/* * ALSA 0.9.x-1.x audio output driver * * Copyright (C) 2004 Alex Beregszaszi * * modified for real ALSA 0.9.0 support by Zsolt Barat * additional AC-3 passthrough support by Andy Lo A Foe * 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. * * 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 #include #include #include #include #include #include "config.h" #include "core/subopt-helper.h" #include "audio/mixer.h" #include "core/mp_msg.h" #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #include #include "ao.h" #include "audio_out_internal.h" #include "audio/format.h" #include "audio/reorder_ch.h" static const ao_info_t info = { "ALSA-0.9.x-1.x audio output", "alsa", "Alex Beregszaszi, Zsolt Barat ", "under development" }; LIBAO_EXTERN(alsa) static snd_pcm_t *alsa_handler; static snd_pcm_format_t alsa_format; #define BUFFER_TIME 500000 // 0.5 s #define FRAGCOUNT 16 static size_t bytes_per_sample; static int alsa_can_pause; static snd_pcm_sframes_t prepause_frames; static float delay_before_pause; #define ALSA_DEVICE_SIZE 256 #define CHECK_ALSA_ERROR(message) \ do { \ if (err < 0) { \ mp_msg(MSGT_VO, MSGL_ERR, "[AO_ALSA] %s: %s\n", \ (message), snd_strerror(err)); \ goto alsa_error; \ } \ } while (0) static void alsa_error_handler(const char *file, int line, const char *function, int err, const char *format, ...) { char tmp[0xc00]; va_list va; va_start(va, format); vsnprintf(tmp, sizeof tmp, format, va); va_end(va); if (err) { mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s: %s\n", file, line, function, tmp, snd_strerror(err)); } else { mp_msg(MSGT_AO, MSGL_ERR, "[AO_ALSA] alsa-lib: %s:%i:(%s) %s\n", file, line, function, tmp); } } /* to set/get/query special features/parameters */ static int control(int cmd, void *arg) { snd_mixer_t *handle = NULL; switch (cmd) { case AOCONTROL_GET_MUTE: case AOCONTROL_SET_MUTE: case AOCONTROL_GET_VOLUME: case AOCONTROL_SET_VOLUME: { int err; snd_mixer_elem_t *elem; snd_mixer_selem_id_t *sid; char *mix_name = "Master"; char *card = "default"; int mix_index = 0; long pmin, pmax; long get_vol, set_vol; float f_multi; if (AF_FORMAT_IS_IEC61937(ao_data.format)) return CONTROL_TRUE; 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_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Invalid mixer index. Defaulting to 0.\n"); mix_index = 0; } } } if (mixer_device) card = mixer_device; //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; } err = snd_mixer_open(&handle, 0); CHECK_ALSA_ERROR("Mixer open error"); err = snd_mixer_attach(handle, card); CHECK_ALSA_ERROR("Mixer attach error"); err = snd_mixer_selem_register(handle, NULL, NULL); CHECK_ALSA_ERROR("Mixer register error"); err = snd_mixer_load(handle); CHECK_ALSA_ERROR("Mixer load error"); elem = snd_mixer_find_selem(handle, sid); if (!elem) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Unable to find simple control '%s',%i.\n", snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid)); goto alsa_error; } snd_mixer_selem_get_playback_volume_range(elem, &pmin, &pmax); f_multi = (100 / (float)(pmax - pmin)); switch (cmd) { case AOCONTROL_SET_VOLUME: { ao_control_vol_t *vol = arg; set_vol = vol->left / f_multi + pmin + 0.5; //setting channels err = snd_mixer_selem_set_playback_volume (elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol); CHECK_ALSA_ERROR("Error setting left channel"); mp_msg(MSGT_AO, MSGL_DBG2, "left=%li, ", set_vol); set_vol = vol->right / f_multi + pmin + 0.5; err = snd_mixer_selem_set_playback_volume (elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol); CHECK_ALSA_ERROR("Error setting right channel"); mp_msg(MSGT_AO, MSGL_DBG2, "right=%li, pmin=%li, pmax=%li, mult=%f\n", set_vol, pmin, pmax, f_multi); break; } case AOCONTROL_GET_VOLUME: { ao_control_vol_t *vol = arg; snd_mixer_selem_get_playback_volume (elem, SND_MIXER_SCHN_FRONT_LEFT, &get_vol); vol->left = (get_vol - pmin) * f_multi; snd_mixer_selem_get_playback_volume (elem, SND_MIXER_SCHN_FRONT_RIGHT, &get_vol); vol->right = (get_vol - pmin) * f_multi; mp_msg(MSGT_AO, MSGL_DBG2, "left=%f, right=%f\n", vol->left, vol->right); break; } case AOCONTROL_SET_MUTE: { bool *mute = arg; if (!snd_mixer_selem_has_playback_switch(elem)) goto alsa_error; if (!snd_mixer_selem_has_playback_switch_joined(elem)) { snd_mixer_selem_set_playback_switch (elem, SND_MIXER_SCHN_FRONT_RIGHT, !