From 68310c6b63d68c31df728e6f47ec73dc7f7e089b Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Rafa=C3=ABl=20Carr=C3=A9?=
<rafael.carre@savoirfairelinux.com>
Date: Thu, 1 Sep 2011 14:46:56 -0400
Subject: [PATCH] * #6629: simplify AlsaLayer::alsa_set_params()
remove unneeded class members
fix a bug with samplerate
---
daemon/src/audio/alsa/alsalayer.cpp | 141 +++++++---------------------
daemon/src/audio/alsa/alsalayer.h | 27 +-----
daemon/src/audio/audiolayer.h | 2 +-
3 files changed, 39 insertions(+), 131 deletions(-)
diff --git a/daemon/src/audio/alsa/alsalayer.cpp b/daemon/src/audio/alsa/alsalayer.cpp
index a9ffb4ea16..b4afbe660d 100644
--- a/daemon/src/audio/alsa/alsalayer.cpp
+++ b/daemon/src/audio/alsa/alsalayer.cpp
@@ -80,7 +80,6 @@ AlsaLayer::AlsaLayer ()
, playbackHandle_ (NULL)
, ringtoneHandle_ (NULL)
, captureHandle_ (NULL)
- , periodSize_ (160)
, audioPlugin_ (audioPref.getPlugin())
, IDSoundCards_ ()
, is_playback_prepared_ (false)
@@ -314,115 +313,45 @@ void AlsaLayer::preparePlaybackStream (void)
bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle)
{
- snd_pcm_hw_params_t *hwparams = NULL;
- snd_pcm_sw_params_t *swparams = NULL;
- int format;
- int periods = 4;
- int periodsize = 160;
-
- /* Allocate the snd_pcm_hw_params_t struct */
- snd_pcm_hw_params_malloc (&hwparams);
-
- periodSize_ = periodsize;
- /* Full configuration space */
-
- int err;
- if ((err = snd_pcm_hw_params_any (pcm_handle, hwparams)) < 0) {
- _debug ("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) {
- _debug ("Audio: Error: Cannot set access type (%s)", snd_strerror (err));
- return false;
- }
-
- /* 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) {
- _debug ("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 */
- int dir = 0;
-
- unsigned int exact_ivalue = audioSampleRate_;
-
- if ((err = snd_pcm_hw_params_set_rate_near (pcm_handle, hwparams, &exact_ivalue, &dir) < 0)) {
- _error("Alsa: Cannot set sample rate (%s)", snd_strerror (err));
- return false;
- } else
- _debug ("Alsa: Set audio rate to %d", audioSampleRate_);
-
- if (dir != 0) {
- _error("Alsa: The chosen rate %d Hz is not supported by your hardware.Using %d Hz instead. ", audioSampleRate_, exact_ivalue);
- //audioSampleRate_ = exact_ivalue;
- // FIXME
- }
-
- /* Set the number of channels */
- if ((err = snd_pcm_hw_params_set_channels (pcm_handle, hwparams, 1)) < 0) {
- _debug ("Audio: Error: Cannot set channel count (%s)", snd_strerror (err));
- return false;
- }
-
- /* Set the buffer size in frames */
- unsigned long exact_lvalue = periodsize;
-
- dir = 0;
+#define TRY(call, error) do { \
+ int err = call; \
+ if (err < 0) { \
+ _error("ALSA: Cannot set "error": %s", snd_strerror(err)); \
+ return false; \
+ } \
+ } while(0)
+
+ snd_pcm_hw_params_t *hwparams;
+ snd_pcm_hw_params_alloca(&hwparams);
+
+ snd_pcm_uframes_t periodSize = 160;
+ unsigned int periods = 4;
+
+#define HW pcm_handle, hwparams /* hardware parameters */
+ TRY(snd_pcm_hw_params_any (HW), "hwparams init");
+ TRY(snd_pcm_hw_params_set_access (HW, SND_PCM_ACCESS_RW_INTERLEAVED), "access type");
+ TRY(snd_pcm_hw_params_set_format (HW, SND_PCM_FORMAT_S16_LE), "sample format");
+ TRY(snd_pcm_hw_params_set_rate_near (HW, &audioSampleRate_, NULL), "sample rate");
+ TRY(snd_pcm_hw_params_set_channels (HW, 1), "channel count");
+ TRY(snd_pcm_hw_params_set_period_size_near (HW, &periodSize, NULL), "period time");
+ TRY(snd_pcm_hw_params_set_periods_near (HW, &periods, NULL), "periods number");
+ TRY(snd_pcm_hw_params (HW), "hwparams");
+#undef HW
+
+ _debug ("ALSA: Using sampling rate %dHz", audioSampleRate_);
- if ((err = snd_pcm_hw_params_set_period_size_near (pcm_handle, hwparams, &exact_lvalue, &dir)) < 0) {
- _debug ("Audio: Error: Cannot set period time (%s)", snd_strerror (err));
- return false;
- }
-
- if (dir != 0)
- _warn("Alsa: The chosen period size %lu bytes is not supported by your hardware.Using %lu instead. ", periodsize, exact_lvalue);
-
- periodSize_ = exact_lvalue;
