[outputs] Remove unused old alsa output

This commit is contained in:
chme 2016-10-25 21:24:04 +02:00
parent 65732ccaf6
commit 4f2d994151
1 changed files with 0 additions and 738 deletions

View File

@ -1,738 +0,0 @@
/*
* Copyright (C) 2010 Julien BLACHE <jb@jblache.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <errno.h>
#include <stdint.h>
#include <inttypes.h>
#include <asoundlib.h>
#include "conffile.h"
#include "logger.h"
#include "player.h"
#include "laudio.h"
struct pcm_packet
{
uint8_t samples[STOB(AIRTUNES_V2_PACKET_SAMPLES)];
uint64_t rtptime;
size_t offset;
struct pcm_packet *next;
};
static uint64_t pcm_pos;
static uint64_t pcm_start_pos;
static int pcm_last_error;
static int pcm_recovery;
static int pcm_buf_threshold;
static int pcm_period_size;
static struct pcm_packet *pcm_pkt_head;
static struct pcm_packet *pcm_pkt_tail;
static char *card_name;
static char *mixer_name;
static snd_pcm_t *hdl;
static snd_mixer_t *mixer_hdl;
static snd_mixer_elem_t *vol_elem;
static long vol_min;
static long vol_max;
static enum laudio_state pcm_status;
static laudio_status_cb status_cb;
static void
update_status(enum laudio_state status)
{
pcm_status = status;
status_cb(status);
}
static int
laudio_alsa_xrun_recover(int err)
{
int ret;
if (err != 0)
pcm_last_error = err;
/* Buffer underrun */
if (err == -EPIPE)
{
DPRINTF(E_WARN, L_LAUDIO, "PCM buffer underrun\n");
ret = snd_pcm_prepare(hdl);
if (ret < 0)
{
DPRINTF(E_WARN, L_LAUDIO, "Couldn't recover from underrun: %s\n", snd_strerror(ret));
return 1;
}
return 0;
}
/* Device suspended */
else if (pcm_last_error == -ESTRPIPE)
{
ret = snd_pcm_resume(hdl);
if (ret == -EAGAIN)
{
pcm_recovery++;
return 2;
}
else if (ret < 0)
{
pcm_recovery = 0;
ret = snd_pcm_prepare(hdl);
if (ret < 0)
{
DPRINTF(E_WARN, L_LAUDIO, "Couldn't recover from suspend: %s\n", snd_strerror(ret));
return 1;
}
}
pcm_recovery = 0;
return 0;
}
return err;
}
static int
laudio_alsa_set_start_threshold(snd_pcm_uframes_t threshold)
{
snd_pcm_sw_params_t *sw_params;
int ret;
ret = snd_pcm_sw_params_malloc(&sw_params);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not allocate sw params: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_sw_params_current(hdl, sw_params);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not retrieve current sw params: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_sw_params_set_start_threshold(hdl, sw_params, threshold);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set start threshold: %s\n", snd_strerror(ret));
goto out_fail;
}
ret = snd_pcm_sw_params(hdl, sw_params);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set sw params: %s\n", snd_strerror(ret));
goto out_fail;
}
return 0;
out_fail:
snd_pcm_sw_params_free(sw_params);
return -1;
}
static void
laudio_alsa_write(uint8_t *buf, uint64_t rtptime)
{
struct pcm_packet *pkt;
snd_pcm_sframes_t nsamp;
int ret;
pkt = (struct pcm_packet *)malloc(sizeof(struct pcm_packet));
if (!pkt)
{
DPRINTF(E_LOG, L_LAUDIO, "Out of memory for PCM pkt\n");
update_status(LAUDIO_FAILED);
return;
}
memcpy(pkt->samples, buf, sizeof(pkt->samples));
pkt->rtptime = rtptime;
pkt->offset = 0;
pkt->next = NULL;
if (pcm_pkt_tail)
{
