From e3dbfbccfd8c412dd4c80660e2ea58bef450e775 Mon Sep 17 00:00:00 2001
From: Tristan Matthews <tristan.matthews@savoirfairelinux.com>
Date: Fri, 19 Aug 2011 11:46:30 -0400
Subject: [PATCH] * #6692: alsalayer cleanup
---
daemon/src/audio/alsa/alsalayer.cpp | 236 +++++++++++-----------------
daemon/src/audio/alsa/alsalayer.h | 12 +-
2 files changed, 96 insertions(+), 152 deletions(-)
diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index a24aeef2e8..067b7a8ddd 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -41,7 +41,7 @@ AlsaLayer::AlsaLayer (ManagerImpl* manager)
, _CaptureHandle (NULL)
, _periodSize (160)
, _audioPlugin ("default")
- , IDSoundCards()
+ , IDSoundCards_()
, _is_prepared_playback (false)
, _is_prepared_capture (false)
, _is_running_playback (false)
@@ -118,7 +118,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
_audioSampleRate = sampleRate;
_frameSize = frameSize;
- _audioPlugin = std::string (plugin);
+ _audioPlugin = plugin;
_debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", _indexIn, _indexOut, _indexRing);
_debugAlsa (" : alsa plugin=%s", _audioPlugin.c_str());
@@ -129,9 +129,8 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate,
// use 1 sec buffer for resampling
_converter = new SamplerateConverter (_audioSampleRate);
-
- AudioLayer::_dcblocker = new DcBlocker();
- AudioLayer::_audiofilter = new AudioProcessing (_dcblocker);
+ _dcblocker = new DcBlocker;
+ _audiofilter = new AudioProcessing (_dcblocker);
}
void
@@ -163,13 +162,11 @@ AlsaLayer::startStream (void)
_debug ("pcmr: %s, index %d", pcmr.c_str(), _indexRing);
_debug ("pcmc: %s, index %d", pcmc.c_str(), _indexIn);
- if (!is_capture_open()) {
+ if (!is_capture_open())
open_device (pcmp, pcmc, pcmr, SFL_PCM_CAPTURE);
- }
- if (!is_playback_open()) {
+ if (!is_playback_open())
open_device (pcmp, pcmc, pcmr, SFL_PCM_PLAYBACK);
- }
prepareCaptureStream ();
preparePlaybackStream ();
@@ -243,7 +240,6 @@ void AlsaLayer::setNoiseSuppressState (bool state)
}
-
//////////////////////////////////////////////////////////////////////////////////////////////
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
@@ -255,40 +251,36 @@ void AlsaLayer::stopCaptureStream (void)
if (_CaptureHandle) {
_debug ("Audio: Stop ALSA capture");
- if ( (err = snd_pcm_drop (_CaptureHandle)) < 0) {
+ if ((err = snd_pcm_drop (_CaptureHandle)) < 0)
_debug ("Audio: Error: stopping ALSA capture: %s", snd_strerror (err));
- } else {
+ else
stop_capture ();
- }
}
}
void AlsaLayer::closeCaptureStream (void)
{
- int err;
-
if (is_capture_prepared() == true && is_capture_running() == true)
stopCaptureStream ();
if (is_capture_open()) {
_debug ("Audio: Close ALSA capture");
- if ( (err = snd_pcm_close (_CaptureHandle)) < 0) {
+ int err;
+ if ((err = snd_pcm_close (_CaptureHandle)) < 0)
_debug ("Audio: Error: Closing ALSA capture: %s", snd_strerror (err));
- } else {
+ else
close_capture ();
- }
}
}
void AlsaLayer::startCaptureStream (void)
{
- int err;
-
if (_CaptureHandle && !is_capture_running()) {
_debug ("Audio: Start ALSA capture");
- if ( (err = snd_pcm_start (_CaptureHandle)) < 0)
+ int err;
+ if ((err = snd_pcm_start (_CaptureHandle)) < 0)
_debug ("Error starting ALSA capture: %s", snd_strerror (err));
else
start_capture();
@@ -311,20 +303,19 @@ void AlsaLayer::prepareCaptureStream (void)
void AlsaLayer::stopPlaybackStream (void)
{
- int err;
-
if (_RingtoneHandle && is_playback_running()) {
_debug ("Audio: Stop ALSA ringtone");
- if ( (err = snd_pcm_drop (_RingtoneHandle)) < 0) {
+ int err;
+ if ((err = snd_pcm_drop (_RingtoneHandle)) < 0)
_debug ("Audio: Error: Stop ALSA ringtone: %s", snd_strerror (err));
- }
}
if (_PlaybackHandle && is_playback_running()) {
_debug ("Audio: Stop ALSA playback");
- if ( (err = snd_pcm_drop (_PlaybackHandle)) < 0)
+ int err;
