diff --git a/engine/client/snd_alsa.c b/engine/client/snd_alsa.c new file mode 100755 index 00000000..6b77b790 --- /dev/null +++ b/engine/client/snd_alsa.c @@ -0,0 +1,352 @@ +/* + snd_alsa.c + + Support for the ALSA 1.0.1 sound driver + + Copyright (C) 1999,2000 contributors of the QuakeForge project + Please see the file "AUTHORS" for a list of contributors + + 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: + + Free Software Foundation, Inc. + 59 Temple Place - Suite 330 + Boston, MA 02111-1307, USA + +*/ +//actually stolen from darkplaces. +//I guess noone can be arsed to write it themselves. :/ + +#include + +#include "quakedef.h" + +static int snd_inited; +static snd_pcm_uframes_t buffer_size; + +static const char *pcmname = NULL; +static snd_pcm_t *pcm; + +soundcardinfo_t *sndcardinfo; + +qboolean snd_firsttime; + +int SNDDMA_Init (soundcardinfo_t *sc) +{ + int err, i; + int bps = -1, stereo = -1; + unsigned int rate = 0; + snd_pcm_hw_params_t *hw; + snd_pcm_sw_params_t *sw; + snd_pcm_uframes_t frag_size; + + snd_pcm_hw_params_alloca (&hw); + snd_pcm_sw_params_alloca (&sw); + +// COMMANDLINEOPTION: Linux ALSA Sound: -sndpcm selects which pcm device to us, default is "default" + if ((i=COM_CheckParm("-sndpcm"))!=0) + pcmname=com_argv[i+1]; + if (!pcmname) + pcmname = "default"; + +// COMMANDLINEOPTION: Linux ALSA Sound: -sndbits sets sound precision to 8 or 16 bit (email me if you want others added) + if ((i=COM_CheckParm("-sndbits")) != 0) + { + bps = atoi(com_argv[i+1]); + if (bps != 16 && bps != 8) + { + Con_Printf("Error: invalid sample bits: %d\n", bps); + return false; + } + } + +// COMMANDLINEOPTION: Linux ALSA Sound: -sndspeed chooses 44100 hz, 22100 hz, or 11025 hz sound output rate + if ((i=COM_CheckParm("-sndspeed")) != 0) + { + rate = atoi(com_argv[i+1]); + if (rate!=44100 && rate!=22050 && rate!=11025) + { + Con_Printf("Error: invalid sample rate: %d\n", rate); + return false; + } + } + +// COMMANDLINEOPTION: Linux ALSA Sound: -sndmono sets sound output to mono + if ((i=COM_CheckParm("-sndmono")) != 0) + stereo=0; +// COMMANDLINEOPTION: Linux ALSA Sound: -sndstereo sets sound output to stereo + if ((i=COM_CheckParm("-sndstereo")) != 0) + stereo=1; + + err = snd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK, + SND_PCM_NONBLOCK); + if (0 > err) { + Con_Printf ("Error: audio open error: %s\n", snd_strerror (err)); + return 0; + } + Con_Printf ("ALSA: Using PCM %s.\n", pcmname); + + err = snd_pcm_hw_params_any (pcm, hw); + if (0 > err) { + Con_Printf ("ALSA: error setting hw_params_any. %s\n", + snd_strerror (err)); + goto error; + } + + err = snd_pcm_hw_params_set_access (pcm, hw, + SND_PCM_ACCESS_MMAP_INTERLEAVED); + if (0 > err) { + Con_Printf ("ALSA: Failure to set noninterleaved PCM access. %s\n" + "Note: Interleaved is not supported\n", + snd_strerror (err)); + goto error; + } + + switch (bps) { + case -1: + err = snd_pcm_hw_params_set_format (pcm, hw, + SND_PCM_FORMAT_S16); + if (0 <= err) { + bps = 16; + } else if (0 <= (err = snd_pcm_hw_params_set_format (pcm, hw, + SND_PCM_FORMAT_U8))) { + bps = 8; + } else { + Con_Printf ("ALSA: no useable formats. %s\n", + snd_strerror (err)); + goto error; + } + break; + case 8: + case 16: + err = snd_pcm_hw_params_set_format (pcm, hw, bps == 8 ? + SND_PCM_FORMAT_U8 : + SND_PCM_FORMAT_S16); + if (0 > err) { + Con_Printf ("ALSA: no usable formats. %s\n", + snd_strerror (err)); + goto error; + } + break; + default: + Con_Printf ("ALSA: desired format not supported\n"); + goto error; + } + + switch (stereo) { + case -1: + err = snd_pcm_hw_params_set_channels (pcm, hw, 2); + if (0 <= err) { + stereo = 1; + } else if (0 <= (err = snd_pcm_hw_params_set_channels (pcm, hw, + 1))) { + stereo = 0; + } else { + Con_Printf ("ALSA: no usable channels. %s\n", + snd_strerror (err)); + goto error; + } + break; + case 0: + case 1: + err = snd_pcm_hw_params_set_channels (pcm, hw, stereo ? 