/*
- 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
+ Copyright (C) 2006 Mathieu Olivier
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
*/
-#include <alsa/asoundlib.h>
+// ALSA module, used by Linux
#include "quakedef.h"
+
+#include <alsa/asoundlib.h>
+
#include "snd_main.h"
-static int snd_inited;
-static snd_pcm_uframes_t buffer_size;
-static const char *pcmname = NULL;
-static snd_pcm_t *pcm;
+#define NB_PERIODS 2
-qboolean SNDDMA_Init (void)
-{
- 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;
+static snd_pcm_t* pcm_handle = NULL;
+static snd_pcm_sframes_t expected_delay = 0;
+static unsigned int alsasoundtime;
- snd_pcm_hw_params_alloca (&hw);
- snd_pcm_sw_params_alloca (&sw);
+static snd_seq_t* seq_handle = NULL;
-// COMMANDLINEOPTION: Linux ALSA Sound: -sndpcm <devicename> selects which pcm device to us, default is "default"
- if ((i=COM_CheckParm("-sndpcm"))!=0)
- pcmname=com_argv[i+1];
- if (!pcmname)
- pcmname = "default";
+/*
+====================
+SndSys_Init
-// COMMANDLINEOPTION: Linux ALSA Sound: -sndbits <number> sets sound precision to 8 or 16 bit (email me if you want others added)
- if ((i=COM_CheckParm("-sndbits")) != 0)
+Create "snd_renderbuffer" with the proper sound format if the call is successful
+May return a suggested format if the requested format isn't available
+====================
+*/
+qboolean SndSys_Init (const snd_format_t* requested, snd_format_t* suggested)
+{
+ const char* pcm_name, *seq_name;
+ int i, err, seq_client, seq_port;
+ snd_pcm_hw_params_t* hw_params = NULL;
+ snd_pcm_format_t snd_pcm_format;
+ snd_pcm_uframes_t buffer_size;
+
+ Con_Print ("SndSys_Init: using the ALSA module\n");
+
+ seq_name = NULL;
+ i = COM_CheckParm ("-sndseqin"); // TODO turn this into a cvar, maybe
+ if (i != 0 && i < com_argc - 1)
+ seq_name = com_argv[i + 1];
+ if(seq_name)
{
- bps = atoi(com_argv[i+1]);
- if (bps != 16 && bps != 8)
+ seq_client = atoi(seq_name);
+ seq_port = 0;
+ if(strchr(seq_name, ':'))
+ seq_port = atoi(strchr(seq_name, ':') + 1);
+ Con_Printf ("SndSys_Init: seq input port has been set to \"%d:%d\". Enabling sequencer input...\n", seq_client, seq_port);
+ err = snd_seq_open (&seq_handle, "default", SND_SEQ_OPEN_INPUT, 0);
+ if (err < 0)
{
- Con_Printf("Error: invalid sample bits: %d\n", bps);
- return false;
+ Con_Print ("SndSys_Init: can't open seq device\n");
+ goto seqdone;
}
+ err = snd_seq_set_client_name(seq_handle, gamename);
+ if (err < 0)
+ {
+ Con_Print ("SndSys_Init: can't set name of seq device\n");
+ goto seqerror;
+ }
+ err = snd_seq_create_simple_port(seq_handle, gamename, SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE, SND_SEQ_PORT_TYPE_MIDI_GENERIC | SND_SEQ_PORT_TYPE_APPLICATION);
+ if(err < 0)
+ {
+ Con_Print ("SndSys_Init: can't create seq port\n");
+ goto seqerror;
+ }
+ err = snd_seq_connect_from(seq_handle, 0, seq_client, seq_port);
+ if(err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't connect to seq port \"%d:%d\"\n", seq_client, seq_port);
+ goto seqerror;
+ }
+ err = snd_seq_nonblock(seq_handle, 1);
+ if(err < 0)
+ {
+ Con_Print ("SndSys_Init: can't make seq nonblocking\n");
+ goto seqerror;
+ }
+
+ goto seqdone;
+
+seqerror:
+ snd_seq_close(seq_handle);
+ seq_handle = NULL;
}
-// COMMANDLINEOPTION: Linux ALSA Sound: -sndspeed <hz> chooses 44100 hz, 22100 hz, or 11025 hz sound output rate
- if ((i=COM_CheckParm("-sndspeed")) != 0)
+seqdone:
+ // Check the requested sound format
+ if (requested->width < 1 || requested->width > 2)
{
- rate = atoi(com_argv[i+1]);
- if (rate!=44100 && rate!=22050 && rate!=11025)
+ Con_Printf ("SndSys_Init: invalid sound width (%hu)\n",
+ requested->width);
+
+ if (suggested != NULL)
{
- Con_Printf("Error: invalid sample rate: %d\n", rate);
- return false;
- }
- }
+ memcpy (suggested, requested, sizeof (*suggested));
-// 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);
+ if (requested->width < 1)
+ suggested->width = 1;
+ else
+ suggested->width = 2;
- 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;
- }
+ Con_Printf ("SndSys_Init: suggesting sound width = %hu\n",
+ suggested->width);
+ }
- 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;
+ return false;
+ }
+
+ if (pcm_handle != NULL)
+ {
+ Con_Print ("SndSys_Init: WARNING: Init called before Shutdown!\n");
+ SndSys_Shutdown ();
}
- 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;
- }
+ // Determine the name of the PCM handle we'll use
+ switch (requested->channels)
+ {
+ case 4:
+ pcm_name = "surround40";
+ break;
+ case 6:
+ pcm_name = "surround51";
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;
- }
+ pcm_name = "surround71";
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;
- }
+ pcm_name = "default";
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;
+// COMMANDLINEOPTION: Linux ALSA Sound: -sndpcm <devicename> selects which pcm device to us, default is "default"
+ i = COM_CheckParm ("-sndpcm");
+ if (i != 0 && i < com_argc - 1)
+ pcm_name = com_argv[i + 1];
+
+ // Open the audio device
+ Con_Printf ("SndSys_Init: PCM device is \"%s\"\n", pcm_name);
+ err = snd_pcm_open (&pcm_handle, pcm_name, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't open audio device \"%s\" (%s)\n",
+ pcm_name, snd_strerror (err));
+ return false;
}
- 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",
+ // Allocate the hardware parameters
+ err = snd_pcm_hw_params_malloc (&hw_params);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't allocate hardware parameters (%s)\n",
snd_strerror (err));
- goto error;
+ goto init_error;
}
- err = snd_pcm_sw_params_current (pcm, sw);
- if (0 > err) {
- Con_Printf ("ALSA: unable to determine current sw params. %s\n",
+ err = snd_pcm_hw_params_any (pcm_handle, hw_params);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't initialize hardware parameters (%s)\n",
snd_strerror (err));
- goto error;
+ goto init_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",
+
+ // Set the access type
+ err = snd_pcm_hw_params_set_access (pcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set access type (%s)\n",
snd_strerror (err));
- goto error;
+ goto init_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;
+
+ // Set the sound width
+ if (requested->width == 1)
+ snd_pcm_format = SND_PCM_FORMAT_U8;
+ else
+ snd_pcm_format = SND_PCM_FORMAT_S16;
+ err = snd_pcm_hw_params_set_format (pcm_handle, hw_params, snd_pcm_format);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set sound width to %hu (%s)\n",
+ requested->width, snd_strerror (err));
+ goto init_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;
+
+ // Set the sound channels
+ err = snd_pcm_hw_params_set_channels (pcm_handle, hw_params, requested->channels);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set sound channels to %hu (%s)\n",
+ requested->channels, snd_strerror (err));
