Mercurial > audlegacy-plugins
view src/alsa/audio.c @ 866:d8b8b36d2709 trunk
[svn] - prevent skin list overlap when it's first populated and user quickly switchs to other tabs in the prefswin
author | giacomo |
---|---|
date | Fri, 16 Mar 2007 18:47:24 -0700 |
parents | 279846578fdc |
children | c7daa70ffe7e |
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/* XMMS - ALSA output plugin * Copyright (C) 2001-2003 Matthieu Sozeau <mattam@altern.org> * Copyright (C) 1998-2003 Peter Alm, Mikael Alm, Olle Hallnas, * Thomas Nilsson and 4Front Technologies * Copyright (C) 1999-2005 Haavard Kvaalen * Copyright (C) 2005 Takashi Iwai * * This program 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. * * This program 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 this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. */ /* * CHANGES * * 2005.01.05 Takashi Iwai <tiwai@suse.de> * Impelemented the multi-threaded mode with an audio-thread. * Many fixes and cleanups. */ #include "alsa.h" #include <ctype.h> #include <audacious/xconvert.h> static snd_pcm_t *alsa_pcm; static snd_output_t *logs; /* Number of bytes that we have received from the input plugin */ static guint64 alsa_total_written; /* Number of bytes that we have sent to the sound card */ static guint64 alsa_hw_written; static guint64 output_time_offset; /* device buffer/period sizes in bytes */ static int hw_buffer_size, hw_period_size; /* in output bytes */ static int hw_buffer_size_in, hw_period_size_in; /* in input bytes */ /* Set/Get volume */ static snd_mixer_elem_t *pcm_element; static snd_mixer_t *mixer; static gboolean going, paused, mixer_start = TRUE; static gboolean prebuffer, remove_prebuffer; static gboolean alsa_can_pause; /* for audio thread */ static GThread *audio_thread; /* audio loop thread */ static int thread_buffer_size; /* size of intermediate buffer in bytes */ static char *thread_buffer; /* audio intermediate buffer */ static int rd_index, wr_index; /* current read/write position in int-buffer */ static gboolean pause_request; /* pause status currently requested */ static int flush_request; /* flush status (time) currently requested */ static int prebuffer_size; GMutex *alsa_mutex; static guint mixer_timeout; struct snd_format { unsigned int rate; unsigned int channels; snd_pcm_format_t format; AFormat xmms_format; int sample_bits; int bps; }; static struct snd_format *inputf = NULL; static struct snd_format *effectf = NULL; static struct snd_format *outputf = NULL; static int alsa_setup(struct snd_format *f); static void alsa_write_audio(char *data, int length); static void alsa_cleanup_mixer(void); static int get_thread_buffer_filled(void); static struct snd_format * snd_format_from_xmms(AFormat fmt, int rate, int channels); static struct xmms_convert_buffers *convertb; static convert_func_t alsa_convert_func; static convert_channel_func_t alsa_stereo_convert_func; static convert_freq_func_t alsa_frequency_convert_func; static const struct { AFormat xmms; snd_pcm_format_t alsa; } format_table[] = {{FMT_S16_LE, SND_PCM_FORMAT_S16_LE}, {FMT_S16_BE, SND_PCM_FORMAT_S16_BE}, {FMT_S16_NE, SND_PCM_FORMAT_S16}, {FMT_U16_LE, SND_PCM_FORMAT_U16_LE}, {FMT_U16_BE, SND_PCM_FORMAT_U16_BE}, {FMT_U16_NE, SND_PCM_FORMAT_U16}, {FMT_U8, SND_PCM_FORMAT_U8}, {FMT_S8, SND_PCM_FORMAT_S8}, }; static