[MPlayer-dev-eng] [PATCH] alsa-merge
joy at pingfm.org
joy at pingfm.org
Wed Apr 28 22:49:47 CEST 2004
So a 'big' alsa-patch it will replace alsa9 and alsa1x. cause i think
alsa5 is nearly obsolet i want to call it alsa. it changes configure and
libao2/audio_out.c so i post it here for discussion.
patch includes:
gcc-3.4 fix (improper select statement)
default device will be default (as recommend by users). users would be
able now to use dmix-plugin without defining the subdevice.
printfs converted to mp_msg's
please test it. i only run it on a few boxes with alsa9 and alsa1.0.4, but
no multichannel tests, wired formats etc.
regards
joy
________________________________________
do interactive tv on www.remote-tv.de
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diff -Nur -x CVS -x DOC -x '.*' -x '*~' clean_cvs/configure src-merge2/configure
--- clean_cvs/configure 2004-04-26 11:44:06.000000000 +0200
+++ src-merge2/configure 2004-04-28 17:15:27.000000000 +0200
@@ -4053,25 +4053,25 @@
_def_sys_asoundlib_h='#define HAVE_SYS_ASOUNDLIB_H 1'
echores "yes (using alsa 0.5.x and sys/asoundlib.h)"
elif test "$_alsaver" = '0.9.x-sys' ; then
- _aosrc="$_aosrc ao_alsa9.c"
+ _aosrc="$_aosrc ao_alsa.c"
_aomodules="alsa9 $_aomodules"
_def_alsa9='#define HAVE_ALSA9 1'
_def_sys_asoundlib_h='#define HAVE_SYS_ASOUNDLIB_H 1'
echores "yes (using alsa 0.9.x and sys/asoundlib.h)"
elif test "$_alsaver" = '0.9.x-alsa' ; then
- _aosrc="$_aosrc ao_alsa9.c"
+ _aosrc="$_aosrc ao_alsa.c"
_aomodules="alsa9 $_aomodules"
_def_alsa9='#define HAVE_ALSA9 1'
_def_alsa_asoundlib_h='#define HAVE_ALSA_ASOUNDLIB_H 1'
echores "yes (using alsa 0.9.x and alsa/asoundlib.h)"
elif test "$_alsaver" = '1.0.x-sys' ; then
- _aosrc="$_aosrc ao_alsa1x.c"
+ _aosrc="$_aosrc ao_alsa.c"
_aomodules="alsa1x $_aomodules"
_def_alsa1x="#define HAVE_ALSA1X 1"
_def_alsa_asoundlib_h='#define HAVE_SYS_ASOUNDLIB_H 1'
echores "yes (using alsa 1.0.x and sys/asoundlib.h)"
elif test "$_alsaver" = '1.0.x-alsa' ; then
- _aosrc="$_aosrc ao_alsa1x.c"
+ _aosrc="$_aosrc ao_alsa.c"
_aomodules="alsa1x $_aomodules"
_def_alsa1x="#define HAVE_ALSA1X 1"
_def_alsa_asoundlib_h='#define HAVE_ALSA_ASOUNDLIB_H 1'
diff -Nur -x CVS -x DOC -x '.*' -x '*~' clean_cvs/libao2/ao_alsa.c src-merge2/libao2/ao_alsa.c
--- clean_cvs/libao2/ao_alsa.c 1970-01-01 01:00:00.000000000 +0100
+++ src-merge2/libao2/ao_alsa.c 2004-04-28 16:58:32.000000000 +0200
@@ -0,0 +1,1177 @@
+/*
+ ao_alsa9/1.x - ALSA-0.9.x-1.x output plugin for MPlayer
+
+ (C) Alex Beregszaszi
+
+ modified for real alsa-0.9.0-support by Zsolt Barat <joy at streamminister.de>
+ additional AC3 passthrough support by Andy Lo A Foe <andy at alsaplayer.org>
+ 08/22/2002 iec958-init rewritten and merged with common init, zsolt
+ 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
+ 04/25/2004 printfs converted to mp_msg, Zsolt.
+
+ Any bugreports regarding to this driver are welcome.
