mirror of
https://github.com/owntone/owntone-server.git
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[outputs] Remove unused old alsa output
This commit is contained in:
chme
parent
65732ccaf6
commit
4f2d994151
@ -1,738 +0,0 @@
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/*
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* Copyright (C) 2010 Julien BLACHE <jb@jblache.org>
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*
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* This program 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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* This program 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
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#ifdef HAVE_CONFIG_H
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# include <config.h>
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#endif
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <string.h>
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#include <errno.h>
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#include <stdint.h>
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#include <inttypes.h>
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#include <asoundlib.h>
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#include "conffile.h"
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#include "logger.h"
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#include "player.h"
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#include "laudio.h"
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struct pcm_packet
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{
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uint8_t samples[STOB(AIRTUNES_V2_PACKET_SAMPLES)];
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uint64_t rtptime;
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size_t offset;
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struct pcm_packet *next;
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};
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static uint64_t pcm_pos;
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static uint64_t pcm_start_pos;
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static int pcm_last_error;
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static int pcm_recovery;
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static int pcm_buf_threshold;
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static int pcm_period_size;
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static struct pcm_packet *pcm_pkt_head;
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static struct pcm_packet *pcm_pkt_tail;
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static char *card_name;
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static char *mixer_name;
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static snd_pcm_t *hdl;
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static snd_mixer_t *mixer_hdl;
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static snd_mixer_elem_t *vol_elem;
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static long vol_min;
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static long vol_max;
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static enum laudio_state pcm_status;
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static laudio_status_cb status_cb;
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static void
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update_status(enum laudio_state status)
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{
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pcm_status = status;
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status_cb(status);
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}
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static int
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laudio_alsa_xrun_recover(int err)
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{
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int ret;
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if (err != 0)
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pcm_last_error = err;
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/* Buffer underrun */
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if (err == -EPIPE)
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{
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DPRINTF(E_WARN, L_LAUDIO, "PCM buffer underrun\n");
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ret = snd_pcm_prepare(hdl);
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if (ret < 0)
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{
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DPRINTF(E_WARN, L_LAUDIO, "Couldn't recover from underrun: %s\n", snd_strerror(ret));
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return 1;
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}
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return 0;
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}
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/* Device suspended */
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else if (pcm_last_error == -ESTRPIPE)
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{
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ret = snd_pcm_resume(hdl);
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if (ret == -EAGAIN)
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{
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pcm_recovery++;
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return 2;
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}
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else if (ret < 0)
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{
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pcm_recovery = 0;
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ret = snd_pcm_prepare(hdl);
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if (ret < 0)
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{
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DPRINTF(E_WARN, L_LAUDIO, "Couldn't recover from suspend: %s\n", snd_strerror(ret));
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return 1;
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}
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}
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pcm_recovery = 0;
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return 0;
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}
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return err;
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}
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static int
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laudio_alsa_set_start_threshold(snd_pcm_uframes_t threshold)
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{
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snd_pcm_sw_params_t *sw_params;
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int ret;
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ret = snd_pcm_sw_params_malloc(&sw_params);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Could not allocate sw params: %s\n", snd_strerror(ret));
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goto out_fail;
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}
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ret = snd_pcm_sw_params_current(hdl, sw_params);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Could not retrieve current sw params: %s\n", snd_strerror(ret));
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goto out_fail;
