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madplay: add alsa build variant 

Signed-off-by: John Crispin <blogic@openwrt.org>
lilik-openwrt-22.03
John Crispin 9 years ago
parent
2e57574f30
commit
f2e8c6a80f
2 changed files with 193 additions and 5 deletions
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  1. +20
    -5
      sound/madplay/Makefile
  2. +173
    -0
      sound/madplay/patches/0001-switch-to-new-alsa-api.patch

+ 20
- 5
sound/madplay/Makefile View File

@ -24,14 +24,18 @@ PKG_FIXUP:=autoreconf
include $(INCLUDE_DIR)/package.mk
include $(INCLUDE_DIR)/nls.mk
define Package/madplay
define Package/madplay/default
SECTION:=sound
CATEGORY:=Sound
DEPENDS:=+libid3tag +libmad $(INTL_DEPENDS)
TITLE:=MPEG audio player in fixed point
TITLE:=MPEG audio player in fixed point - $(1)
VARIANT:=$(1)
URL:=http://sourceforge.net/projects/mad
endef
Package/madplay-alsa=$(call Package/madplay/default,alsa)
Package/madplay-oss=$(call Package/madplay/default,oss)
define Package/madplay/description
MAD is an MPEG audio decoder. It currently only supports the MPEG 1
standard, but fully implements all three audio layers (Layer I, Layer II,
@ -48,16 +52,27 @@ define Build/Configure
--disable-experimental \
--without-libiconv-prefix \
--without-libintl-prefix \
--without-alsa \
--without-esd \
, \
LIBS="-lz" \
)
endef
define Package/madplay/install
ifeq ($(BUILD_VARIANT),madplay-alsa)
CONFIGURE_ARGS += \
--without-oss \
--with-alsa
endif
ifeq ($(BUILD_VARIANT),madplay-oss)
CONFIGURE_ARGS += \
--without-alsa
endif
define Package/madplay-$(BUILD_VARIANT)/install
$(INSTALL_DIR) $(1)/usr/bin
$(INSTALL_BIN) $(PKG_BUILD_DIR)/madplay $(1)/usr/bin/
endef
$(eval $(call BuildPackage,madplay))
$(eval $(call BuildPackage,madplay-alsa))
$(eval $(call BuildPackage,madplay-oss))

+ 173
- 0
sound/madplay/patches/0001-switch-to-new-alsa-api.patch View File

@ -0,0 +1,173 @@
Switch madplay to the new API. This is done thanks to a patch written
by Micha Nelissen <micha@neli.hopto.org> and available at
http://article.gmane.org/gmane.comp.audio.mad.devel/729.
--- 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;

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