+*/
+
+#include <errno.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#include <math.h>
+#include <string.h>
+#include <sys/poll.h>
+
+#include "../config.h"
+#include "../mixer.h"
+#include "../mp_msg.h"
+
+#define ALSA_PCM_NEW_HW_PARAMS_API
+#define ALSA_PCM_NEW_SW_PARAMS_API
+
+#if HAVE_SYS_ASOUNDLIB_H
+#include <sys/asoundlib.h>
+#elif HAVE_ALSA_ASOUNDLIB_H
+#include <alsa/asoundlib.h>
+#else
+#error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
+#endif
+
+
+#include "audio_out.h"
+#include "audio_out_internal.h"
+#include "afmt.h"
+
+static ao_info_t info =
+{
+ "ALSA-0.9.x-1.x audio output",
+ "alsa",
+ "Alex Beregszaszi, Zsolt Barat <joy at streamminister.de>",
+ "under developement"
+};
+
+LIBAO_EXTERN(alsa)
+
+static snd_pcm_t *alsa_handler;
+static snd_pcm_format_t alsa_format;
+static snd_pcm_hw_params_t *alsa_hwparams;
+static snd_pcm_sw_params_t *alsa_swparams;
+static char *alsa_device;
+
+/* possible 4096, original 8192
+ * was only needed for calculating chunksize? */
+static int alsa_fragsize = 4096;
+/* 16 sets buffersize to 16 * chunksize is as default 1024
+ * which seems to be good avarge for most situations
+ * so buffersize is 16384 frames by default */
+static int alsa_fragcount = 16;
+static snd_pcm_uframes_t chunk_size = 1024;//is alsa_fragsize / 4
+
+#define MIN_CHUNK_SIZE 1024
+
+static size_t bits_per_sample, bytes_per_sample, bits_per_frame;
+static size_t chunk_bytes;
+
+int ao_mmap = 0;
+int ao_noblock = 0;
+int first = 1;
+
+static int open_mode;
+static int set_block_mode;
+static int alsa_can_pause = 0;
+
+#define ALSA_DEVICE_SIZE 48
+
+#undef BUFFERTIME
+#undef SET_CHUNKSIZE
+#undef USE_POLL
+
+/* to set/get/query special features/parameters */
+static int control(int cmd, void *arg)
+{
+ switch(cmd) {
+ case AOCONTROL_QUERY_FORMAT:
+ return CONTROL_TRUE;
+#ifndef WORDS_BIGENDIAN
+ case AOCONTROL_GET_VOLUME:
+ case AOCONTROL_SET_VOLUME:
+ {
+ ao_control_vol_t *vol = (ao_control_vol_t *)arg;
+
+ int err;
+ snd_mixer_t *handle;
+ snd_mixer_elem_t *elem;
+ snd_mixer_selem_id_t *sid;
+
+ static char *mix_name = NULL;
+ static char *card = NULL;
+
+ long pmin, pmax;
+ long get_vol, set_vol;
+ float calc_vol, diff, f_multi;
+
+ if(mix_name == NULL){
+ if(mixer_device) {
+ card = strdup(mixer_device);
+ mix_name = strchr(card, '/');
+ if(mix_name) {
+ *mix_name++ = 0;
+ } else {
+ mix_name = "PCM";
+ }
+ } else {
+ mix_name = "PCM";
+ card = "default";
+ }
+ }
+
+ if(ao_data.format == AFMT_AC3)
+ return CONTROL_TRUE;
+
+ //allocate simple id
+ snd_mixer_selem_id_alloca(&sid);
+
+ //sets simple-mixer index and name
+ snd_mixer_selem_id_set_index(sid, 0);
+ snd_mixer_selem_id_set_name(sid, mix_name);
+
+ if ((err = snd_mixer_open(&handle, 0)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer open error: %s\n", snd_strerror(err));
+ return CONTROL_ERROR;
+ }
+
+ if ((err = snd_mixer_attach(handle, card)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer attach %s error: %s\n",
+ card, snd_strerror(err));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer register error: %s\n", snd_strerror(err));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+ err = snd_mixer_load(handle);
+ if (err < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer load error: %s\n", snd_strerror(err));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ elem = snd_mixer_find_selem(handle, sid);
+ if (!elem) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: unable to find simple control '%s',%i\n",
+ snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