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}
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ret = snd_pcm_sw_params_set_start_threshold(hdl, sw_params, threshold);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Could not set start threshold: %s\n", snd_strerror(ret));
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goto out_fail;
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}
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ret = snd_pcm_sw_params(hdl, sw_params);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Could not set sw params: %s\n", snd_strerror(ret));
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goto out_fail;
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}
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return 0;
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out_fail:
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snd_pcm_sw_params_free(sw_params);
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return -1;
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}
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static void
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laudio_alsa_write(uint8_t *buf, uint64_t rtptime)
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{
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struct pcm_packet *pkt;
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snd_pcm_sframes_t nsamp;
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int ret;
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pkt = (struct pcm_packet *)malloc(sizeof(struct pcm_packet));
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if (!pkt)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Out of memory for PCM pkt\n");
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update_status(LAUDIO_FAILED);
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return;
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}
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memcpy(pkt->samples, buf, sizeof(pkt->samples));
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pkt->rtptime = rtptime;
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pkt->offset = 0;
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pkt->next = NULL;
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if (pcm_pkt_tail)
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{
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pcm_pkt_tail->next = pkt;
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pcm_pkt_tail = pkt;
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}
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else
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{
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pcm_pkt_head = pkt;
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pcm_pkt_tail = pkt;
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}
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if (pcm_pos < pcm_pkt_head->rtptime)
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{
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pcm_pos += AIRTUNES_V2_PACKET_SAMPLES;
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return;
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}
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else if ((pcm_status != LAUDIO_RUNNING) && (pcm_pos >= pcm_start_pos))
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{
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update_status(LAUDIO_RUNNING);
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}
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pkt = pcm_pkt_head;
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while (pkt)
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{
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if (pcm_recovery)
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{
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ret = laudio_alsa_xrun_recover(0);
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if ((ret == 2) && (pcm_recovery < 10))
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return;
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else
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{
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if (ret == 2)
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DPRINTF(E_LOG, L_LAUDIO, "Couldn't recover PCM device after 10 tries, aborting\n");
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update_status(LAUDIO_FAILED);
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return;
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}
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}
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nsamp = snd_pcm_writei(hdl, pkt->samples + pkt->offset, BTOS(sizeof(pkt->samples) - pkt->offset));
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if ((nsamp == -EPIPE) || (nsamp == -ESTRPIPE))
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{
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ret = laudio_alsa_xrun_recover(nsamp);
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if ((ret < 0) || (ret == 1))
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{
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if (ret < 0)
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DPRINTF(E_LOG, L_LAUDIO, "PCM write error: %s\n", snd_strerror(ret));
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update_status(LAUDIO_FAILED);
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return;
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}
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else if (ret != 0)
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return;
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continue;
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}
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else if (nsamp < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "PCM write error: %s\n", snd_strerror(nsamp));
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update_status(LAUDIO_FAILED);
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return;
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}
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pcm_last_error = 0;
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pcm_pos += nsamp;
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pkt->offset += STOB(nsamp);
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if (pkt->offset == sizeof(pkt->samples))
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{
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pcm_pkt_head = pkt->next;
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if (pkt == pcm_pkt_tail)
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pcm_pkt_tail = NULL;
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free(pkt);
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pkt = pcm_pkt_head;
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}
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/* Don't let ALSA fill up the buffer too much */
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if (nsamp == AIRTUNES_V2_PACKET_SAMPLES)
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return;
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}
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}
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static uint64_t
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laudio_alsa_get_pos(void)
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{
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snd_pcm_sframes_t delay;
