Madplay improved patch for alsa audio

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Madplay improved patch for alsa audio

by Micha Nelissen :: Rate this Message:

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Hi,

Attached is an improved patch for alsa backend of madplay.
* fix allocation of buffer, buffer_size is number of frames
* use default 16 bit depth (some hardware has small buffers, 16/24 bit
difference is very small)

Regards,

Micha


--- madplay-0.15.2b/audio_alsa.c 2008-10-18 15:10:16.000000000 +0200
+++ madplay-0.15.2b/audio_alsa.c.new 2008-10-18 15:03:27.000000000 +0200
@@ -28,31 +28,30 @@
 
 #include <errno.h>
 
-#define ALSA_PCM_OLD_HW_PARAMS_API
-#define ALSA_PCM_OLD_SW_PARAMS_API
 #include <alsa/asoundlib.h>
 
 #include <mad.h>
 
 #include "audio.h"
 
-char *buf = NULL;
-int paused = 0;
+#define BUFFER_TIME_MAX     500000
 
-int rate = -1;
-int channels = -1;
-int bitdepth = -1;
-int sample_size = -1;
-
-int buffer_time = 500000;
-int period_time = 100000;
-char *defaultdev = "plughw:0,0";
+unsigned char *buf  = NULL;
+int paused    = 0;
+
+unsigned int rate           = 0;
+unsigned int channels    = -1;
+unsigned int bitdepth    = -1;
+unsigned int sample_size    = -1;
+
+unsigned int buffer_time;
+unsigned int period_time;
+char *defaultdev    = "plughw:0,0";
 
 snd_pcm_hw_params_t *alsa_hwparams;
 snd_pcm_sw_params_t *alsa_swparams;
 
-snd_pcm_sframes_t buffer_size;
-snd_pcm_sframes_t period_size;
+snd_pcm_uframes_t buffer_size;
 
 snd_pcm_format_t  alsa_format = -1;
 snd_pcm_access_t  alsa_access = SND_PCM_ACCESS_MMAP_INTERLEAVED;
@@ -66,14 +65,20 @@
  snd_pcm_hw_params_t *params,
  snd_pcm_access_t access)
 {
- int err, dir;
-
+ int err;
+
  /* choose all parameters */
  err = snd_pcm_hw_params_any(handle,params);
  if (err < 0) {
  printf("Access type not available for playback: %s\n", snd_strerror(err));
  return err;
  }
+ /* set the access type */
+ err = snd_pcm_hw_params_set_access(handle, params, alsa_access);
+ if (err < 0) {
+ printf("Sample format not available for playback: %s\n", snd_strerror(err));
+ return err;
+ }
  /* set the sample format */
  err = snd_pcm_hw_params_set_format(handle, params, alsa_format);
  if (err < 0) {
@@ -87,29 +92,38 @@
  return err;
  }
  /* set the stream rate */
- err = snd_pcm_hw_params_set_rate_near(handle, params, rate, 0);
+ err = snd_pcm_hw_params_set_rate(handle, params, rate, 0);
  if (err < 0) {
  printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err));
  return err;
  }
- if (err != rate) {
- printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err);
- return -EINVAL;
- }
+ err = snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, NULL);
+        if (err < 0) {
+                printf("Unable to retrieve buffer time: %s\n", snd_strerror(err));
+                return err;
+        }
+        if (buffer_time > BUFFER_TIME_MAX)
+                buffer_time = BUFFER_TIME_MAX;
  /* set buffer time */
- err = snd_pcm_hw_params_set_buffer_time_near(handle, params, buffer_time, &dir);
+ err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, 0);
  if (err < 0) {
  printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err));
  return err;
  }
- buffer_size = snd_pcm_hw_params_get_buffer_size(params);
+        if (period_time * 4 > buffer_time)
+                period_time = buffer_time / 4;
  /* set period time */
- err = snd_pcm_hw_params_set_period_time_near(handle, params, period_time, &dir);
+ err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, NULL);
  if (err < 0) {
  printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err));
  return err;
  }
- period_size = snd_pcm_hw_params_get_period_size(params, &dir);
+        /* retrieve buffer size */
+ err = snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
+        if (err < 0) {
+                printf("Unable to retrieve buffer size: %s\n", snd_strerror(err));
+                return err;
+        }
  /* write the parameters to device */
  err = snd_pcm_hw_params(handle, params);
  if (err < 0) {
@@ -123,6 +137,7 @@
 int set_swparams(snd_pcm_t *handle,
  snd_pcm_sw_params_t *params)
 {
+        unsigned int start_threshold;
  int err;
 
         /* get current swparams */
@@ -136,13 +151,7 @@
         if (err < 0) {
                 printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err));
                 return err;
-        }
-        /* allow transfer when at least period_size samples can be processed */
-        err = snd_pcm_sw_params_set_avail_min(handle, params, period_size);
-        if (err < 0) {
-                printf("Unable to set avail min for playback: %s\n", snd_strerror(err));
-                return err;
-        }
+ }
         /* align all transfers to 1 samples */
         err = snd_pcm_sw_params_set_xfer_align(handle, params, 1);
         if (err < 0) {
@@ -190,7 +199,7 @@
  rate = config->speed;
 
  if ( bitdepth == 0 )
- config->precision = bitdepth = 32;
+ config->precision = bitdepth = 16;
 
  switch (bitdepth)
  {
@@ -241,7 +250,7 @@
  return -1;
  }
 
- buf = malloc(buffer_size);
+ buf = malloc(buffer_size * sample_size);
  if (buf == NULL) {
  audio_error="unable to allocate output buffer table";
  return -1;
@@ -279,7 +288,7 @@
 int play(struct audio_play *play)
 {
  int err, len;
- char *ptr;
+ unsigned char *ptr;
 
  ptr = buf;
  len = play->nsamples;