From e32ac62fbb47038f10335c48d9b85dbda3489682 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Rafa=C3=ABl=20Carr=C3=A9?=
<rafael.carre@savoirfairelinux.com>
Date: Tue, 30 Aug 2011 10:00:38 -0400
Subject: [PATCH] * #6629 : simplify AudioLayer creation
Move ALSA specific options to alsalayer only
---
daemon/src/audio/alsa/alsalayer.cpp | 225 ++++++++-------------
daemon/src/audio/alsa/alsalayer.h | 82 ++++----
daemon/src/audio/audiolayer.cpp | 12 +-
daemon/src/audio/audiolayer.h | 79 +-------
daemon/src/audio/pulseaudio/pulselayer.cpp | 117 ++---------
daemon/src/audio/pulseaudio/pulselayer.h | 26 ---
daemon/src/audio/samplerateconverter.cpp | 1 +
daemon/src/managerimpl.cpp | 183 +++++------------
daemon/test/audiolayertest.cpp | 6 -
9 files changed, 215 insertions(+), 516 deletions(-)
diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index 3201c117da..1ab1574e4c 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -74,11 +74,14 @@ void AlsaThread::run (void)
// Constructor
AlsaLayer::AlsaLayer ()
: AudioLayer (ALSA)
+ , indexIn_ (audioPref.getCardin())
+ , indexOut_ (audioPref.getCardout())
+ , indexRing_ (audioPref.getCardring())
, playbackHandle_ (NULL)
, ringtoneHandle_ (NULL)
, captureHandle_ (NULL)
, periodSize_ (160)
- , audioPlugin_ ("default")
+ , audioPlugin_ (audioPref.getPlugin())
, IDSoundCards_ ()
, is_playback_prepared_ (false)
, is_capture_prepared_ (false)
@@ -88,85 +91,30 @@ AlsaLayer::AlsaLayer ()
, is_capture_open_ (false)
, trigger_request_ (false)
, audioThread_ (NULL)
- , converter_ (0)
-
{
- _debug ("Audio: Build ALSA layer");
- urgentRingBuffer_.createReadPointer();
-
- audioPlugin_ = audioPref.getPlugin();
}
// Destructor
AlsaLayer::~AlsaLayer (void)
{
- _debug ("Audio: Destroy of ALSA layer");
- closeLayer();
-
- delete converter_;
-}
-
-void
-AlsaLayer::closeLayer()
-{
- _debug ("Audio: Close ALSA streams");
-
- try {
- /* Stop the audio thread first */
- if (audioThread_) {
- _debug ("Audio: Stop Audio Thread");
- delete audioThread_;
- audioThread_ = NULL;
- }
- } catch (...) {
- _debug ("Audio: Exception: when stopping audiortp");
- throw;
- }
+ delete audioThread_;
/* Then close the audio devices */
closeCaptureStream();
closePlaybackStream();
-
- captureHandle_ = NULL;
- playbackHandle_ = NULL;
- ringtoneHandle_ = NULL;
}
-void
-AlsaLayer::openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream , const std::string &plugin)
+void AlsaLayer::setPlugin(const std::string &plugin)
{
- /* Close the devices before open it */
- if (stream == SFL_PCM_BOTH and is_capture_open_ and is_playback_open_) {
- closeCaptureStream();
- closePlaybackStream();
- } else if ( (stream == SFL_PCM_CAPTURE or stream == SFL_PCM_BOTH) and is_capture_open_)
- closeCaptureStream ();
- else if ( (stream == SFL_PCM_PLAYBACK or stream == SFL_PCM_BOTH) and is_playback_open_)
- closePlaybackStream ();
-
- indexIn_ = indexIn;
- indexOut_ = indexOut;
- indexRing_ = indexRing;
-
- audioSampleRate_ = sampleRate;
-
audioPlugin_ = plugin;
-
- _debug (" Setting AlsaLayer: device in=%2d, out=%2d, ring=%2d", indexIn_, indexOut_, indexRing_);
- _debug (" : alsa plugin=%s", audioPlugin_.c_str());
- _debug (" : sample rate=%5d, format=%s", audioSampleRate_, SFLDataFormatString);
-
- audioThread_ = NULL;
-
- // use 1 sec buffer for resampling
- converter_ = new SamplerateConverter (audioSampleRate_);
+ delete audioThread_;
+ audioThread_ = NULL;
+ // FIXME : restart audio thread
}
void
AlsaLayer::startStream (void)
{
- _debug ("Audio: Start stream");
-
dcblocker_.reset();
if (is_playback_running_ and is_capture_running_)
@@ -190,12 +138,55 @@ AlsaLayer::startStream (void)
_debug ("pcmr: %s, index %d", pcmr.c_str(), indexRing_);
_debug ("pcmc: %s, index %d", pcmc.c_str(), indexIn_);
- if (not is_capture_open_)
- open_device (pcmp, pcmc, pcmr, SFL_PCM_CAPTURE);
+ if (not is_capture_open_) {
+ _debug ("Audio: Open capture device");
- if (not is_playback_open_)
- open_device (pcmp, pcmc, pcmr, SFL_PCM_PLAYBACK);
+ if (snd_pcm_open (&captureHandle_, pcmc.c_str(), SND_PCM_STREAM_CAPTURE, 0) < 0) {
+ _warn ("Audio: Error: Opening capture device %s", pcmc.c_str());
+ Manager::instance().getDbusManager()->getConfigurationManager()->errorAlert(ALSA_CAPTURE_DEVICE);
+ is_capture_open_ = false;
+ }
+
+ if (!alsa_set_params (captureHandle_, 0)) {
+ _warn ("Audio: Error: Capture failed");
+ snd_pcm_close (captureHandle_);
+ is_capture_open_ = false;
+ }
+
