Mercurial > audlegacy-plugins
view src/alsa-ng/alsa-core.c @ 3181:dc22f5fa8f2f
alsa-ng: Remove another debug message.
(http://jira.atheme.org/browse/AUDPLUG-15)
author | William Pitcock <nenolod@atheme.org> |
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date | Thu, 11 Jun 2009 20:21:16 -0500 |
parents | 2e7a64bb33cd |
children | 631d217913e0 |
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/* * Audacious ALSA Plugin (-ng) * Copyright (c) 2009 William Pitcock <nenolod@dereferenced.org> * * 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. */ #define ALSA_DEBUG #include "alsa-stdinc.h" static snd_pcm_t *pcm_handle = NULL; static alsaplug_ringbuf_t pcm_ringbuf; static gboolean pcm_going = FALSE; static GThread *audio_thread = NULL; static gint bps; static gsize wr_total = 0; static gsize wr_hwframes = 0; static gint flush_request, paused; static GMutex *pcm_pause_mutex, *pcm_state_mutex; static GCond *pcm_pause_cond, *pcm_state_cond, *pcm_flush_cond; /******************************************************************************** * ALSA Mixer setting functions. * ********************************************************************************/ static snd_mixer_t *amixer = NULL; static gboolean mixer_ready = FALSE; static snd_mixer_elem_t * alsaplug_get_mixer_elem_by_name(snd_mixer_t *mixer, gchar *name) { snd_mixer_selem_id_t *selem_id; snd_mixer_elem_t *elem; g_return_val_if_fail(mixer != NULL, NULL); g_return_val_if_fail(name != NULL, NULL); snd_mixer_selem_id_alloca(&selem_id); snd_mixer_selem_id_set_name(selem_id, name); elem = snd_mixer_find_selem(mixer, selem_id); if (elem == NULL) return NULL; snd_mixer_selem_set_playback_volume_range(elem, 0, 100); return elem; } /* try to determine the best choice... may need tweaking. --nenolod */ static snd_mixer_elem_t * alsaplug_guess_mixer_elem(snd_mixer_t *mixer) { gchar *elem_names[] = { "Wave", "PCM", "Front", "Master" }; gint i; snd_mixer_elem_t *elem; for (i = 0; i < G_N_ELEMENTS(elem_names); i++) { elem = alsaplug_get_mixer_elem_by_name(mixer, elem_names[i]); if (elem != NULL) return elem; } return NULL; } static gint alsaplug_mixer_new(snd_mixer_t **mixer) { gint ret; ret = snd_mixer_open(mixer, 0); if (ret < 0) { _ERROR("mixer initialization failed: %s", snd_strerror(ret)); return ret; } ret = snd_mixer_attach(*mixer, "default"); if (ret < 0) { snd_mixer_close(*mixer); _ERROR("failed to attach to hardware mixer: %s", snd_strerror(ret)); return ret; } ret = snd_mixer_selem_register(*mixer, NULL, NULL); if (ret < 0) { snd_mixer_detach(*mixer, "default"); snd_mixer_close(*mixer); _ERROR("failed to register hardware mixer: %s", snd_strerror(ret)); return ret; } ret = snd_mixer_load(*mixer); if (ret < 0) { snd_mixer_detach(*mixer, "default"); snd_mixer_close(*mixer); _ERROR("failed to load hardware mixer controls: %s", snd_strerror(ret)); return ret; } return 0; } static void alsaplug_set_volume(gint l, gint r) { snd_mixer_elem_t *elem = alsaplug_guess_mixer_elem(amixer); if (elem == NULL) return; if (snd_mixer_selem_is_playback_mono(elem)) { gint vol = (l > r) ? l : r; snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_MONO, vol); if (snd_mixer_selem_has_playback_switch(elem)) snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_MONO, vol != 0); } else { snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, l); snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, r); if (snd_mixer_selem_has_playback_switch(elem) && !snd_mixer_selem_has_playback_switch_joined(elem)) { snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_LEFT, l != 0); snd_mixer_selem_set_playback_switch(elem, SND_MIXER_SCHN_FRONT_RIGHT, r != 0); } } snd_mixer_handle_events(amixer); } static void alsaplug_get_volume(gint *l, gint *r) { snd_mixer_elem_t *elem = alsaplug_guess_mixer_elem(amixer); if (elem == NULL) return; snd_mixer_handle_events(amixer); *l = 0; *r = 0; if (snd_mixer_selem_is_playback_mono(elem)) { snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_MONO, (glong *) l); snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_MONO, (glong *) r); } else { snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, (glong *) l); snd_mixer_selem_get_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, (glong *) r); } } /******************************************************************************** * ALSA PCM I/O functions. * ********************************************************************************/ static void alsaplug_write_buffer(gpointer data, gint length) { snd_pcm_sframes_t wr_frames; while (length > 0) { gint frames = snd_pcm_bytes_to_frames(pcm_handle, length); wr_frames = snd_pcm_writei(pcm_handle, data, frames); if (wr_frames > 0) { gint written = snd_pcm_frames_to_bytes(pcm_handle, wr_frames); length -= written; data += written; } else { gint err = snd_pcm_recover(pcm_handle, wr_frames, 1); if (err < 0) _ERROR("(write) snd_pcm_recover: %s", snd_strerror(err)); return; } } } static gpointer alsaplug_loop(gpointer unused) { gchar buf[2048]; while (pcm_going) { if (flush_request != -1) { snd_pcm_drop(pcm_handle); snd_pcm_prepare(pcm_handle); wr_total = flush_request * (bps / 1000); flush_request = -1; g_cond_broadcast(pcm_flush_cond); } if (alsaplug_ringbuffer_read(&pcm_ringbuf, buf, 2048) == -1) { /* less than 2048 bytes to go...? */ gint remain = alsaplug_ringbuffer_used(&pcm_ringbuf); if (remain <= 2048 && remain > 0) { alsaplug_ringbuffer_read(&pcm_ringbuf, buf, remain); alsaplug_write_buffer(buf, remain); } else { g_mutex_lock(pcm_state_mutex); g_cond_wait(pcm_state_cond, pcm_state_mutex); g_mutex_unlock(pcm_state_mutex); } continue; } alsaplug_write_buffer(buf, 2048); } snd_pcm_drain(pcm_handle); snd_pcm_close(pcm_handle); pcm_handle = NULL; audio_thread = NULL; alsaplug_ringbuffer_destroy(&pcm_ringbuf); return NULL; } /******************************************************************************** * Output Plugin API implementation. * ********************************************************************************/ static OutputPluginInitStatus alsaplug_init(void) { gint card = -1; pcm_pause_mutex = g_mutex_new(); pcm_pause_cond = g_cond_new(); pcm_state_mutex = g_mutex_new(); pcm_state_cond = g_cond_new(); pcm_flush_cond = g_cond_new(); if (snd_card_next(&card) != 0) return OUTPUT_PLUGIN_INIT_NO_DEVICES; if (!alsaplug_mixer_new(&amixer)) mixer_ready = TRUE; return OUTPUT_PLUGIN_INIT_FOUND_DEVICES; } static gint alsaplug_open_audio(AFormat fmt, gint rate, gint nch) { gint err, bitwidth, ringbuf_size; snd_pcm_format_t afmt; snd_pcm_hw_params_t *hwparams = NULL; afmt = alsaplug_format_convert(fmt); if ((err = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_PLAYBACK, 0)) < 0) { _ERROR("snd_pcm_open: %s", snd_strerror(err)); pcm_handle = NULL; return -1; } snd_pcm_hw_params_alloca(&hwparams); snd_pcm_hw_params_any(pcm_handle, hwparams); snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(pcm_handle, hwparams, afmt); snd_pcm_hw_params_set_channels(pcm_handle, hwparams, nch); snd_pcm_hw_params_set_rate(pcm_handle, hwparams, rate, 