void debug(char *str, ...) G_GNUC_PRINTF(1, 2); static void debug(char *str, ...) { va_list args; if (alsa_cfg.debug) { va_start(args, str); g_logv(NULL, G_LOG_LEVEL_MESSAGE, str, args); va_end(args); } } int alsa_playing(void) { if (!going || paused || alsa_pcm == NULL) return FALSE; return snd_pcm_state(alsa_pcm) == SND_PCM_STATE_RUNNING; } static int xrun_recover(void) { if (alsa_cfg.debug) { snd_pcm_status_t *alsa_status; snd_pcm_status_alloca(&alsa_status); if (snd_pcm_status(alsa_pcm, alsa_status) < 0) g_warning("xrun_recover(): snd_pcm_status() failed"); else { printf("Status:\n"); snd_pcm_status_dump(alsa_status, logs); } } return snd_pcm_prepare(alsa_pcm); } static int suspend_recover(void) { int err; while ((err = snd_pcm_resume(alsa_pcm)) == -EAGAIN) /* wait until suspend flag is released */ xmms_usleep(1000000); if (err < 0) { g_warning("alsa_handle_error(): " "snd_pcm_resume() failed."); return snd_pcm_prepare(alsa_pcm); } return err; } /* handle generic errors */ static int alsa_handle_error(int err) { switch (err) { case -EPIPE: return xrun_recover(); case -ESTRPIPE: return suspend_recover(); } return err; } /* update and get the available space on h/w buffer (in frames) */ static snd_pcm_sframes_t alsa_get_avail(void) { snd_pcm_sframes_t ret; if (alsa_pcm == NULL) return 0; while ((ret = snd_pcm_avail_update(alsa_pcm)) < 0) { ret = alsa_handle_error(ret); if (ret < 0) { g_warning("alsa_get_avail(): snd_pcm_avail_update() failed: %s", snd_strerror(-ret)); return 0; } } return ret; } /* get the free space on buffer */ int alsa_free(void) { if (remove_prebuffer && prebuffer) { prebuffer = FALSE; remove_prebuffer = FALSE; } if (prebuffer) remove_prebuffer = TRUE; return thread_buffer_size - get_thread_buffer_filled() - 1; } /* do pause operation */ static void alsa_do_pause(gboolean p) { if (paused == p) return; if (alsa_pcm) { if (alsa_can_pause) snd_pcm_pause(alsa_pcm, p); else if (p) { snd_pcm_drop(alsa_pcm); snd_pcm_prepare(alsa_pcm); } } paused = p; } void alsa_pause(short p) { debug("alsa_pause"); pause_request = p; } /* close PCM and release associated resources */ static void alsa_close_pcm(void) { if (alsa_pcm) { int err; snd_pcm_drop(alsa_pcm); if ((err = snd_pcm_close(alsa_pcm)) < 0) g_warning("alsa_pcm_close() failed: %s", snd_strerror(-err)); alsa_pcm = NULL; } } void alsa_close(void) { if (!going) return; debug("Closing device"); going = 0; g_thread_join(audio_thread); g_mutex_lock(alsa_mutex); /* alsa_loop locks alsa_mutex! */ alsa_cleanup_mixer(); xmms_convert_buffers_destroy(convertb); convertb = NULL; g_free(inputf); inputf = NULL; g_free(effectf); effectf = NULL; g_free(outputf); outputf = NULL; alsa_save_config(); if (alsa_cfg.debug) snd_output_close(logs); debug("Device closed"); g_mutex_unlock(alsa_mutex); } /* reopen ALSA PCM */ static int alsa_reopen(struct snd_format *f) { /* remember the current position */ output_time_offset += (alsa_hw_written * 1000) / outputf->bps; alsa_hw_written = 0; alsa_close_pcm(); return alsa_setup(f); } /* do flush (drop) operation */ static void alsa_do_flush(int time) { if (alsa_pcm) { snd_pcm_drop(alsa_pcm); snd_pcm_prepare(alsa_pcm); } /* correct the offset */ output_time_offset = time; alsa_total_written = (guint64) time * inputf->bps / 1000; rd_index = wr_index = alsa_hw_written = 0; } void alsa_flush(int time) { flush_request = time; while (flush_request != -1) xmms_usleep(10000); } static void parse_mixer_name(char *str, char **name, int *index) { char *end; while (isspace(*str)) str++; if ((end = strchr(str, ',')) != NULL) { *name = g_strndup(str, end - str); end++; *index = atoi(end); } else { *name = g_strdup(str); *index = 0; } } int alsa_get_mixer(snd_mixer_t **mixer, int card) { char *dev; int err; debug("alsa_get_mixer"); dev = g_strdup_printf("hw:%i", card); if ((err = snd_mixer_open(mixer, 0)) < 0) { g_warning("alsa_get_mixer(): Failed to open empty mixer: %s", snd_strerror(-err)); mixer = NULL; return -1; } if ((err = snd_mixer_attach(*mixer, dev)) < 0) { g_warning("alsa_get_mixer(): Attaching to mixer %s failed: %s", dev, snd_strerror(-err)); return -1; } if ((err = snd_mixer_selem_register(*mixer, NULL, NULL)) < 0) { g_warning("alsa_get_mixer(): Failed to register mixer: %s", snd_strerror(-err)); return -1; } if ((err = snd_mixer_load(*mixer)) < 0) { g_warning("alsa_get_mixer(): Failed to load mixer: %s", snd_strerror(-err)); return -1; } g_free(dev); return (*mixer != NULL); } static snd_mixer_elem_t* alsa_get_mixer_elem(snd_mixer_t *mixer, char *name, int index) { snd_mixer_selem_id_t *selem_id; snd_mixer_elem_t* elem; snd_mixer_selem_id_alloca(&selem_id); if (index != -1) snd_mixer_selem_id_set_index(selem_id, index); if (name != NULL) snd_mixer_selem_id_set_name(selem_id, name); elem = snd_mixer_find_selem(mixer, selem_id); return elem; } static int alsa_setup_mixer(void) { char *name; long int a, b; long alsa_min_vol, alsa_max_vol; int err, index; debug("alsa_setup_mixer"); if ((err = alsa_get_mixer(&mixer, alsa_cfg.mixer_card)) < 0) return err; parse_mixer_name(alsa_cfg.mixer_device, &name, &index); pcm_element = alsa_get_mixer_elem(mixer, name, index); g_free(name); if (!pcm_element) { g_warning("alsa_setup_mixer(): Failed to find mixer element: %s", alsa_cfg.mixer_device); return -1; } /* * Work around a bug in alsa-lib up to 1.0.0rc2 where the * new range don't take effect until the volume is changed. * This hack should be removed once we depend on Alsa 1.0.0. */ snd_mixer_selem_get_playback_volume(pcm_element, SND_MIXER_SCHN_FRONT_LEFT, &a); snd_mixer_selem_get_playback_volume(pcm_element, SND_MIXER_SCHN_FRONT_RIGHT, &b); snd_mixer_selem_get_playback_volume_range(pcm_element, &alsa_min_vol, &alsa_max_vol); snd_mixer_selem_set_playback_volume_range(pcm_element, 0, 100); if (alsa_max_vol == 0) { pcm_element = NULL; return -1; } if (!alsa_cfg.soft_volume) alsa_set_volume(a * 100 / alsa_max_vol, b * 100 / alsa_max_vol); debug("alsa_setup_mixer: end"); return 0; } static int alsa_mixer_timeout(void *data) { if (mixer) { snd_mixer_close(mixer); mixer = NULL; pcm_element = NULL; } mixer_timeout = 0; mixer_start = TRUE; g_message("alsa mixer timed out"); return FALSE; } static void alsa_cleanup_mixer(void) { pcm_element = NULL; if (mixer) { snd_mixer_close(mixer); mixer = NULL; } } void alsa_get_volume(int *l, int *r) { long ll = *l, lr = *r; if (mixer_start) { alsa_setup_mixer(); mixer_start = FALSE; } if (alsa_cfg.soft_volume) { *l = alsa_cfg.vol.left; *r = alsa_cfg.vol.right; } if (!pcm_element) return; snd_mixer_handle_events(mixer); if (!alsa_cfg.soft_volume) { snd_mixer_selem_get_playback_volume(pcm_element, SND_MIXER_SCHN_FRONT_LEFT, &ll); snd_mixer_selem_get_playback_volume(pcm_element, SND_MIXER_SCHN_FRONT_RIGHT, &lr); *l = ll; *r = lr; } if (mixer_timeout) gtk_timeout_remove(mixer_timeout); mixer_timeout = gtk_timeout_add(5000, alsa_mixer_timeout, NULL); } void alsa_set_volume(int l, int r) { if (alsa_cfg.soft_volume) { alsa_cfg.vol.left = l; alsa_cfg.vol.right = r; return; } if (!pcm_element) return; if (snd_mixer_selem_is_playback_mono(pcm_element)) { if (l > r) snd_mixer_selem_set_playback_volume(pcm_element, SND_MIXER_SCHN_MONO, l); else snd_mixer_selem_set_playback_volume(pcm_element, SND_MIXER_SCHN_MONO, r); } else { snd_mixer_selem_set_playback_volume(pcm_element, SND_MIXER_SCHN_FRONT_LEFT, l); snd_mixer_selem_set_playback_volume(pcm_element, SND_MIXER_SCHN_FRONT_RIGHT, r); } if (snd_mixer_selem_has_playback_switch(pcm_element)) { snd_mixer_selem_set_playback_switch(pcm_element, SND_MIXER_SCHN_FRONT_LEFT, l != 0); snd_mixer_selem_set_playback_switch(pcm_element, SND_MIXER_SCHN_FRONT_RIGHT, r != 0); } } /* * audio stuff */ /* return the size of audio data filled in the audio thread buffer */ static int get_thread_buffer_filled(void) { if (wr_index >= rd_index) return wr_index - rd_index; return thread_buffer_size - (rd_index - wr_index); } int alsa_get_output_time(void) { snd_pcm_sframes_t delay; guint64 bytes = alsa_hw_written; if (!going || alsa_pcm == NULL) return 0; if (!snd_pcm_delay(alsa_pcm, &delay)) { unsigned int d = snd_pcm_frames_to_bytes(alsa_pcm, delay); if (bytes < d) bytes = 0; else bytes -= d; } return output_time_offset + (bytes * 1000) / outputf->bps; } int alsa_get_written_time(void) { if (!going) return 0; return (alsa_total_written * 1000) / inputf->bps; } #define STEREO_ADJUST(type, type2, endian) \ do { \ type *ptr = data; \ for (i = 0; i < length; i += 4) \ { \ *ptr = type2##_TO_##endian(type2##_FROM_## endian(*ptr) * \ alsa_cfg.vol.left / 100); \ ptr++; \ *ptr = type2##_TO_##endian(type2##_FROM_##endian(*ptr) * \ alsa_cfg.vol.right / 100); \ ptr++; \ } \ } while (0) #define MONO_ADJUST(type, type2, endian) \ do { \ type *ptr = data; \ for (i = 0; i < length; i += 2) \ { \ *ptr = type2##_TO_##endian(type2##_FROM_## endian(*ptr) * \ vol / 100); \ ptr++; \ } \ } while (0) #define VOLUME_ADJUST(type, type2, endian) \ do { \ if (channels == 2) \ STEREO_ADJUST(type, type2, endian); \ else \ MONO_ADJUST(type, type2, endian); \ } while (0) #define STEREO_ADJUST8(type) \ do { \ type *ptr = data; \ for (i = 0; i < length; i += 2) \ { \ *ptr = *ptr * alsa_cfg.vol.left / 100; \ ptr++; \ *ptr = *ptr * alsa_cfg.vol.right / 100; \ ptr++; \ } \ } while (0) #define MONO_ADJUST8(type) \ do { \ type *ptr = data; \ for (i = 0; i < length; i++) \ { \ *ptr = *ptr * vol / 100; \ ptr++; \ } \ } while (0) #define VOLUME_ADJUST8(type) \ do { \ if (channels == 2) \ STEREO_ADJUST8(type); \ else \ MONO_ADJUST8(type); \ } while (0) static void volume_adjust(void* data, int length, AFormat fmt, int channels) { int