+ is_capture_open_ = true;
+ }
+
+ if (not is_playback_open_) {
+
+ _debug ("Audio: Open playback device (and ringtone)");
+
+ int err;
+ if ((err = snd_pcm_open (&playbackHandle_, pcmp.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+ _warn ("Audio: Error while opening playback device %s", pcmp.c_str());
+ Manager::instance().getDbusManager()->getConfigurationManager()->errorAlert(ALSA_PLAYBACK_DEVICE);
+ is_playback_open_ = false;
+ }
+
+ if (!alsa_set_params (playbackHandle_, 1)) {
+ _warn ("Audio: Error: Playback failed");
+ snd_pcm_close (playbackHandle_);
+ is_playback_open_ = false;
+ }
+
+ if (getIndexOut() != getIndexRing()) {
+
+ if ((err = snd_pcm_open (&ringtoneHandle_, pcmr.c_str(), SND_PCM_STREAM_PLAYBACK, 0)) < 0)
+ _warn ("Audio: Error: Opening ringtone device %s", pcmr.c_str());
+
+ if (!alsa_set_params (ringtoneHandle_, 1)) {
+ _warn ("Audio: Error: Ringtone failed");
+ snd_pcm_close (ringtoneHandle_);
+ }
+ }
+
+ is_playback_open_ = true;
+ }
prepareCaptureStream ();
preparePlaybackStream ();
@@ -225,17 +216,8 @@ AlsaLayer::stopStream (void)
isStarted_ = false;
- try {
- /* Stop the audio thread first */
- if (audioThread_) {
- _debug ("Audio: Stop audio thread");
- delete audioThread_;
- audioThread_ = NULL;
- }
- } catch (...) {
- _debug ("Audio: Exception: when stopping audiortp");
- throw;
- }
+ delete audioThread_;
+ audioThread_ = NULL;
closeCaptureStream ();
closePlaybackStream ();
@@ -268,6 +250,32 @@ void AlsaLayer::stopCaptureStream (void)
}
}
+void AlsaLayer::setIndexRing(int index)
+{
+ indexRing_ = index;
+ delete audioThread_;
+ audioThread_ = NULL;
+}
+
+
+void AlsaLayer::setIndexOut(int index)
+{
+ indexOut_ = index;
+ if (is_playback_open_)
+ closePlaybackStream ();
+ delete audioThread_;
+ audioThread_ = NULL;
+}
+
+void AlsaLayer::setIndexIn(int index)
+{
+ indexIn_ = index;
+ if (is_capture_open_)
+ closeCaptureStream ();
+ delete audioThread_;
+ audioThread_ = NULL;
+}
+
void AlsaLayer::closeCaptureStream (void)
{
if (is_capture_prepared_ and is_capture_running_)
@@ -427,7 +435,8 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
if (dir != 0) {
_debug ("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;
+ //audioSampleRate_ = exact_ivalue;
+ // FIXME
}
/* Set the number of channels */
@@ -493,68 +502,6 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type)
return true;
}
-
-bool
-AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, std::string pcm_r, int flag)
-{
- if (flag == SFL_PCM_BOTH or flag == SFL_PCM_PLAYBACK) {
-
- _debug ("Audio: Open playback device (and ringtone)");
-
- 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());
- Manager::instance().getDbusManager()->getConfigurationManager()->errorAlert(ALSA_PLAYBACK_DEVICE);
- is_playback_open_ = false;
- return false;
- }
-
- if (!alsa_set_params (playbackHandle_, 1)) {
- _warn ("Audio: Error: Playback failed");
- snd_pcm_close (playbackHandle_);
- is_playback_open_ = false;
- return false;
- }
-
- if (getIndexOut() != getIndexRing()) {
-
- 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)) {
- _warn ("Audio: Error: Ringtone failed");
- snd_pcm_close (ringtoneHandle_);
- }
- }
-
- is_playback_open_ = true;
- }
-
- if (flag == SFL_PCM_BOTH or flag == SFL_PCM_CAPTURE) {
-
- _debug ("Audio: Open capture device");
-
- 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());
-
- Manager::instance().getDbusManager()->getConfigurationManager()->errorAlert(ALSA_CAPTURE_DEVICE);
- is_capture_open_ = false;
- return false;
- }
-
- if (!alsa_set_params (captureHandle_, 0)) {
- _warn ("Audio: Error: Capture failed");
- snd_pcm_close (captureHandle_);
- is_capture_open_ = false;
- return false;
- }
-
- is_capture_open_ = true;
- }
-
- return true;
-}
-
//TODO first frame causes broken pipe (underrun) because not enough data are send --> make the handle wait to be ready
int
AlsaLayer::write (void* buffer, int length, snd_pcm_t * handle)
diff --git a/daemon/src/audio/alsa/alsalayer.h b/daemon/src/audio/alsa/alsalayer.h
index 49d2382bf1..c29ccda940 100644
--- a/daemon/src/audio/alsa/alsalayer.h
+++ b/daemon/src/audio/alsa/alsalayer.h
@@ -35,7 +35,6 @@
#include "audio/audiolayer.h"
#include <alsa/asoundlib.h>
-class SamplerateConverter;
class RingBuffer;
class ManagerImpl;
class AlsaThread;
@@ -58,20 +57,6 @@ class AlsaLayer : public AudioLayer
*/
~AlsaLayer (void);
- /**
- * Check if no devices are opened, otherwise close them.
