mirror of https://github.com/mpv-player/mpv
732 lines
23 KiB
C
732 lines
23 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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* Zsolt Barat <joy@streamminister.de>
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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 "config.h"
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#include "options/options.h"
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#include "options/m_option.h"
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#include "common/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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#include <alsa/asoundlib.h>
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#include "ao.h"
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#include "internal.h"
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#include "audio/format.h"
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#include "audio/reorder_ch.h"
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struct priv {
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snd_pcm_t *alsa;
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snd_pcm_format_t alsa_fmt;
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int can_pause;
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snd_pcm_sframes_t prepause_frames;
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float delay_before_pause;
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int buffersize; // in frames
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int outburst; // in frames
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int cfg_block;
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char *cfg_device;
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char *cfg_mixer_device;
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char *cfg_mixer_name;
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int cfg_mixer_index;
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int cfg_resample;
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};
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#define BUFFER_TIME 500000 // 0.5 s
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#define FRAGCOUNT 16
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#define CHECK_ALSA_ERROR(message) \
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do { \
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if (err < 0) { \
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MP_ERR(ao, "%s: %s\n", (message), snd_strerror(err)); \
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goto alsa_error; \
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} \
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} while (0)
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static float get_delay(struct ao *ao);
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static void uninit(struct ao *ao, bool immed);
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/* to set/get/query special features/parameters */
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static int control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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struct priv *p = ao->priv;
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snd_mixer_t *handle = NULL;
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switch (cmd) {
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case AOCONTROL_GET_MUTE:
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case AOCONTROL_SET_MUTE:
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case AOCONTROL_GET_VOLUME:
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case AOCONTROL_SET_VOLUME:
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{
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int err;
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snd_mixer_elem_t *elem;
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snd_mixer_selem_id_t *sid;
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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_IEC61937(ao->format))
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return CONTROL_TRUE;
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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, p->cfg_mixer_index);
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snd_mixer_selem_id_set_name(sid, p->cfg_mixer_name);
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err = snd_mixer_open(&handle, 0);
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CHECK_ALSA_ERROR("Mixer open error");
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err = snd_mixer_attach(handle, p->cfg_mixer_device);
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CHECK_ALSA_ERROR("Mixer attach error");
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err = snd_mixer_selem_register(handle, NULL, NULL);
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CHECK_ALSA_ERROR("Mixer register error");
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err = snd_mixer_load(handle);
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CHECK_ALSA_ERROR("Mixer load error");
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elem = snd_mixer_find_selem(handle, sid);
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if (!elem) {
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MP_VERBOSE(ao, "Unable to find simple control '%s',%i.\n",
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snd_mixer_selem_id_get_name(sid),
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snd_mixer_selem_id_get_index(sid));
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goto alsa_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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switch (cmd) {
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case AOCONTROL_SET_VOLUME: {
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ao_control_vol_t *vol = arg;
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set_vol = vol->left / f_multi + pmin + 0.5;
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//setting channels
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err = snd_mixer_selem_set_playback_volume
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(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol);
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CHECK_ALSA_ERROR("Error setting left channel");
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MP_DBG(ao, "left=%li, ", set_vol);
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set_vol = vol->right / f_multi + pmin + 0.5;
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err = snd_mixer_selem_set_playback_volume
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(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol);
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CHECK_ALSA_ERROR("Error setting right channel");
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MP_DBG(ao, "right=%li, pmin=%li, pmax=%li, mult=%f\n",