+ goto init_error;
}
- shm->format.channels = stereo + 1;
- shm->samplepos = 0;
- shm->format.width = bps / 8;
+ // Set the sound speed
+ err = snd_pcm_hw_params_set_rate (pcm_handle, hw_params, requested->speed, 0);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set sound speed to %u (%s)\n",
+ requested->speed, snd_strerror (err));
+ goto init_error;
+ }
+
+ buffer_size = requested->speed / 5;
+ err = snd_pcm_hw_params_set_buffer_size_near (pcm_handle, hw_params, &buffer_size);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set sound buffer size to %lu (%s)\n",
+ buffer_size, snd_strerror (err));
+ goto init_error;
+ }
- err = snd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
- if (0 > err) {
- Con_Printf ("ALSA: unable to get buffer size. %s\n",
+ buffer_size /= NB_PERIODS;
+ err = snd_pcm_hw_params_set_period_size_near(pcm_handle, hw_params, &buffer_size, 0);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set sound period size to %lu (%s)\n",
+ buffer_size, snd_strerror (err));
+ goto init_error;
+ }
+
+ err = snd_pcm_hw_params (pcm_handle, hw_params);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Init: can't set hardware parameters (%s)\n",
snd_strerror (err));
- goto error;
+ goto init_error;
}
- shm->samples = buffer_size * shm->format.channels; // mono samples in buffer
- shm->format.speed = rate;
- SNDDMA_GetDMAPos (); // sets shm->buffer
+ snd_pcm_hw_params_free (hw_params);
+
+ snd_renderbuffer = Snd_CreateRingBuffer(requested, 0, NULL);
+ expected_delay = 0;
+ alsasoundtime = 0;
+ if (snd_channellayout.integer == SND_CHANNELLAYOUT_AUTO)
+ Cvar_SetValueQuick (&snd_channellayout, SND_CHANNELLAYOUT_ALSA);
- snd_inited = 1;
return true;
-error:
- snd_pcm_close (pcm);
+
+// It's not very clean, but it avoids a lot of duplicated code.
+init_error:
+
+ if (hw_params != NULL)
+ snd_pcm_hw_params_free (hw_params);
+
+ snd_pcm_close(pcm_handle);
+ pcm_handle = NULL;
+
return false;
}
-int SNDDMA_GetDMAPos (void)
+
+/*
+====================
+SndSys_Shutdown
+
+Stop the sound card, delete "snd_renderbuffer" and free its other resources
+====================
+*/
+void SndSys_Shutdown (void)
{
- const snd_pcm_channel_area_t *areas;
- snd_pcm_uframes_t offset;
- snd_pcm_uframes_t nframes = shm->samples/shm->format.channels;
+ if (seq_handle != NULL)
+ {
+ snd_seq_close(seq_handle);
+ seq_handle = NULL;
+ }
- if (!snd_inited)
- return 0;
+ if (pcm_handle != NULL)
+ {
+ snd_pcm_close(pcm_handle);
+ pcm_handle = NULL;
+ }
- snd_pcm_avail_update (pcm);
- snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
- offset *= shm->format.channels;
- nframes *= shm->format.channels;
- shm->samplepos = offset;
- shm->buffer = (qbyte *)areas->addr;
- return shm->samplepos;
+ if (snd_renderbuffer != NULL)
+ {
+ Mem_Free(snd_renderbuffer->ring);
+ Mem_Free(snd_renderbuffer);
+ snd_renderbuffer = NULL;
+ }
}
-void SNDDMA_Shutdown (void)
+
+/*
+====================
+SndSys_Recover
+
+Try to recover from errors
+====================
+*/
+static qboolean SndSys_Recover (int err_num)
{
- if (snd_inited) {
- snd_pcm_close (pcm);
- snd_inited = 0;
+ int err;
+
+ // We can only do something on underrun ("broken pipe") errors
+ if (err_num != -EPIPE)
+ return false;
+
+ err = snd_pcm_prepare (pcm_handle);
+ if (err < 0)
+ {
+ Con_Printf ("SndSys_Recover: unable to recover (%s)\n",
+ snd_strerror (err));
+
+ // TOCHECK: should we stop the playback ?