- * Then open the specified devices by calling the private functions open_device
- * @param indexIn The number of the card chosen for capture
- * @param indexOut The number of the card chosen for playback
- * @param sampleRate The sample rate
- * @param stream To indicate which kind of stream you want to open
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @param plugin The alsa plugin ( dmix , default , front , surround , ...)
- */
- void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream, const std::string &plugin);
-
/**
* Start the capture stream and prepare the playback stream.
* The playback starts accordingly to its threshold
@@ -123,7 +108,7 @@ class AlsaLayer : public AudioLayer
* @return bool True if it exists and supports the mode
* false otherwise
*/
- bool soundCardIndexExist (int card , int stream);
+ static bool soundCardIndexExist (int card , int stream);
/**
* An index is associated with its string description
@@ -142,9 +127,55 @@ class AlsaLayer : public AudioLayer
void audioCallback (void);
+ /**
+ * Get the index of the audio card for capture
+ * @return int The index of the card used for capture
+ * 0 for the first available card on the system, 1 ...
+ */
+ int getIndexIn() const {
+ return indexIn_;
+ }
+ void setIndexIn(int);
+
+ /**
+ * Get the index of the audio card for playback
+ * @return int The index of the card used for playback
+ * 0 for the first available card on the system, 1 ...
+ */
+ int getIndexOut() const {
+ return indexOut_;
+ }
+ void setIndexOut(int);
+
+ /**
+ * Get the index of the audio card for ringtone (could be differnet from playback)
+ * @return int The index of the card used for ringtone
+ * 0 for the first available card on the system, 1 ...
+ */
+ int getIndexRing() const {
+ return indexRing_;
+ }
+ void setIndexRing(int);
+
+ void setPlugin(const std::string &plugin);
private:
- void closeLayer (void);
+
+
+ /**
+ * Number of audio cards on which capture stream has been opened
+ */
+ int indexIn_;
+
+ /**
+ * Number of audio cards on which playback stream has been opened
+ */
+ int indexOut_;
+
+ /**
+ * Number of audio cards on which ringtone stream has been opened
+ */
+ int indexRing_;
/** Associate a sound card index to its string description */
typedef std::pair<int , std::string> HwIDPair;
@@ -170,20 +201,6 @@ class AlsaLayer : public AudioLayer
void startPlaybackStream (void);
void preparePlaybackStream (void);
- /**
- * Open the specified device.