i, vol; if ((alsa_cfg.vol.left == 100 && alsa_cfg.vol.right == 100) || (channels == 1 && (alsa_cfg.vol.left == 100 || alsa_cfg.vol.right == 100))) return; vol = MAX(alsa_cfg.vol.left, alsa_cfg.vol.right); switch (fmt) { case FMT_S16_LE: VOLUME_ADJUST(gint16, GINT16, LE); break; case FMT_U16_LE: VOLUME_ADJUST(guint16, GUINT16, LE); break; case FMT_S16_BE: VOLUME_ADJUST(gint16, GINT16, BE); break; case FMT_U16_BE: VOLUME_ADJUST(guint16, GUINT16, BE); break; case FMT_S8: VOLUME_ADJUST8(gint8); break; case FMT_U8: VOLUME_ADJUST8(guint8); break; default: g_warning("volue_adjust(): unhandled format: %d", fmt); break; } } /* transfer data to audio h/w; length is given in bytes * * data can be modified via effect plugin, rate conversion or * software volume before passed to audio h/w */ static void alsa_do_write(gpointer data, int length) { EffectPlugin *ep = NULL; int new_freq; int new_chn; AFormat f; if (paused) return; new_freq = inputf->rate; new_chn = inputf->channels; f = inputf->xmms_format; if (effects_enabled() && (ep = get_current_effect_plugin()) && ep->query_format) ep->query_format(&f, &new_freq, &new_chn); if (f != effectf->xmms_format || (unsigned int)new_freq != effectf->rate || (unsigned int)new_chn != effectf->channels) { debug("Changing audio format for effect plugin"); g_free(effectf); effectf = snd_format_from_xmms(f, new_freq, new_chn); if (alsa_reopen(effectf) < 0) { /* fatal error... */ alsa_close(); return; } } if (ep) length = ep->mod_samples(&data, length, inputf->xmms_format, inputf->rate, inputf->channels); if (alsa_convert_func != NULL) length = alsa_convert_func(convertb, &data, length); if (alsa_stereo_convert_func != NULL) length = alsa_stereo_convert_func(convertb, &data, length); if (alsa_frequency_convert_func != NULL) length = alsa_frequency_convert_func(convertb, &data, length, effectf->rate, outputf->rate); if (alsa_cfg.soft_volume) volume_adjust(data, length, outputf->xmms_format, outputf->channels); alsa_write_audio(data, length); } /* write callback */ void alsa_write(gpointer data, int length) { int cnt; char *src = (char *)data; remove_prebuffer = FALSE; alsa_total_written += length; while (length > 0) { int wr; cnt = MIN(length, thread_buffer_size - wr_index); memcpy(thread_buffer + wr_index, src, cnt); wr = (wr_index + cnt) % thread_buffer_size; wr_index = wr; length -= cnt; src += cnt; } } /* transfer data to audio h/w via normal write */ static void alsa_write_audio(char *data, int length) { snd_pcm_sframes_t written_frames; while (length > 0) { int frames = snd_pcm_bytes_to_frames(alsa_pcm, length); written_frames = snd_pcm_writei(alsa_pcm, data, frames); if (written_frames > 0) { int written = snd_pcm_frames_to_bytes(alsa_pcm, written_frames); length -= written; data += written; alsa_hw_written += written; } else { int err = alsa_handle_error((int)written_frames); if (err < 0) { g_warning("alsa_write_audio(): write error: %s", snd_strerror(-err)); break; } } } } /* transfer audio data from thread buffer to h/w */ static void alsa_write_out_thread_data(void) { gint length, cnt, avail; length = MIN(hw_period_size_in, get_thread_buffer_filled()); avail = snd_pcm_frames_to_bytes(alsa_pcm, alsa_get_avail()); length = MIN(length, avail); while (length > 0) { int rd; cnt = MIN(length, thread_buffer_size - rd_index); alsa_do_write(thread_buffer + rd_index, cnt); rd = (rd_index + cnt) % thread_buffer_size; rd_index = rd; length -= cnt; } } /* audio thread loop */ /* FIXME: proper lock? */ static void *alsa_loop(void *arg) { int npfds = snd_pcm_poll_descriptors_count(alsa_pcm); struct pollfd *pfds; unsigned short *revents; g_mutex_lock(alsa_mutex); if (npfds <= 0) goto _error; pfds = alloca(sizeof(*pfds) * npfds); revents = alloca(sizeof(*revents) * npfds); while (going && alsa_pcm) { if (get_thread_buffer_filled() > prebuffer_size) prebuffer = FALSE; if (!paused && !prebuffer && get_thread_buffer_filled() > hw_period_size_in) { snd_pcm_poll_descriptors(alsa_pcm, pfds, npfds); if (poll(pfds, npfds, 10) > 0) { /* * need to check revents. poll() with * dmix returns a postive value even * if no data is available */ int i; snd_pcm_poll_descriptors_revents(alsa_pcm, pfds, npfds, revents); for (i = 0; i < npfds; i++) if (revents[i] & POLLOUT) { alsa_write_out_thread_data(); break; } } } else xmms_usleep(10000); if (pause_request != paused) alsa_do_pause(pause_request); if (flush_request != -1) { alsa_do_flush(flush_request); flush_request = -1; prebuffer = TRUE; } } _error: g_mutex_unlock(alsa_mutex); alsa_close_pcm(); g_free(thread_buffer); thread_buffer = NULL; g_thread_exit(NULL); return(NULL); } /* open callback */ int alsa_open(AFormat fmt, int rate, int nch) { debug("Opening device"); inputf = snd_format_from_xmms(fmt, rate, nch); effectf = snd_format_from_xmms(fmt, rate, nch); if (alsa_cfg.debug) snd_output_stdio_attach(&logs, stdout, 0); if (alsa_setup(inputf) < 0) { alsa_close(); return 0; } g_mutex_lock(alsa_mutex); if (!mixer) alsa_setup_mixer(); convertb = xmms_convert_buffers_new(); output_time_offset = 0; alsa_total_written = alsa_hw_written = 0; going = TRUE; paused = FALSE; prebuffer = TRUE; remove_prebuffer = FALSE; thread_buffer_size = (guint64)cfg.output_buffer_size * inputf->bps / 1000; if (thread_buffer_size < hw_buffer_size) thread_buffer_size = hw_buffer_size * 2; if (thread_buffer_size < 8192) thread_buffer_size = 8192; prebuffer_size = thread_buffer_size / 2; if (prebuffer_size < 8192) prebuffer_size = 8192; thread_buffer_size += hw_buffer_size; thread_buffer_size -= thread_buffer_size % hw_period_size; thread_buffer = g_malloc0(thread_buffer_size); wr_index = rd_index = 0; pause_request = FALSE; flush_request = -1; g_mutex_unlock(alsa_mutex); audio_thread = g_thread_create((GThreadFunc)alsa_loop, NULL, TRUE, NULL); return 1; } static struct snd_format * snd_format_from_xmms(AFormat fmt, int rate, int channels) { struct snd_format *f = g_malloc(sizeof(struct snd_format)); size_t i; f->xmms_format = fmt; f->format = SND_PCM_FORMAT_UNKNOWN; for (i = 0; i < sizeof(format_table) / sizeof(format_table[0]); i++) if (format_table[i].xmms == fmt) { f->format = format_table[i].alsa; break; } /* Get rid of _NE */ for (i = 0; i < sizeof(format_table) / sizeof(format_table[0]); i++) if (format_table[i].alsa == f->format) { f->xmms_format = format_table[i].xmms; break; } f->rate = rate; f->channels = channels; f->sample_bits = snd_pcm_format_physical_width(f->format); f->bps = (rate * f->sample_bits * channels) >> 3; return f; } static int format_from_alsa(snd_pcm_format_t fmt) { size_t i; for (i = 0; i < sizeof(format_table) / sizeof(format_table[0]); i++) if (format_table[i].alsa == fmt) return format_table[i].xmms; g_warning("Unsupported format: %s", snd_pcm_format_name(fmt)); return -1; } static int alsa_setup(struct snd_format *f) { int err; snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; uint alsa_buffer_time; unsigned int alsa_period_time; snd_pcm_uframes_t alsa_buffer_size, alsa_period_size; debug("alsa_setup"); alsa_convert_func = NULL; alsa_stereo_convert_func = NULL; alsa_frequency_convert_func = NULL; g_free(outputf); outputf = snd_format_from_xmms(f->xmms_format, f->rate, f->channels); debug("Opening device: %s", alsa_cfg.pcm_device); /* FIXME: Can snd_pcm_open() return EAGAIN? */ if ((err = snd_pcm_open(&alsa_pcm, alsa_cfg.pcm_device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK)) < 0) { g_warning("alsa_setup(): Failed to open pcm device (%s): %s", alsa_cfg.pcm_device, snd_strerror(-err)); alsa_pcm = NULL; g_free(outputf); outputf = NULL; return -1; } /* doesn't care about non-blocking */ /* snd_pcm_nonblock(alsa_pcm, 0); */ if (alsa_cfg.debug) { snd_pcm_info_t *info; int alsa_card, alsa_device, alsa_subdevice; snd_pcm_info_alloca(&info); snd_pcm_info(alsa_pcm, info); alsa_card = snd_pcm_info_get_card(info); alsa_device = snd_pcm_info_get_device(info); alsa_subdevice = snd_pcm_info_get_subdevice(info); printf("Card %i, Device %i, Subdevice %i\n", alsa_card, alsa_device, alsa_subdevice); } snd_pcm_hw_params_alloca(&hwparams); if ((err = snd_pcm_hw_params_any(alsa_pcm, hwparams)) < 0) { g_warning("alsa_setup(): No configuration available for " "playback: %s", snd_strerror(-err)); return -1; } if ((err = snd_pcm_hw_params_set_access(alsa_pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { g_warning("alsa_setup(): Cannot set direct write mode: %s", snd_strerror(-err)); return -1; } if ((err = snd_pcm_hw_params_set_format(alsa_pcm, hwparams, outputf->format)) < 0) { /* * Try if one of these format work (one of them should work * on almost all soundcards) */ snd_pcm_format_t formats[] = {SND_PCM_FORMAT_S16_LE, SND_PCM_FORMAT_S16_BE, SND_PCM_FORMAT_U8}; size_t i; for (i = 0; i < sizeof(formats) / sizeof(formats[0]); i++) { if (snd_pcm_hw_params_set_format(alsa_pcm, hwparams, formats[i]) == 0) { outputf->format = formats[i]; break; } } if (outputf->format != f->format) { outputf->xmms_format = format_from_alsa(outputf->format); debug("Converting format from %d to %d", f->xmms_format, outputf->xmms_format); alsa_convert_func = xmms_convert_get_func(outputf->xmms_format, f->xmms_format); if (alsa_convert_func == NULL) return -1; } else { g_warning("alsa_setup(): Sample format not " "available for playback: %s", snd_strerror(-err)); return -1; } } snd_pcm_hw_params_set_channels_near(alsa_pcm, hwparams, &outputf->channels); if (outputf->channels != f->channels) { debug("Converting channels from %d to %d", f->channels, outputf->channels); alsa_stereo_convert_func = xmms_convert_get_channel_func(outputf->xmms_format, outputf->channels, f->channels); if (alsa_stereo_convert_func == NULL) return -1; } snd_pcm_hw_params_set_rate_near(alsa_pcm, hwparams, &outputf->rate, 