mpv/libao2/ao_esd.c

460 lines
11 KiB
C

/*
* ao_esd - EsounD audio output driver for MPlayer
*
* Juergen Keil <jk@tools.de>
*
* This driver is distributed under the terms of the GPL
*
* TODO / known problems:
* - does not work well when the esd daemon has autostandby disabled
* (workaround: run esd with option "-as 2" - fortunatelly this is
* the default)
* - plays noise on a linux 2.4.4 kernel with a SB16PCI card, when using
* a local tcp connection to the esd daemon; there is no noise when using
* a unix domain socket connection.
* (there are EIO errors reported by the sound card driver, so this is
* most likely a linux sound card driver problem)
*/
#include <sys/types.h>
#include <sys/time.h>
#include <sys/socket.h>
#include <stdio.h>
#include <string.h>
#include <unistd.h>
#include <errno.h>
#include <fcntl.h>
#include <time.h>
#ifdef __svr4__
#include <stropts.h>
#endif
#include <esd.h>
#include "config.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
#include "help_mp.h"
#undef ESD_DEBUG
#if ESD_DEBUG
#define dprintf(...) printf(__VA_ARGS__)
#else
#define dprintf(...) /**/
#endif
#define ESD_CLIENT_NAME "MPlayer"
#define ESD_MAX_DELAY (1.0f) /* max amount of data buffered in esd (#sec) */
static ao_info_t info =
{
"EsounD audio output",
"esd",
"Juergen Keil <jk@tools.de>",
""
};
LIBAO_EXTERN(esd)
static int esd_fd = -1;
static int esd_play_fd = -1;
static esd_server_info_t *esd_svinfo;
static int esd_latency;
static int esd_bytes_per_sample;
static unsigned long esd_samples_written;
static struct timeval esd_play_start;
extern float audio_delay;
/*
* to set/get/query special features/parameters
*/
static int control(int cmd, void *arg)
{
esd_player_info_t *esd_pi;
esd_info_t *esd_i;
time_t now;
static time_t vol_cache_time;
static ao_control_vol_t vol_cache;
switch (cmd) {
case AOCONTROL_GET_VOLUME:
time(&now);
if (now == vol_cache_time) {
*(ao_control_vol_t *)arg = vol_cache;
return CONTROL_OK;
}
dprintf("esd: get vol\n");
if ((esd_i = esd_get_all_info(esd_fd)) == NULL)
return CONTROL_ERROR;
for (esd_pi = esd_i->player_list; esd_pi != NULL; esd_pi = esd_pi->next)
if (strcmp(esd_pi->name, ESD_CLIENT_NAME) == 0)
break;
if (esd_pi != NULL) {
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
vol->left = esd_pi->left_vol_scale * 100 / ESD_VOLUME_BASE;
vol->right = esd_pi->right_vol_scale * 100 / ESD_VOLUME_BASE;
vol_cache = *vol;
vol_cache_time = now;
}
esd_free_all_info(esd_i);
return CONTROL_OK;
case AOCONTROL_SET_VOLUME:
dprintf("esd: set vol\n");
if ((esd_i = esd_get_all_info(esd_fd)) == NULL)
return CONTROL_ERROR;
for (esd_pi = esd_i->player_list; esd_pi != NULL; esd_pi = esd_pi->next)
if (strcmp(esd_pi->name, ESD_CLIENT_NAME) == 0)
break;
if (esd_pi != NULL) {
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
esd_set_stream_pan(esd_fd, esd_pi->source_id,
vol->left * ESD_VOLUME_BASE / 100,
vol->right * ESD_VOLUME_BASE / 100);
vol_cache = *vol;
time(&vol_cache_time);
}
esd_free_all_info(esd_i);
return CONTROL_OK;
default:
return CONTROL_UNKNOWN;
}
}
/*
* open & setup audio device
* return: 1=success 0=fail
*/
static int init(int rate_hz, int channels, int format, int flags)
{
esd_format_t esd_fmt;
int bytes_per_sample;
int fl;
char *server = ao_subdevice; /* NULL for localhost */
float lag_seconds, lag_net, lag_serv;
struct timeval proto_start, proto_end;
if (esd_fd < 0) {
esd_fd = esd_open_sound(server);
if (esd_fd < 0) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenSound,
strerror(errno));
return 0;
}