- * ALSA Library API
- * @param pcm_p The string name for the playback device
- * @param pcm_c The string name for the capture device
- * @param flag To indicate which kind of stream you want to open
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @return true if successful
- * false otherwise
- */
- 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);
/**
@@ -256,9 +273,6 @@ class AlsaLayer : public AudioLayer
bool trigger_request_;
AlsaThread* audioThread_;
-
- /** Sample rate converter object */
- SamplerateConverter* converter_;
};
#endif // _ALSA_LAYER_H_
diff --git a/daemon/src/audio/audiolayer.cpp b/daemon/src/audio/audiolayer.cpp
index 1edee1a967..cb2c30df80 100644
--- a/daemon/src/audio/audiolayer.cpp
+++ b/daemon/src/audio/audiolayer.cpp
@@ -37,20 +37,20 @@
AudioLayer::AudioLayer (int type)
: isStarted_ (false)
, urgentRingBuffer_ (SIZEBUF, Call::DEFAULT_ID)
- , recorder_ (0)
- , indexIn_ (0)
- , indexOut_ (0)
- , indexRing_ (0)
- , audioSampleRate_ (0)
+ , audioSampleRate_(Manager::instance().getMainBuffer()->getInternalSamplingRate())
, mutex_ ()
, audioPref(Manager::instance().audioPreference)
+ , converter_ (new SamplerateConverter(audioSampleRate_))
, layerType_ (type)
, lastNotificationTime_ (0)
-{}
+{
+ urgentRingBuffer_.createReadPointer();
+}
AudioLayer::~AudioLayer ()
{
+ delete converter_;
}
void AudioLayer::flushMain (void)
diff --git a/daemon/src/audio/audiolayer.h b/daemon/src/audio/audiolayer.h
index 27a2cdedb1..56b47e2a8c 100644
--- a/daemon/src/audio/audiolayer.h
+++ b/daemon/src/audio/audiolayer.h
@@ -40,6 +40,7 @@
#include "manager.h"
#include "ringbuffer.h"
#include "dcblocker.h"
+#include "samplerateconverter.h"
/**
* @file audiolayer.h
@@ -72,20 +73,6 @@ class AudioLayer
*/
virtual ~AudioLayer (void);
- /**
- * Check if no devices are opened, otherwise close them.
- * Then open the specified devices by calling the private functions open_device
- * @param indexIn The number of the card chosen for capture
- * @param indexOut The number of the card chosen for playback
- * @param sampleRate The sample rate
- * @param stream To indicate which kind of stream you want to open
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @param plugin The alsa plugin ( dmix , default , front , surround , ...)
- */
- virtual void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream , const std::string &plugin) = 0;
-
/**
* Start the capture stream and prepare the playback stream.
* The playback starts accordingly to its threshold
@@ -114,32 +101,6 @@ class AudioLayer
void flushUrgent (void);
- /**
- * Get the index of the audio card for capture
- * @return int The index of the card used for capture
- * 0 for the first available card on the system, 1 ...
- */
- int getIndexIn() const {
- return indexIn_;
- }
-
- /**
- * Get the index of the audio card for playback
- * @return int The index of the card used for playback
- * 0 for the first available card on the system, 1 ...
- */
- int getIndexOut() const {
- return indexOut_;
- }
-
- /**
- * Get the index of the audio card for ringtone (could be differnet from playback)
- * @return int The index of the card used for ringtone
- * 0 for the first available card on the system, 1 ...
- */
- int getIndexRing() const {
- return indexRing_;
- }
/**
* Get the sample rate of the audio layer
@@ -171,20 +132,6 @@ class AudioLayer
return Manager::instance().getMainBuffer();
}
- /**
- * Set the audio recorder
- */
- void setRecorderInstance (Recordable* rec) {
- recorder_ = rec;
- }
-
- /**
- * Get the audio recorder
- */
- Recordable* getRecorderInstance (void) const {
- return recorder_;
- }
-
/**
* Get the mutex lock for the entire audio layer
*/
@@ -216,31 +163,11 @@ class AudioLayer
*/
RingBuffer urgentRingBuffer_;
- /**
- * A pointer to the recordable instance (may be a call or a conference)
- */
- Recordable* recorder_;
-
- /**
- * Number of audio cards on which capture stream has been opened
- */
- int indexIn_;
-
- /**
- * Number of audio cards on which playback stream has been opened
- */
- int indexOut_;
-
- /**
- * Number of audio cards on which ringtone stream has been opened
- */
- int indexRing_;
-
/**
* Sample Rate SFLphone should send sound data to the sound card
* The value can be set in the user config file- now: 44100HZ
*/
- unsigned int audioSampleRate_;
+ const unsigned int audioSampleRate_;
/**
* Lock for the entire audio layer
@@ -251,6 +178,8 @@ class AudioLayer
AudioPreference &audioPref;
+ SamplerateConverter *converter_;
+
private:
const int layerType_;
diff --git a/daemon/src/audio/pulseaudio/pulselayer.cpp b/daemon/src/audio/pulseaudio/pulselayer.cpp
index cbdff8273f..90f09b3e19 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.cpp