pcm_pkt_tail->next = pkt;
pcm_pkt_tail = pkt;
}
else
{
pcm_pkt_head = pkt;
pcm_pkt_tail = pkt;
}
if (pcm_pos < pcm_pkt_head->rtptime)
{
pcm_pos += AIRTUNES_V2_PACKET_SAMPLES;
return;
}
else if ((pcm_status != LAUDIO_RUNNING) && (pcm_pos >= pcm_start_pos))
{
update_status(LAUDIO_RUNNING);
}
pkt = pcm_pkt_head;
while (pkt)
{
if (pcm_recovery)
{
ret = laudio_alsa_xrun_recover(0);
if ((ret == 2) && (pcm_recovery < 10))
return;
else
{
if (ret == 2)
DPRINTF(E_LOG, L_LAUDIO, "Couldn't recover PCM device after 10 tries, aborting\n");
update_status(LAUDIO_FAILED);
return;
}
}
nsamp = snd_pcm_writei(hdl, pkt->samples + pkt->offset, BTOS(sizeof(pkt->samples) - pkt->offset));
if ((nsamp == -EPIPE) || (nsamp == -ESTRPIPE))
{
ret = laudio_alsa_xrun_recover(nsamp);
if ((ret < 0) || (ret == 1))
{
if (ret < 0)
DPRINTF(E_LOG, L_LAUDIO, "PCM write error: %s\n", snd_strerror(ret));
update_status(LAUDIO_FAILED);
return;
}
else if (ret != 0)
return;
continue;
}
else if (nsamp < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "PCM write error: %s\n", snd_strerror(nsamp));
update_status(LAUDIO_FAILED);
return;
}
pcm_last_error = 0;
pcm_pos += nsamp;
pkt->offset += STOB(nsamp);
if (pkt->offset == sizeof(pkt->samples))
{
pcm_pkt_head = pkt->next;
if (pkt == pcm_pkt_tail)
pcm_pkt_tail = NULL;
free(pkt);
pkt = pcm_pkt_head;
}
/* Don't let ALSA fill up the buffer too much */
if (nsamp == AIRTUNES_V2_PACKET_SAMPLES)
return;
}
}
static uint64_t
laudio_alsa_get_pos(void)
{
snd_pcm_sframes_t delay;
int ret;
if (pcm_pos == 0)
return 0;
if (pcm_last_error != 0)
return pcm_pos;
if (snd_pcm_state(hdl) != SND_PCM_STATE_RUNNING)
return pcm_pos;
ret = snd_pcm_delay(hdl, &delay);
if (ret < 0)
{
DPRINTF(E_WARN, L_LAUDIO, "Could not obtain PCM delay: %s\n", snd_strerror(ret));
return pcm_pos;
}
return pcm_pos - delay;
}
static void
laudio_alsa_set_volume(int vol)
{
int pcm_vol;
if (!mixer_hdl || !vol_elem)
return;
snd_mixer_handle_events(mixer_hdl);
if (!snd_mixer_selem_is_active(vol_elem))
return;
switch (vol)
{
case 0:
pcm_vol = vol_min;
break;
case 100:
pcm_vol = vol_max;
break;
default:
pcm_vol = vol_min + (vol * (vol_max - vol_min)) / 100;
break;
}
DPRINTF(E_DBG, L_LAUDIO, "Setting PCM volume to %d (%d)\n", pcm_vol, vol);
snd_mixer_selem_set_playback_volume_all(vol_elem, pcm_vol);
}
static int
laudio_alsa_start(uint64_t cur_pos, uint64_t next_pkt)
{
snd_output_t *output;
char *debug_pcm_cfg;
int ret;
ret = snd_pcm_prepare(hdl);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not prepare PCM device: %s\n", snd_strerror(ret));
return -1;
}
DPRINTF(E_DBG, L_LAUDIO, "Start local audio curpos %" PRIu64 ", next_pkt %" PRIu64 "\n", cur_pos, next_pkt);
/* Make pcm_pos the rtptime of the packet containing cur_pos */
pcm_pos = next_pkt;
while (pcm_pos > cur_pos)
pcm_pos -= AIRTUNES_V2_PACKET_SAMPLES;
pcm_start_pos = next_pkt + pcm_period_size;
/* Compensate period size, otherwise get_pos won't be correct */
pcm_pos += pcm_period_size;
DPRINTF(E_DBG, L_LAUDIO, "PCM pos %" PRIu64 ", start pos %" PRIu64 "\n", pcm_pos, pcm_start_pos);
pcm_pkt_head = NULL;
pcm_pkt_tail = NULL;
pcm_last_error = 0;
pcm_recovery = 0;
// alsa doesn't actually seem to wait for this threshold?
