback->getStreamState() == PA_STREAM_READY)
@@ -577,12 +608,14 @@ void PulseLayer::writeToSpeaker (void)
if((tone == 0) && (file_tone == 0)) {
// _debug("maxNbBytesToGet: %i\n", maxNbBytesToGet);
+
SFLDataFormat* zeros = (SFLDataFormat*)pa_xmalloc (framesPerBuffer*sizeof(SFLDataFormat));
bzero (zeros, framesPerBuffer*sizeof(SFLDataFormat));
pa_stream_write(playback->pulseStream(), zeros, framesPerBuffer*sizeof(SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (zeros);
+
}
}
@@ -603,7 +636,7 @@ void PulseLayer::readFromMic (void)
size_t r;
if (pa_stream_peek (record->pulseStream() , (const void**) &data , &r) < 0 || !data) {
- //_debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
+ _debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
}
if (data != 0) {
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.h b/sflphone-common/src/audio/pulseaudio/pulselayer.h
index e4a4981e57..4e7e30e816 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.h
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.h
@@ -133,6 +133,8 @@ class PulseLayer : public AudioLayer {
void processPlaybackData( void );
void processCaptureData( void );
+
+ void processData(void);
private:
// Copy Constructor
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 95293f6dc5..21b400a822 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -1651,6 +1651,8 @@ ManagerImpl::playDtmf (char code, bool isTalking)
AudioLayer *audiolayer;
SFLDataFormat *buf;
+ _debug("ManagerImpl::playDtmf\n");
+
stopTone (false);
bool hasToPlayTone = getConfigBool (SIGNALISATION, PLAY_DTMF);
@@ -1689,6 +1691,9 @@ ManagerImpl::playDtmf (char code, bool isTalking)
// ms/s
size = (int) ((pulselen * (float) audiolayer->getSampleRate()) / 1000);
+ _debug("DTMF pulselen: %i\n", pulselen);
+ _debug("DTMF size: %i\n", size);
+
// this buffer is for mono
// TODO <-- this should be global and hide if same size
buf = new SFLDataFormat[size];
@@ -2023,6 +2028,8 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
AudioLayer *audiolayer;
unsigned int nbSamples;
+ _debug("ManagerImpl::playATone\n");
+
hasToPlayTone = getConfigBool (SIGNALISATION, PLAY_TONES);
if (!hasToPlayTone)
@@ -2213,6 +2220,8 @@ void ManagerImpl::notificationIncomingCall (void)
audiolayer = getAudioDriver();
+ _debug("ManagerImpl::notificationIncomingCall\n");
+
if (audiolayer != 0) {
samplerate = audiolayer->getSampleRate();
frequency << "440/" << FRAME_PER_BUFFER;
--
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