+ snd_mixer_close(handle);
+ return CONTROL_ERROR;
+ }
+
+ snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
+ f_multi = (100 / (float)pmax);
+
+ if (cmd == AOCONTROL_SET_VOLUME) {
+
+ diff = (vol->left+vol->right) / 2;
+ set_vol = rint(diff / f_multi);
+
+ if (set_vol < 0)
+ set_vol = 0;
+ else if (set_vol > pmax)
+ set_vol = pmax;
+
+ //setting channels
+ if ((err = snd_mixer_selem_set_playback_volume(elem, 0, set_vol)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting left channel, %s\n",
+ snd_strerror(err));
+ return CONTROL_ERROR;
+ }
+ if ((err = snd_mixer_selem_set_playback_volume(elem, 1, set_vol)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting right channel, %s\n",
+ snd_strerror(err));
+ return CONTROL_ERROR;
+ }
+
+ mp_msg(MSGT_AO,MSGL_DBG2,"diff=%f, set_vol=%li, pmax=%li, mult=%f\n",
+ diff, set_vol, pmax, f_multi);
+ }
+ else {
+ snd_mixer_selem_get_playback_volume(elem, 0, &get_vol);
+ calc_vol = get_vol;
+ calc_vol = rintf(calc_vol * f_multi);
+
+ vol->left = vol->right = (int)calc_vol;
+
+ mp_msg(MSGT_AO,MSGL_DBG2,"get_vol = %li, calc=%f\n",get_vol, calc_vol);
+ }
+ snd_mixer_close(handle);
+ return CONTROL_OK;
+ }
+#endif
+
+ } //end switch
+ return(CONTROL_UNKNOWN);
+}
+
+
+/*
+ open & setup audio device
+ return: 1=success 0=fail
+*/
+static int init(int rate_hz, int channels, int format, int flags)
+{
+ int err;
+ int cards = -1;
+ int period_val;
+ snd_pcm_info_t *alsa_info;
+ char *str_block_mode;
+ int device_set = 0;
+ int dir = 0;
+ snd_pcm_uframes_t bufsize;
+
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
+ channels, audio_out_format_name(format));
+ alsa_handler = NULL;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
+
+ if ((err = snd_card_next(&cards)) < 0 || cards < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: no soundcards found: %s\n", snd_strerror(err));
+ return(0);
+ }
+
+ ao_data.samplerate = rate_hz;
+ ao_data.bps = channels * rate_hz;
+ ao_data.format = format;
+ ao_data.channels = channels;
+ ao_data.outburst = OUTBURST;
+
+ switch (format)
+ {
+ case AFMT_S8:
+ alsa_format = SND_PCM_FORMAT_S8;
+ break;
+ case AFMT_U8:
+ alsa_format = SND_PCM_FORMAT_U8;
+ break;
+ case AFMT_U16_LE:
+ alsa_format = SND_PCM_FORMAT_U16_LE;
+ break;
+ case AFMT_U16_BE:
+ alsa_format = SND_PCM_FORMAT_U16_BE;
+ break;
+#ifndef WORDS_BIGENDIAN
+ case AFMT_AC3:
+#endif
+ case AFMT_S16_LE:
+ alsa_format = SND_PCM_FORMAT_S16_LE;
+ break;
+#ifdef WORDS_BIGENDIAN
+ case AFMT_AC3:
+#endif
+ case AFMT_S16_BE:
+ alsa_format = SND_PCM_FORMAT_S16_BE;
+ break;
+ case AFMT_S32_LE:
+ alsa_format = SND_PCM_FORMAT_S32_LE;
+ break;
+ case AFMT_S32_BE:
+ alsa_format = SND_PCM_FORMAT_S32_BE;
+ break;
+
+ default:
+ alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
+ break;
+ }
+
+ //setting bw according to the input-format. resolution seems to be always s16_le or
+ //u16_le so 32bit is probably obsolet.
+ switch(alsa_format)
+ {
+ case SND_PCM_FORMAT_S16_LE:
+ case SND_PCM_FORMAT_U16_LE:
+ ao_data.bps *= 2;
+ break;
+ case SND_PCM_FORMAT_S32_LE:
+ case SND_PCM_FORMAT_S32_BE:
+ ao_data.bps *= 4;
+ break;
+ case -1:
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%s) requested - output disabled\n",
+ audio_out_format_name(format));
+ return(0);
+ break;
+ default:
+ ao_data.bps *= 2;
+ mp_msg(MSGT_AO,MSGL_WARN,"alsa-init: couldn't convert to right format. setting bps to: %d", ao_data.bps);
+ }
+
+ if (ao_subdevice) {
+ //start parsing ao_subdevice, ugly and not thread safe!
+ //maybe there's a better way?
