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-rw-r--r--libao2/ao_arts.c148
-rw-r--r--libao2/ao_esd.c477
-rw-r--r--libao2/audio_out.c6
3 files changed, 0 insertions, 631 deletions
diff --git a/libao2/ao_arts.c b/libao2/ao_arts.c
deleted file mode 100644
index d828e7953e..0000000000
--- a/libao2/ao_arts.c
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * aRts audio output driver for MPlayer
- *
- * copyright (c) 2002 Michele Balistreri <brain87@gmx.net>
- *
- * This file is part of MPlayer.
- *
- * MPlayer 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.
- *
- * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <artsc.h>
-#include <stdio.h>
-
-#include "config.h"
-#include "audio_out.h"
-#include "audio_out_internal.h"
-#include "libaf/af_format.h"
-#include "mp_msg.h"
-
-#define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8)
-
-/* Feel free to experiment with the following values: */
-#define ARTS_PACKETS 10 /* Number of audio packets */
-#define ARTS_PACKET_SIZE_LOG2 11 /* Log2 of audio packet size */
-
-static arts_stream_t stream;
-
-static const ao_info_t info =
-{
- "aRts audio output",
- "arts",
- "Michele Balistreri <brain87@gmx.net>",
- ""
-};
-
-LIBAO_EXTERN(arts)
-
-static int control(int cmd, void *arg)
-{
- return CONTROL_UNKNOWN;
-}
-
-static int init(int rate_hz, int channels, int format, int flags)
-{
- int err;
- int frag_spec;
-
- if( (err=arts_init()) ) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ARTS] %s\n", arts_error_text(err));
- return 0;
- }
- mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] Connected to sound server.\n");
-
- /*
- * arts supports 8bit unsigned and 16bit signed sample formats
- * (16bit apparently in little endian format, even in the case
- * when artsd runs on a big endian cpu).
- *
- * Unsupported formats are translated to one of these two formats
- * using mplayer's audio filters.
- */
- switch (format) {
- case AF_FORMAT_U8:
- case AF_FORMAT_S8:
- format = AF_FORMAT_U8;
- break;
- default:
- format = AF_FORMAT_S16_LE; /* artsd always expects little endian?*/
- break;
- }
-
- ao_data.format = format;
- ao_data.channels = channels;
- ao_data.samplerate = rate_hz;
- ao_data.bps = (rate_hz*channels);
-
- if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
- ao_data.bps*=2;
-
- stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer");
-
- if(stream == NULL) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ARTS] Unable to open a stream.\n");
- arts_free();
- return 0;
- }
-
- /* Set the stream to blocking: it will not block anyway, but it seems */
- /* to be working better */
- arts_stream_set(stream, ARTS_P_BLOCKING, 1);
- frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16;
- arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec);
- ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE);
- mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] Stream opened.\n");
-
- mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] buffer size: %d\n",
- ao_data.buffersize);
- mp_tmsg(MSGT_AO, MSGL_INFO, "[AO ARTS] buffer size: %d\n",
- arts_stream_get(stream, ARTS_P_PACKET_SIZE));
-
- return 1;
-}
-
-static void uninit(int immed)
-{
- arts_close_stream(stream);
- arts_free();
-}
-
-static int play(void* data,int len,int flags)
-{
- return arts_write(stream, data, len);
-}
-
-static void audio_pause(void)
-{
-}
-
-static void audio_resume(void)
-{
-}
-
-static void reset(void)
-{
-}
-
-static int get_space(void)
-{
- return arts_stream_get(stream, ARTS_P_BUFFER_SPACE);
-}
-
-static float get_delay(void)
-{
- return ((float) (ao_data.buffersize - arts_stream_get(stream,
- ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps);
-}
diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c
deleted file mode 100644
index e7c6701aa0..0000000000
--- a/libao2/ao_esd.c
+++ /dev/null
@@ -1,477 +0,0 @@
-/*
- * EsounD audio output driver for MPlayer
- *
- * copyright (c) 2002 Juergen Keil <jk@tools.de>
- *
- * This file is part of MPlayer.
- *
- * MPlayer 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.
- *
- * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
- /*
- * 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"
-
-
-#define ESD_RESAMPLES 0
-#define ESD_DEBUG 0
-
-#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 const 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 = 0., lag_serv;
- struct timeval proto_start, proto_end;
-
- global_ao->no_persistent_volume = true;
-
- if (esd_fd < 0) {
- esd_fd = esd_open_sound(server);
- if (esd_fd < 0) {
- mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ESD] esd_open_sound failed: %s\n",
- 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 CONFIG_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_tmsg(MSGT_AO, MSGL_INFO,"[AO ESD] latency: [server: %0.2fs, net: %0.2fs] (adjust %0.2fs)\n",
- 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_tmsg(MSGT_AO, MSGL_ERR, "[AO ESD] failed to open ESD playback stream: %s\n", 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;
-}
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index 5dcb10c027..36b3e2d7c6 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -90,12 +90,6 @@ static const struct ao_driver * const audio_out_drivers[] = {
&audio_out_sun,
#endif
// wrappers:
-#ifdef CONFIG_ARTS
- &audio_out_arts,
-#endif
-#ifdef CONFIG_ESD
- &audio_out_esd,
-#endif
#ifdef CONFIG_JACK
&audio_out_jack,
#endif
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