*mute); } snd_mixer_selem_set_playback_switch (elem, SND_MIXER_SCHN_FRONT_LEFT, !*mute); break; } case AOCONTROL_GET_MUTE: { bool *mute = arg; if (!snd_mixer_selem_has_playback_switch(elem)) goto alsa_error; int tmp = 1; snd_mixer_selem_get_playback_switch (elem, SND_MIXER_SCHN_FRONT_LEFT, &tmp); *mute = !tmp; if (!snd_mixer_selem_has_playback_switch_joined(elem)) { snd_mixer_selem_get_playback_switch (elem, SND_MIXER_SCHN_FRONT_RIGHT, &tmp); *mute &= !tmp; } break; } } snd_mixer_close(handle); return CONTROL_OK; } } //end switch return CONTROL_UNKNOWN; alsa_error: if (handle) snd_mixer_close(handle); return CONTROL_ERROR; } static void parse_device(char *dest, const char *src, int len) { char *tmp; memmove(dest, src, len); dest[len] = 0; while ((tmp = strrchr(dest, '.'))) tmp[0] = ','; while ((tmp = strrchr(dest, '='))) tmp[0] = ':'; } static void print_help(void) { mp_tmsg(MSGT_AO, MSGL_FATAL, "\n[AO_ALSA] -ao alsa commandline help:\n" \ "[AO_ALSA] Example: mpv -ao alsa:device=hw=0.3\n" \ "[AO_ALSA] Sets first card fourth hardware device.\n\n" \ "[AO_ALSA] Options:\n" \ "[AO_ALSA] noblock\n" \ "[AO_ALSA] Opens device in non-blocking mode.\n" \ "[AO_ALSA] device=\n" \ "[AO_ALSA] Sets device (change , to . and : to =)\n"); } static int str_maxlen(void *strp) { strarg_t *str = strp; return str->len <= ALSA_DEVICE_SIZE; } static int try_open_device(const char *device, int open_mode, int try_ac3) { int err, len; char *ac3_device, *args; if (try_ac3) { /* to set the non-audio bit, use AES0=6 */ len = strlen(device); ac3_device = malloc(len + 7 + 1); if (!ac3_device) return -ENOMEM; strcpy(ac3_device, device); args = strchr(ac3_device, ':'); if (!args) { /* no existing parameters: add it behind device name */ strcat(ac3_device, ":AES0=6"); } else { do { ++args; } while (isspace(*args)); if (*args == '\0') { /* ":" but no parameters */ strcat(ac3_device, "AES0=6"); } else if (*args != '{') { /* a simple list of parameters: add it at the end of the list */ strcat(ac3_device, ",AES0=6"); } else { /* parameters in config syntax: add it inside the { } block */ do { --len; } while (len > 0 && isspace(ac3_device[len])); if (ac3_device[len] == '}') strcpy(ac3_device + len, " AES0=6}"); } } err = snd_pcm_open (&alsa_handler, ac3_device, SND_PCM_STREAM_PLAYBACK, open_mode); free(ac3_device); if (!err) return 0; } return snd_pcm_open (&alsa_handler, device, SND_PCM_STREAM_PLAYBACK, open_mode); } /* open & setup audio device return: 1=success 0=fail */ static int init(int rate_hz, const struct mp_chmap *channels, int format, int flags) { int err; int block; strarg_t device; snd_pcm_uframes_t chunk_size; snd_pcm_uframes_t bufsize; snd_pcm_uframes_t boundary; const opt_t subopts[] = { {"block", OPT_ARG_BOOL, &block, NULL}, {"device", OPT_ARG_STR, &device, str_maxlen}, {NULL} }; 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, %x\n", rate_hz, ao_data.channels.num, format); alsa_handler = NULL; mp_msg(MSGT_AO, MSGL_V, "alsa-init: using ALSA %s\n", snd_asoundlib_version()); prepause_frames = 0; delay_before_pause = 0; snd_lib_error_set_handler(alsa_error_handler); switch (format) { case AF_FORMAT_S8: alsa_format = SND_PCM_FORMAT_S8; break; case AF_FORMAT_U8: alsa_format = SND_PCM_FORMAT_U8; break; case AF_FORMAT_U16_LE: alsa_format = SND_PCM_FORMAT_U16_LE; break; case AF_FORMAT_U16_BE: alsa_format = SND_PCM_FORMAT_U16_BE; break; case AF_FORMAT_AC3_LE: case AF_FORMAT_S16_LE: case AF_FORMAT_IEC61937_LE: alsa_format = SND_PCM_FORMAT_S16_LE; break; case AF_FORMAT_AC3_BE: case AF_FORMAT_S16_BE: case AF_FORMAT_IEC61937_BE: alsa_format = SND_PCM_FORMAT_S16_BE; break; case AF_FORMAT_U32_LE: alsa_format = SND_PCM_FORMAT_U32_LE; break; case AF_FORMAT_U32_BE: alsa_format = SND_PCM_FORMAT_U32_BE; break; case AF_FORMAT_S32_LE: alsa_format = SND_PCM_FORMAT_S32_LE; break; case AF_FORMAT_S32_BE: alsa_format = SND_PCM_FORMAT_S32_BE; break; case AF_FORMAT_U24_LE: alsa_format = SND_PCM_FORMAT_U24_3LE; break; case AF_FORMAT_U24_BE: alsa_format = SND_PCM_FORMAT_U24_3BE; break; case AF_FORMAT_S24_LE: alsa_format = SND_PCM_FORMAT_S24_3LE; break; case AF_FORMAT_S24_BE: alsa_format = SND_PCM_FORMAT_S24_3BE; break; case AF_FORMAT_FLOAT_LE: alsa_format = SND_PCM_FORMAT_FLOAT_LE; break; case AF_FORMAT_FLOAT_BE: alsa_format = SND_PCM_FORMAT_FLOAT_BE; break; default: alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1 break; } //subdevice parsing // set defaults block = 1; /* 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 (AF_FORMAT_IS_IEC61937(format)) { device.str = "iec958"; mp_msg(MSGT_AO, MSGL_V, "alsa-spdif-init: playing AC3/iec61937/iec958, %i channels\n", ao_data.channels.num); } else { /* in any case for multichannel playback we should select * appropriate device */ switch (ao_data.channels.num) { case 1: case 2: device.str = "default"; 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 device.str = "plug:surround40"; else device.str = "surround40"; mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround40\n"); break; case 6: if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) device.str = "plug:surround51"; else device.str = "surround51"; mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround51\n"); break; case 8: if (alsa_format == SND_PCM_FORMAT_FLOAT_LE) device.str = "plug:surround71"; else device.str = "surround71"; mp_msg(MSGT_AO, MSGL_V, "alsa-init: device set to surround71\n"); break; default: device.str = "default"; mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] %d channels are not supported.\n", ao_data.channels.num); } } device.len = strlen(device.str); if (subopt_parse(ao_subdevice, subopts) != 0) { print_help(); return 0; } parse_device(alsa_device, device.str, device.len); mp_msg(MSGT_AO, MSGL_V, "alsa-init: using device %s\n", alsa_device); alsa_can_pause = 1; if (!alsa_handler) { int open_mode = block ? 0 : SND_PCM_NONBLOCK; int isac3 = AF_FORMAT_IS_IEC61937(format); //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC err = try_open_device(alsa_device, open_mode, isac3); if (err < 0) { if (err != -EBUSY && !block) { mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Open in nonblock-mode " "failed, trying to open in block-mode.\n"); err = try_open_device(alsa_device, 0, isac3); } CHECK_ALSA_ERROR("Playback open error"); } err = snd_pcm_nonblock(alsa_handler, 0); if (err < 0) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AL_ALSA] Error setting block-mode %s.\n", snd_strerror(err)); } else { mp_msg(MSGT_AO, MSGL_V, "alsa-init: pcm opened in blocking mode\n"); } snd_pcm_hw_params_t *alsa_hwparams; snd_pcm_sw_params_t *alsa_swparams; snd_pcm_hw_params_alloca(&alsa_hwparams); snd_pcm_sw_params_alloca(&alsa_swparams); // setting hw-parameters err = snd_pcm_hw_params_any(alsa_handler, alsa_hwparams); CHECK_ALSA_ERROR("Unable to get initial parameters"); err = snd_pcm_hw_params_set_access (alsa_handler, alsa_hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); CHECK_ALSA_ERROR("Unable to set access type"); /* workaround for nonsupported formats sets default format to S16_LE if the given formats aren't supported */ err = snd_pcm_hw_params_test_format (alsa_handler, alsa_hwparams, alsa_format); if (err < 0) { mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Format %s is not supported " "by hardware, trying default.