+ int i2 = 1;
+ int i3 = 0;
+ char *sub_str;
+
+ char *token_str[3];
+ char* test_str = strdup(ao_subdevice);
+
+
+ if ((strcspn(ao_subdevice, ":")) > 0) {
+
+ sub_str = strtok(test_str, ":");
+ *(token_str) = sub_str;
+
+ while (((sub_str = strtok(NULL, ":")) != NULL) && (i2 <= 3)) {
+ *(token_str+i2) = sub_str;
+ i2 += 1;
+ }
+
+ for (i3=0; i3 <= i2-1; i3++) {
+ if (strcmp(*(token_str + i3), "mmap") == 0) {
+ ao_mmap = 1;
+ }
+ else if (strcmp(*(token_str+i3), "noblock") == 0) {
+ ao_noblock = 1;
+ }
+ else if (strcmp(*(token_str+i3), "hw") == 0) {
+ if ((i3 < i2-1) && (strcmp(*(token_str+i3+1), "noblock") != 0) && (strcmp(*(token_str+i3+1), "mmap") != 0)) {
+ char *tmp;
+
+ alsa_device = alloca(ALSA_DEVICE_SIZE);
+ snprintf(alsa_device, ALSA_DEVICE_SIZE, "hw:%s", *(token_str+(i3+1)));
+ if ((tmp = strrchr(alsa_device, '.')) && isdigit(*(tmp+1)))
+ *tmp = ',';
+ device_set = 1;
+ }
+ else {
+ alsa_device = *(token_str+i3);
+ device_set = 1;
+ }
+ }
+ else if (device_set == 0 && (!ao_mmap || !ao_noblock)) {
+ alsa_device = *(token_str+i3);
+ device_set = 1;
+ }
+ }
+ }
+ } else { //end parsing ao_subdevice
+ /* in any case for multichannel playback we should select
+ * appropriate device
+ */
+ char devstr[128];
+
+ switch (channels) {
+ case 1:
+ case 2:
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
+ break;
+ case 4:
+ strcpy(devstr, "surround40");
+ alsa_device = devstr;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
+ break;
+ case 6:
+ strcpy(devstr, "surround51");
+ alsa_device = devstr;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
+ break;
+ default:
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: %d channels are not supported\n",channels);
+ }
+ }
+
+ /* switch for spdif
+ * sets opening sequence for SPDIF
+ * sets also the playback and other switches 'on the fly'
+ * while opening the abstract alias for the spdif subdevice
+ * 'iec958'
+ */
+ if (format == AFMT_AC3) {
+ char devstr[128];
+ unsigned char s[4];
+ //int err, c; //unused
+
+ switch (channels) {
+ case 1:
+ case 2:
+
+ s[0] = IEC958_AES0_NONAUDIO |
+ IEC958_AES0_CON_EMPHASIS_NONE;
+ s[1] = IEC958_AES1_CON_ORIGINAL |
+ IEC958_AES1_CON_PCM_CODER;
+ s[2] = 0;
+ s[3] = IEC958_AES3_CON_FS_48000;
+
+ sprintf(devstr, "iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x",
+ s[0], s[1], s[2], s[3]);
+
+ mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
+ break;
+ case 4:
+ strcpy(devstr, "surround40");
+ break;
+
+ case 6:
+ strcpy(devstr, "surround51");
+ break;
+
+ default:
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-spdif-init: %d channels are not supported\n", channels);
+ }
+
+ alsa_device = devstr;
+ }
+
+ if (alsa_device == NULL)
+ {
+ int tmp_device, tmp_subdevice, err;
+
+ if ((err = snd_pcm_info_malloc(&alsa_info)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: memory allocation error: %s\n", snd_strerror(err));
+ }
+
+ if ((alsa_device = alloca(ALSA_DEVICE_SIZE)) == NULL)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: memory allocation error: %s\n", strerror(errno));
+ }
+
+ if ((tmp_device = snd_pcm_info_get_device(alsa_info)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't get device\n");
+ }
+
+ if ((tmp_subdevice = snd_pcm_info_get_subdevice(alsa_info)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't get subdevice\n");
+ }
+
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: got device=%i, subdevice=%i\n",
+ tmp_device, tmp_subdevice);
+
+ //we are setting here device to default cause it could be configured by the user
+ //if its not set by the user, it defaults to hw:0,0
+ if ((err = snprintf(alsa_device, ALSA_DEVICE_SIZE, "default")) <= 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't write device-id\n");
+ }
+
+ snd_pcm_info_free(alsa_info);
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: %d soundcard%s found, using: %s\n", cards+1,(cards >= 0) ? "" : "s", alsa_device);
+ } else if (strcmp(alsa_device, "help") == 0) {
+ printf("alsa-help: available options are:\n");
+ printf(" mmap: sets mmap-mode\n");
+ printf(" noblock: sets noblock-mode\n");
+ printf(" device-name: sets device name (change comma to point)\n");
+ printf(" example -ao alsa9:mmap:noblock:hw:0.3 sets noblock-mode,\n");
+ printf(" mmap-mode and the device-name as first card fourth device\n");
+ return(0);
+ } else {
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: soundcard set to %s\n", alsa_device);
+ }
+
+ //setting modes for block or nonblock-mode
+ if (ao_noblock) {
+ open_mode = SND_PCM_NONBLOCK;
+ set_block_mode = 1;
+ str_block_mode = "nonblock-mode";
+ }
+ else {
+ open_mode = 0;
+ set_block_mode = 0;
+ str_block_mode = "block-mode";
+ }
+
+ //sets buff/chunksize if its set manually
+ if (ao_data.buffersize) {
+ switch (ao_data.buffersize)
+ {
+ case 1:
+ alsa_fragcount = 16;
+ chunk_size = 512;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
+ break;
