mirror of
https://github.com/mpv-player/mpv
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00323c06e2
Remove the help/ subdirectory, configure code to create toplevel help_mp.h, and all the '#include "help_mp.h"' lines from .c files.
864 lines
25 KiB
C
864 lines
25 KiB
C
/*
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* ALSA 0.9.x-1.x audio output driver
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*
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* Copyright (C) 2004 Alex Beregszaszi
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*
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* modified for real ALSA 0.9.0 support by Zsolt Barat <joy@streamminister.de>
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* additional AC-3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org>
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* 08/22/2002 iec958-init rewritten and merged with common init, zsolt
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* 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka
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* 04/25/2004 printfs converted to mp_msg, Zsolt.
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <errno.h>
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#include <sys/time.h>
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#include <stdlib.h>
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#include <stdarg.h>
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#include <ctype.h>
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#include <math.h>
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#include <string.h>
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#include <alloca.h>
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#include "config.h"
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#include "subopt-helper.h"
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#include "mixer.h"
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#include "mp_msg.h"
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#define ALSA_PCM_NEW_SW_PARAMS_API
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#ifdef HAVE_SYS_ASOUNDLIB_H
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#include <sys/asoundlib.h>
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#elif defined(HAVE_ALSA_ASOUNDLIB_H)
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#include <alsa/asoundlib.h>
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#else
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#error "asoundlib.h is not in sys/ or alsa/ - please bugreport"
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#endif
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "libaf/af_format.h"
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static const ao_info_t info =
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{
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"ALSA-0.9.x-1.x audio output",
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"alsa",
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"Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>",
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"under developement"
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};
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LIBAO_EXTERN(alsa)
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static snd_pcm_t *alsa_handler;
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static snd_pcm_format_t alsa_format;
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static snd_pcm_hw_params_t *alsa_hwparams;
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static snd_pcm_sw_params_t *alsa_swparams;
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#define BUFFER_TIME 500000 // 0.5 s
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#define FRAGCOUNT 16
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static size_t bytes_per_sample;
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static int alsa_can_pause;
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static snd_pcm_sframes_t prepause_frames;
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#define ALSA_DEVICE_SIZE 256
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static void alsa_error_handler(const char *file, int line, const char *function,
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int err, const char *format, ...)
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{
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char tmp[0xc00];
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va_list va;
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va_start(va, format);
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vsnprintf(tmp, sizeof tmp, format, va);
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va_end(va);
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tmp[sizeof tmp - 1] = '\0';
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if (err)
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mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n",
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file, line, function, tmp, snd_strerror(err));
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else
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mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n",
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file, line, function, tmp);
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}
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/* to set/get/query special features/parameters */
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static int control(int cmd, void *arg)
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{
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switch(cmd) {
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case AOCONTROL_QUERY_FORMAT:
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return CONTROL_TRUE;
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case AOCONTROL_GET_VOLUME:
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case AOCONTROL_SET_VOLUME:
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{
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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int err;
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snd_mixer_t *handle;
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snd_mixer_elem_t *elem;
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snd_mixer_selem_id_t *sid;
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char *mix_name = "PCM";
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char *card = "default";
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int mix_index = 0;
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long pmin, pmax;
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long get_vol, set_vol;
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float f_multi;
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if(AF_FORMAT_IS_AC3(ao_data.format))
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return CONTROL_TRUE;
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if(mixer_channel) {
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char *test_mix_index;
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mix_name = strdup(mixer_channel);
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if ((test_mix_index = strchr(mix_name, ','))){
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*test_mix_index = 0;
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test_mix_index++;
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mix_index = strtol(test_mix_index, &test_mix_index, 0);
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if (*test_mix_index){
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mp_tmsg(MSGT_AO,MSGL_ERR,
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"[AO_ALSA] Invalid mixer index. Defaulting to 0.\n");
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mix_index = 0 ;
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}
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}
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}
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if(mixer_device) card = mixer_device;
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//allocate simple id
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snd_mixer_selem_id_alloca(&sid);
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//sets simple-mixer index and name
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snd_mixer_selem_id_set_index(sid, mix_index);
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snd_mixer_selem_id_set_name(sid, mix_name);
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if (mixer_channel) {
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free(mix_name);
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mix_name = NULL;
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}
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if ((err = snd_mixer_open(&handle, 0)) < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer open error: %s\n", snd_strerror(err));
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return CONTROL_ERROR;
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}
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if ((err = snd_mixer_attach(handle, card)) < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
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card, snd_strerror(err));
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer register error: %s\n", snd_strerror(err));
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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err = snd_mixer_load(handle);
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if (err < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer load error: %s\n", snd_strerror(err));
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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elem = snd_mixer_find_selem(handle, sid);
