Mercurial > audlegacy-plugins
diff src/alsa-ng/alsa-core.c @ 3162:e387614b9be9
alsa-ng: Import rewritten ALSA plugin. This is still woefully incomplete, but supports basic playback.
This driver uses the "safe" ALSA API subset, including use of blocking I/O.
Right now, it is hardcoded to use "default".
Do not complain about bugs in this plugin.
author | William Pitcock <nenolod@atheme.org> |
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date | Thu, 14 May 2009 21:05:11 -0500 |
parents | |
children | 26a2c237ef53 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/src/alsa-ng/alsa-core.c Thu May 14 21:05:11 2009 -0500 @@ -0,0 +1,250 @@ +/* + * 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 GMutex *pcm_mutex; +static GCond *pcm_cond; + +static gsize wr_total = 0; +static gsize wr_hwframes = 0; + +static gint flush_request; + +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; + } + + if (alsaplug_ringbuffer_read(&pcm_ringbuf, buf, 2048) == -1) + { + GTimeVal pcm_abs_time; + + g_get_current_time(&pcm_abs_time); + g_time_val_add(&pcm_abs_time, 10000); + + g_mutex_lock(pcm_mutex); + g_cond_timed_wait(pcm_cond, pcm_mutex, &pcm_abs_time); + g_mutex_unlock(pcm_mutex); + + continue; + } + + alsaplug_write_buffer(buf, 2048); + } + + snd_pcm_close(pcm_handle); + pcm_handle = NULL; + + return NULL; +} + +/******************************************************************************** + * Output Plugin API implementation. * + ********************************************************************************/ + +static OutputPluginInitStatus +alsaplug_init(void) +{ + gint card = -1; + + pcm_mutex = g_mutex_new(); + pcm_cond = g_cond_new(); + + if (snd_card_next(&card) != 0) + return OUTPUT_PLUGIN_INIT_NO_DEVICES; + + 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) +{ + pcm_going = FALSE; + + g_thread_join(audio_thread); + + wr_total = 0; + wr_hwframes = 0; + bps = 0; +} + +static void +alsaplug_write_audio(gpointer data, gint length) +{ + wr_total += length; + alsaplug_ringbuffer_write(&pcm_ringbuf, data, length); +} + +static gint +alsaplug_output_time(void) +{ + snd_pcm_sframes_t delay; + gsize bytes = wr_total; + + 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; + } + + return (bytes * 1000) / bps; + } + + return 0; +} + +static gint +alsaplug_written_time(void) +{ + if (pcm_going) + return (wr_total * 1000) / bps; + + return 0; +} + +static gint +alsaplug_buffer_free(void) +{ + return alsaplug_ringbuffer_free(&pcm_ringbuf); +} + +static void +alsaplug_flush(gint time) +{ + flush_request = time; + while (flush_request != -1 && pcm_going) + g_usleep(10000); +} + +static gint +alsaplug_buffer_playing(void) +{ + return pcm_going; +} + +/******************************************************************************** + * 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, +}; + +OutputPlugin *alsa_oplist[] = { &alsa_op, NULL }; +SIMPLE_OUTPUT_PLUGIN(alsa, alsa_oplist);