+ case 2:
+ alsa_fragcount = 8;
+ chunk_size = 1024;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n");
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
+ break;
+ case 3:
+ alsa_fragcount = 32;
+ chunk_size = 512;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n");
+ break;
+ case 4:
+ alsa_fragcount = 16;
+ chunk_size = 1024;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n");
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n");
+ break;
+ default:
+ alsa_fragcount = 16;
+ if (ao_mmap)
+ chunk_size = 512;
+ else
+ chunk_size = 1024;
+ break;
+ }
+ }
+
+ if (!alsa_handler) {
+ //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
+ if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, open_mode)) < 0)
+ {
+ if (err != -EBUSY && ao_noblock) {
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: open in nonblock-mode failed, trying to open in block-mode\n");
+ if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err));
+ return(0);
+ } else {
+ set_block_mode = 0;
+ str_block_mode = "block-mode";
+ }
+ } else {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err));
+ return(0);
+ }
+ }
+
+ if ((err = snd_pcm_nonblock(alsa_handler, set_block_mode)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: error set block-mode %s\n", snd_strerror(err));
+ } else {
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opend in %s\n", str_block_mode);
+ }
+
+ snd_pcm_hw_params_alloca(&alsa_hwparams);
+ snd_pcm_sw_params_alloca(&alsa_swparams);
+
+ // setting hw-parameters
+ if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get initial parameters: %s\n",
+ snd_strerror(err));
+ return(0);
+ }
+
+ if (ao_mmap) {
+ snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof());
+ snd_pcm_access_mask_none(mask);
+ snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED);
+ snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
+ snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX);
+ err = snd_pcm_hw_params_set_access_mask(alsa_handler, alsa_hwparams, mask);
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: mmap set\n");
+ } else {
+ err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+ }
+ if (err < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set access type: %s\n",
+ snd_strerror(err));
+ return (0);
+ }
+
+ /* workaround for nonsupported formats
+ sets default format to S16_LE if the given formats aren't supported */
+ if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams,
+ alsa_format)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_INFO,
+ "alsa-init: format %s are not supported by hardware, trying default\n",
+ audio_out_format_name(format));
+ alsa_format = SND_PCM_FORMAT_S16_LE;
+ ao_data.format = AFMT_S16_LE;
+ ao_data.bps = channels * rate_hz * 2;
+ }
+
+ bytes_per_sample = ao_data.bps / ao_data.samplerate; //it should be here
+
+
+ if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
+ alsa_format)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set format: %s\n",
+ snd_strerror(err));
+ }
+
+ if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams,
+ ao_data.channels)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set channels: %s\n",
+ snd_strerror(err));
+ }
+
+ if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
+ &ao_data.samplerate, &dir)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set samplerate-2: %s\n",
+ snd_strerror(err));
+ return(0);
+ }
+
+#ifdef BUFFERTIME
+ {
+ int alsa_buffer_time = 500000; /* original 60 */
+ int alsa_period_time;
+ alsa_period_time = alsa_buffer_time/4;
+ if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
+ &alsa_buffer_time, &dir)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set buffer time near: %s\n",
+ snd_strerror(err));
+ return(0);
+ } else
+ alsa_buffer_time = err;
+
+ if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
+ &alsa_period_time, &dir)) < 0)
+ /* original: alsa_buffer_time/ao_data.bps */
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set period time: %s\n",
+ snd_strerror(err));
+ }
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: buffer_time: %d, period_time :%d\n",
+ alsa_buffer_time, err);
+ }
+#endif//end SET_BUFFERTIME
+
+#ifdef SET_CHUNKSIZE
+ {
+ //set chunksize
+ dir=0;
+ if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
+ &chunk_size, &dir)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periodsize(%d): %s\n",
+ chunk_size, snd_strerror(err));
+ }
+ else {
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %i\n", chunk_size);
+ }
+ if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
+ &alsa_fragcount, &dir)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periods: %s\n",
+ snd_strerror(err));
+ }
+ else {
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount);
+ }
+ }
+#endif//end SET_CHUNKSIZE
+
+ /* finally install hardware parameters */
+ if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set hw-parameters: %s\n",