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int ret;
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if (pcm_pos == 0)
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return 0;
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if (pcm_last_error != 0)
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return pcm_pos;
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if (snd_pcm_state(hdl) != SND_PCM_STATE_RUNNING)
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return pcm_pos;
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ret = snd_pcm_delay(hdl, &delay);
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if (ret < 0)
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{
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DPRINTF(E_WARN, L_LAUDIO, "Could not obtain PCM delay: %s\n", snd_strerror(ret));
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return pcm_pos;
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}
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return pcm_pos - delay;
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}
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static void
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laudio_alsa_set_volume(int vol)
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{
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int pcm_vol;
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if (!mixer_hdl || !vol_elem)
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return;
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snd_mixer_handle_events(mixer_hdl);
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if (!snd_mixer_selem_is_active(vol_elem))
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return;
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switch (vol)
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{
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case 0:
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pcm_vol = vol_min;
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break;
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case 100:
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pcm_vol = vol_max;
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break;
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default:
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pcm_vol = vol_min + (vol * (vol_max - vol_min)) / 100;
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break;
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}
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DPRINTF(E_DBG, L_LAUDIO, "Setting PCM volume to %d (%d)\n", pcm_vol, vol);
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snd_mixer_selem_set_playback_volume_all(vol_elem, pcm_vol);
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}
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static int
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laudio_alsa_start(uint64_t cur_pos, uint64_t next_pkt)
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{
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snd_output_t *output;
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char *debug_pcm_cfg;
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int ret;
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ret = snd_pcm_prepare(hdl);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Could not prepare PCM device: %s\n", snd_strerror(ret));
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return -1;
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}
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DPRINTF(E_DBG, L_LAUDIO, "Start local audio curpos %" PRIu64 ", next_pkt %" PRIu64 "\n", cur_pos, next_pkt);
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/* Make pcm_pos the rtptime of the packet containing cur_pos */
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pcm_pos = next_pkt;
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while (pcm_pos > cur_pos)
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pcm_pos -= AIRTUNES_V2_PACKET_SAMPLES;
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pcm_start_pos = next_pkt + pcm_period_size;
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/* Compensate period size, otherwise get_pos won't be correct */
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pcm_pos += pcm_period_size;
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DPRINTF(E_DBG, L_LAUDIO, "PCM pos %" PRIu64 ", start pos %" PRIu64 "\n", pcm_pos, pcm_start_pos);
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pcm_pkt_head = NULL;
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pcm_pkt_tail = NULL;
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pcm_last_error = 0;
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pcm_recovery = 0;
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// alsa doesn't actually seem to wait for this threshold?
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ret = laudio_alsa_set_start_threshold(pcm_buf_threshold);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Could not set PCM start threshold for local audio start\n");
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return -1;
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}
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// Dump PCM config data for E_DBG logging
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ret = snd_output_buffer_open(&output);
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if (ret == 0)
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{
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if (snd_pcm_dump_setup(hdl, output) == 0)
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{
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snd_output_buffer_string(output, &debug_pcm_cfg);
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DPRINTF(E_DBG, L_LAUDIO, "Dump of sound device config:\n%s\n", debug_pcm_cfg);
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}
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snd_output_close(output);
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}
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update_status(LAUDIO_STARTED);
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return 0;
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}
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static void
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laudio_alsa_stop(void)
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{
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struct pcm_packet *pkt;
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update_status(LAUDIO_STOPPING);
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snd_pcm_drop(hdl);
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for (pkt = pcm_pkt_head; pcm_pkt_head; pkt = pcm_pkt_head)
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{
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pcm_pkt_head = pkt->next;
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free(pkt);
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}
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pcm_pkt_head = NULL;
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pcm_pkt_tail = NULL;
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update_status(LAUDIO_OPEN);
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}
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static int
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mixer_open(void)
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{
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snd_mixer_elem_t *elem;