+ if ((err = snd_pcm_drop (_PlaybackHandle)) < 0)
_debug ("Audio: Error: Stopping ALSA playback: %s", snd_strerror (err));
else
stop_playback ();
@@ -334,23 +325,18 @@ void AlsaLayer::stopPlaybackStream (void)
void AlsaLayer::closePlaybackStream (void)
{
- int err;
-
if (is_playback_prepared() == true && is_playback_running() == true)
stopPlaybackStream ();
-
if (is_playback_open()) {
_debug ("Audio: Close ALSA playback");
- if (_RingtoneHandle) {
- if ( (err = snd_pcm_close (_RingtoneHandle)) < 0) {
+ int err;
+ if (_RingtoneHandle and (err = snd_pcm_close (_RingtoneHandle)) < 0)
_warn ("Audio: Error: Closing ALSA ringtone: %s", snd_strerror (err));
- }
- }
- if ( (err = snd_pcm_close (_PlaybackHandle)) < 0)
+ if ((err = snd_pcm_close (_PlaybackHandle)) < 0)
_warn ("Audio: Error: Closing ALSA playback: %s", snd_strerror (err));
else
close_playback ();
@@ -360,12 +346,11 @@ void AlsaLayer::closePlaybackStream (void)
void AlsaLayer::startPlaybackStream (void)
{
- int err;
-
if (_PlaybackHandle && !is_playback_running()) {
_debug ("Audio: Start ALSA playback");
- if ( (err = snd_pcm_start (_PlaybackHandle)) < 0)
+ int err;
+ if ((err = snd_pcm_start (_PlaybackHandle)) < 0)
_debug ("Audio: Error: Starting ALSA playback: %s", snd_strerror (err));
else
start_playback();
@@ -374,27 +359,21 @@ void AlsaLayer::startPlaybackStream (void)
void AlsaLayer::preparePlaybackStream (void)
{
- int err;
-
if (is_playback_open() && !is_playback_prepared()) {
_debug ("Audio: Prepare playback stream");
- if ( (err = snd_pcm_prepare (_PlaybackHandle)) < 0)
+ int err;
+ if ((err = snd_pcm_prepare (_PlaybackHandle)) < 0)
_debug ("Audio: Preparing the device: %s", snd_strerror (err));
else
prepare_playback ();
}
}
-bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
+bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
{
-
snd_pcm_hw_params_t *hwparams = NULL;
snd_pcm_sw_params_t *swparams = NULL;
- unsigned int exact_ivalue;
- unsigned long exact_lvalue;
- int dir;
- int err;
int format;
int periods = 4;
int periodsize = 160;
@@ -405,12 +384,13 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
_periodSize = periodsize;
/* Full configuration space */
- if ( (err = snd_pcm_hw_params_any (pcm_handle, hwparams)) < 0) {
+ int err;
+ if ((err = snd_pcm_hw_params_any (pcm_handle, hwparams)) < 0) {
_debugAlsa ("Audio: Error: Cannot initialize hardware parameter structure (%s)", snd_strerror (err));
return false;
}
- if ( (err = snd_pcm_hw_params_set_access (pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
+ if ((err = snd_pcm_hw_params_set_access (pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
_debugAlsa ("Audio: Error: Cannot set access type (%s)", snd_strerror (err));
return false;
}
@@ -418,69 +398,64 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
/* Set sample format */
format = SND_PCM_FORMAT_S16_LE;
- if ( (err = snd_pcm_hw_params_set_format (pcm_handle, hwparams, (snd_pcm_format_t) format)) < 0) {
+ if ((err = snd_pcm_hw_params_set_format (pcm_handle, hwparams, (snd_pcm_format_t) format)) < 0) {
_debugAlsa ("Audio: Error: Cannot set sample format (%s)", snd_strerror (err));
return false;
}
/* Set sample rate. If we can't set to the desired exact value, we set to the nearest acceptable */
- dir=0;
-
- rate = _audioSampleRate;
+ int dir = 0;
- exact_ivalue = rate;
+ unsigned int exact_ivalue = _audioSampleRate;
- if ( (err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) {
+ if ((err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) {
_debugAlsa ("Audio: Error: Cannot set sample rate (%s)", snd_strerror (err));
return false;
} else
- _debug ("Audio: Set audio rate to %d", rate);
+ _debug ("Audio: Set audio rate to %d", _audioSampleRate);
- if (dir!= 0) {
- _debugAlsa ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ",type ,rate, exact_ivalue);