+ snd_strerror(err));
+ }
+ // end setting hw-params
+
+
+ // gets buffersize for control
+ if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get buffersize: %s\n", snd_strerror(err));
+ }
+ else {
+ ao_data.buffersize = bufsize * bytes_per_sample;
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
+ }
+
+ // setting sw-params (only avail-min) if noblocking mode was choosed
+ if (ao_noblock)
+ {
+
+ if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get parameters: %s\n",
+ snd_strerror(err));
+
+ }
+
+ //set min available frames to consider pcm ready (4)
+ //increased for nonblock-mode should be set dynamically later
+ if ((err = snd_pcm_sw_params_set_avail_min(alsa_handler, alsa_swparams, 4)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set avail_min %s\n",
+ snd_strerror(err));
+ }
+
+ if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to install sw-params\n");
+ }
+
+ bits_per_sample = snd_pcm_format_physical_width(alsa_format);
+ bits_per_frame = bits_per_sample * channels;
+ chunk_bytes = chunk_size * bits_per_frame / 8;
+
+ mp_msg(MSGT_AO,MSGL_V,"alsa-init: bits per sample (bps)=%i, bits per frame (bpf)=%i, chunk_bytes=%i\n",bits_per_sample,bits_per_frame,chunk_bytes);}
+ //end swparams
+
+ if ((err = snd_pcm_prepare(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: pcm prepare error: %s\n", snd_strerror(err));
+ }
+
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
+ ao_data.samplerate, ao_data.channels, bytes_per_sample, ao_data.buffersize,
+ snd_pcm_format_description(alsa_format));
+
+ } // end switch alsa_handler (spdif)
+ alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
+ return(1);
+} // end init
+
+
+/* close audio device */
+static void uninit(int immed)
+{
+
+ if (alsa_handler) {
+ int err;
+
+ if (!ao_noblock) {
+ if ((err = snd_pcm_drop(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm drop error: %s\n", snd_strerror(err));
+ return;
+ }
+ }
+
+ if ((err = snd_pcm_close(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm close error: %s\n", snd_strerror(err));
+ return;
+ }
+ else {
+ alsa_handler = NULL;
+ alsa_device = NULL;
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-uninit: pcm closed\n");
+ }
+ }
+ else {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: no handler defined!\n");
+ }
+}
+
+static void audio_pause()
+{
+ int err;
+
+ if (alsa_can_pause) {
+ if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm pause error: %s\n", snd_strerror(err));
+ return;
+ }
+ mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
+ } else {
+ if ((err = snd_pcm_drop(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm drop error: %s\n", snd_strerror(err));
+ return;
+ }
+ }
+}
+
+static void audio_resume()
+{
+ int err;
+
+ if (alsa_can_pause) {
+ if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm resume error: %s\n", snd_strerror(err));
+ return;
+ }
+ mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n");
+ } else {
+ if ((err = snd_pcm_prepare(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm prepare error: %s\n", snd_strerror(err));
+ return;
+ }
+ }
+}
+
+/* stop playing and empty buffers (for seeking/pause) */
+static void reset()
+{
+ int err;
+
+ if ((err = snd_pcm_drop(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm drop error: %s\n", snd_strerror(err));
+ return;
+ }
+ if ((err = snd_pcm_prepare(alsa_handler)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm prepare error: %s\n", snd_strerror(err));
+ return;
+ }
+ return;
+}
+
+#ifdef USE_POLL
+static int wait_for_poll(snd_pcm_t *handle, struct pollfd *ufds, unsigned int count)
+{
+ unsigned short revents;
+
+ while (1) {
+ poll(ufds, count, -1);
+ snd_pcm_poll_descriptors_revents(handle, ufds, count, &revents);
+ if (revents & POLLERR)
+ return -EIO;
+ if (revents & POLLOUT)
+ return 0;
+ }
+}
+#endif
+
+#ifndef timersub
+#define timersub(a, b, result) \
+do { \
+ (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \
+ (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \
+ if ((result)->tv_usec < 0) { \
+ --(result)->tv_sec; \
+ (result)->tv_usec += 1000000; \
+ } \
+} while (0)
+#endif
+
+/* I/O error handler */
+static int xrun(u_char *str_mode)
+{
+ int err;
+ snd_pcm_status_t *status;
+
+ snd_pcm_status_alloca(&status);
+
+ if ((err = snd_pcm_status(alsa_handler, status))<0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"status error: %s", snd_strerror(err));
+ return(0);
+ }
+
+ if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) {
+ struct timeval now, diff, tstamp;
+ gettimeofday(&now, 0);
+ snd_pcm_status_get_trigger_tstamp(status, &tstamp);
+ timersub(&now, &tstamp, &diff);
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-%s: xrun of at least %.3f msecs. resetting stream\n",
+ str_mode,
+ diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
+ }
+