+++ b/daemon/src/audio/pulseaudio/pulselayer.cpp
@@ -219,45 +219,41 @@ void playback_overflow_callback (pa_stream* s UNUSED, void* userdata UNUSED)
PulseLayer::PulseLayer ()
: AudioLayer (PULSEAUDIO)
- , context_(0)
- , mainloop_(0)
, playback_(0)
, record_(0)
, ringtone_(0)
- , converter_(0)
{
- urgentRingBuffer_.createReadPointer();
+ setenv ("PULSE_PROP_media.role", "phone", 1);
- openLayer();
-}
+ mainloop_ = pa_threaded_mainloop_new();
+ if (!mainloop_)
+ _error("Couldn't create pulseaudio mainloop");
-// Destructor
-PulseLayer::~PulseLayer (void)
-{
- closeLayer ();
- delete converter_;
-}
+ context_ = pa_context_new (pa_threaded_mainloop_get_api (mainloop_) , "SFLphone");
+ if (!context_)
+ _error("Couldn't create pulseaudio context");
-void
-PulseLayer::openLayer (void)
-{
- if (isStarted_)
- return;
+ pa_context_set_state_callback (context_, context_state_callback, this);
+
+ if (pa_context_connect (context_, NULL , PA_CONTEXT_NOAUTOSPAWN , NULL) < 0)
+ _error("Could not connect pulseaudio context to the server");
+
+ pa_threaded_mainloop_lock (mainloop_);
+
+ if (pa_threaded_mainloop_start (mainloop_) < 0)
+ _error("Failed to start pulseaudio mainloop");
+ pa_threaded_mainloop_wait (mainloop_);
- _info ("Audio: Open Pulseaudio layer");
+ if (pa_context_get_state (context_) != PA_CONTEXT_READY)
+ _error("Couldn't connect to pulse audio server");
- connectPulseAudioServer();
+ pa_threaded_mainloop_unlock (mainloop_);
isStarted_ = true;
}
-void
-PulseLayer::closeLayer (void)
+PulseLayer::~PulseLayer (void)
{
- _info ("Audio: Close Pulseaudio layer");
-
- isStarted_ = false;
-
disconnectAudioStream();
if (mainloop_)
@@ -266,69 +262,10 @@ PulseLayer::closeLayer (void)
if (context_) {
pa_context_disconnect (context_);
pa_context_unref (context_);
- context_ = NULL;
}
- if (mainloop_) {
+ if (mainloop_)
pa_threaded_mainloop_free (mainloop_);
- mainloop_ = NULL;
- }
-}
-
-void
-PulseLayer::connectPulseAudioServer (void)
-{
- _info ("Audio: Connect to Pulseaudio server");
-
- setenv ("PULSE_PROP_media.role", "phone", 1);
-
- pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
-
- if (!mainloop_) {
-
- // Instantiate a mainloop
- _info ("Audio: Creating PulseAudio mainloop");
-
- if (! (mainloop_ = pa_threaded_mainloop_new()))
- _warn ("Audio: Error: while creating pulseaudio mainloop");
-
- assert (mainloop_);
- }
-
- if (!context_) {
-
- // Instantiate a context
- if (! (context_ = pa_context_new (pa_threaded_mainloop_get_api (mainloop_) , "SFLphone")))
- _warn ("Audio: Error: while creating pulseaudio context");
-
- assert (context_);
- }
-
- // set context state callback before starting the mainloop
- pa_context_set_state_callback (context_, context_state_callback, this);
-
- _info ("Audio: Connect the context to the server");
-
- if (pa_context_connect (context_, NULL , flag , NULL) < 0) {
- _warn ("Audio: Error: Could not connect context to the server");
- }
-
- // Lock the loop before starting it
- pa_threaded_mainloop_lock (mainloop_);
-
- if (pa_threaded_mainloop_start (mainloop_) < 0)
- _warn ("Audio: Error: Failed to start pulseaudio mainloop");
-
- pa_threaded_mainloop_wait (mainloop_);
-
- // Run the main loop
- if (pa_context_get_state (context_) != PA_CONTEXT_READY) {
- _warn ("Audio: Error: connecting to pulse audio server");
- }
-
- pa_threaded_mainloop_unlock (mainloop_);
-
- _info ("Audio: Context creation done");
}
void PulseLayer::context_state_callback (pa_context* c, void* user_data)
@@ -370,16 +307,6 @@ void PulseLayer::context_state_callback (pa_context* c, void* user_data)
}
}
-void PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int indexRing UNUSED, int sampleRate, int stream UNUSED, const std::string &plugin UNUSED)
-{
- audioSampleRate_ = sampleRate;
-
- flushUrgent();
-
- // use 1 sec buffer for resampling
- converter_ = new SamplerateConverter (audioSampleRate_);
-}
-
void PulseLayer::updateSinkList (void)
{
diff --git a/daemon/src/audio/pulseaudio/pulselayer.h b/daemon/src/audio/pulseaudio/pulselayer.h
index 290f6b8c7e..f06ce86539 100644
--- a/daemon/src/audio/pulseaudio/pulselayer.h
+++ b/daemon/src/audio/pulseaudio/pulselayer.h
@@ -56,19 +56,6 @@ class PulseLayer : public AudioLayer
public:
PulseLayer ();
~PulseLayer (void);
- /**
- * Check if no devices are opened, otherwise close them.
- * Then open the specified devices by calling the private functions open_device
- * @param indexIn The number of the card chosen for capture
- * @param indexOut The number of the card chosen for playback
- * @param sampleRate The sample rate
- * @param stream To indicate which kind of stream you want to open
- * SFL_PCM_CAPTURE
- * SFL_PCM_PLAYBACK
- * SFL_PCM_BOTH
- * @param plugin The alsa plugin ( dmix , default , front , surround , ...)