+ if (dir != 0) {
+ _debugAlsa ("Audio: Error: (%i) The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", type , _audioSampleRate, exact_ivalue);
+ _audioSampleRate = exact_ivalue;
}
/* Set the number of channels */
- if ( (err = snd_pcm_hw_params_set_channels (pcm_handle, hwparams, 1)) < 0) {
+ if ((err = snd_pcm_hw_params_set_channels (pcm_handle, hwparams, 1)) < 0) {
_debugAlsa ("Audio: Error: Cannot set channel count (%s)", snd_strerror (err));
return false;
}
/* Set the buffer size in frames */
- exact_lvalue = periodsize;
+ unsigned long exact_lvalue = periodsize;
- dir=0;
+ dir = 0;
- if ( (err = snd_pcm_hw_params_set_period_size_near (pcm_handle, hwparams, &exact_lvalue , &dir)) < 0) {
+ if ((err = snd_pcm_hw_params_set_period_size_near (pcm_handle, hwparams, &exact_lvalue, &dir)) < 0) {
_debugAlsa ("Audio: Error: Cannot set period time (%s)", snd_strerror (err));
return false;
}
- if (dir!=0) {
- _debugAlsa ("Audio: Warning: (%i) The chosen period size %d bytes is not supported by your hardware.Using %d instead. ", type, (int) periodsize, (int) exact_lvalue);
- }
-
- periodsize = exact_lvalue;
+ if (dir != 0)
+ _debugAlsa ("Audio: Warning: (%i) The chosen period size %lu bytes is not supported by your hardware.Using %lu instead. ", type, periodsize, exact_lvalue);
_periodSize = exact_lvalue;
/* Set the number of fragments */
exact_ivalue = periods;
- dir=0;
+ dir = 0;
- if ( (err = snd_pcm_hw_params_set_periods_near (pcm_handle, hwparams, &exact_ivalue, &dir)) < 0) {
+ if ((err = snd_pcm_hw_params_set_periods_near (pcm_handle, hwparams, &exact_ivalue, &dir)) < 0) {
_debugAlsa ("Audio: Error: Cannot set periods number (%s)", snd_strerror (err));
return false;
}
- if (dir!=0) {
+ if (dir != 0)
_debugAlsa ("Audio: Warning: The chosen period number %i bytes is not supported by your hardware.Using %i instead. ", periods, exact_ivalue);
- }
- periods=exact_ivalue;
+ periods = exact_ivalue;
/* Set the hw parameters */
- if ( (err = snd_pcm_hw_params (pcm_handle, hwparams)) < 0) {
+ if ((err = snd_pcm_hw_params (pcm_handle, hwparams)) < 0) {
_debugAlsa ("Audio: Error: Cannot set hw parameters (%s)", snd_strerror (err));
return false;
}
@@ -493,12 +468,12 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
/* Set the start threshold */
- if ( (err = snd_pcm_sw_params_set_start_threshold (pcm_handle, swparams, _periodSize*2)) < 0) {
+ if ((err = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, _periodSize * 2)) < 0) {
_debugAlsa ("Audio: Error: Cannot set start threshold (%s)", snd_strerror (err));
return false;
}
- if ( (err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) {
+ if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) {
_debugAlsa ("Audio: Error: Cannot set sw parameters (%s)", snd_strerror (err));
return false;
}
@@ -511,20 +486,19 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
bool
AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag)
{
- int err;
-
- if (flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK) {
+ if (flag == SFL_PCM_BOTH or flag == SFL_PCM_PLAYBACK) {
_debug ("Audio: Open playback device (and ringtone)");
- if ( (err = snd_pcm_open (&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+ int err;
+ if ((err = snd_pcm_open (&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
_warn ("Audio: Error while opening playback device %s", pcm_p.c_str());
setErrorMessage (ALSA_PLAYBACK_DEVICE);
close_playback ();
return false;
}
- if (!alsa_set_params (_PlaybackHandle, 1, getSampleRate())) {
+ if (!alsa_set_params (_PlaybackHandle, 1)) {
_warn ("Audio: Error: Playback failed");
snd_pcm_close (_PlaybackHandle);
close_playback ();
@@ -533,25 +507,23 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
if (getIndexOut() != getIndexRing()) {