\n", af_fmt2str_short(format)); alsa_format = SND_PCM_FORMAT_S16_LE; if (AF_FORMAT_IS_AC3(ao_data.format)) ao_data.format = AF_FORMAT_AC3_LE; else if (AF_FORMAT_IS_IEC61937(ao_data.format)) ao_data.format = AF_FORMAT_IEC61937_LE; else ao_data.format = AF_FORMAT_S16_LE; } err = snd_pcm_hw_params_set_format (alsa_handler, alsa_hwparams, alsa_format); CHECK_ALSA_ERROR("Unable to set format"); int num_channels = ao_data.channels.num; err = snd_pcm_hw_params_set_channels_near (alsa_handler, alsa_hwparams, &num_channels); CHECK_ALSA_ERROR("Unable to set channels"); mp_chmap_from_channels(&ao_data.channels, num_channels); if (!AF_FORMAT_IS_IEC61937(format)) mp_chmap_reorder_to_alsa(&ao_data.channels); /* workaround for buggy rate plugin (should be fixed in ALSA 1.0.11) prefer our own resampler, since that allows users to choose the resampler, even per file if desired */ err = snd_pcm_hw_params_set_rate_resample (alsa_handler, alsa_hwparams, 0); CHECK_ALSA_ERROR("Unable to disable resampling"); err = snd_pcm_hw_params_set_rate_near (alsa_handler, alsa_hwparams, &ao_data.samplerate, NULL); CHECK_ALSA_ERROR("Unable to set samplerate-2"); bytes_per_sample = af_fmt2bits(ao_data.format) / 8; bytes_per_sample *= ao_data.channels.num; ao_data.bps = ao_data.samplerate * bytes_per_sample; err = snd_pcm_hw_params_set_buffer_time_near (alsa_handler, alsa_hwparams, &(unsigned int){BUFFER_TIME}, NULL); CHECK_ALSA_ERROR("Unable to set buffer time near"); err = snd_pcm_hw_params_set_periods_near (alsa_handler, alsa_hwparams, &(unsigned int){FRAGCOUNT}, NULL); CHECK_ALSA_ERROR("Unable to set periods"); /* finally install hardware parameters */ err = snd_pcm_hw_params(alsa_handler, alsa_hwparams); CHECK_ALSA_ERROR("Unable to set hw-parameters"); // end setting hw-params // gets buffersize for control err = snd_pcm_hw_params_get_buffer_size(alsa_hwparams, &bufsize); CHECK_ALSA_ERROR("Unable to get buffersize"); ao_data.buffersize = bufsize * bytes_per_sample; mp_msg(MSGT_AO, MSGL_V, "alsa-init: got buffersize=%i\n", ao_data.buffersize); err = snd_pcm_hw_params_get_period_size (alsa_hwparams, &chunk_size, NULL); CHECK_ALSA_ERROR("Unable to get period size"); mp_msg(MSGT_AO, MSGL_V, "alsa-init: got period size %li\n", chunk_size); ao_data.outburst = chunk_size * bytes_per_sample; /* setting software parameters */ err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams); CHECK_ALSA_ERROR("Unable to get sw-parameters"); err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary); CHECK_ALSA_ERROR("Unable to get boundary"); /* start playing when one period has been written */ err = snd_pcm_sw_params_set_start_threshold (alsa_handler, alsa_swparams, chunk_size); CHECK_ALSA_ERROR("Unable to set start threshold"); /* disable underrun reporting */ err = snd_pcm_sw_params_set_stop_threshold (alsa_handler, alsa_swparams, boundary); CHECK_ALSA_ERROR("Unable to set stop threshold"); /* play silence when there is an underrun */ err = snd_pcm_sw_params_set_silence_size (alsa_handler, alsa_swparams, boundary); CHECK_ALSA_ERROR("Unable to set silence size"); err = snd_pcm_sw_params(alsa_handler, alsa_swparams); CHECK_ALSA_ERROR("Unable to get sw-parameters"); /* end setting sw-params */ alsa_can_pause = snd_pcm_hw_params_can_pause(alsa_hwparams); mp_msg(MSGT_AO, MSGL_V, "alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n", ao_data.samplerate, ao_data.channels.num, (int)bytes_per_sample, ao_data.buffersize, snd_pcm_format_description(alsa_format)); } // end switch alsa_handler (spdif) return 1; alsa_error: return 0; } // end init /* close audio device */ static void uninit(int immed) { if (alsa_handler) { int err; if (!immed) snd_pcm_drain(alsa_handler); err = snd_pcm_close(alsa_handler); CHECK_ALSA_ERROR("pcm close error"); alsa_handler = NULL; mp_msg(MSGT_AO, MSGL_V, "alsa-uninit: pcm closed\n"); } else { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] No handler defined!