0); if (outputf->rate == 0) { g_warning("alsa_setup(): No usable samplerate available."); return -1; } if (outputf->rate != f->rate) { debug("Converting samplerate from %d to %d", f->rate, outputf->rate); alsa_frequency_convert_func = xmms_convert_get_frequency_func(outputf->xmms_format, outputf->channels); if (alsa_frequency_convert_func == NULL) return -1; } outputf->sample_bits = snd_pcm_format_physical_width(outputf->format); outputf->bps = (outputf->rate * outputf->sample_bits * outputf->channels) >> 3; alsa_buffer_time = alsa_cfg.buffer_time * 1000; if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_pcm, hwparams, &alsa_buffer_time, 0)) < 0) { g_warning("alsa_setup(): Set buffer time failed: %s.", snd_strerror(-err)); return -1; } alsa_period_time = alsa_cfg.period_time * 1000; if ((err = snd_pcm_hw_params_set_period_time_near(alsa_pcm, hwparams, &alsa_period_time, 0)) < 0) { g_warning("alsa_setup(): Set period time failed: %s.", snd_strerror(-err)); return -1; } if (snd_pcm_hw_params(alsa_pcm, hwparams) < 0) { if (alsa_cfg.debug) snd_pcm_hw_params_dump(hwparams, logs); g_warning("alsa_setup(): Unable to install hw params"); return -1; } if ((err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size)) < 0) { g_warning("alsa_setup(): snd_pcm_hw_params_get_buffer_size() " "failed: %s", snd_strerror(-err)); return -1; } if ((err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0)) < 0) { g_warning("alsa_setup(): snd_pcm_hw_params_get_period_size() " "failed: %s", snd_strerror(-err)); return -1; } alsa_can_pause = snd_pcm_hw_params_can_pause(hwparams); snd_pcm_sw_params_alloca(&swparams); snd_pcm_sw_params_current(alsa_pcm, swparams); /* This has effect for non-mmap only */ if ((err = snd_pcm_sw_params_set_start_threshold(alsa_pcm, swparams, alsa_buffer_size - alsa_period_size) < 0)) g_warning("alsa_setup(): setting start " "threshold failed: %s", snd_strerror(-err)); if (snd_pcm_sw_params(alsa_pcm, swparams) < 0) { g_warning("alsa_setup(): Unable to install sw params"); return -1; } if (alsa_cfg.debug) { snd_pcm_sw_params_dump(swparams, logs); snd_pcm_dump(alsa_pcm, logs); } hw_buffer_size = snd_pcm_frames_to_bytes(alsa_pcm, alsa_buffer_size); hw_period_size = snd_pcm_frames_to_bytes(alsa_pcm, alsa_period_size); if (inputf->bps != outputf->bps) { int align = (inputf->sample_bits * inputf->channels) / 8; hw_buffer_size_in = ((guint64)hw_buffer_size * inputf->bps + outputf->bps/2) / outputf->bps; hw_period_size_in = ((guint64)hw_period_size * inputf->bps + outputf->bps/2) / outputf->bps; hw_buffer_size_in -= hw_buffer_size_in % align; hw_period_size_in -= hw_period_size_in % align; } else { hw_buffer_size_in = hw_buffer_size; hw_period_size_in = hw_period_size; } debug("Device setup: buffer time: %i, size: %i.", alsa_buffer_time, hw_buffer_size); debug("Device setup: period time: %i, size: %i.", alsa_period_time, hw_period_size); debug("bits per sample: %i; frame size: %i; Bps: %i", snd_pcm_format_physical_width(outputf->format), (int)snd_pcm_frames_to_bytes(alsa_pcm, 1), outputf->bps); return 0; } void alsa_tell(AFormat * fmt, gint * rate, gint * nch) { (*fmt) = inputf->xmms_format; (*rate) = inputf->rate; (*nch) = inputf->channels; }