- if ( (err = snd_pcm_open (&_RingtoneHandle, pcm_r.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+ if ((err = snd_pcm_open (&_RingtoneHandle, pcm_r.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0)
_warn ("Audio: Error: Opening ringtone device %s", pcm_r.c_str());
- }
- if (!alsa_set_params (_RingtoneHandle, 1, getSampleRate())) {
+ if (!alsa_set_params (_RingtoneHandle, 1)) {
_warn ("Audio: Error: Ringtone failed");
snd_pcm_close (_RingtoneHandle);
-
}
}
open_playback ();
}
- if (flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE) {
+ if (flag == SFL_PCM_BOTH or flag == SFL_PCM_CAPTURE) {
_debug ("Audio: Open capture device");
- if ( (err = snd_pcm_open (&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0)) < 0) {
+ if (snd_pcm_open (&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0) < 0) {
_warn ("Audio: Error: Opening capture device %s", pcm_c.c_str());
setErrorMessage (ALSA_CAPTURE_DEVICE);
@@ -559,7 +531,7 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
return false;
}
- if (!alsa_set_params (_CaptureHandle, 0, 8000)) {
+ if (!alsa_set_params (_CaptureHandle, 0)) {
_warn ("Audio: Error: Capture failed");
snd_pcm_close (_CaptureHandle);
close_capture ();
@@ -576,7 +548,7 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r,
int
AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
{
- if (_trigger_request == true) {
+ if (_trigger_request) {
_trigger_request = false;
startPlaybackStream ();
}
@@ -584,20 +556,16 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames (handle, length);
int err;
-
- if ( (err=snd_pcm_writei (handle, buffer , frames)) <0) {
+ if ((err = snd_pcm_writei (handle, buffer , frames)) < 0) {
switch (err) {
case -EPIPE:
-
case -ESTRPIPE:
-
case -EIO:
handle_xrun_playback (handle);
- if (snd_pcm_writei (handle, buffer , frames) <0) {
+ if (snd_pcm_writei (handle, buffer , frames) < 0)
_debugAlsa ("Audio: XRUN handling failed");
- }
_trigger_request = true;
@@ -616,8 +584,6 @@ AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
int
AlsaLayer::read (void* buffer, int toCopy)
{
- int samples;
-
if (snd_pcm_state (_CaptureHandle) == SND_PCM_STATE_XRUN) {
prepareCaptureStream ();
startCaptureStream ();
@@ -625,20 +591,19 @@ AlsaLayer::read (void* buffer, int toCopy)
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames (_CaptureHandle, toCopy);
- if ( (samples = snd_pcm_readi (_CaptureHandle, buffer, frames)) < 0) {
- switch (samples) {
+ int err;
+ if ((err = snd_pcm_readi (_CaptureHandle, buffer, frames)) < 0) {
+ switch (err) {
case -EPIPE:
-
case -ESTRPIPE:
-
case -EIO:
- _debugAlsa ("Audio: XRUN capture ignored (%s)", snd_strerror (samples));
+ _debugAlsa ("Audio: XRUN capture ignored (%s)", snd_strerror (err));
handle_xrun_capture();
break;
case EPERM:
- _debugAlsa ("Audio: Capture EPERM (%s)", snd_strerror (samples));
+ _debugAlsa ("Audio: Capture EPERM (%s)", snd_strerror (err));
prepareCaptureStream ();
startCaptureStream ();
break;
@@ -661,9 +626,7 @@ AlsaLayer::handle_xrun_capture (void)
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
- int res = snd_pcm_status (_CaptureHandle, status);
-
- if (res <= 0) {
+ if (snd_pcm_status(_CaptureHandle, status) <= 0) {
if (snd_pcm_status_get_state (status) == SND_PCM_STATE_XRUN) {
stopCaptureStream ();
prepareCaptureStream ();
@@ -678,14 +641,14 @@ AlsaLayer::handle_xrun_playback (snd_pcm_t *handle)
{
_debugAlsa ("Audio: Handle xrun playback");
- int state;
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
- if ( (state = snd_pcm_status (handle, status)) < 0)
+ int state;
+ if ((state = snd_pcm_status (handle, status)) < 0)
_debugAlsa ("Audio: Error: Cannot get playback handle status (%s)" , snd_strerror (state));
else {
- state = snd_pcm_status_get_state (status);
+ int state = snd_pcm_status_get_state (status);