ret = laudio_alsa_set_start_threshold(pcm_buf_threshold);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Could not set PCM start threshold for local audio start\n");
return -1;
}
// Dump PCM config data for E_DBG logging
ret = snd_output_buffer_open(&output);
if (ret == 0)
{
if (snd_pcm_dump_setup(hdl, output) == 0)
{
snd_output_buffer_string(output, &debug_pcm_cfg);
DPRINTF(E_DBG, L_LAUDIO, "Dump of sound device config:\n%s\n", debug_pcm_cfg);
}
snd_output_close(output);
}
update_status(LAUDIO_STARTED);
return 0;
}
static void
laudio_alsa_stop(void)
{
struct pcm_packet *pkt;
update_status(LAUDIO_STOPPING);
snd_pcm_drop(hdl);
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_OPEN);
}
static int
mixer_open(void)
{
snd_mixer_elem_t *elem;
snd_mixer_elem_t *master;
snd_mixer_elem_t *pcm;
snd_mixer_elem_t *custom;
snd_mixer_selem_id_t *sid;
int ret;
ret = snd_mixer_open(&mixer_hdl, 0);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Failed to open mixer: %s\n", snd_strerror(ret));
mixer_hdl = NULL;
return -1;
}
ret = snd_mixer_attach(mixer_hdl, card_name);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Failed to attach mixer: %s\n", snd_strerror(ret));
goto out_close;
}
ret = snd_mixer_selem_register(mixer_hdl, NULL, NULL);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Failed to register mixer: %s\n", snd_strerror(ret));
goto out_detach;
}
ret = snd_mixer_load(mixer_hdl);
if (ret < 0)
{
DPRINTF(E_LOG, L_LAUDIO, "Failed to load mixer: %s\n", snd_strerror(ret));
goto out_detach;
}
/* Grab interesting elements */
snd_mixer_selem_id_alloca(&sid);
pcm = NULL;
master = NULL;
custom = NULL;
for (elem = snd_mixer_first_elem(mixer_hdl); elem; elem = snd_mixer_elem_next(elem))
{
snd_mixer_selem_get_id(elem, sid);
if (mixer_name && (strcmp(snd_mixer_selem_id_get_name(sid), mixer_name) == 0))
{
custom = elem;
break;
}
else if (strcmp(snd_mixer_selem_id_get_name(sid), "PCM") == 0)
pcm = elem;
else if (strcmp(snd_mixer_selem_id_get_name(sid), "Master") == 0)
master = elem;
}
if (mixer_name)
{
if (custom)
vol_elem = custom;
else
{
DPRINTF(E_LOG, L_LAUDIO, "Failed to open configured mixer element '%s'\n", mixer_name);
goto out_detach;
}
}
else if (pcm)
vol_elem = pcm;
else if (master)
vol_elem = master;
else
{
DPRINTF(E_LOG, L_LAUDIO, "Failed to open PCM or Master mixer element\n");
goto out_detach;
}
/* Get min & max volume */
snd_mixer_selem_get_playback_volume_range(vol_elem, &vol_min, &vol_max);
return 0;
out_detach:
snd_mixer_detach(mixer_hdl, card_name);
out_close:
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,
};