+ if ((err = snd_pcm_prepare(alsa_handler))<0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"xrun: prepare error: %s", snd_strerror(err));
+ return(0);
+ }
+
+ return(1); /* ok, data should be accepted again */
+}
+
+static int play_normal(void* data, int len);
+static int play_mmap(void* data, int len);
+
+static int play(void* data, int len, int flags)
+{
+ int result;
+ if (ao_mmap)
+ result = play_mmap(data, len);
+ else
+ result = play_normal(data, len);
+
+ return result;
+}
+
+/*
+ plays 'len' bytes of 'data'
+ returns: number of bytes played
+ modified last at 29.06.02 by jp
+ thanxs for marius <marius at rospot.com> for giving us the light ;)
+*/
+
+static int play_normal(void* data, int len)
+{
+
+ //bytes_per_sample is always 4 for 2 chn S16_LE
+ int num_frames = len / bytes_per_sample;
+ char *output_samples = (char *)data;
+ snd_pcm_sframes_t res = 0;
+
+ //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
+
+ if (!alsa_handler) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: device configuration error");
+ return 0;
+ }
+
+ while (num_frames > 0) {
+
+ res = snd_pcm_writei(alsa_handler, (void *)output_samples, num_frames);
+
+ if (res == -EAGAIN) {
+ snd_pcm_wait(alsa_handler, 1000);
+ }
+ else if (res == -EPIPE) { /* underrun */
+ if (xrun("play") <= 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: xrun reset error");
+ return(0);
+ }
+ }
+ else if (res == -ESTRPIPE) { /* suspend */
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: pcm in suspend mode. trying to resume\n");
+ while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
+ sleep(1);
+ }
+ else if (res < 0) {
+ mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: unknown status, trying to reset soundcard\n");
+ if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: snd prepare error");
+ return(0);
+ break;
+ }
+ }
+
+ if (res > 0) {
+
+ /* output_samples += ao_data.channels * res; */
+ output_samples += res * bytes_per_sample;
+
+ num_frames -= res;
+ }
+
+ } //end while
+
+ if (res < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: write error %s", snd_strerror(res));
+ return 0;
+ }
+ return res < 0 ? (int)res : len - len % bytes_per_sample;
+}
+
+/* mmap-mode mainly based on descriptions by Joshua Haberman <joshua at haberman.com>
+ * 'An overview of the ALSA API' http://people.debian.org/~joshua/x66.html
+ * and some help by Paul Davis <pbd at op.net> */
+
+static int play_mmap(void* data, int len)
+{
+ snd_pcm_sframes_t commitres, frames_available;
+ snd_pcm_uframes_t frames_transmit, size, offset;
+ const snd_pcm_channel_area_t *area;
+ void *outbuffer;
+ int err, result;
+
+#ifdef USE_POLL //seems not really be needed
+ struct pollfd *ufds;
+ int count;
+
+ count = snd_pcm_poll_descriptors_count (alsa_handler);
+ ufds = malloc(sizeof(struct pollfd) * count);
+ snd_pcm_poll_descriptors(alsa_handler, ufds, count);
+
+ //first wait_for_poll
+ if (err = (wait_for_poll(alsa_handler, ufds, count) < 0)) {
+ if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_XRUN ||
+ snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
+ xrun("play");
+ }
+ }
+#endif
+
+ outbuffer = alloca(ao_data.buffersize);
+
+ //don't trust get_space() ;)
+ frames_available = snd_pcm_avail_update(alsa_handler) * bytes_per_sample;
+ if (frames_available < 0)
+ xrun("play");
+
+ if (frames_available < 4) {
+ if (first) {
+ first = 0;
+ snd_pcm_start(alsa_handler);
+ }
+ else { //FIXME should break and return 0?
+ snd_pcm_wait(alsa_handler, -1);
+ first = 1;
+ }
+ }
+
+ /* len is simply the available bufferspace got by get_space()
+ * but real avail_buffer in frames is ab/bytes_per_sample */
+ size = len / bytes_per_sample;
+
+ //mp_msg(MSGT_AO,MSGL_V,"len: %i size %i, f_avail %i, bps %i ...\n", len, size, frames_available, bytes_per_sample);
+
+ frames_transmit = size;
+
+ /* prepare areas and set sw-pointers
+ * frames_transmit returns the real available buffer-size
+ * sometimes != frames_available cause of ringbuffer 'emulation' */
+ snd_pcm_mmap_begin(alsa_handler, &area, &offset, &frames_transmit);
+
+ /* this is specific to interleaved streams (or non-interleaved
+ * streams with only one channel) */
+ outbuffer = ((char *) area->addr + (area->first + area->step * offset) / 8); //8
+
+ //write data
+ memcpy(outbuffer, data, (frames_transmit * bytes_per_sample));
+
+ commitres = snd_pcm_mmap_commit(alsa_handler, offset, frames_transmit);
+
+ if (commitres < 0 || commitres != frames_transmit) {
+ if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_XRUN ||
+ snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
+ xrun("play");
+ }
+ }
+
+ //mp_msg(MSGT_AO,MSGL_V,"mmap ft: %i, cres: %i\n", frames_transmit, commitres);
+
+ /* err = snd_pcm_area_copy(&area, offset, &data, offset, len, alsa_format); */
+ /* if (err < 0) { */
+ /* mp_msg(MSGT_AO,MSGL_ERR,"area-copy-error\n"); */
+ /* return 0; */
+ /* } */
+
+
+ //calculate written frames!
