Mercurial > mplayer.hg
view libao2/ao_alsa.c @ 13822:9e9b5e6b0513
preliminary pre6 changes from Reimar
author | diego |
---|---|
date | Sun, 31 Oct 2004 01:29:17 +0000 |
parents | 07dc40f25068 |
children | a92101a7eb49 |
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/* ao_alsa9/1.x - ALSA-0.9.x-1.x output plugin for MPlayer (C) Alex Beregszaszi modified for real alsa-0.9.0-support by Zsolt Barat <joy@streamminister.de> additional AC3 passthrough support by Andy Lo A Foe <andy@alsaplayer.org> 08/22/2002 iec958-init rewritten and merged with common init, zsolt 04/13/2004 merged with ao_alsa1.x, fixes provided by Jindrich Makovicka 04/25/2004 printfs converted to mp_msg, Zsolt. Any bugreports regarding to this driver are welcome. */ #include <errno.h> #include <sys/time.h> #include <stdlib.h> #include <math.h> #include <string.h> #include <sys/poll.h> #include "../config.h" #include "../mixer.h" #include "../mp_msg.h" #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #if HAVE_SYS_ASOUNDLIB_H #include <sys/asoundlib.h> #elif HAVE_ALSA_ASOUNDLIB_H #include <alsa/asoundlib.h> #else #error "asoundlib.h is not in sys/ or alsa/ - please bugreport" #endif #include "audio_out.h" #include "audio_out_internal.h" #include "afmt.h" static ao_info_t info = { "ALSA-0.9.x-1.x audio output", "alsa", "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>", "under developement" }; LIBAO_EXTERN(alsa) static snd_pcm_t *alsa_handler; static snd_pcm_format_t alsa_format; static snd_pcm_hw_params_t *alsa_hwparams; static snd_pcm_sw_params_t *alsa_swparams; /* possible 4096, original 8192 * was only needed for calculating chunksize? */ static int alsa_fragsize = 4096; /* 16 sets buffersize to 16 * chunksize is as default 1024 * which seems to be good avarge for most situations * so buffersize is 16384 frames by default */ static int alsa_fragcount = 16; static snd_pcm_uframes_t chunk_size = 1024;//is alsa_fragsize / 4 #define MIN_CHUNK_SIZE 1024 static size_t bits_per_sample, bytes_per_sample, bits_per_frame; static size_t chunk_bytes; int ao_mmap = 0; int ao_noblock = 0; int first = 1; static int open_mode; static int set_block_mode; static int alsa_can_pause = 0; #define ALSA_DEVICE_SIZE 256 #undef BUFFERTIME #define SET_CHUNKSIZE #undef USE_POLL /* to set/get/query special features/parameters */ static int control(int cmd, void *arg) { switch(cmd) { case AOCONTROL_QUERY_FORMAT: return CONTROL_TRUE; #ifndef WORDS_BIGENDIAN case AOCONTROL_GET_VOLUME: case AOCONTROL_SET_VOLUME: { ao_control_vol_t *vol = (ao_control_vol_t *)arg; int err; snd_mixer_t *handle; snd_mixer_elem_t *elem; snd_mixer_selem_id_t *sid; static char *mix_name = "PCM"; static char *card = "default"; static int mix_index = 0; long pmin, pmax; long get_vol, set_vol; float f_multi; if(mixer_channel) { char *test_mix_index; mix_name = strdup(mixer_channel); if (test_mix_index = strchr(mix_name, ',')){ *test_mix_index = 0; test_mix_index++; mix_index = strtol(test_mix_index, &test_mix_index, 0); if (*test_mix_index){ mp_msg(MSGT_AO,MSGL_ERR, "alsa-control: invalid mixer index. Defaulting to 0\n"); mix_index = 0 ; } } } if(mixer_device) card = mixer_device; if(ao_data.format == AFMT_AC3) return CONTROL_TRUE; //allocate simple id snd_mixer_selem_id_alloca(&sid); //sets simple-mixer index and name snd_mixer_selem_id_set_index(sid, mix_index); snd_mixer_selem_id_set_name(sid, mix_name); if (mixer_channel) { free(mix_name); mix_name = NULL; } if ((err = snd_mixer_open(&handle, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer open error: %s\n", snd_strerror(err)); return CONTROL_ERROR; } if ((err = snd_mixer_attach(handle, card)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer attach %s error: %s\n", card, snd_strerror(err)); snd_mixer_close(handle); return CONTROL_ERROR; } if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer register error: %s\n", snd_strerror(err)); snd_mixer_close(handle); return CONTROL_ERROR; } err = snd_mixer_load(handle); if (err < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: mixer load error: %s\n", snd_strerror(err)); snd_mixer_close(handle); return CONTROL_ERROR; } elem = snd_mixer_find_selem(handle, sid); if (!elem) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: unable to find simple control '%s',%i\n", snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid)); snd_mixer_close(handle); return CONTROL_ERROR; } snd_mixer_selem_get_playback_volume_range(elem,&pmin,&pmax); f_multi = (100 / (float)(pmax - pmin)); if (cmd == AOCONTROL_SET_VOLUME) { set_vol = vol->left / f_multi + pmin + 0.5; //setting channels if ((err = snd_mixer_selem_set_playback_volume(elem, 0, set_vol)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting left channel, %s\n", snd_strerror(err)); return CONTROL_ERROR; } mp_msg(MSGT_AO,MSGL_DBG2,"left=%li, ", set_vol); set_vol = vol->right / f_multi + pmin + 0.5; if ((err = snd_mixer_selem_set_playback_volume(elem, 1, set_vol)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-control: error setting right channel, %s\n", snd_strerror(err)); return CONTROL_ERROR; } mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n", set_vol, pmin, pmax, f_multi); } else { snd_mixer_selem_get_playback_volume(elem, 0, &get_vol); vol->left = (get_vol - pmin) * f_multi; snd_mixer_selem_get_playback_volume(elem, 1, &get_vol); vol->right = (get_vol - pmin) * f_multi; mp_msg(MSGT_AO,MSGL_DBG2,"left=%f, right=%f\n",vol->left,vol->right); } snd_mixer_close(handle); return CONTROL_OK; } #endif } //end switch return(CONTROL_UNKNOWN); } static void parse_device (char *dest, char *src, int len) { char *tmp; strncpy (dest, src, len); while ((tmp = strrchr(dest, '.'))) tmp[0] = ','; while ((tmp = strrchr(dest, '='))) tmp[0] = ':'; } static void print_help () { mp_msg (MSGT_AO, MSGL_FATAL, "\n-ao alsa commandline help:\n" "Example: mplayer -ao alsa:mmap:device=hw=0.3\n" " sets mmap-mode and first card fourth device\n" "\nOptions:\n" " mmap\n" " Set memory-mapped mode, experimental\n" " noblock\n" " Sets non-blocking mode\n" " device=<device-name>\n" " Sets device (change , to . and : to =)\n"); } /* open & setup audio device return: 1=success 0=fail */ static int init(int rate_hz, int channels, int format, int flags) { int err; int cards = -1; snd_pcm_info_t *alsa_info; char *str_block_mode; int device_set = 0; int dir = 0; snd_pcm_uframes_t bufsize; char alsa_device[ALSA_DEVICE_SIZE + 1]; // make sure alsa_device is null-terminated even when using strncpy etc. memset(alsa_device, 0, ALSA_DEVICE_SIZE + 1); mp_msg(MSGT_AO,MSGL_V,"alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz, channels, audio_out_format_name(format)); alsa_handler = NULL; mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR); if ((err = snd_card_next(&cards)) < 0 || cards < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: no soundcards found: %s\n", snd_strerror(err)); return(0); } ao_data.samplerate = rate_hz; ao_data.bps = channels * rate_hz; ao_data.format = format; ao_data.channels = channels; ao_data.outburst = OUTBURST; switch (format) { case AFMT_S8: alsa_format = SND_PCM_FORMAT_S8; break; case AFMT_U8: alsa_format = SND_PCM_FORMAT_U8; break; case AFMT_U16_LE: alsa_format = SND_PCM_FORMAT_U16_LE; break; case AFMT_U16_BE: alsa_format = SND_PCM_FORMAT_U16_BE; break; #ifndef WORDS_BIGENDIAN case AFMT_AC3: #endif case AFMT_S16_LE: alsa_format = SND_PCM_FORMAT_S16_LE; break; #ifdef WORDS_BIGENDIAN case AFMT_AC3: #endif case AFMT_S16_BE: alsa_format = SND_PCM_FORMAT_S16_BE; break; case AFMT_S32_LE: alsa_format = SND_PCM_FORMAT_S32_LE; break; case AFMT_S32_BE: alsa_format = SND_PCM_FORMAT_S32_BE; break; case AFMT_FLOAT: alsa_format = SND_PCM_FORMAT_FLOAT_LE; break; default: alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 break; } //setting bw according to the input-format. resolution seems to be always s16_le or //u16_le so 32bit is probably obsolet. switch(alsa_format) { case SND_PCM_FORMAT_S8: case SND_PCM_FORMAT_U8: ao_data.bps *= 1; break; case SND_PCM_FORMAT_S16_LE: case SND_PCM_FORMAT_U16_LE: case SND_PCM_FORMAT_S16_BE: case SND_PCM_FORMAT_U16_BE: ao_data.bps *= 2; break; case SND_PCM_FORMAT_S32_LE: case SND_PCM_FORMAT_S32_BE: case SND_PCM_FORMAT_FLOAT_LE: ao_data.bps *= 4; break; case -1: mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: invalid