\n"); } alsa_error: ; } static void audio_pause(void) { int err; if (alsa_can_pause) { delay_before_pause = get_delay(); err = snd_pcm_pause(alsa_handler, 1); CHECK_ALSA_ERROR("pcm pause error"); mp_msg(MSGT_AO, MSGL_V, "alsa-pause: pause supported by hardware\n"); } else { if (snd_pcm_delay(alsa_handler, &prepause_frames) < 0 || prepause_frames < 0) prepause_frames = 0; delay_before_pause = prepause_frames / (float)ao_data.samplerate; err = snd_pcm_drop(alsa_handler); CHECK_ALSA_ERROR("pcm drop error"); } alsa_error: ; } static void audio_resume(void) { int err; if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) { mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); } if (alsa_can_pause) { err = snd_pcm_pause(alsa_handler, 0); CHECK_ALSA_ERROR("pcm resume error"); mp_msg(MSGT_AO, MSGL_V, "alsa-resume: resume supported by hardware\n"); } else { err = snd_pcm_prepare(alsa_handler); CHECK_ALSA_ERROR("pcm prepare error"); if (prepause_frames) { void *silence = calloc(prepause_frames, bytes_per_sample); play(silence, prepause_frames * bytes_per_sample, 0); free(silence); } } alsa_error: ; } /* stop playing and empty buffers (for seeking/pause) */ static void reset(void) { int err; prepause_frames = 0; delay_before_pause = 0; err = snd_pcm_drop(alsa_handler); CHECK_ALSA_ERROR("pcm prepare error"); err = snd_pcm_prepare(alsa_handler); CHECK_ALSA_ERROR("pcm prepare error"); alsa_error: ; } /* plays 'len' bytes of 'data' returns: number of bytes played modified last at 29.06.02 by jp thanxs for marius for giving us the light ;) */ static int play(void *data, int len, int flags) { int num_frames; snd_pcm_sframes_t res = 0; if (!(flags & AOPLAY_FINAL_CHUNK)) len = len / ao_data.outburst * ao_data.outburst; num_frames = len / bytes_per_sample; //mp_msg(MSGT_AO,MSGL_ERR,"alsa-play: frames=%i, len=%i\n",num_frames,len); if (!alsa_handler) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Device configuration error."); return 0; } if (num_frames == 0) return 0; do { res = snd_pcm_writei(alsa_handler, data, num_frames); if (res == -EINTR) { /* nothing to do */ res = 0; } else if (res == -ESTRPIPE) { /* suspend */ mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Pcm in suspend mode, trying to resume.\n"); while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); } if (res < 0) { mp_tmsg(MSGT_AO, MSGL_ERR, "[AO_ALSA] Write error: %s\n", snd_strerror(res)); mp_tmsg(MSGT_AO, MSGL_INFO, "[AO_ALSA] Trying to reset soundcard.\n"); res = snd_pcm_prepare(alsa_handler); int err = res; CHECK_ALSA_ERROR("pcm prepare error"); res = 0; } } while (res == 0); return res < 0 ? 0 : res * bytes_per_sample; alsa_error: return 0; } /* how many byes are free in the buffer */ static int get_space(void) { snd_pcm_status_t *status; int err; snd_pcm_status_alloca(&status); err = snd_pcm_status(alsa_handler, status); CHECK_ALSA_ERROR("cannot get pcm status"); unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample; if (space > ao_data.buffersize) // Buffer underrun? space = ao_data.buffersize; return space; alsa_error: return 0; } /* delay in seconds between first and last sample in buffer */ static float get_delay(void) { if (alsa_handler) { snd_pcm_sframes_t delay; if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_PAUSED) return delay_before_pause; if (snd_pcm_delay(alsa_handler, &delay) < 0) return 0; if (delay < 0) { /* underrun - move the application pointer forward to catch up */ snd_pcm_forward(alsa_handler, -delay); delay = 0; } return (float)delay / (float)ao_data.samplerate; } else return 0; }