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set_vol, pmin, pmax,
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f_multi);
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break;
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}
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case AOCONTROL_GET_VOLUME: {
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ao_control_vol_t *vol = arg;
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snd_mixer_selem_get_playback_volume
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(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
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(elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol);
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vol->right = (get_vol - pmin) * f_multi;
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MP_DBG(ao, "left=%f, right=%f\n", vol->left, vol->right);
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break;
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}
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case AOCONTROL_SET_MUTE: {
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bool *mute = arg;
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if (!snd_mixer_selem_has_playback_switch(elem))
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goto alsa_error;
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if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
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snd_mixer_selem_set_playback_switch
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(elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute);
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}
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snd_mixer_selem_set_playback_switch
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(elem, SND_MIXER_SCHN_FRONT_LEFT, !*mute);
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break;
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}
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case AOCONTROL_GET_MUTE: {
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bool *mute = arg;
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if (!snd_mixer_selem_has_playback_switch(elem))
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goto alsa_error;
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int tmp = 1;
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snd_mixer_selem_get_playback_switch
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(elem, SND_MIXER_SCHN_FRONT_LEFT, &tmp);
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*mute = !tmp;
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if (!snd_mixer_selem_has_playback_switch_joined(elem)) {
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snd_mixer_selem_get_playback_switch
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(elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp);
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*mute &= !tmp;
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}
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break;
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}
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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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alsa_error:
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if (handle)
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snd_mixer_close(handle);
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return CONTROL_ERROR;
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}
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static const int mp_to_alsa_format[][2] = {
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{AF_FORMAT_S8, SND_PCM_FORMAT_S8},
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{AF_FORMAT_U8, SND_PCM_FORMAT_U8},
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{AF_FORMAT_U16_LE, SND_PCM_FORMAT_U16_LE},
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{AF_FORMAT_U16_BE, SND_PCM_FORMAT_U16_BE},
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{AF_FORMAT_S16_LE, SND_PCM_FORMAT_S16_LE},
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{AF_FORMAT_S16_BE, SND_PCM_FORMAT_S16_BE},
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{AF_FORMAT_U32_LE, SND_PCM_FORMAT_U32_LE},
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{AF_FORMAT_U32_BE, SND_PCM_FORMAT_U32_BE},
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{AF_FORMAT_S32_LE, SND_PCM_FORMAT_S32_LE},
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{AF_FORMAT_S32_BE, SND_PCM_FORMAT_S32_BE},
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{AF_FORMAT_U24_LE, SND_PCM_FORMAT_U24_3LE},
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{AF_FORMAT_U24_BE, SND_PCM_FORMAT_U24_3BE},
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{AF_FORMAT_S24_LE, SND_PCM_FORMAT_S24_3LE},
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{AF_FORMAT_S24_BE, SND_PCM_FORMAT_S24_3BE},
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{AF_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_LE},
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{AF_FORMAT_FLOAT_BE, SND_PCM_FORMAT_FLOAT_BE},
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{AF_FORMAT_AC3_LE, SND_PCM_FORMAT_S16_LE},
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{AF_FORMAT_AC3_BE, SND_PCM_FORMAT_S16_BE},
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{AF_FORMAT_IEC61937_LE, SND_PCM_FORMAT_S16_LE},
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{AF_FORMAT_IEC61937_BE, SND_PCM_FORMAT_S16_BE},
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{AF_FORMAT_MPEG2, SND_PCM_FORMAT_MPEG},
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{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
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};
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static int find_alsa_format(int af_format)
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{
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af_format = af_fmt_from_planar(af_format);
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for (int n = 0; mp_to_alsa_format[n][0] != AF_FORMAT_UNKNOWN; n++) {
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if (mp_to_alsa_format[n][0] == af_format)
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return mp_to_alsa_format[n][1];
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}
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return SND_PCM_FORMAT_UNKNOWN;
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}
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// Lists device names and their implied channel map.
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// The second item must be resolvable with mp_chmap_from_str().
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// Source: http://www.alsa-project.org/main/index.php/DeviceNames
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// (Speaker names are slightly different from mpv's.)