format (%s) requested - output disabled\n", audio_out_format_name(format)); return(0); break; default: ao_data.bps *= 2; mp_msg(MSGT_AO,MSGL_WARN,"alsa-init: couldn't convert to right format. setting bps to: %d", ao_data.bps); } //subdevice parsing if (ao_subdevice) { int parse_err = 0; char *parse_pos = &ao_subdevice[0]; while (parse_pos[0] && !parse_err) { if (strncmp (parse_pos, "mmap", 4) == 0) { parse_pos = &parse_pos[4]; ao_mmap = 1; } else if (strncmp (parse_pos, "noblock", 7) == 0) { parse_pos = &parse_pos[7]; ao_noblock = 1; } else if (strncmp (parse_pos, "device=", 7) == 0) { int name_len; parse_pos = &parse_pos[7]; name_len = strcspn (parse_pos, ":"); if (name_len < 0 || name_len > ALSA_DEVICE_SIZE) { parse_err = 1; break; } parse_device (alsa_device, parse_pos, name_len); parse_pos = &parse_pos[name_len]; device_set = 1; } if (parse_pos[0] == ':') parse_pos = &parse_pos[1]; else if (parse_pos[0]) parse_err = 1; } if (parse_err) { print_help(); return 0; } } else { //end parsing ao_subdevice /* in any case for multichannel playback we should select * appropriate device */ switch (channels) { case 1: case 2: mp_msg(MSGT_AO,MSGL_V,"alsa-init: setup for 1/2 channel(s)\n"); break; case 4: if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) // hack - use the converter plugin strncpy(alsa_device, "plug:surround40", ALSA_DEVICE_SIZE); else strncpy(alsa_device, "surround40", ALSA_DEVICE_SIZE); device_set = 1; mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround40\n"); break; case 6: if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) strncpy(alsa_device, "plug:surround51", ALSA_DEVICE_SIZE); else strncpy(alsa_device, "surround51", ALSA_DEVICE_SIZE); device_set = 1; mp_msg(MSGT_AO,MSGL_V,"alsa-init: device set to surround51\n"); break; default: mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: %d channels are not supported\n",channels); } } if (!device_set) { /* switch for spdif * sets opening sequence for SPDIF * sets also the playback and other switches 'on the fly' * while opening the abstract alias for the spdif subdevice * 'iec958' */ if (format == AFMT_AC3) { unsigned char s[4]; switch (channels) { case 1: case 2: s[0] = IEC958_AES0_NONAUDIO | IEC958_AES0_CON_EMPHASIS_NONE; s[1] = IEC958_AES1_CON_ORIGINAL | IEC958_AES1_CON_PCM_CODER; s[2] = 0; s[3] = IEC958_AES3_CON_FS_48000; snprintf(alsa_device, ALSA_DEVICE_SIZE, "iec958:AES0=0x%x,AES1=0x%x,AES2=0x%x,AES3=0x%x", s[0], s[1], s[2], s[3]); mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels); break; case 4: strncpy(alsa_device, "surround40", ALSA_DEVICE_SIZE); break; case 6: strncpy(alsa_device, "surround51", ALSA_DEVICE_SIZE); break; default: mp_msg(MSGT_AO,MSGL_ERR,"alsa-spdif-init: %d channels are not supported\n", channels); } } else { int tmp_device, tmp_subdevice, err; snd_pcm_info_alloca(&alsa_info); if ((tmp_device = snd_pcm_info_get_device(alsa_info)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't get device\n"); } if ((tmp_subdevice = snd_pcm_info_get_subdevice(alsa_info)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't get subdevice\n"); } mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: got device=%i, subdevice=%i\n", tmp_device, tmp_subdevice); //we are setting here device to default cause it could be configured by the user //if its not set by the user, it defaults to hw:0,0 if ((err = snprintf(alsa_device, ALSA_DEVICE_SIZE, "default")) <= 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: can't write device-id\n"); } mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: %d soundcard%s found, using: %s\n", cards+1,(cards >= 0) ? "" : "s", alsa_device); } } else { mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: soundcard set to %s\n", alsa_device); } //setting modes for block or nonblock-mode if (ao_noblock) { open_mode = SND_PCM_NONBLOCK; set_block_mode = 1; str_block_mode = "nonblock-mode"; } else { open_mode = 0; set_block_mode = 0; str_block_mode = "block-mode"; } //sets buff/chunksize if its set manually if (ao_data.buffersize) { switch (ao_data.buffersize) { case 1: alsa_fragcount = 16; chunk_size = 512; mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n"); mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n"); break; case 2: alsa_fragcount = 8; chunk_size = 1024; mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 8192\n"); mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n"); break; case 3: alsa_fragcount = 32; chunk_size = 512; mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n"); mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 512\n"); break; case 4: alsa_fragcount = 16; chunk_size = 1024; mp_msg(MSGT_AO,MSGL_V,"alsa-init: buffersize set manually to 16384\n"); mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set manually to 1024\n"); break; default: alsa_fragcount = 16; if (ao_mmap) chunk_size = 512; else chunk_size = 1024; break; } } if (!alsa_handler) { //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, open_mode)) < 0) { if (err != -EBUSY && ao_noblock) { mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: open in nonblock-mode failed, trying to open in block-mode\n"); if ((err = snd_pcm_open(&alsa_handler, alsa_device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err)); return(0); } else { set_block_mode = 0; str_block_mode = "block-mode"; } } else { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: playback open error: %s\n", snd_strerror(err)); return(0); } } if ((err = snd_pcm_nonblock(alsa_handler, set_block_mode)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: error set block-mode %s\n", snd_strerror(err)); } else { mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opend in %s\n", str_block_mode); } snd_pcm_hw_params_alloca(&alsa_hwparams); snd_pcm_sw_params_alloca(&alsa_swparams); // setting hw-parameters if ((err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get initial parameters: %s\n", snd_strerror(err)); return(0); } if (ao_mmap) { snd_pcm_access_mask_t *mask = alloca(snd_pcm_access_mask_sizeof()); snd_pcm_access_mask_none(mask); snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_INTERLEAVED); snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_NONINTERLEAVED); snd_pcm_access_mask_set(mask, SND_PCM_ACCESS_MMAP_COMPLEX); err = snd_pcm_hw_params_set_access_mask(alsa_handler, alsa_hwparams, mask); mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: mmap set\n"); } else { err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); } if (err < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set access type: %s\n", snd_strerror(err)); return (0); } /* workaround for nonsupported formats sets default format to S16_LE if the given formats aren't supported */ if ((err = snd_pcm_hw_params_test_format(alsa_handler, alsa_hwparams, alsa_format)) < 0) { mp_msg(MSGT_AO,MSGL_INFO, "alsa-init: format %s are not supported by hardware, trying default\n", audio_out_format_name(format)); alsa_format = SND_PCM_FORMAT_S16_LE; ao_data.format = AFMT_S16_LE; ao_data.bps = channels * rate_hz * 2; } bytes_per_sample = ao_data.bps / ao_data.samplerate; //it should be here if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams, alsa_format)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set format: %s\n", snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams, ao_data.channels)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set channels: %s\n", snd_strerror(err)); } if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, &ao_data.samplerate, &dir)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set samplerate-2: %s\n", snd_strerror(err)); return(0); } #ifdef BUFFERTIME { int alsa_buffer_time = 500000; /* original 60 */ int alsa_period_time; alsa_period_time = alsa_buffer_time/4; if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams, &alsa_buffer_time, &dir)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set buffer time near: %s\n", snd_strerror(err)); return(0); } else alsa_buffer_time = err; if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, &alsa_period_time, &dir)) < 0) /* original: alsa_buffer_time/ao_data.bps */ { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set period time: %s\n", snd_strerror(err)); } mp_msg(MSGT_AO,MSGL_INFO,"alsa-init: buffer_time: %d, period_time :%d\n", alsa_buffer_time, err); } #endif//end SET_BUFFERTIME #ifdef SET_CHUNKSIZE { //set chunksize dir=0; if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams, &chunk_size, &dir)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periodsize(%d): %s\n", chunk_size, snd_strerror(err)); } else { mp_msg(MSGT_AO,MSGL_V,"alsa-init: chunksize set to %i\n", chunk_size); } if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams, &alsa_fragcount, &dir)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set periods: %s\n", snd_strerror(err)); } else { mp_msg(MSGT_AO,MSGL_V,"alsa-init: fragcount=%i\n", alsa_fragcount); } } #endif//end SET_CHUNKSIZE /* finally install hardware parameters */ if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set hw-parameters: %s\n", snd_strerror(err)); } // end setting hw-params // gets buffersize for control if ((err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get buffersize: %s\n", snd_strerror(err)); } else { ao_data.buffersize = bufsize * bytes_per_sample; mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize); } // setting sw-params (only avail-min) if noblocking mode was choosed if (ao_noblock) { if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to get parameters: %s\n", snd_strerror(err)); } //set min available frames to consider pcm ready (4) //increased for nonblock-mode should be set dynamically later if ((err = snd_pcm_sw_params_set_avail_min(alsa_handler, alsa_swparams, 4)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to set avail_min %s\n", snd_strerror(err)); } if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: unable to install sw-params\n"); } bits_per_sample = snd_pcm_format_physical_width(alsa_format); bits_per_frame = bits_per_sample * channels; chunk_bytes = chunk_size * bits_per_frame / 8; mp_msg(MSGT_AO,MSGL_V,"alsa-init: bits per sample (bps)=%i, bits per frame (bpf)=%i, chunk_bytes=%i\n",bits_per_sample,bits_per_frame,chunk_bytes);} //end swparams if ((err = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-init: pcm prepare error: %s\n", snd_strerror(err)); } mp_msg(MSGT_AO,MSGL_INFO,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n", ao_data.samplerate, ao_data.channels, bytes_per_sample, ao_data.buffersize, snd_pcm_format_description(alsa_format)); } // end switch alsa_handler (spdif) alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); return(1); } // end init /* close audio device */ static void uninit(int immed) { if (alsa_handler) { int err; if (!ao_noblock) { if ((err = snd_pcm_drop(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm drop error: %s\n", snd_strerror(err)); return; } } if ((err = snd_pcm_close(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: pcm close error: %s\n", snd_strerror(err)); return; } else { alsa_handler = NULL; mp_msg(MSGT_AO,MSGL_INFO,"alsa-uninit: pcm closed\n"); } } else { mp_msg(MSGT_AO,MSGL_ERR,"alsa-uninit: no handler defined!\n"); } } static void audio_pause() { int err; if (alsa_can_pause) { if ((err = snd_pcm_pause(alsa_handler, 1)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm pause error: %s\n", snd_strerror(err)); return; } mp_msg(MSGT_AO,MSGL_V,"alsa-pause: pause supported by hardware\n"); } else { if ((err = snd_pcm_drop(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-pause: pcm drop error: %s\n", snd_strerror(err)); return; } } } static void audio_resume() { int err; if (alsa_can_pause) { if ((err = snd_pcm_pause(alsa_handler, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm resume error: %s\n", snd_strerror(err)); return; } mp_msg(MSGT_AO,MSGL_V,"alsa-resume: resume supported by hardware\n"); } else { if ((err = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-resume: pcm prepare error: %s\n", snd_strerror(err)); return; } } } /* stop playing and empty buffers (for seeking/pause) */ static void reset() { int err; if ((err = snd_pcm_drop(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm drop error: %s\n", snd_strerror(err)); return; } if ((err = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-reset: pcm prepare error: %s\n", snd_strerror(err)); return; } return; } #ifdef USE_POLL static int wait_for_poll(snd_pcm_t *handle, struct pollfd *ufds, unsigned int count) { unsigned short revents; while (1) { poll(ufds, count, -1); snd_pcm_poll_descriptors_revents(handle, ufds, count, &revents); if (revents & POLLERR) return -EIO; if (revents & POLLOUT) return 0; } } #endif #ifndef timersub #define timersub(a, b, result) \ do { \ (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \ (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \ if ((result)->tv_usec < 0) { \ --(result)->tv_sec; \ (result)->tv_usec += 1000000; \ } \ } while (0) #endif /* I/O error handler */ static int xrun(u_char *str_mode) { int err; snd_pcm_status_t *status; snd_pcm_status_alloca(&status); if ((err = snd_pcm_status(alsa_handler, status))<0) { mp_msg(MSGT_AO,MSGL_ERR,"status error: %s", snd_strerror(err)); return(0); } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) { struct timeval now, diff, tstamp; gettimeofday(&now, 0); snd_pcm_status_get_trigger_tstamp(status, &tstamp); timersub(&now, &tstamp, &diff); mp_msg(MSGT_AO,MSGL_INFO,"alsa-%s: xrun of at least %.3f msecs. resetting stream\n", str_mode, diff.tv_sec * 1000 + diff.tv_usec / 1000.0); } if ((err = snd_pcm_prepare(alsa_handler))<0) { mp_msg(MSGT_AO,MSGL_ERR,"xrun: prepare error: %s", snd_strerror(err)); return(0); } return(1); /* ok, data should be accepted again */ } static int play_normal(void* data, int len); static int play_mmap(void* data, int len); static int play(void* data, int len, int flags) { int result; if (ao_mmap) result = play_mmap(data, len); else result = play_normal(data, len); return result; } /* plays 'len' bytes of 'data' returns: number of bytes played modified last at 29.06.02 by jp thanxs for marius <marius@rospot.com> for giving us the light ;) */ static int play_normal(void* data, int len) { //bytes_per_sample is always 4 for 2 chn S16_LE int num_frames = len / bytes_per_sample; char *output_samples = (char *)data; snd_pcm_sframes_t res = 0; //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len); if (!alsa_handler) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: device configuration error"); return 0; } while (num_frames > 0) { res = snd_pcm_writei(alsa_handler, (void *)output_samples, num_frames); if (res == -EAGAIN) { snd_pcm_wait(alsa_handler, 1000); } else if (res == -EPIPE) { /* underrun */ if (xrun("play") <= 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: xrun reset error"); return(0); } } else if (res == -ESTRPIPE) { /* suspend */ mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: pcm in suspend mode. trying to resume\n"); while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); } else if (res < 0) { mp_msg(MSGT_AO,MSGL_INFO,"alsa-play: unknown status, trying to reset soundcard\n"); if ((res = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: snd prepare error"); return(0); break; } } if (res > 0) { /* output_samples += ao_data.channels * res; */ output_samples += res * bytes_per_sample; num_frames -= res; } } //end while if (res < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: write error %s", snd_strerror(res)); return 0; } return len - len % bytes_per_sample; } /* mmap-mode mainly based on descriptions by Joshua Haberman <joshua@haberman.com> * 'An overview of the ALSA API' http://people.debian.org/~joshua/x66.html * and some help by Paul Davis <pbd@op.net> */ static int play_mmap(void* data, int len) { snd_pcm_sframes_t commitres, frames_available; snd_pcm_uframes_t frames_transmit, size, offset; const snd_pcm_channel_area_t *area; void *outbuffer; int result; #ifdef USE_POLL //seems not really be needed struct pollfd *ufds; int count; count = snd_pcm_poll_descriptors_count (alsa_handler); ufds = malloc(sizeof(struct pollfd) * count); snd_pcm_poll_descriptors(alsa_handler, ufds, count); //first wait_for_poll if (err = (wait_for_poll(alsa_handler, ufds, count) < 0)) { if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_XRUN || snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) { xrun("play"); } } #endif outbuffer = alloca(ao_data.buffersize); //don't trust get_space() ;) frames_available = snd_pcm_avail_update(alsa_handler) * bytes_per_sample; if (frames_available < 0) xrun("play"); if (frames_available < 4) { if (first) { first = 0; snd_pcm_start(alsa_handler); } else { //FIXME should break and return 0? snd_pcm_wait(alsa_handler, -1); first = 1; } } /* len is simply the available bufferspace got by get_space() * but real avail_buffer in frames is ab/bytes_per_sample */ size = len / bytes_per_sample; //mp_msg(MSGT_AO,MSGL_V,"len: %i size %i, f_avail %i, bps %i ...