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if (!elem) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to find simple control '%s',%i.\n",
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snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid));
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax);
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f_multi = (100 / (float)(pmax - pmin));
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if (cmd == AOCONTROL_SET_VOLUME) {
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set_vol = vol->left / f_multi + pmin + 0.5;
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//setting channels
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if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
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snd_strerror(err));
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol);
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set_vol = vol->right / f_multi + pmin + 0.5;
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if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
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snd_strerror(err));
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
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set_vol, pmin, pmax, f_multi);
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if (snd_mixer_selem_has_playback_switch(elem)) {
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int lmute = (vol->left == 0.0);
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int rmute = (vol->right == 0.0);
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if (snd_mixer_selem_has_playback_switch_joined(elem)) {
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lmute = rmute = lmute && rmute;
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} else {
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snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute);
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}
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snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute);
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}
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}
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else {
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snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol);
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vol->left = (get_vol - pmin) * f_multi;
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snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
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vol->right = (get_vol - pmin) * f_multi;
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mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right);
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}
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snd_mixer_close(handle);
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return CONTROL_OK;
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}
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} //end switch
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return CONTROL_UNKNOWN;
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}
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static void parse_device (char *dest, const char *src, int len)
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{
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char *tmp;
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memmove(dest, src, len);
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dest[len] = 0;
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while ((tmp = strrchr(dest, '.')))
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tmp[0] = ',';
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while ((tmp = strrchr(dest, '=')))
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tmp[0] = ':';
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}
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static void print_help (void)
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{
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mp_tmsg (MSGT_AO, MSGL_FATAL,
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"\n[AO_ALSA] -ao alsa commandline help:\n"\
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"[AO_ALSA] Example: mplayer -ao alsa:device=hw=0.3\n"\
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"[AO_ALSA] Sets first card fourth hardware device.\n\n"\
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"[AO_ALSA] Options:\n"\
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"[AO_ALSA] noblock\n"\
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"[AO_ALSA] Opens device in non-blocking mode.\n"\
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"[AO_ALSA] device=<device-name>\n"\
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"[AO_ALSA] Sets device (change , to . and : to =)\n");
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}
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static int str_maxlen(void *strp) {
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strarg_t *str = strp;
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return str->len <= ALSA_DEVICE_SIZE;
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}
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static int try_open_device(const char *device, int open_mode, int try_ac3)
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{
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int err, len;
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char *ac3_device, *args;
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if (try_ac3) {
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/* to set the non-audio bit, use AES0=6 */
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len = strlen(device);
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ac3_device = malloc(len + 7 + 1);
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if (!ac3_device)
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return -ENOMEM;
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strcpy(ac3_device, device);
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args = strchr(ac3_device, ':');
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if (!args) {
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/* no existing parameters: add it behind device name */
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strcat(ac3_device, ":AES0=6");
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} else {
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do
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++args;
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while (isspace(*args));
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if (*args == '\0') {
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/* ":" but no parameters */
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strcat(ac3_device, "AES0=6");
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} else if (*args != '{') {
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/* a simple list of parameters: add it at the end of the list */
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strcat(ac3_device, ",AES0=6");
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} else {
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/* parameters in config syntax: add it inside the { } block */
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do
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--len;
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while (len > 0 && isspace(ac3_device[len]));
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if (ac3_device[len] == '}')
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strcpy(ac3_device + len, " AES0=6}");
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}
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}
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err = snd_pcm_open(&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK,
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open_mode);
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free(ac3_device);
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}
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if (!try_ac3 || err < 0)
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err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK,
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open_mode);
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return err;
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}
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/*
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open & setup audio device
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return: 1=success 0=fail
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*/
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static int init(int rate_hz, int channels, int format, int flags)
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{
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int err;
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int block;
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strarg_t device;
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snd_pcm_uframes_t chunk_size;
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snd_pcm_uframes_t bufsize;
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snd_pcm_uframes_t boundary;
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const opt_t subopts[] = {
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{"block", OPT_ARG_BOOL, &block, NULL},
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{"device", OPT_ARG_STR, &device, str_maxlen},
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{NULL}
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};
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char alsa_device[ALSA_DEVICE_SIZE + 1];
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// make sure alsa_device is null-terminated even when using strncpy etc.