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static const char *device_channel_layouts[][2] = {
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{"default", "fc"},
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{"default", "fl-fr"},
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{"rear", "bl-br"},
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{"center_lfe", "fc-lfe"},
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{"side", "sl-sr"},
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{"surround40", "fl-fr-bl-br"},
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{"surround50", "fl-fr-bl-br-fc"},
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{"surround41", "fl-fr-bl-br-lfe"},
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{"surround51", "fl-fr-bl-br-fc-lfe"},
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{"surround71", "fl-fr-bl-br-fc-lfe-sl-sr"},
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};
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#define ARRAY_LEN(x) (sizeof(x) / sizeof((x)[0]))
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#define NUM_ALSA_CHMAPS ARRAY_LEN(device_channel_layouts)
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static const char *select_chmap(struct ao *ao)
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{
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struct mp_chmap_sel sel = {0};
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struct mp_chmap maps[NUM_ALSA_CHMAPS];
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for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
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mp_chmap_from_str(&maps[n], bstr0(device_channel_layouts[n][1]));
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mp_chmap_sel_add_map(&sel, &maps[n]);
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};
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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return NULL;
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for (int n = 0; n < NUM_ALSA_CHMAPS; n++) {
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if (mp_chmap_equals(&ao->channels, &maps[n]))
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return device_channel_layouts[n][0];
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}
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char *name = mp_chmap_to_str(&ao->channels);
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MP_ERR(ao, "channel layout %s (%d ch) not supported.\n",
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name, ao->channels.num);
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talloc_free(name);
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return "default";
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}
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static int map_iec958_srate(int srate)
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{
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switch (srate) {
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case 44100: return IEC958_AES3_CON_FS_44100;
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case 48000: return IEC958_AES3_CON_FS_48000;
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case 32000: return IEC958_AES3_CON_FS_32000;
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case 22050: return IEC958_AES3_CON_FS_22050;
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case 24000: return IEC958_AES3_CON_FS_24000;
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case 88200: return IEC958_AES3_CON_FS_88200;
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case 768000: return IEC958_AES3_CON_FS_768000;
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case 96000: return IEC958_AES3_CON_FS_96000;
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case 176400: return IEC958_AES3_CON_FS_176400;
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case 192000: return IEC958_AES3_CON_FS_192000;
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default: return IEC958_AES3_CON_FS_NOTID;
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}
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}
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static int try_open_device(struct ao *ao, const char *device, int open_mode)
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{
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struct priv *p = ao->priv;
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if (AF_FORMAT_IS_IEC61937(ao->format)) {
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void *tmp = talloc_new(NULL);
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/* to set the non-audio bit, use AES0=6 */
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char *params = talloc_asprintf(tmp,
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"AES0=%d,AES1=%d,AES2=0,AES3=%d",
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IEC958_AES0_NONAUDIO | IEC958_AES0_PRO_EMPHASIS_NONE,
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IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER,
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map_iec958_srate(ao->samplerate));
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const char *ac3_device = device;
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int len = strlen(device);
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char *end = strchr(device, ':');
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if (!end) {
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/* no existing parameters: add it behind device name */
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ac3_device = talloc_asprintf(tmp, "%s:%s", device, params);
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} else if (end[1] == '\0') {
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/* ":" but no parameters */
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ac3_device = talloc_asprintf(tmp, "%s%s", device, params);
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} else if (end[1] == '{' && device[len - 1] == '}') {
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/* parameters in config syntax: add it inside the { } block */
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ac3_device = talloc_asprintf(tmp, "%.*s %s}", len - 1, device, params);
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} else {
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/* a simple list of parameters: add it at the end of the list */
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ac3_device = talloc_asprintf(tmp, "%s,%s", device, params);
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}
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int err = snd_pcm_open
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(&p->alsa, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode);
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talloc_free(tmp);
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if (!err)
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return 0;
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}
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return snd_pcm_open(&p->alsa, device, SND_PCM_STREAM_PLAYBACK, open_mode);
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}
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/*
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open & setup audio device
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return: 0=success -1=fail
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*/
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static int init(struct ao *ao)
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{
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int err;
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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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struct priv *p = ao->priv;
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p->prepause_frames = 0;
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p->delay_before_pause = 0;
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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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const char *device;
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if (AF_FORMAT_IS_IEC61937(ao->format)) {
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device = "iec958";
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MP_VERBOSE(ao, "playing AC3/iec61937/iec958, %i channels\n",
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ao->channels.num);
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} else {
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device = select_chmap(ao);
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if (strcmp(device, "default") != 0 && (ao->format & AF_FORMAT_F)) {
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// hack - use the converter plugin (why the heck?)