if (state == SND_PCM_STATE_XRUN) {
_debug ("Audio: audio device in state SND_PCM_STATE_XRUN, restart device");
@@ -698,10 +661,10 @@ AlsaLayer::handle_xrun_playback (snd_pcm_t *handle)
}
std::string
-AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice)
+AlsaLayer::buildDeviceTopo (const std::string &plugin, int card, int subdevice)
{
std::stringstream ss,ss1;
- std::string pcm = plugin;
+ std::string pcm(plugin);
if (pcm == PCM_DEFAULT)
return pcm;
@@ -726,9 +689,6 @@ AlsaLayer::buildDeviceTopo (std::string plugin, int card, int subdevice)
std::vector<std::string>
AlsaLayer::getSoundCardsInfo (int stream)
{
- std::vector<std::string> cards_id;
- HwIDPair p;
-
_debug ("Audio: Get sound cards info: ");
snd_ctl_t* handle;
@@ -740,6 +700,7 @@ AlsaLayer::getSoundCardsInfo (int stream)
int numCard = -1 ;
std::string description;
+ std::vector<std::string> cards_id;
if (snd_card_next (&numCard) < 0 || numCard < 0)
return cards_id;
@@ -765,27 +726,23 @@ AlsaLayer::getSoundCardsInfo (int stream)
description.append (snd_pcm_info_get_name (pcminfo));
cards_id.push_back (description);
// The number of the sound card is associated with a string description
- p = HwIDPair (numCard , description);
- IDSoundCards.push_back (p);
-
+ IDSoundCards_.push_back(HwIDPair(numCard , description));
}
}
snd_ctl_close (handle);
}
- if (snd_card_next (&numCard) < 0) {
+ if (snd_card_next (&numCard) < 0)
break;
- }
}
return cards_id;
}
-
bool
-AlsaLayer::soundCardIndexExist (int card , int stream)
+AlsaLayer::soundCardIndexExist (int card, int stream)
{
snd_ctl_t* handle;
snd_pcm_info_t *pcminfo;
@@ -799,22 +756,17 @@ AlsaLayer::soundCardIndexExist (int card , int stream)
return false;
snd_pcm_info_set_stream (pcminfo , (stream == SFL_PCM_PLAYBACK) ? SND_PCM_STREAM_PLAYBACK : SND_PCM_STREAM_CAPTURE);
- bool ret = snd_ctl_pcm_info (handle , pcminfo) >= 0;
+ bool ret = snd_ctl_pcm_info (handle , pcminfo) >= 0;
snd_ctl_close(handle);
return ret;
}
int
-AlsaLayer::soundCardGetIndex (std::string description)
+AlsaLayer::soundCardGetIndex (const std::string &description)
{
- unsigned int i;
-
- for (i = 0 ; i < IDSoundCards.size() ; i++) {
- HwIDPair p = IDSoundCards[i];
-
- if (p.second == description)
- return p.first ;
- }
+ for (std::vector<HwIDPair>::const_iterator iter = IDSoundCards_.begin(); iter != IDSoundCards_.end(); ++iter)
+ if (iter->second == description)
+ return iter->first;
// else return the default one
return 0;
@@ -822,33 +774,27 @@ AlsaLayer::soundCardGetIndex (std::string description)
void AlsaLayer::audioCallback (void)
{
-
- int urgentAvailBytes;
- unsigned short spkrVolume, micVolume;
- AudioLoop *tone;
- AudioLoop *file_tone;
unsigned int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
bool resample = _audioSampleRate != _mainBufferSampleRate;
notifyincomingCall();
- spkrVolume = _manager->getSpkrVolume();
- micVolume = _manager->getMicVolume();
+ unsigned short spkrVolume = _manager->getSpkrVolume();
- tone = _manager->getTelephoneTone();
- file_tone = _manager->getTelephoneFile();
+ AudioLoop *tone = _manager->getTelephoneTone();
+ AudioLoop *file_tone = _manager->getTelephoneFile();
// AvailForGet tell the number of chars inside the buffer
// framePerBuffer are the number of data for one channel (left)
- urgentAvailBytes = _urgentRingBuffer.AvailForGet();
+ int urgentAvailBytes = _urgentRingBuffer.AvailForGet();
- if (!_PlaybackHandle || !_CaptureHandle)
+ if (!_PlaybackHandle or !_CaptureHandle)
return;
snd_pcm_wait (_PlaybackHandle, 20);
int playbackAvailSmpl = snd_pcm_avail_update (_PlaybackHandle);
- int playbackAvailBytes = playbackAvailSmpl*sizeof (SFLDataFormat);
+ int playbackAvailBytes = playbackAvailSmpl * sizeof (SFLDataFormat);
if (urgentAvailBytes > 0) {
// Urgent data (dtmf, incoming call signal) come first.