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memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1);
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %x\n", rate_hz,
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channels, format);
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alsa_handler = NULL;
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#if SND_LIB_VERSION >= 0x010005
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version());
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#else
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR);
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#endif
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prepause_frames = 0;
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snd_lib_error_set_handler(alsa_error_handler);
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ao_data.samplerate = rate_hz;
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ao_data.format = format;
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ao_data.channels = channels;
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switch (format)
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{
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case AF_FORMAT_S8:
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alsa_format = SND_PCM_FORMAT_S8;
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break;
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case AF_FORMAT_U8:
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alsa_format = SND_PCM_FORMAT_U8;
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break;
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case AF_FORMAT_U16_LE:
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alsa_format = SND_PCM_FORMAT_U16_LE;
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break;
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case AF_FORMAT_U16_BE:
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alsa_format = SND_PCM_FORMAT_U16_BE;
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break;
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case AF_FORMAT_AC3_LE:
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case AF_FORMAT_S16_LE:
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alsa_format = SND_PCM_FORMAT_S16_LE;
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break;
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case AF_FORMAT_AC3_BE:
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case AF_FORMAT_S16_BE:
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alsa_format = SND_PCM_FORMAT_S16_BE;
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break;
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case AF_FORMAT_U32_LE:
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alsa_format = SND_PCM_FORMAT_U32_LE;
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break;
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case AF_FORMAT_U32_BE:
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alsa_format = SND_PCM_FORMAT_U32_BE;
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break;
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case AF_FORMAT_S32_LE:
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alsa_format = SND_PCM_FORMAT_S32_LE;
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break;
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case AF_FORMAT_S32_BE:
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alsa_format = SND_PCM_FORMAT_S32_BE;
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break;
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case AF_FORMAT_U24_LE:
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alsa_format = SND_PCM_FORMAT_U24_3LE;
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break;
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case AF_FORMAT_U24_BE:
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alsa_format = SND_PCM_FORMAT_U24_3BE;
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break;
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case AF_FORMAT_S24_LE:
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alsa_format = SND_PCM_FORMAT_S24_3LE;
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break;
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case AF_FORMAT_S24_BE:
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alsa_format = SND_PCM_FORMAT_S24_3BE;
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break;
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case AF_FORMAT_FLOAT_LE:
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alsa_format = SND_PCM_FORMAT_FLOAT_LE;
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break;
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case AF_FORMAT_FLOAT_BE:
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alsa_format = SND_PCM_FORMAT_FLOAT_BE;
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break;
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case AF_FORMAT_MU_LAW:
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alsa_format = SND_PCM_FORMAT_MU_LAW;
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break;
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case AF_FORMAT_A_LAW:
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alsa_format = SND_PCM_FORMAT_A_LAW;
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break;
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default:
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alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
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break;
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}
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//subdevice parsing
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// set defaults
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block = 1;
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/* switch for spdif
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* sets opening sequence for SPDIF
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* sets also the playback and other switches 'on the fly'
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* while opening the abstract alias for the spdif subdevice
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* 'iec958'