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device = talloc_asprintf(ao, "plug:%s", device);
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}
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}
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if (p->cfg_device && p->cfg_device[0])
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device = p->cfg_device;
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MP_VERBOSE(ao, "using device: %s\n", device);
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p->can_pause = 1;
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MP_VERBOSE(ao, "using ALSA version: %s\n", snd_asoundlib_version());
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int open_mode = p->cfg_block ? 0 : SND_PCM_NONBLOCK;
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//modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC
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err = try_open_device(ao, device, open_mode);
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if (err < 0) {
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if (err != -EBUSY && !p->cfg_block) {
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MP_WARN(ao, "Open in nonblock-mode "
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"failed, trying to open in block-mode.\n");
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err = try_open_device(ao, device, 0);
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}
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CHECK_ALSA_ERROR("Playback open error");
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}
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err = snd_pcm_nonblock(p->alsa, 0);
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if (err < 0) {
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MP_ERR(ao, "Error setting block-mode: %s.\n", snd_strerror(err));
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} else {
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MP_VERBOSE(ao, "pcm opened in blocking mode\n");
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}
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snd_pcm_hw_params_t *alsa_hwparams;
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snd_pcm_sw_params_t *alsa_swparams;
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snd_pcm_hw_params_alloca(&alsa_hwparams);
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snd_pcm_sw_params_alloca(&alsa_swparams);
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// setting hw-parameters
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err = snd_pcm_hw_params_any(p->alsa, alsa_hwparams);
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CHECK_ALSA_ERROR("Unable to get initial parameters");
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p->alsa_fmt = find_alsa_format(ao->format);
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if (p->alsa_fmt == SND_PCM_FORMAT_UNKNOWN) {
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p->alsa_fmt = SND_PCM_FORMAT_S16;
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ao->format = AF_FORMAT_S16;
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}
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err = snd_pcm_hw_params_test_format(p->alsa, alsa_hwparams, p->alsa_fmt);
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if (err < 0) {
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MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
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af_fmt_to_str(ao->format));
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p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
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if (AF_FORMAT_IS_AC3(ao->format))
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ao->format = AF_FORMAT_AC3_LE;
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else if (AF_FORMAT_IS_IEC61937(ao->format))
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ao->format = AF_FORMAT_IEC61937_LE;
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else
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ao->format = AF_FORMAT_S16_LE;
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}
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err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
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CHECK_ALSA_ERROR("Unable to set format");