+ result = commitres * bytes_per_sample;
+
+
+ /* if (verbose) { */
+ /* if (len == result) */
+ /* mp_msg(MSGT_AO,MSGL_V,"result: %i, frames written: %i ...\n", result, frames_transmit); */
+ /* else */
+ /* mp_msg(MSGT_AO,MSGL_V,"result: %i, frames written: %i, result != len ...\n", result, frames_transmit); */
+ /* } */
+
+ //mplayer doesn't like -result
+ if (result < 0)
+ result = 0;
+
+#ifdef USE_POLL
+ free(ufds);
+#endif
+
+ return result;
+}
+
+/* how many byes are free in the buffer */
+static int get_space()
+{
+ snd_pcm_status_t *status;
+ int ret;
+ char *str_status;
+
+ //snd_pcm_sframes_t avail_frames = 0;
+
+ if ((ret = snd_pcm_status_malloc(&status)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-space: memory allocation error: %s\n", snd_strerror(ret));
+ return(0);
+ }
+
+ if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-space: cannot get pcm status: %s\n", snd_strerror(ret));
+ return(0);
+ }
+
+ switch(snd_pcm_status_get_state(status))
+ {
+ case SND_PCM_STATE_OPEN:
+ str_status = "open";
+ case SND_PCM_STATE_PREPARED:
+ if (str_status != "open") {
+ str_status = "prepared";
+ first = 1;
+ ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
+ if (ret == 0) //ugly workaround for hang in mmap-mode
+ ret = 10;
+ break;
+ }
+ case SND_PCM_STATE_RUNNING:
+ ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
+ //avail_frames = snd_pcm_avail_update(alsa_handler) * bytes_per_sample;
+ if (str_status != "open" && str_status != "prepared")
+ str_status = "running";
+ break;
+ case SND_PCM_STATE_PAUSED:
+ mp_msg(MSGT_AO,MSGL_V,"alsa-space: paused");
+ str_status = "paused";
+ ret = 0;
+ break;
+ case SND_PCM_STATE_XRUN:
+ xrun("space");
+ str_status = "xrun";
+ first = 1;
+ ret = 0;
+ break;
+ default:
+ str_status = "undefined";
+ ret = snd_pcm_status_get_avail(status) * bytes_per_sample;
+ if (ret <= 0) {
+ xrun("space");
+ }
+ }
+
+ if (str_status != "running")
+ mp_msg(MSGT_AO,MSGL_V,"alsa-space: free space = %i, status=%i, %s --\n", ret, status, str_status);
+ snd_pcm_status_free(status);
+
+ if (ret < 0) {
+ mp_msg(MSGT_AO,MSGL_ERR,"negative value!!\n");
+ ret = 0;
+ }
+
+ // workaround for too small value returned
+ if (ret < MIN_CHUNK_SIZE)
+ ret = 0;
+
+ return(ret);
+}
+
+/* delay in seconds between first and last sample in buffer */
+static float get_delay()
+{
+
+ if (alsa_handler) {
+
+ snd_pcm_status_t *status;
+ float ret;
+
+ if ((ret = snd_pcm_status_malloc(&status)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-delay: memory allocation error: %s\n", snd_strerror(ret));
+ }
+
+ if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
+ {
+ mp_msg(MSGT_AO,MSGL_ERR,"alsa-delay: cannot get pcm status: %s\n", snd_strerror(ret));
+ }
+
+ switch(snd_pcm_status_get_state(status))
+ {
+ case SND_PCM_STATE_OPEN:
+ case SND_PCM_STATE_PREPARED:
+ case SND_PCM_STATE_RUNNING:
+ ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
+ break;
+ default:
+ ret = 0;
+ }
+
+ snd_pcm_status_free(status);
+
+ if (ret < 0)
+ ret = 0;
+ return(ret);
+
+ } else {
+ return(0);
+ }
+}
diff -Nur -x CVS -x DOC -x '.*' -x '*~' clean_cvs/libao2/audio_out.c src-merge2/libao2/audio_out.c
--- clean_cvs/libao2/audio_out.c 2004-01-11 18:07:32.000000000 +0100
+++ src-merge2/libao2/audio_out.c 2004-04-28 17:16:55.000000000 +0200
@@ -27,10 +27,10 @@
extern ao_functions_t audio_out_alsa5;
#endif
#ifdef HAVE_ALSA9
- extern ao_functions_t audio_out_alsa9;
+ extern ao_functions_t audio_out_alsa;
#endif
#ifdef HAVE_ALSA1X
- extern ao_functions_t audio_out_alsa1x;
+ extern ao_functions_t audio_out_alsa;
#endif
#ifdef HAVE_NAS
extern ao_functions_t audio_out_nas;
@@ -76,10 +76,10 @@
&audio_out_oss,
#endif
#ifdef HAVE_ALSA1X
- &audio_out_alsa1x,
+ &audio_out_alsa,
#endif
#ifdef HAVE_ALSA9
- &audio_out_alsa9,
+ &audio_out_alsa,
#endif
#ifdef HAVE_ALSA5
&audio_out_alsa5,
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