- */
- void openDevice (int indexIn, int indexOut, int indexRing, int sampleRate, int stream, const std::string &plugin) ;
DeviceList* getSinkList (void) {
return &sinkList_;
@@ -203,11 +190,6 @@ class PulseLayer : public AudioLayer
*/
void closePlaybackStream (void);
- /**
- * Establishes the connection with the local pulseaudio server
- */
- void connectPulseAudioServer (void);
-
/**
* Close the connection with the local pulseaudio server
*/
@@ -218,11 +200,6 @@ class PulseLayer : public AudioLayer
*/
void serverinfo (void);
- void openLayer (void);
-
- void closeLayer (void);
-
-
/** PulseAudio context and asynchronous loop */
pa_context* context_;
pa_threaded_mainloop* mainloop_;
@@ -242,9 +219,6 @@ class PulseLayer : public AudioLayer
*/
AudioStream* ringtone_;
- /** Sample rate converter object */
- SamplerateConverter * converter_;
-
int spkrVolume_;
int micVolume_;
diff --git a/daemon/src/audio/samplerateconverter.cpp b/daemon/src/audio/samplerateconverter.cpp
index 8bfcbeb687..d9fae5b7ba 100644
--- a/daemon/src/audio/samplerateconverter.cpp
+++ b/daemon/src/audio/samplerateconverter.cpp
@@ -34,6 +34,7 @@
SamplerateConverter::SamplerateConverter (int freq) : _maxFreq(freq)
{
+ printf("FREQ %d\n", freq);
int err;
_src_state = src_new (SRC_LINEAR, 1, &err);
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index e6ed18e229..9ad7b54b9c 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -116,7 +116,6 @@ void ManagerImpl::init (std::string config_file)
initVolume();
initAudioDriver();
- selectAudioDriver();
// Initialize the list of supported audio codecs
_audioCodecFactory.init();
@@ -2053,11 +2052,10 @@ void ManagerImpl::setAudioPlugin (const std::string& audioPlugin)
audioPreference.setPlugin (audioPlugin);
- if (_audiodriver -> getLayerType() == ALSA) {
- _audiodriver -> openDevice (_audiodriver->getIndexIn(), _audiodriver->getIndexOut(),
- _audiodriver->getIndexRing(), _audiodriver -> getSampleRate(),
- SFL_PCM_BOTH, audioPlugin);
- }
+ AlsaLayer *alsa = dynamic_cast<AlsaLayer*>(_audiodriver);
+ if (alsa)
+ alsa->setPlugin(audioPlugin);
+
audioLayerMutexUnlock();
}
@@ -2066,46 +2064,30 @@ void ManagerImpl::setAudioPlugin (const std::string& audioPlugin)
*/
void ManagerImpl::setAudioDevice (const int index, int streamType)
{
- _debug ("Manager: Set audio device: %d", index);
audioLayerMutexLock();
- if(_audiodriver == NULL) {
- _warn ("Manager: Error: No audio driver");
- audioLayerMutexUnlock();
- return;
- }
-
AlsaLayer *alsaLayer = dynamic_cast<AlsaLayer*>(_audiodriver);
if (!alsaLayer) {
- _error("Cannot set audio output for non-alsa device");
+ _error("Can't find alsa device");
audioLayerMutexUnlock();
return ;
}
- const std::string alsaplugin(alsaLayer->getAudioPlugin());
-
- _debug ("Manager: Set ALSA plugin: %s", alsaplugin.c_str());
switch (streamType) {
case SFL_PCM_PLAYBACK:
- _debug ("Manager: Set output device");
- _audiodriver->openDevice (_audiodriver->getIndexIn(), index, _audiodriver->getIndexRing(),
- _audiodriver->getSampleRate(), SFL_PCM_PLAYBACK, alsaplugin);
+ alsaLayer->setIndexOut(index);
audioPreference.setCardout (index);
break;
case SFL_PCM_CAPTURE:
- _debug ("Manager: Set input device");
- _audiodriver->openDevice (index, _audiodriver->getIndexOut(), _audiodriver->getIndexRing(),
- _audiodriver->getSampleRate(), SFL_PCM_CAPTURE, alsaplugin);
+ alsaLayer->setIndexIn(index);
audioPreference.setCardin (index);
break;
case SFL_PCM_RINGTONE:
- _debug ("Manager: Set ringtone device");
- _audiodriver->openDevice (_audiodriver->getIndexOut(), _audiodriver->getIndexOut(), index,
- _audiodriver->getSampleRate(), SFL_PCM_RINGTONE, alsaplugin);
+ alsaLayer->setIndexRing(index);
audioPreference.setCardring (index);
break;
default:
- _warn ("Unknown stream type");
+ break;
}
audioLayerMutexUnlock();
@@ -2116,7 +2098,6 @@ void ManagerImpl::setAudioDevice (const int index, int streamType)
*/
std::vector<std::string> ManagerImpl::getAudioOutputDeviceList (void)
{
- _debug ("Manager: Get audio output device list");
std::vector<std::string> devices;
audioLayerMutexLock();
@@ -2141,15 +2122,9 @@ std::vector<std::string> ManagerImpl::getAudioInputDeviceList (void)
audioLayerMutexLock();
- AlsaLayer *alsalayer = dynamic_cast<AlsaLayer *>(_audiodriver);
-
- if (alsalayer == NULL) {
- _error("Manager: Error: Audio layer not initialized");
- audioLayerMutexUnlock();
- return devices;
- }
-
- devices = alsalayer->getSoundCardsInfo (SFL_PCM_CAPTURE);
+ AlsaLayer *alsalayer = dynamic_cast<AlsaLayer *> (_audiodriver);
+ if (alsalayer)
+ devices = alsalayer->getSoundCardsInfo (SFL_PCM_CAPTURE);
audioLayerMutexUnlock();
@@ -2165,20 +2140,17 @@ std::vector<std::string> ManagerImpl::getCurrentAudioDevicesIndex ()
std::vector<std::string> v;
- if (_audiodriver == NULL) {
- _error("Manager: Error: Audio layer not initialized");
- audioLayerMutexUnlock();
- return v;
+ AlsaLayer *alsa = dynamic_cast<AlsaLayer*>(_audiodriver);
+ if (alsa) {
+ std::stringstream ssi, sso, ssr;
+ sso << alsa->getIndexOut();
+ v.push_back (sso.str());
+ ssi << alsa->getIndexIn();
+ v.push_back (ssi.str());
+ ssr << alsa->getIndexRing();
+ v.push_back (ssr.str());
}
- std::stringstream ssi, sso, ssr;
- sso << _audiodriver->getIndexOut();
- v.push_back (sso.str());
- ssi << _audiodriver->getIndexIn();
- v.push_back (ssi.str());
- ssr << _audiodriver->getIndexRing();
- v.push_back (ssr.str());
-
audioLayerMutexUnlock();
return v;
@@ -2205,7 +2177,6 @@ void ManagerImpl::ringtoneEnabled (const std::string& id)
}
account->getRingtoneEnabled() ? account->setRingtoneEnabled (false) : account->setRingtoneEnabled (true);
-
}
std::string ManagerImpl::getRecordPath (void) const
@@ -2458,63 +2429,35 @@ void ManagerImpl::initAudioDriver (void)
{
audioLayerMutexLock();
- if (preferences.getAudioApi() == PULSEAUDIO && system("ps -C pulseaudio") == 0) {
- _audiodriver = new PulseLayer;
- } else {
- preferences.setAudioApi (ALSA);
- _audiodriver = new AlsaLayer;
- }
-
- audioLayerMutexUnlock();
-}
-
-/**
- * Initialization: Main Thread and gui
- */
-void ManagerImpl::selectAudioDriver (void)
-{
- audioLayerMutexLock();
-
- if (_audiodriver == NULL) {
- _debug("Manager: Audio layer not initialized");
- audioLayerMutexUnlock();
- return;
- }
-
/* Retrieve the global devices info from the user config */
- std::string alsaPlugin(audioPreference.getPlugin());
int numCardIn = audioPreference.getCardin();
int numCardOut = audioPreference.getCardout();
int numCardRing = audioPreference.getCardring();
- int sampleRate = getMainBuffer()->getInternalSamplingRate();
-
- /* Only for the ALSA layer, we check the sound card information */
-
- AlsaLayer *alsalayer = dynamic_cast<AlsaLayer*>(_audiodriver);
- if (alsalayer) {
-
- if (!alsalayer->soundCardIndexExist (numCardIn, SFL_PCM_CAPTURE)) {
- _debug (" Card with index %d doesn't exist or cannot capture. Switch to 0.", numCardIn);
- numCardIn = ALSA_DFT_CARD_ID;
- audioPreference.setCardin (ALSA_DFT_CARD_ID);
- }
+ if (!AlsaLayer::soundCardIndexExist (numCardIn, SFL_PCM_CAPTURE)) {
+ _debug (" Card with index %d doesn't exist or cannot capture. Switch to 0.", numCardIn);
+ numCardIn = ALSA_DFT_CARD_ID;
+ audioPreference.setCardin (ALSA_DFT_CARD_ID);
+ }
- if (!alsalayer->soundCardIndexExist (numCardOut, SFL_PCM_PLAYBACK)) {
- _debug (" Card with index %d doesn't exist or cannot playback. Switch to 0.", numCardOut);
- numCardOut = ALSA_DFT_CARD_ID;
- audioPreference.setCardout (ALSA_DFT_CARD_ID);
- }