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snd_mixer_elem_t *master;
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snd_mixer_elem_t *pcm;
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snd_mixer_elem_t *custom;
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snd_mixer_selem_id_t *sid;
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int ret;
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ret = snd_mixer_open(&mixer_hdl, 0);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Failed to open mixer: %s\n", snd_strerror(ret));
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mixer_hdl = NULL;
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return -1;
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}
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ret = snd_mixer_attach(mixer_hdl, card_name);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Failed to attach mixer: %s\n", snd_strerror(ret));
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goto out_close;
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}
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ret = snd_mixer_selem_register(mixer_hdl, NULL, NULL);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Failed to register mixer: %s\n", snd_strerror(ret));
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goto out_detach;
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}
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ret = snd_mixer_load(mixer_hdl);
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if (ret < 0)
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{
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DPRINTF(E_LOG, L_LAUDIO, "Failed to load mixer: %s\n", snd_strerror(ret));
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goto out_detach;
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}
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/* Grab interesting elements */
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snd_mixer_selem_id_alloca(&sid);
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pcm = NULL;
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master = NULL;
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custom = NULL;
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for (elem = snd_mixer_first_elem(mixer_hdl); elem; elem = snd_mixer_elem_next(elem))
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{
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snd_mixer_selem_get_id(elem, sid);
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if (mixer_name && (strcmp(snd_mixer_selem_id_get_name(sid), mixer_name) == 0))
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{
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custom = elem;
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break;
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}
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else if (strcmp(snd_mixer_selem_id_get_name(sid), "PCM") == 0)
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pcm = elem;
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else if (strcmp(snd_mixer_selem_id_get_name(sid), "Master") == 0)
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master = elem;
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}
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if (mixer_name)
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{
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if (custom)
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vol_elem = custom;
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else
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{
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DPRINTF(E_LOG, L_LAUDIO, "Failed to open configured mixer element '%s'\n", mixer_name);
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goto out_detach;
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}
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}
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else if (pcm)
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vol_elem = pcm;
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else if (master)
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vol_elem = master;
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else
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{
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DPRINTF(E_LOG, L_LAUDIO, "Failed to open PCM or Master mixer element\n");
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goto out_detach;
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}
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/* Get min & max volume */
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snd_mixer_selem_get_playback_volume_range(vol_elem, &vol_min, &vol_max);
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return 0;
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out_detach:
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snd_mixer_detach(mixer_hdl, card_name);
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out_close:
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snd_mixer_close(mixer_hdl);
|
||||
mixer_hdl = NULL;
|
||||
vol_elem = NULL;
|
||||
|
||||
return -1;
|
||||
}
|
||||
|
||||
static int
|
||||
laudio_alsa_open(void)
|
||||
{
|
||||
snd_pcm_hw_params_t *hw_params;
|
||||
snd_pcm_uframes_t bufsize;
|
||||
snd_pcm_uframes_t period_size;
|
||||
int ret;
|
||||
|
||||
hw_params = NULL;
|
||||
|
||||
ret = snd_pcm_open(&hdl, card_name, SND_PCM_STREAM_PLAYBACK, 0);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not open playback device: %s\n", snd_strerror(ret));
|
||||
|
||||
return -1;
|
||||
}
|
||||
|
||||
/* HW params */
|
||||
ret = snd_pcm_hw_params_malloc(&hw_params);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not allocate hw params: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
ret = snd_pcm_hw_params_any(hdl, hw_params);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not retrieve hw params: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
ret = snd_pcm_hw_params_set_access(hdl, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not set access method: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
ret = snd_pcm_hw_params_set_format(hdl, hw_params, SND_PCM_FORMAT_S16_LE);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not set S16LE format: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
ret = snd_pcm_hw_params_set_channels(hdl, hw_params, 2);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not set stereo output: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
ret = snd_pcm_hw_params_set_rate(hdl, hw_params, 44100, 0);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Hardware doesn't support 44.1 kHz: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
ret = snd_pcm_hw_params_get_buffer_size_max(hw_params, &bufsize);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not get max buffer size: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
DPRINTF(E_DBG, L_LAUDIO, "Max buffer size is %lu samples\n", bufsize);
|
||||
|
||||
ret = snd_pcm_hw_params_set_buffer_size_max(hdl, hw_params, &bufsize);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not set buffer size to max: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
// With a small period size we seem to get underruns because the period time
|
||||
// passes before we manage to feed with samples (if the player is slightly
|
||||
// behind - especially critical during startup when the buffer is low)
|
||||
// Internet suggests period_size should be /2 bufsize, but default seems to be
|
||||
// much lower, so compromise on /4 (but not more than 65536 frames = almost 2 sec).
