Mercurial > mplayer.hg
view libao2/ao_alsa.c @ 32302:6d3212dd47b2
Drop two unnecessary Xlib.h #includes and use helper functions.
author | diego |
---|---|
date | Mon, 27 Sep 2010 17:18:39 +0000 |
parents | 64ba1daa147a |
children | f60b1fd07712 |
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/* * 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 <joy@streamminister.de> * additional AC-3 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. * * 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 <errno.h> #include <sys/time.h> #include <stdlib.h> #include <stdarg.h> #include <ctype.h> #include <math.h> #include <string.h> #include <alloca.h> #include "config.h" #include "subopt-helper.h" #include "mixer.h" #include "mp_msg.h" #include "help_mp.h" #define ALSA_PCM_NEW_HW_PARAMS_API #define ALSA_PCM_NEW_SW_PARAMS_API #ifdef HAVE_SYS_ASOUNDLIB_H #include <sys/asoundlib.h> #elif defined(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 "libaf/af_format.h" static const ao_info_t info = { "ALSA-0.9.x-1.x audio output", "alsa", "Alex Beregszaszi, Zsolt Barat <joy@streamminister.de>", "under development" }; 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; static size_t bytes_per_sample; static int alsa_can_pause; #define ALSA_DEVICE_SIZE 256 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); tmp[sizeof tmp - 1] = '\0'; 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) { switch(cmd) { case AOCONTROL_QUERY_FORMAT: return CONTROL_TRUE; 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; char *mix_name = "PCM"; char *card = "default"; int mix_index = 0; long pmin, pmax; long get_vol, set_vol; float f_multi; if(AF_FORMAT_IS_AC3(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_msg(MSGT_AO,MSGL_ERR, MSGTR_AO_ALSA_InvalidMixerIndexDefaultingToZero); 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; } if ((err = snd_mixer_open(&handle, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerOpenError, snd_strerror(err)); return CONTROL_ERROR; } if ((err = snd_mixer_attach(handle, card)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerAttachError, 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,MSGTR_AO_ALSA_MixerRegisterError, snd_strerror(err)); snd_mixer_close(handle); return CONTROL_ERROR; } err = snd_mixer_load(handle); if (err < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_MixerLoadError, 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,MSGTR_AO_ALSA_UnableToFindSimpleControl, 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, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingLeftChannel, snd_strerror(err)); snd_mixer_close(handle); 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, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSettingRightChannel, snd_strerror(err)); snd_mixer_close(handle); return CONTROL_ERROR; } mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n", set_vol, pmin, pmax, f_multi); if (snd_mixer_selem_has_playback_switch(elem)) { int lmute = (vol->left == 0.0); int rmute = (vol->right == 0.0); if (snd_mixer_selem_has_playback_switch_joined(elem)) { lmute = rmute = lmute && rmute; } else { snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, !rmute); } snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, !lmute); } } else { 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); } snd_mixer_close(handle); return CONTROL_OK; } } //end switch return CONTROL_UNKNOWN; } 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_msg (MSGT_AO, MSGL_FATAL, MSGTR_AO_ALSA_CommandlineHelp); } 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 (!try_ac3 || err < 0) err = snd_pcm_open(&alsa_handler, device, SND_PCM_STREAM_PLAYBACK, open_mode); return err; } /* open & setup audio device return: 1=success 0=fail */ static int init(int rate_hz, int channels, int format, int flags) { unsigned int alsa_buffer_time = 500000; /* 0.5 s */ unsigned int alsa_fragcount = 16; 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, channels, format); alsa_handler = NULL; #if SND_LIB_VERSION >= 0x010005 mp_msg(MSGT_AO,MSGL_V,"alsa-init: using ALSA %s\n", snd_asoundlib_version()); #else mp_msg(MSGT_AO,MSGL_V,"alsa-init: compiled for ALSA-%s\n", SND_LIB_VERSION_STR); #endif snd_lib_error_set_handler(alsa_error_handler); ao_data.samplerate = rate_hz; ao_data.format = format; ao_data.channels = channels; 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: alsa_format = SND_PCM_FORMAT_S16_LE; break; case AF_FORMAT_AC3_BE: case AF_FORMAT_S16_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; case AF_FORMAT_MU_LAW: alsa_format = SND_PCM_FORMAT_MU_LAW; break; case AF_FORMAT_A_LAW: alsa_format = SND_PCM_FORMAT_A_LAW; 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_AC3(format)) { device.str = "iec958"; mp_msg(MSGT_AO,MSGL_V,"alsa-spdif-init: playing AC3, %i channels\n", channels); } else /* in any case for multichannel playback we should select * appropriate device */ switch (channels) { 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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ChannelsNotSupported,channels); } 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); if (!alsa_handler) { int open_mode = block ? 0 : SND_PCM_NONBLOCK; int isac3 = AF_FORMAT_IS_AC3(format); //modes = 0, SND_PCM_NONBLOCK, SND_PCM_ASYNC if ((err = try_open_device(alsa_device, open_mode, isac3)) < 0) { if (err != -EBUSY && !block) { mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_OpenInNonblockModeFailed); if ((err = try_open_device(alsa_device, 0, isac3)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err)); return 0; } } else { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PlaybackOpenError, snd_strerror(err)); return 0; } } if ((err = snd_pcm_nonblock(alsa_handler, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_ErrorSetBlockMode, snd_strerror(err)); } else { mp_msg(MSGT_AO,MSGL_V,"alsa-init: pcm opened in blocking mode\n"); } 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,MSGTR_AO_ALSA_UnableToGetInitialParameters, snd_strerror(err)); return 0; } 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,MSGTR_AO_ALSA_UnableToSetAccessType, 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, MSGTR_AO_ALSA_FormatNotSupportedByHardware, 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 ao_data.format = AF_FORMAT_S16_LE; } if ((err = snd_pcm_hw_params_set_format(alsa_handler, alsa_hwparams, alsa_format)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetFormat, snd_strerror(err)); return 0; } if ((err = snd_pcm_hw_params_set_channels_near(alsa_handler, alsa_hwparams, &ao_data.channels)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetChannels, snd_strerror(err)); return 0; } /* 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 */ #if SND_LIB_VERSION >= 0x010009 if ((err = snd_pcm_hw_params_set_rate_resample(alsa_handler, alsa_hwparams, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToDisableResampling, snd_strerror(err)); return 0; } #endif if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, &ao_data.samplerate, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSamplerate2, snd_strerror(err)); return 0; } bytes_per_sample = af_fmt2bits(ao_data.format) / 8; bytes_per_sample *= ao_data.channels; ao_data.bps = ao_data.samplerate * bytes_per_sample; if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams, &alsa_buffer_time, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetBufferTimeNear, snd_strerror(err)); return 0; } if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams, &alsa_fragcount, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetPeriods, snd_strerror(err)); return 0; } /* finally install hardware parameters */ if ((err = snd_pcm_hw_params(alsa_handler, alsa_hwparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetHwParameters, snd_strerror(err)); return 0; } // 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,MSGTR_AO_ALSA_UnableToGetBufferSize, snd_strerror(err)); return 0; } else { ao_data.buffersize = bufsize * bytes_per_sample; mp_msg(MSGT_AO,MSGL_V,"alsa-init: got buffersize=%i\n", ao_data.buffersize); } if ((err = snd_pcm_hw_params_get_period_size(alsa_hwparams, &chunk_size, NULL)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetPeriodSize, snd_strerror(err)); return 0; } else { 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 */ if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters, snd_strerror(err)); return 0; } #if SND_LIB_VERSION >= 0x000901 if ((err = snd_pcm_sw_params_get_boundary(alsa_swparams, &boundary)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetBoundary, snd_strerror(err)); return 0; } #else boundary = 0x7fffffff; #endif /* start playing when one period has been written */ if ((err = snd_pcm_sw_params_set_start_threshold(alsa_handler, alsa_swparams, chunk_size)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStartThreshold, snd_strerror(err)); return 0; } /* disable underrun reporting */ if ((err = snd_pcm_sw_params_set_stop_threshold(alsa_handler, alsa_swparams, boundary)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetStopThreshold, snd_strerror(err)); return 0; } #if SND_LIB_VERSION >= 0x000901 /* play silence when there is an underrun */ if ((err = snd_pcm_sw_params_set_silence_size(alsa_handler, alsa_swparams, boundary)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToSetSilenceSize, snd_strerror(err)); return 0; } #endif if ((err = snd_pcm_sw_params(alsa_handler, alsa_swparams)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_UnableToGetSwParameters, snd_strerror(err)); return 0; } /* end setting sw-params */ mp_msg(MSGT_AO,MSGL_V,"alsa: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n", ao_data.samplerate, ao_data.channels, (int)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 (!immed) snd_pcm_drain(alsa_handler); if ((err = snd_pcm_close(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmCloseError, snd_strerror(err)); return; } else { alsa_handler = NULL; mp_msg(MSGT_AO,MSGL_V,"alsa-uninit: pcm closed\n"); } } else { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_NoHandlerDefined); } } static void audio_pause(void) { int err; if (alsa_can_pause) { if ((err = snd_pcm_pause(alsa_handler, 1)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPauseError, 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,MSGTR_AO_ALSA_PcmDropError, snd_strerror(err)); return; } } } static void audio_resume(void) { int err; if (snd_pcm_state(alsa_handler) == SND_PCM_STATE_SUSPENDED) { mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume); while ((err = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); } if (alsa_can_pause) { if ((err = snd_pcm_pause(alsa_handler, 0)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmResumeError, 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,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err)); return; } } } /* stop playing and empty buffers (for seeking/pause) */ static void reset(void) { int err; if ((err = snd_pcm_drop(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err)); return; } if ((err = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(err)); return; } return; } /* 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(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_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_DeviceConfigurationError); 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_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_PcmInSuspendModeTryingResume); while ((res = snd_pcm_resume(alsa_handler)) == -EAGAIN) sleep(1); } if (res < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_WriteError, snd_strerror(res)); mp_msg(MSGT_AO,MSGL_INFO,MSGTR_AO_ALSA_TryingToResetSoundcard); if ((res = snd_pcm_prepare(alsa_handler)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_PcmPrepareError, snd_strerror(res)); return 0; break; } } } while (res == 0); return res < 0 ? res : res * bytes_per_sample; } /* how many byes are free in the buffer */ static int get_space(void) { snd_pcm_status_t *status; int ret; snd_pcm_status_alloca(&status); if ((ret = snd_pcm_status(alsa_handler, status)) < 0) { mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_ALSA_CannotGetPcmStatus, snd_strerror(ret)); return 0; } ret = snd_pcm_status_get_avail(status) * bytes_per_sample; if (ret > ao_data.buffersize) // Buffer underrun? ret = ao_data.buffersize; return ret; } /* 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_delay(alsa_handler, &delay) < 0) return 0; if (delay < 0) { /* underrun - move the application pointer forward to catch up */ #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */ snd_pcm_forward(alsa_handler, -delay); #endif delay = 0; } return (float)delay / (float)ao_data.samplerate; } else { return 0; } }