+ if (!AlsaLayer::soundCardIndexExist (numCardOut, SFL_PCM_PLAYBACK)) {
+ _debug (" Card with index %d doesn't exist or cannot playback. Switch to 0.", numCardOut);
+ numCardOut = ALSA_DFT_CARD_ID;
+ audioPreference.setCardout (ALSA_DFT_CARD_ID);
+ }
- if (!alsalayer->soundCardIndexExist (numCardRing, SFL_PCM_RINGTONE)) {
- _debug (" Card with index %d doesn't exist or cannot ringtone. Switch to 0.", numCardRing);
- numCardRing = ALSA_DFT_CARD_ID;
- audioPreference.setCardring (ALSA_DFT_CARD_ID);
- }
+ if (!AlsaLayer::soundCardIndexExist (numCardRing, SFL_PCM_RINGTONE)) {
+ _debug (" Card with index %d doesn't exist or cannot ringtone. Switch to 0.", numCardRing);
+ numCardRing = ALSA_DFT_CARD_ID;
+ audioPreference.setCardring (ALSA_DFT_CARD_ID);
}
- /* Open the audio devices */
- _audiodriver->openDevice (numCardIn, numCardOut, numCardRing, sampleRate, SFL_PCM_BOTH, alsaPlugin);
+ if (preferences.getAudioApi() == PULSEAUDIO && system("ps -C pulseaudio") == 0) {
+ _audiodriver = new PulseLayer;
+ } else {
+ preferences.setAudioApi (ALSA);
+ _audiodriver = new AlsaLayer;
+ }
audioLayerMutexUnlock();
}
@@ -2525,34 +2468,15 @@ void ManagerImpl::switchAudioManager (void)
audioLayerMutexLock();
- if (_audiodriver == NULL) {
- audioLayerMutexUnlock();
- return;
- }
-
bool wasStarted = _audiodriver->isStarted();
- int type = _audiodriver->getLayerType();
-
- int samplerate = _mainBuffer.getInternalSamplingRate();
-
- std::string alsaPlugin(audioPreference.getPlugin());
-
- int numCardIn = audioPreference.getCardin();
- int numCardOut = audioPreference.getCardout();
- int numCardRing = audioPreference.getCardring();
-
- _debug ("Manager: Deleting current layer... ");
-
+ int newType = (_audiodriver->getLayerType()) == PULSEAUDIO ? ALSA : PULSEAUDIO;
delete _audiodriver;
- if (type == PULSEAUDIO)
- _audiodriver = new PulseLayer();
- else
- _audiodriver = new AlsaLayer();
- _audiodriver->openDevice (numCardIn, numCardOut, numCardRing, samplerate, SFL_PCM_BOTH, alsaPlugin);
-
- _debug ("Manager: Current device: %d ", type);
+ if (newType == ALSA)
+ _audiodriver = new AlsaLayer;
+ else
+ _audiodriver = new PulseLayer();
if (wasStarted)
_audiodriver->startStream();
@@ -2585,15 +2509,10 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
_debug ("Manager: New samplerate: %d", samplerate);
- std::string alsaPlugin(audioPreference.getPlugin());
-
- int numCardIn = audioPreference.getCardin();
- int numCardOut = audioPreference.getCardout();
- int numCardRing = audioPreference.getCardring();
+ bool wasActive = _audiodriver->isStarted();
- _debug ("Manager: Deleting current layer...");
- bool wasActive = _audiodriver->isStarted();
+ _mainBuffer.setInternalSamplingRate(samplerate);
delete _audiodriver;
if (type == PULSEAUDIO)
@@ -2601,12 +2520,6 @@ void ManagerImpl::audioSamplingRateChanged (int samplerate)
else
_audiodriver = new PulseLayer;
- _audiodriver->openDevice (numCardIn, numCardOut, numCardRing, samplerate, SFL_PCM_BOTH, alsaPlugin);
-
- _debug ("Manager: Current device: %d ", type);
-
- _mainBuffer.setInternalSamplingRate(samplerate);
-
unsigned int sampleRate = _audiodriver->getSampleRate();
delete _telephoneTone;
diff --git a/daemon/test/audiolayertest.cpp b/daemon/test/audiolayertest.cpp
index ca177d06e6..258916a162 100644
--- a/daemon/test/audiolayertest.cpp
+++ b/daemon/test/audiolayertest.cpp
@@ -108,12 +108,6 @@ void AudioLayerTest::testPulseConnect()
CPPUNIT_ASSERT (_pulselayer->getPlaybackStream() == NULL);
CPPUNIT_ASSERT (_pulselayer->getRecordStream() == NULL);
- try {
- _pulselayer->openDevice (sampleRate);
- } catch (...) {
- _debug ("Exception occured wile opening device! ");
- }
-
sleep (1);
CPPUNIT_ASSERT (_pulselayer->getPlaybackStream() == NULL);
--
GitLab