/* get server info, and measure network latency */
gettimeofday(&proto_start, NULL);
esd_svinfo = esd_get_server_info(esd_fd);
if(server) {
gettimeofday(&proto_end, NULL);
lag_net = (proto_end.tv_sec - proto_start.tv_sec) +
(proto_end.tv_usec - proto_start.tv_usec) / 1000000.0;
lag_net /= 2.0; /* round trip -> one way */
} else
lag_net = 0.0; /* no network lag */
/*
if (esd_svinfo) {
mp_msg(MSGT_AO, MSGL_INFO, "AO: [esd] server info:\n");
esd_print_server_info(esd_svinfo);
}
*/
}
esd_fmt = ESD_STREAM | ESD_PLAY;
#if ESD_RESAMPLES
/* let the esd daemon convert sample rate */
#else
/* let mplayer's audio filter convert the sample rate */
if (esd_svinfo != NULL)
rate_hz = esd_svinfo->rate;
#endif
ao_data.samplerate = rate_hz;
/* EsounD can play mono or stereo */
switch (channels) {
case 1:
esd_fmt |= ESD_MONO;
ao_data.channels = bytes_per_sample = 1;
break;
default:
esd_fmt |= ESD_STEREO;
ao_data.channels = bytes_per_sample = 2;
break;
}
/* EsounD can play 8bit unsigned and 16bit signed native */
switch (format) {
case AF_FORMAT_S8:
case AF_FORMAT_U8:
esd_fmt |= ESD_BITS8;
ao_data.format = AF_FORMAT_U8;
break;
default:
esd_fmt |= ESD_BITS16;
ao_data.format = AF_FORMAT_S16_NE;
bytes_per_sample *= 2;
break;
}
/* modify audio_delay depending on esd_latency
* latency is number of samples @ 44.1khz stereo 16 bit
* adjust according to rate_hz & bytes_per_sample
*/
#ifdef HAVE_ESD_LATENCY
esd_latency = esd_get_latency(esd_fd);
#else
esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE *
(ESD_BUF_SIZE + 64 * (4.0f / bytes_per_sample))
) / rate_hz;
esd_latency += ESD_BUF_SIZE * 2;
#endif
if(esd_latency > 0) {
lag_serv = (esd_latency * 4.0f) / (bytes_per_sample * rate_hz);
lag_seconds = lag_net + lag_serv;
audio_delay += lag_seconds;
mp_msg(MSGT_AO, MSGL_INFO,MSGTR_AO_ESD_LatencyInfo,
lag_serv, lag_net, lag_seconds);
}
esd_play_fd = esd_play_stream_fallback(esd_fmt, rate_hz,
server, ESD_CLIENT_NAME);
if (esd_play_fd < 0) {
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ESD_CantOpenPBStream, strerror(errno));
return 0;
}
/* enable non-blocking i/o on the socket connection to the esd server */
if ((fl = fcntl(esd_play_fd, F_GETFL)) >= 0)
fcntl(esd_play_fd, F_SETFL, O_NDELAY|fl);
#if ESD_DEBUG
{
int sbuf, rbuf, len;
len = sizeof(sbuf);
getsockopt(esd_play_fd, SOL_SOCKET, SO_SNDBUF, &sbuf, &len);
len = sizeof(rbuf);
getsockopt(esd_play_fd, SOL_SOCKET, SO_RCVBUF, &rbuf, &len);
dprintf("esd: send/receive socket buffer space %d/%d bytes\n",
sbuf, rbuf);
}
#endif
ao_data.bps = bytes_per_sample * rate_hz;
ao_data.outburst = ao_data.bps > 100000 ? 4*ESD_BUF_SIZE : 2*ESD_BUF_SIZE;
esd_play_start.tv_sec = 0;
esd_samples_written = 0;
esd_bytes_per_sample = bytes_per_sample;
return 1;
}
/*
* close audio device
*/
static void uninit(int immed)
{
if (esd_play_fd >= 0) {
esd_close(esd_play_fd);
esd_play_fd = -1;
}
if (esd_svinfo) {
esd_free_server_info(esd_svinfo);
esd_svinfo = NULL;
}
if (esd_fd >= 0) {
esd_close(esd_fd);
esd_fd = -1;
}
}
/*
* plays 'len' bytes of 'data'
* it should round it down to outburst*n
* return: number of bytes played
*/
static int play(void* data, int len, int flags)
{
int offs;
int nwritten;
int nsamples;
int n;
/* round down buffersize to a multiple of ESD_BUF_SIZE bytes */
len = len / ESD_BUF_SIZE * ESD_BUF_SIZE;
if (len <= 0)
return 0;
#define SINGLE_WRITE 0
#if SINGLE_WRITE
nwritten = write(esd_play_fd, data, len);
#else
for (offs = 0, nwritten=0; offs + ESD_BUF_SIZE <= len; offs += ESD_BUF_SIZE) {
/*
* note: we're writing to a non-blocking socket here.