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snd_pcm_access_t access = af_fmt_is_planar(ao->format)
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? SND_PCM_ACCESS_RW_NONINTERLEAVED
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: SND_PCM_ACCESS_RW_INTERLEAVED;
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err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
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if (err < 0 && af_fmt_is_planar(ao->format)) {
|
|
ao->format = af_fmt_from_planar(ao->format);
|
|
access = SND_PCM_ACCESS_RW_INTERLEAVED;
|
|
err = snd_pcm_hw_params_set_access(p->alsa, alsa_hwparams, access);
|
|
}
|
|
CHECK_ALSA_ERROR("Unable to set access type");
|
|
|
|
int num_channels = ao->channels.num;
|
|
err = snd_pcm_hw_params_set_channels_near
|
|
(p->alsa, alsa_hwparams, &num_channels);
|
|
CHECK_ALSA_ERROR("Unable to set channels");
|
|
|
|
if (num_channels != ao->channels.num) {
|
|
MP_ERR(ao, "Couldn't get requested number of channels.\n");
|
|
mp_chmap_from_channels_alsa(&ao->channels, num_channels);
|
|
}
|
|
|
|
// Some ALSA drivers have broken delay reporting, so disable the ALSA
|
|
// resampling plugin by default.
|
|
if (!p->cfg_resample) {
|
|
err = snd_pcm_hw_params_set_rate_resample(p->alsa, alsa_hwparams, 0);
|
|
CHECK_ALSA_ERROR("Unable to disable resampling");
|
|
}
|
|
|
|
err = snd_pcm_hw_params_set_rate_near
|
|
(p->alsa, alsa_hwparams, &ao->samplerate, NULL);
|
|
CHECK_ALSA_ERROR("Unable to set samplerate-2");
|
|
|
|
err = snd_pcm_hw_params_set_buffer_time_near
|
|
(p->alsa, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL);
|
|
CHECK_ALSA_ERROR("Unable to set buffer time near");
|
|
|
|
err = snd_pcm_hw_params_set_periods_near
|
|
(p->alsa, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL);
|
|
CHECK_ALSA_ERROR("Unable to set periods");
|
|
|
|
/* finally install hardware parameters */
|
|
err = snd_pcm_hw_params(p->alsa, alsa_hwparams);
|
|
CHECK_ALSA_ERROR("Unable to set hw-parameters");
|
|
|
|
// end setting hw-params
|
|
|
|
// gets buffersize for control
|
|
err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize);
|
|
CHECK_ALSA_ERROR("Unable to get buffersize");
|
|
|
|
p->buffersize = bufsize;
|
|
MP_VERBOSE(ao, "got buffersize=%i samples\n", p->buffersize);
|
|
|
|
err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL);
|
|
CHECK_ALSA_ERROR("Unable to get period size");
|
|
|
|
MP_VERBOSE(ao, "got period size %li\n", chunk_size);
|
|
p->outburst = chunk_size;
|
|
|
|
/* setting software parameters */
|
|
err = snd_pcm_sw_params_current(p->alsa, alsa_swparams);
|
|
CHECK_ALSA_ERROR("Unable to get sw-parameters");
|
|
|
|
err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary);
|
|
CHECK_ALSA_ERROR("Unable to get boundary");
|
|
|
|
/* start playing when one period has been written */
|
|
err = snd_pcm_sw_params_set_start_threshold
|
|
(p->alsa, alsa_swparams, chunk_size);
|
|
CHECK_ALSA_ERROR("Unable to set start threshold");
|
|
|
|
/* disable underrun reporting */
|
|
err = snd_pcm_sw_params_set_stop_threshold
|
|
(p->alsa, alsa_swparams, boundary);
|
|
CHECK_ALSA_ERROR("Unable to set stop threshold");
|
|
|
|
/* play silence when there is an underrun */
|
|
err = snd_pcm_sw_params_set_silence_size
|
|
(p->alsa, alsa_swparams, boundary);
|
|
CHECK_ALSA_ERROR("Unable to set silence size");
|
|
|
|
err = snd_pcm_sw_params(p->alsa, alsa_swparams);
|
|
CHECK_ALSA_ERROR("Unable to get sw-parameters");
|
|
|
|
/* end setting sw-params */
|
|
|
|
p->can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams);
|
|
|
|
MP_VERBOSE(ao, "opened: %d Hz/%d channels/%d bps/%d samples buffer/%s\n",
|
|
ao->samplerate, ao->channels.num, af_fmt2bits(ao->format),
|
|
p->buffersize, snd_pcm_format_description(p->alsa_fmt));
|
|
|
|
return 0;
|
|
|
|
alsa_error:
|
|
uninit(ao, true);
|
|
return -1;
|
|
} // end init
|
|
|
|
|
|
/* close audio device */
|
|
static void uninit(struct ao *ao, bool immed)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
|
|
if (p->alsa) {
|
|
int err;
|
|
|
|
if (!immed)
|
|
snd_pcm_drain(p->alsa);
|
|
|
|
err = snd_pcm_close(p->alsa);
|
|
CHECK_ALSA_ERROR("pcm close error");
|
|
|
|
MP_VERBOSE(ao, "uninit: pcm closed\n");
|
|
}
|
|
|
|
alsa_error:
|
|
p->alsa = NULL;
|
|
}
|
|
|
|
static void audio_pause(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
int err;
|