|
||||
period_size = bufsize / 4;
|
||||
if (period_size > 65536)
|
||||
period_size = 65536;
|
||||
|
||||
ret = snd_pcm_hw_params_set_period_size_near(hdl, hw_params, &period_size, NULL);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not set period size: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
DPRINTF(E_DBG, L_LAUDIO, "Buffer size is %lu samples, period size is %lu samples\n", bufsize, period_size);
|
||||
|
||||
ret = snd_pcm_hw_params(hdl, hw_params);
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not set hw params: %s\n", snd_strerror(ret));
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
snd_pcm_hw_params_free(hw_params);
|
||||
hw_params = NULL;
|
||||
|
||||
pcm_pos = 0;
|
||||
pcm_last_error = 0;
|
||||
pcm_recovery = 0;
|
||||
pcm_buf_threshold = ((bufsize - period_size) / AIRTUNES_V2_PACKET_SAMPLES) * AIRTUNES_V2_PACKET_SAMPLES;
|
||||
pcm_period_size = period_size;
|
||||
|
||||
ret = mixer_open();
|
||||
if (ret < 0)
|
||||
{
|
||||
DPRINTF(E_LOG, L_LAUDIO, "Could not open mixer\n");
|
||||
|
||||
goto out_fail;
|
||||
}
|
||||
|
||||
update_status(LAUDIO_OPEN);
|
||||
|
||||
return 0;
|
||||
|
||||
out_fail:
|
||||
if (hw_params)
|
||||
snd_pcm_hw_params_free(hw_params);
|
||||
|
||||
snd_pcm_close(hdl);
|
||||
hdl = NULL;
|
||||
|
||||
return -1;
|
||||
}
|
||||
|
||||
static void
|
||||
laudio_alsa_close(void)
|
||||
{
|
||||
struct pcm_packet *pkt;
|
||||
|
||||
snd_pcm_close(hdl);
|
||||
hdl = NULL;
|
||||
|
||||
if (mixer_hdl)
|
||||
{
|
||||
snd_mixer_detach(mixer_hdl, card_name);
|
||||
snd_mixer_close(mixer_hdl);
|
||||
|
||||
mixer_hdl = NULL;
|
||||
vol_elem = NULL;
|
||||
}
|
||||
|
||||
for (pkt = pcm_pkt_head; pcm_pkt_head; pkt = pcm_pkt_head)
|
||||
{
|
||||
pcm_pkt_head = pkt->next;
|
||||
|
||||
free(pkt);
|
||||
}
|
||||
|
||||
pcm_pkt_head = NULL;
|
||||
pcm_pkt_tail = NULL;
|
||||
|
||||
update_status(LAUDIO_CLOSED);
|
||||
}
|
||||
|
||||
|
||||
static int
|
||||
laudio_alsa_init(laudio_status_cb cb, cfg_t *cfg_audio)
|
||||
{
|
||||
snd_lib_error_set_handler(logger_alsa);
|
||||
|
||||
status_cb = cb;
|
||||
|
||||
card_name = cfg_getstr(cfg_audio, "card");
|
||||
mixer_name = cfg_getstr(cfg_audio, "mixer");
|
||||
|
||||
hdl = NULL;
|
||||
mixer_hdl = NULL;
|
||||
vol_elem = NULL;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void
|
||||
laudio_alsa_deinit(void)
|
||||
{
|
||||
snd_lib_error_set_handler(NULL);
|
||||
}
|
||||
|
||||
audio_output audio_alsa = {
|
||||
.name = "alsa",
|
||||
.init = &laudio_alsa_init,
|
||||
.deinit = &laudio_alsa_deinit,
|
||||
.start = &laudio_alsa_start,
|
||||
.stop = &laudio_alsa_stop,
|
||||
.open = &laudio_alsa_open,
|
||||
.close = &laudio_alsa_close,
|
||||
.pos = &laudio_alsa_get_pos,
|
||||
.write = &laudio_alsa_write,
|
||||
.volume = &laudio_alsa_set_volume,
|
||||
};
|
||||
Loading…
x
Reference in New Issue
Block a user