\n", len, size, frames_available, bytes_per_sample); frames_transmit = size; /* prepare areas and set sw-pointers * frames_transmit returns the real available buffer-size * sometimes != frames_available cause of ringbuffer 'emulation' */ snd_pcm_mmap_begin(alsa_handler, &area, &offset, &frames_transmit); /* this is specific to interleaved streams (or non-interleaved * streams with only one channel) */ outbuffer = ((char *) area->addr + (area->first + area->step * offset) / 8); //8 //write data memcpy(outbuffer, data, (frames_transmit * bytes_per_sample)); commitres = snd_pcm_mmap_commit(alsa_handler, offset, frames_transmit); if (commitres < 0 || commitres != frames_transmit) { if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_XRUN || snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) { xrun("play"); } } //mp_msg(MSGT_AO,MSGL_V,"mmap ft: %i, cres: %i\n", frames_transmit, commitres); /* err = snd_pcm_area_copy(&area, offset, &data, offset, len, alsa_format); */ /* if (err < 0) { */ /* mp_msg(MSGT_AO,MSGL_ERR,"area-copy-error\n"); */ /* return 0; */ /* } */ //calculate written frames! result = commitres * bytes_per_sample; /* if (verbose) { */ /* if (len == result) */ /* mp_msg(MSGT_AO,MSGL_V,"result: %i, frames written: %i ...\n", result, frames_transmit); */ /* else */ /* mp_msg(MSGT_AO,MSGL_V,"result: %i, frames written: %i, result != len ...\n", result, frames_transmit); */ /* } */ //mplayer doesn't like -result if (result < 0) result = 0; #ifdef USE_POLL free(ufds); #endif return result; } /* how many byes are free in the buffer */ static int get_space() { snd_pcm_status_t *status; int ret; char *str_status; //snd_pcm_sframes_t avail_frames = 0; snd_pcm_status_alloca(&status); if ((ret = snd_pcm_status(alsa_handler, status)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-space: cannot get pcm status: %s\n", snd_strerror(ret)); return(0); } switch(snd_pcm_status_get_state(status)) { case SND_PCM_STATE_OPEN: str_status = "open"; ret = snd_pcm_status_get_avail(status) * bytes_per_sample; break; case SND_PCM_STATE_PREPARED: str_status = "prepared"; first = 1; ret = snd_pcm_status_get_avail(status) * bytes_per_sample; if (ret == 0) //ugly workaround for hang in mmap-mode ret = 10; break; case SND_PCM_STATE_RUNNING: ret = snd_pcm_status_get_avail(status) * bytes_per_sample; //avail_frames = snd_pcm_avail_update(alsa_handler) * bytes_per_sample; if (str_status != "open" && str_status != "prepared") str_status = "running"; break; case SND_PCM_STATE_PAUSED: mp_msg(MSGT_AO,MSGL_V,"alsa-space: paused"); str_status = "paused"; ret = 0; break; case SND_PCM_STATE_XRUN: xrun("space"); str_status = "xrun"; first = 1; ret = 0; break; default: str_status = "undefined"; ret = snd_pcm_status_get_avail(status) * bytes_per_sample; if (ret <= 0) { xrun("space"); } } if (snd_pcm_status_get_state(status) != SND_PCM_STATE_RUNNING) mp_msg(MSGT_AO,MSGL_V,"alsa-space: free space = %i, %s --\n", ret, str_status); if (ret < 0) { mp_msg(MSGT_AO,MSGL_ERR,"negative value!!\n"); ret = 0; } // workaround for too small value returned if (ret < MIN_CHUNK_SIZE) ret = 0; return(ret); } /* delay in seconds between first and last sample in buffer */ static float get_delay() { if (alsa_handler) { snd_pcm_status_t *status; float ret; snd_pcm_status_alloca(&status); if ((ret = snd_pcm_status(alsa_handler, status)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,"alsa-delay: cannot get pcm status: %s\n", snd_strerror(ret)); } switch(snd_pcm_status_get_state(status)) { case SND_PCM_STATE_OPEN: case SND_PCM_STATE_PREPARED: case SND_PCM_STATE_RUNNING: ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate; break; default: ret = 0; } if (ret < 0) ret = 0; return(ret); } else { return(0); } }