2 : 1); + if (0 > err) { + Con_Printf ("ALSA: no usable channels. %s\n", + snd_strerror (err)); + goto error; + } + break; + default: + Con_Printf ("ALSA: desired channels not supported\n"); + goto error; + } + + switch (rate) { + case 0: + rate = 44100; + err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0); + if (0 <= err) { + frag_size = 32 * bps; + } else { + rate = 22050; + err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0); + if (0 <= err) { + frag_size = 16 * bps; + } else { + rate = 11025; + err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, + 0); + if (0 <= err) { + frag_size = 8 * bps; + } else { + Con_Printf ("ALSA: no usable rates. %s\n", + snd_strerror (err)); + goto error; + } + } + } + break; + case 11025: + case 22050: + case 44100: + err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0); + if (0 > err) { + Con_Printf ("ALSA: desired rate %i not supported. %s\n", rate, + snd_strerror (err)); + goto error; + } + frag_size = 8 * bps * rate / 11025; + break; + default: + Con_Printf ("ALSA: desired rate %i not supported.\n", rate); + goto error; + } + + err = snd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0); + if (0 > err) { + Con_Printf ("ALSA: unable to set period size near %i. %s\n", + (int) frag_size, snd_strerror (err)); + goto error; + } + err = snd_pcm_hw_params (pcm, hw); + if (0 > err) { + Con_Printf ("ALSA: unable to install hw params: %s\n", + snd_strerror (err)); + goto error; + } + err = snd_pcm_sw_params_current (pcm, sw); + if (0 > err) { + Con_Printf ("ALSA: unable to determine current sw params. %s\n", + snd_strerror (err)); + goto error; + } + err = snd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U); + if (0 > err) { + Con_Printf ("ALSA: unable to set playback threshold. %s\n", + snd_strerror (err)); + goto error; + } + err = snd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U); + if (0 > err) { + Con_Printf ("ALSA: unable to set playback stop threshold. %s\n", + snd_strerror (err)); + goto error; + } + err = snd_pcm_sw_params (pcm, sw); + if (0 > err) { + Con_Printf ("ALSA: unable to install sw params. %s\n", + snd_strerror (err)); + goto error; + } + + sc->sn.numchannels = stereo + 1; + sc->sn.samplepos = 0; + sc->sn.samplebits = bps; + + err = snd_pcm_hw_params_get_buffer_size (hw, &buffer_size); + if (0 > err) { + Con_Printf ("ALSA: unable to get buffer size. %s\n", + snd_strerror (err)); + goto error; + } + + sc->sn.samples = buffer_size * sc->sn.numchannels; // mono samples in buffer + sc->sn.speed = rate; + SNDDMA_GetDMAPos (sc); // sets shm->buffer + + snd_inited = 1; + return true; + +error: + snd_pcm_close (pcm); + return false; +} + +int SNDDMA_GetDMAPos (soundcardinfo_t *sc) +{ + const snd_pcm_channel_area_t *areas; + snd_pcm_uframes_t offset; + snd_pcm_uframes_t nframes = sc->sn.samples / sc->sn.numchannels; + + if (!snd_inited) + return 0; + + snd_pcm_avail_update (pcm); + snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes); + offset *= sc->sn.numchannels; + nframes *= sc->sn.numchannels; + sc->sn.samplepos = offset; + sc->sn.buffer = areas->addr; + return sc->sn.samplepos; +} + +void SNDDMA_Shutdown (soundcardinfo_t *sc) +{ + if (snd_inited) { + snd_pcm_close (pcm); + snd_inited = 0; + } +} + +/* + SNDDMA_Submit + + Send sound to device if buffer isn't really the dma buffer +*/ +void SNDDMA_Submit (soundcardinfo_t *sc) +{ + extern int soundtime; + int state; + int count = sc->paintedtime - soundtime; + const snd_pcm_channel_area_t *areas; + snd_pcm_uframes_t nframes; + snd_pcm_uframes_t offset; + + nframes = count / sc->sn.numchannels; + + snd_pcm_avail_update (pcm); + snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes); + + state = snd_pcm_state (pcm); + + switch (state) { + case SND_PCM_STATE_PREPARED: + snd_pcm_mmap_commit (pcm, offset, nframes); + snd_pcm_start (pcm); + break; + case SND_PCM_STATE_RUNNING: + snd_pcm_mmap_commit (pcm, offset, nframes); + break; + default: + break; + } +} + +void *S_LockBuffer(soundcardinfo_t *sc) +{ + return sc->sn.buffer; +} + +void S_UnlockBuffer(soundcardinfo_t *sc) +{ +} + +void SNDDMA_SetUnderWater(qboolean underwater) +{ +} + +void S_UpdateCapture(void) +{ +} +