|
|
|
if (p->can_pause) {
|
|
p->delay_before_pause = get_delay(ao);
|
|
err = snd_pcm_pause(p->alsa, 1);
|
|
CHECK_ALSA_ERROR("pcm pause error");
|
|
} else {
|
|
MP_VERBOSE(ao, "pause not supported by hardware\n");
|
|
if (snd_pcm_delay(p->alsa, &p->prepause_frames) < 0
|
|
|| p->prepause_frames < 0)
|
|
p->prepause_frames = 0;
|
|
p->delay_before_pause = p->prepause_frames / (float)ao->samplerate;
|
|
|
|
err = snd_pcm_drop(p->alsa);
|
|
CHECK_ALSA_ERROR("pcm drop error");
|
|
}
|
|
|
|
alsa_error: ;
|
|
}
|
|
|
|
static void audio_resume(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
int err;
|
|
|
|
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_SUSPENDED) {
|
|
MP_INFO(ao, "PCM in suspend mode, trying to resume.\n");
|
|
|
|
while ((err = snd_pcm_resume(p->alsa)) == -EAGAIN)
|
|
sleep(1);
|
|
}
|
|
|
|
if (p->can_pause) {
|
|
err = snd_pcm_pause(p->alsa, 0);
|
|
CHECK_ALSA_ERROR("pcm resume error");
|
|
} else {
|
|
MP_VERBOSE(ao, "resume not supported by hardware\n");
|
|
err = snd_pcm_prepare(p->alsa);
|
|
CHECK_ALSA_ERROR("pcm prepare error");
|
|
if (p->prepause_frames)
|
|
ao_play_silence(ao, p->prepause_frames);
|
|
}
|
|
|
|
alsa_error: ;
|
|
}
|
|
|
|
/* stop playing and empty buffers (for seeking/pause) */
|
|
static void reset(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
int err;
|
|
|
|
p->prepause_frames = 0;
|
|
p->delay_before_pause = 0;
|
|
err = snd_pcm_drop(p->alsa);
|
|
CHECK_ALSA_ERROR("pcm prepare error");
|
|
err = snd_pcm_prepare(p->alsa);
|
|
CHECK_ALSA_ERROR("pcm prepare error");
|
|
|
|
alsa_error: ;
|
|
}
|
|
|
|
static int play(struct ao *ao, void **data, int samples, int flags)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
snd_pcm_sframes_t res = 0;
|
|
if (!(flags & AOPLAY_FINAL_CHUNK))
|
|
samples = samples / p->outburst * p->outburst;
|
|
|
|
if (samples == 0)
|
|
return 0;
|
|
|
|
do {
|
|
if (af_fmt_is_planar(ao->format)) {
|
|
res = snd_pcm_writen(p->alsa, data, samples);
|
|
} else {
|
|
res = snd_pcm_writei(p->alsa, data[0], samples);
|
|
}
|
|
|
|
if (res == -EINTR) {
|
|
/* nothing to do */
|
|
res = 0;
|
|
} else if (res == -ESTRPIPE) { /* suspend */
|
|
audio_resume(ao);
|
|
} else if (res < 0) {
|
|
MP_ERR(ao, "Write error: %s\n", snd_strerror(res));
|
|
res = snd_pcm_prepare(p->alsa);
|
|
int err = res;
|
|
CHECK_ALSA_ERROR("pcm prepare error");
|
|
res = 0;
|
|
}
|
|
} while (res == 0);
|
|
|
|
return res < 0 ? -1 : res;
|
|
|
|
alsa_error:
|
|
return -1;
|
|
}
|
|
|
|
static int get_space(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
snd_pcm_status_t *status;
|
|
int err;
|
|
|
|
snd_pcm_status_alloca(&status);
|
|
|
|
err = snd_pcm_status(p->alsa, status);
|
|
CHECK_ALSA_ERROR("cannot get pcm status");
|
|
|
|
unsigned space = snd_pcm_status_get_avail(status);
|
|
if (space > p->buffersize) // Buffer underrun?
|
|
space = p->buffersize;
|
|
return space;
|
|
|
|
alsa_error:
|
|
return 0;
|
|
}
|
|
|
|
/* delay in seconds between first and last sample in buffer */
|
|
static float get_delay(struct ao *ao)
|
|
{
|
|
struct priv *p = ao->priv;
|
|
snd_pcm_sframes_t delay;
|
|
|
|
if (snd_pcm_state(p->alsa) == SND_PCM_STATE_PAUSED)
|
|
return p->delay_before_pause;
|
|
|
|
if (snd_pcm_delay(p->alsa, &delay) < 0)
|
|
return 0;
|
|
|
|
if (delay < 0) {
|
|
/* underrun - move the application pointer forward to catch up */
|
|
snd_pcm_forward(p->alsa, -delay);
|
|
delay = 0;
|
|
}
|
|
return (float)delay / (float)ao->samplerate;
|
|
}
|
|
|
|
#define OPT_BASE_STRUCT struct priv
|
|
|
|
const struct ao_driver audio_out_alsa = {
|
|
.description = "ALSA-0.9.x-1.x audio output",
|
|
.name = "alsa",
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.control = control,
|
|
.get_space = get_space,
|
|
.play = play,
|
|
.get_delay = get_delay,
|
|
.pause = audio_pause,
|
|
.resume = audio_resume,
|
|
.reset = reset,
|
|
.priv_size = sizeof(struct priv),
|
|
.priv_defaults = &(const struct priv) {
|
|
.cfg_block = 1,
|
|
.cfg_mixer_device = "default",
|
|
.cfg_mixer_name = "Master",
|
|
.cfg_mixer_index = 0,
|
|
},
|
|
.options = (const struct m_option[]) {
|
|
OPT_FLAG("resample", cfg_resample, 0),
|
|
OPT_STRING("device", cfg_device, 0),
|
|
OPT_FLAG("block", cfg_block, 0),
|
|
OPT_STRING("mixer-device", cfg_mixer_device, 0),
|
|
OPT_STRING("mixer-name", cfg_mixer_name, 0),
|
|
OPT_INTRANGE("mixer-index", cfg_mixer_index, 0, 0, 99),
|
|
{0}
|
|
},
|
|
};
|