From e5f8ab3bcacdbaf57827cf2ea238c8bb320857c3 Mon Sep 17 00:00:00 2001 From: Uoti Urpala Date: Sun, 6 May 2012 18:29:48 +0300 Subject: ao_arts, ao_esd: remove these AOs Delete ao_arts and ao_esd. Both have been deprecated upstream. --- libao2/ao_arts.c | 148 ----------------- libao2/ao_esd.c | 477 ----------------------------------------------------- libao2/audio_out.c | 6 - 3 files changed, 631 deletions(-) delete mode 100644 libao2/ao_arts.c delete mode 100644 libao2/ao_esd.c (limited to 'libao2') 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 - * - * 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 -#include - -#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 ", - "" -}; - -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 - * - * 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 -#include -#include -#include -#include -#include -#include -#include -#include -#ifdef __svr4__ -#include -#endif -#include - -#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 ", - "" -}; - -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 -- cgit v1.2.3