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*/
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if (AF_FORMAT_IS_AC3(format)) {
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device.str = "iec958";
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mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels);
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}
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else
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/* in any case for multichannel playback we should select
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* appropriate device
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*/
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switch (channels) {
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case 1:
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case 2:
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device.str = "default";
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n");
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break;
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case 4:
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if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
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// hack - use the converter plugin
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device.str = "plug:surround40";
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else
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device.str = "surround40";
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n");
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break;
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case 6:
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if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
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device.str = "plug:surround51";
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else
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device.str = "surround51";
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n");
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break;
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case 8:
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if (alsa_format == SND_PCM_FORMAT_FLOAT_LE)
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device.str = "plug:surround71";
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else
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device.str = "surround71";
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mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround71\n");
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break;
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default:
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device.str = "default";
|
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mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] %d channels are not supported.\n",channels);
|
|
}
|
|
device.len = strlen(device.str);
|
|
if (subopt_parse(ao_subdevice, subopts) != 0) {
|
|
print_help();
|
|
return 0;
|
|
}
|
|
parse_device(alsa_device, device.str, device.len);
|
|
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa-init: using device %s\n", alsa_device);
|
|
|
|
if (!alsa_handler) {
|
|
int open_mode = block ? 0 : SND_PCM_NONBLOCK;
|
|
int isac3 = AF_FORMAT_IS_AC3(format);
|
|
//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
|
|
if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0)
|
|
{
|
|
if (err != -EBUSY && !block) {
|
|
mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Open in nonblock-mode failed, trying to open in block-mode.\n");
|
|
if ((err = try_open_device(alsa_device, 0, isac3)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
|
|
return 0;
|
|
}
|
|
} else {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Playback open error: %s\n", snd_strerror(err));
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err));
|
|
} else {
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n");
|
|
}
|
|
|
|
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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get initial parameters: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
|
|
SND_PCM_ACCESS_RW_INTERLEAVED);
|
|
if (err < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_INFO,
|
|
"[AO_ALSA] Format %s is not supported by hardware, trying default.\n", af_fmt2str_short(format));
|
|
alsa_format = SND_PCM_FORMAT_S16_LE;
|
|
if (AF_FORMAT_IS_AC3(ao_data.format))
|
|
ao_data.format = AF_FORMAT_AC3_LE;
|
|
else
|
|
ao_data.format = AF_FORMAT_S16_LE;
|
|
}
|
|
|
|
if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams,
|
|
alsa_format)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set format: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
|
|
if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams,
|
|
&ao_data.channels)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set channels: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
|
|
/* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11)
|
|
prefer our own resampler, since that allows users to choose the resampler,
|
|
even per file if desired */
|
|
#if SND_LIB_VERSION >= 0x010009
|
|
if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams,
|
|
0)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to disable resampling: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
|
|
&ao_data.samplerate, NULL)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
|
|
bytes_per_sample = af_fmt2bits(ao_data.format) / 8;
|
|
bytes_per_sample *= ao_data.channels;
|
|
ao_data.bps = ao_data.samplerate * bytes_per_sample;
|
|
|
|
if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
|
|
&(unsigned int){BUFFER_TIME}, NULL)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
|
|
if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
|
|
&(unsigned int){FRAGCOUNT}, NULL)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
|
|
/* finally install hardware parameters */
|
|
if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set hw-parameters: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
// end setting hw-params
|
|
|
|
|
|
// gets buffersize for control