0); err = snd_pcm_hw_params(pcm_handle, hwparams); if (err < 0) { _ERROR("snd_pcm_hw_params failed: %s", snd_strerror(err)); return -1; } bitwidth = snd_pcm_format_physical_width(afmt); bps = (rate * bitwidth * nch) >> 3; ringbuf_size = aud_cfg->output_buffer_size * bps / 1000; alsaplug_ringbuffer_init(&pcm_ringbuf, ringbuf_size); pcm_going = TRUE; flush_request = -1; audio_thread = g_thread_create(alsaplug_loop, NULL, TRUE, NULL); return 1; } static void alsaplug_close_audio(void) { g_mutex_lock(pcm_state_mutex); pcm_going = FALSE; wr_total = 0; wr_hwframes = 0; bps = 0; g_mutex_unlock(pcm_state_mutex); g_cond_broadcast(pcm_state_cond); if (audio_thread != NULL) g_thread_join(audio_thread); audio_thread = NULL; } static void alsaplug_write_audio(gpointer data, gint length) { /* software pause... snd_pcm_pause() is not safe. --nenolod */ if (paused) { g_mutex_lock(pcm_pause_mutex); g_cond_wait(pcm_pause_cond, pcm_pause_mutex); g_mutex_unlock(pcm_pause_mutex); } g_mutex_lock(pcm_state_mutex); wr_total += length; alsaplug_ringbuffer_write(&pcm_ringbuf, data, length); g_mutex_unlock(pcm_state_mutex); g_cond_broadcast(pcm_state_cond); } static gint alsaplug_output_time(void) { gint ret = 0; snd_pcm_sframes_t delay; gsize bytes = wr_total; g_mutex_lock(pcm_state_mutex); if (pcm_going && pcm_handle != NULL) { if (!snd_pcm_delay(pcm_handle, &delay)) { guint d = snd_pcm_frames_to_bytes(pcm_handle, delay); if (bytes < d) bytes = 0; else bytes -= d; } ret = (bytes * 1000) / bps; } g_mutex_unlock(pcm_state_mutex); return ret; } static gint alsaplug_written_time(void) { gint ret = 0; g_mutex_lock(pcm_state_mutex); if (pcm_going) ret = (wr_total * 1000) / bps; g_mutex_unlock(pcm_state_mutex); return ret; } static gint alsaplug_buffer_free(void) { gint ret; g_mutex_lock(pcm_state_mutex); if (pcm_going == FALSE) ret = 0; else ret = alsaplug_ringbuffer_free(&pcm_ringbuf); g_mutex_unlock(pcm_state_mutex); return ret; } static void alsaplug_flush(gint time) { /* make the request... */ g_mutex_lock(pcm_state_mutex); flush_request = time; g_cond_broadcast(pcm_state_cond); /* ...then wait for the transaction to complete. */ g_cond_wait(pcm_flush_cond, pcm_state_mutex); g_mutex_unlock(pcm_state_mutex); } static gint alsaplug_buffer_playing(void) { gint ret; g_mutex_lock(pcm_state_mutex); if (pcm_going == FALSE) ret = 0; else ret = alsaplug_ringbuffer_used(&pcm_ringbuf) != 0; g_mutex_unlock(pcm_state_mutex); return ret; } static void alsaplug_pause(short p) { g_mutex_lock(pcm_pause_mutex); paused = p; g_mutex_unlock(pcm_pause_mutex); g_cond_broadcast(pcm_pause_cond); } /******************************************************************************** * Plugin glue. * ********************************************************************************/ static OutputPlugin alsa_op = { .description = "ALSA Output Plugin (-ng)", .probe_priority = 1, .init = alsaplug_init, .open_audio = alsaplug_open_audio, .close_audio = alsaplug_close_audio, .write_audio = alsaplug_write_audio, .output_time = alsaplug_output_time, .written_time = alsaplug_written_time, .buffer_free = alsaplug_buffer_free, .buffer_playing = alsaplug_buffer_playing, .flush = alsaplug_flush, .pause = alsaplug_pause, .set_volume = alsaplug_set_volume, .get_volume = alsaplug_get_volume, }; OutputPlugin *alsa_oplist[] = { &alsa_op, NULL }; SIMPLE_OUTPUT_PLUGIN(alsa, alsa_oplist);