@@ -917,7 +863,6 @@ void AlsaLayer::audioCallback (void)
}
if (file_tone && _RingtoneHandle) {
-
int ringtoneAvailSmpl = snd_pcm_avail_update (_RingtoneHandle);
int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof (SFLDataFormat);
SFLDataFormat *out = (SFLDataFormat *) malloc (ringtoneAvailBytes);
@@ -925,7 +870,6 @@ void AlsaLayer::audioCallback (void)
write (out, ringtoneAvailBytes, _RingtoneHandle);
free (out);
} else if (_RingtoneHandle) {
-
int ringtoneAvailSmpl = snd_pcm_avail_update (_RingtoneHandle);
int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof (SFLDataFormat);
@@ -979,9 +923,9 @@ void AlsaLayer::audioCallback (void)
free (in);
}
-void AlsaLayer::adjustVolume (SFLDataFormat *src , int samples, int volume)
+void AlsaLayer::adjustVolume (SFLDataFormat *src , int samples, int volumePercentage)
{
- if (volume != 100)
+ if (volumePercentage != 100)
for (int i = 0 ; i < samples; i++)
- src[i] = src[i] * volume / 100 ;
+ src[i] = src[i] * volumePercentage * 0.01;
}
diff --git a/daemon/src/audio/alsa/alsalayer.h b/daemon/src/audio/alsa/alsalayer.h
index f35f322a4b..9084808530 100644
--- a/daemon/src/audio/alsa/alsalayer.h
+++ b/daemon/src/audio/alsa/alsalayer.h
@@ -116,7 +116,7 @@ class AlsaLayer : public AudioLayer
* @param subdevice the subdevice number
* @return std::string the concatenated string
*/
- std::string buildDeviceTopo (std::string plugin, int card, int subdevice);
+ std::string buildDeviceTopo (const std::string &plugin, int card, int subdevice);
/**
* Scan the sound card available on the system
@@ -145,13 +145,13 @@ class AlsaLayer : public AudioLayer
* @param description The string description
* @return int Its index
*/
- int soundCardGetIndex (std::string description);
+ int soundCardGetIndex (const std::string &description);
/**
* Get the current audio plugin.
* @return std::string The name of the audio plugin
*/
- std::string getAudioPlugin (void) {
+ std::string getAudioPlugin (void) const {
return _audioPlugin;
}
@@ -260,7 +260,7 @@ class AlsaLayer : public AudioLayer
*/
bool open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag);
- bool alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate);
+ bool alsa_set_params (snd_pcm_t *pcm_handle, int type);
/**
* Copy a data buffer in the internal ring buffer
@@ -292,7 +292,7 @@ class AlsaLayer : public AudioLayer
*/
void handle_xrun_playback (snd_pcm_t *handle);
- void adjustVolume (SFLDataFormat* buffer , int samples, int volume);
+ void adjustVolume (SFLDataFormat* buffer , int samples, int volumePercentage);
/**
* Handles to manipulate playback stream
@@ -323,7 +323,7 @@ class AlsaLayer : public AudioLayer
std::string _audioPlugin;
/** Vector to manage all soundcard index - description association of the system */
- std::vector<HwIDPair> IDSoundCards;
+ std::vector<HwIDPair> IDSoundCards_;
bool _is_prepared_playback;
bool _is_prepared_capture;
--
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