- /* Set the number of fragments */
- exact_ivalue = periods;
- dir = 0;
-
- if ((err = snd_pcm_hw_params_set_periods_near (pcm_handle, hwparams, &exact_ivalue, &dir)) < 0) {
- _debug ("Audio: Error: Cannot set periods number (%s)", snd_strerror (err));
- return false;
- }
-
- if (dir != 0)
- _debug ("Audio: Warning: The chosen period number %i bytes is not supported by your hardware.Using %i instead. ", periods, exact_ivalue);
-
- periods = exact_ivalue;
-
- /* Set the hw parameters */
-
- if ((err = snd_pcm_hw_params (pcm_handle, hwparams)) < 0) {
- _debug ("Audio: Error: Cannot set hw parameters (%s)", snd_strerror (err));
- return false;
- }
-
- snd_pcm_hw_params_free (hwparams);
-
- /* Set the sw parameters */
- snd_pcm_sw_params_malloc (&swparams);
- snd_pcm_sw_params_current (pcm_handle, swparams);
-
- /* Set the start threshold */
-
- if ((err = snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, periodSize_ * 2)) < 0) {
- _debug ("Audio: Error: Cannot set start threshold (%s)", snd_strerror (err));
- return false;
- }
+ snd_pcm_sw_params_t *swparams = NULL;
+ snd_pcm_sw_params_alloca(&swparams);
- if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) {
- _debug ("Audio: Error: Cannot set sw parameters (%s)", snd_strerror (err));
- return false;
- }
+#define SW pcm_handle, swparams /* software parameters */
+ snd_pcm_sw_params_current (SW);
+ TRY(snd_pcm_sw_params_set_start_threshold (SW, periodSize * 2), "start threshold");
+ TRY(snd_pcm_sw_params (SW), "sw parameters");
+#undef SW
- snd_pcm_sw_params_free (swparams);
return true;
+
+#undef TRY
}
//TODO first frame causes broken pipe (underrun) because not enough data are send --> make the handle wait to be ready
diff --git a/daemon/src/audio/alsa/alsalayer.h b/daemon/src/audio/alsa/alsalayer.h
index d81997e6f4..9c14a1e6c2 100644
--- a/daemon/src/audio/alsa/alsalayer.h
+++ b/daemon/src/audio/alsa/alsalayer.h
@@ -44,6 +44,9 @@ class AlsaThread;
* @brief Main sound class. Manages the data transfers between the application and the hardware.
*/
+/** Associate a sound card index to its string description */
+typedef std::pair<int , std::string> HwIDPair;
+
class AlsaLayer : public AudioLayer
{
public:
@@ -71,14 +74,6 @@ class AlsaLayer : public AudioLayer
*/
void stopStream (void);
- /**
- * Get data from the capture device
- * @param buffer The buffer for data
- * @param toCopy The number of bytes to get
- * @return int The number of bytes acquired ( 0 if an error occured)
- */
- int getMic (void * buffer, int toCopy);
-
/**
* Concatenate two strings. Used to build a valid pcm device name.
* @param plugin the alsa PCM name
@@ -116,14 +111,6 @@ class AlsaLayer : public AudioLayer
*/
int soundCardGetIndex (const std::string &description);
- /**
- * Get the current audio plugin.
- * @return std::string The name of the audio plugin
- */
- std::string getAudioPlugin (void) const {
- return audioPlugin_;
- }
-
void playback(int maxSamples);
void capture(void);
@@ -176,9 +163,6 @@ class AlsaLayer : public AudioLayer
*/
int indexRing_;
- /** Associate a sound card index to its string description */
- typedef std::pair<int , std::string> HwIDPair;
-
// Copy Constructor
AlsaLayer (const AlsaLayer& rh);
@@ -237,11 +221,6 @@ class AlsaLayer : public AudioLayer
*/
snd_pcm_t* captureHandle_;
- /**
- * Alsa parameter - Size of a period in the hardware ring buffer
- */
- snd_pcm_uframes_t periodSize_;
-
/**
* name of the alsa audio plugin used
*/
diff --git a/daemon/src/audio/audiolayer.h b/daemon/src/audio/audiolayer.h
index f6e348b0d4..850922758f 100644
--- a/daemon/src/audio/audiolayer.h
+++ b/daemon/src/audio/audiolayer.h
@@ -157,7 +157,7 @@ class AudioLayer
* Sample Rate SFLphone should send sound data to the sound card
* The value can be set in the user config file- now: 44100HZ
*/
- const unsigned int audioSampleRate_;
+ unsigned int audioSampleRate_;
/**
* Lock for the entire audio layer
--
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