|
|
if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get buffersize: %s\n", snd_strerror(err));
|
|
return 0;
|
|
}
|
|
else {
|
|
ao_data.buffersize = bufsize * bytes_per_sample;
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize);
|
|
}
|
|
|
|
if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to get period size: %s\n", snd_strerror(err));
|
|
return 0;
|
|
} else {
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa-init: got period size %li\n", chunk_size);
|
|
}
|
|
ao_data.outburst = chunk_size * bytes_per_sample;
|
|
|
|
/* setting software parameters */
|
|
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
#if SND_LIB_VERSION >= 0x000901
|
|
if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get boundary: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
#else
|
|
boundary = 0x7fffffff;
|
|
#endif
|
|
/* start playing when one period has been written */
|
|
if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set start threshold: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
/* disable underrun reporting */
|
|
if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set stop threshold: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
#if SND_LIB_VERSION >= 0x000901
|
|
/* play silence when there is an underrun */
|
|
if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set silence size: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
#endif
|
|
if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to get sw-parameters: %s\n",
|
|
snd_strerror(err));
|
|
return 0;
|
|
}
|
|
/* end setting sw-params */
|
|
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
|
|
ao_data.samplerate, ao_data.channels, (int)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 (!immed)
|
|
snd_pcm_drain(alsa_handler);
|
|
|
|
if ((err = snd_pcm_close(alsa_handler)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm close error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
else {
|
|
alsa_handler = NULL;
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n");
|
|
}
|
|
}
|
|
else {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] No handler defined!\n");
|
|
}
|
|
}
|
|
|
|
static void audio_pause(void)
|
|
{
|
|
int err;
|
|
|
|
if (alsa_can_pause) {
|
|
if ((err = snd_pcm_pause(alsa_handler, 1)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm pause error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n");
|
|
} else {
|
|
if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0
|
|
|| prepause_frames < 0)
|
|
prepause_frames = 0;
|
|
|
|
if ((err = snd_pcm_drop(alsa_handler)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm drop error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
static void audio_resume(void)
|
|
{
|
|
int err;
|
|
|
|
if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) {
|
|
mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
|
|
while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1);
|
|
}
|
|
if (alsa_can_pause) {
|
|
if ((err = snd_pcm_pause(alsa_handler, 0)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] 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_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
if (prepause_frames) {
|
|
void *silence = calloc(prepause_frames, bytes_per_sample);
|
|
play(silence, prepause_frames * bytes_per_sample, 0);
|
|
free(silence);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* stop playing and empty buffers (for seeking/pause) */
|
|
static void reset(void)
|
|
{
|
|
int err;
|
|
|
|
prepause_frames = 0;
|
|
if ((err = snd_pcm_drop(alsa_handler)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
if ((err = snd_pcm_prepare(alsa_handler)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
return;
|
|
}
|
|
|
|
/*
|
|
plays 'len' bytes of 'data'
|
|
returns: number of bytes played
|
|
modified last at 29.06.02 by jp
|
|
thanxs for marius <marius@rospot.com> for giving us the light ;)
|
|
*/
|
|
|
|
static int play(void* data, int len, int flags)
|
|
{
|
|
int num_frames;
|
|
snd_pcm_sframes_t res = 0;
|
|
if (!(flags & AOPLAY_FINAL_CHUNK))
|
|
len = len / ao_data.outburst * ao_data.outburst;
|
|
num_frames = len / bytes_per_sample;
|
|
|
|
//mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len);
|
|
|
|
if (!alsa_handler) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Device configuration error.");
|
|
return 0;
|
|
}
|
|
|
|
if (num_frames == 0)
|
|
return 0;
|
|
|
|
do {
|
|
res = snd_pcm_writei(alsa_handler, data, num_frames);
|
|
|
|
if (res == -EINTR) {
|
|
/* nothing to do */
|
|
res = 0;
|
|
}
|
|
else if (res == -ESTRPIPE) { /* suspend */
|
|
mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Pcm in suspend mode, trying to resume.\n");
|
|
while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN)
|
|
sleep(1);
|
|
}
|
|
if (res < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Write error: %s\n", snd_strerror(res));
|
|
mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] Trying to reset soundcard.\n");
|
|
if ((res = snd_pcm_prepare(alsa_handler)) < 0) {
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] pcm prepare error: %s\n", snd_strerror(res));
|
|
return 0;
|
|
break;
|
|
}
|
|
}
|
|
} while (res == 0);
|
|
|
|
return res < 0 ? res : res * bytes_per_sample;
|
|
}
|
|
|
|
/* how many byes are free in the buffer */
|
|
static int get_space(void)
|
|
{
|
|
snd_pcm_status_t *status;
|
|
int ret;
|
|
|
|
snd_pcm_status_alloca(&status);
|
|
|
|
if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
|
|
{
|
|
mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
|
|
return 0;
|
|
}
|
|
|
|
unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
|
|
if (space > ao_data.buffersize) // Buffer underrun?
|
|
space = ao_data.buffersize;
|
|
return space;
|
|
}
|
|
|
|
/* delay in seconds between first and last sample in buffer */
|
|
static float get_delay(void)
|
|
{
|
|
if (alsa_handler) {
|
|
snd_pcm_sframes_t delay;
|
|
|
|
if (snd_pcm_delay(alsa_handler, &delay) < 0)
|
|
return 0;
|
|
|
|
if (delay < 0) {
|
|
/* underrun - move the application pointer forward to catch up */
|
|
#if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
|
|
snd_pcm_forward(alsa_handler, -delay);
|
|
#endif
|
|
delay = 0;
|
|
}
|
|
return (float)delay / (float)ao_data.samplerate;
|
|
} else {
|
|
return 0;
|
|
}
|
|
}
|