* A partial write means, that the socket buffer is full.
*/
n = write(esd_play_fd, (char*)data + offs, ESD_BUF_SIZE);
if ( n < 0 ) {
if ( errno != EAGAIN )
dprintf("esd play: write failed: %s\n", strerror(errno));
break;
} else if ( n != ESD_BUF_SIZE ) {
nwritten += n;
break;
} else
nwritten += n;
}
#endif
if (nwritten > 0) {
if (!esd_play_start.tv_sec)
gettimeofday(&esd_play_start, NULL);
nsamples = nwritten / esd_bytes_per_sample;
esd_samples_written += nsamples;
dprintf("esd play: %d %lu\n", nsamples, esd_samples_written);
} else {
dprintf("esd play: blocked / %lu\n", esd_samples_written);
}
return nwritten;
}
/*
* stop playing, keep buffers (for pause)
*/
static void audio_pause(void)
{
/*
* not possible with esd. the esd daemom will continue playing
* buffered data (not more than ESD_MAX_DELAY seconds of samples)
*/
}
/*
* resume playing, after audio_pause()
*/
static void audio_resume(void)
{
/*
* not possible with esd.
*
* Let's hope the pause was long enough that the esd ran out of
* buffered data; we restart our time based delay computation
* for an audio resume.
*/
esd_play_start.tv_sec = 0;
esd_samples_written = 0;
}
/*
* stop playing and empty buffers (for seeking/pause)
*/
static void reset(void)
{
#ifdef __svr4__
/* throw away data buffered in the esd connection */
if (ioctl(esd_play_fd, I_FLUSH, FLUSHW))
perror("I_FLUSH");
#endif
}
/*
* return: how many bytes can be played without blocking
*/
static int get_space(void)
{
struct timeval tmout;
fd_set wfds;
float current_delay;
int space;
/*
* Don't buffer too much data in the esd daemon.
*
* If we send too much, esd will block in write()s to the sound
* device, and the consequence is a huge slow down for things like
* esd_get_all_info().
*/
if ((current_delay = get_delay()) >= ESD_MAX_DELAY) {
dprintf("esd get_space: too much data buffered\n");
return 0;
}
FD_ZERO(&wfds);
FD_SET(esd_play_fd, &wfds);
tmout.tv_sec = 0;
tmout.tv_usec = 0;
if (select(esd_play_fd + 1, NULL, &wfds, NULL, &tmout) != 1)
return 0;
if (!FD_ISSET(esd_play_fd, &wfds))
return 0;
/* try to fill 50% of the remaining "free" buffer space */
space = (ESD_MAX_DELAY - current_delay) * ao_data.bps * 0.5f;
/* round up to next multiple of ESD_BUF_SIZE */
space = (space + ESD_BUF_SIZE-1) / ESD_BUF_SIZE * ESD_BUF_SIZE;
dprintf("esd get_space: %d\n", space);
return space;
}
/*
* return: delay in seconds between first and last sample in buffer
*/
static float get_delay(void)
{
struct timeval now;
double buffered_samples_time;
double play_time;
if (!esd_play_start.tv_sec)
return 0;
buffered_samples_time = (float)esd_samples_written / ao_data.samplerate;
gettimeofday(&now, NULL);
play_time = now.tv_sec - esd_play_start.tv_sec;
play_time += (now.tv_usec - esd_play_start.tv_usec) / 1000000.;
/* dprintf("esd delay: %f %f\n", play_time, buffered_samples_time); */
if (play_time > buffered_samples_time) {
dprintf("esd: underflow\n");
esd_play_start.tv_sec = 0;
esd_samples_written = 0;
return 0;
}
dprintf("esd: get_delay %f\n", buffered_samples_time - play_time);
return buffered_samples_time - play_time;
}