+
+ return false;
}
+
+ return true;
}
+
/*
- SNDDMA_Submit
+====================
+SndSys_Write
+====================
+*/
+static snd_pcm_sframes_t SndSys_Write (const unsigned char* buffer, unsigned int nbframes)
+{
+ snd_pcm_sframes_t written;
+
+ written = snd_pcm_writei (pcm_handle, buffer, nbframes);
+ if (written < 0)
+ {
+ if (developer.integer >= 1000 && vid_activewindow)
+ Con_Printf ("SndSys_Write: audio write returned %ld (%s)!\n",
+ written, snd_strerror (written));
- Send sound to device if buffer isn't really the dma buffer
+ if (SndSys_Recover (written))
+ {
+ written = snd_pcm_writei (pcm_handle, buffer, nbframes);
+ if (written < 0)
+ Con_DPrintf ("SndSys_Write: audio write failed again (error %ld: %s)!\n",
+ written, snd_strerror (written));
+ }
+ }
+ if (written > 0)
+ {
+ snd_renderbuffer->startframe += written;
+ expected_delay += written;
+ }
+
+ return written;
+}
+
+
+/*
+====================
+SndSys_Submit
+
+Submit the contents of "snd_renderbuffer" to the sound card
+====================
*/
-void SNDDMA_Submit (void)
+void SndSys_Submit (void)
{
- int state;
- int count = paintedtime - soundtime;
- const snd_pcm_channel_area_t *areas;
- snd_pcm_uframes_t nframes;
- snd_pcm_uframes_t offset;
+ unsigned int startoffset, factor;
+ snd_pcm_uframes_t limit, nbframes;
+ snd_pcm_sframes_t written;
- nframes = count / shm->format.channels;
+ if (pcm_handle == NULL ||
+ snd_renderbuffer->startframe == snd_renderbuffer->endframe)
+ return;
- snd_pcm_avail_update (pcm);
- snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
+ startoffset = snd_renderbuffer->startframe % snd_renderbuffer->maxframes;
+ factor = snd_renderbuffer->format.width * snd_renderbuffer->format.channels;
+ limit = snd_renderbuffer->maxframes - startoffset;
+ nbframes = snd_renderbuffer->endframe - snd_renderbuffer->startframe;
- state = snd_pcm_state (pcm);
+ if (nbframes > limit)
+ {
+ written = SndSys_Write (&snd_renderbuffer->ring[startoffset * factor], limit);
+ if (written < 0 || (snd_pcm_uframes_t)written != limit)
+ return;
- 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;
+ nbframes -= limit;
+ startoffset = 0;
}
+
+ written = SndSys_Write (&snd_renderbuffer->ring[startoffset * factor], nbframes);
+ if (written < 0)
+ return;
}
-void *S_LockBuffer(void)
+
+/*
+====================
+SndSys_GetSoundTime
+
+Returns the number of sample frames consumed since the sound started
+====================
+*/
+unsigned int SndSys_GetSoundTime (void)
+{
+ snd_pcm_sframes_t delay, timediff;
+ int err;
+
+ if (pcm_handle == NULL)
+ return 0;
+
+ err = snd_pcm_delay (pcm_handle, &delay);
+ if (err < 0)
+ {
+ if (developer.integer >= 1000 && vid_activewindow)
+ Con_DPrintf ("SndSys_GetSoundTime: can't get playback delay (%s)\n",
+ snd_strerror (err));
+
+ if (! SndSys_Recover (err))
+ return 0;
+
+ err = snd_pcm_delay (pcm_handle, &delay);
+ if (err < 0)
+ {
+ Con_DPrintf ("SndSys_GetSoundTime: can't get playback delay, again (%s)\n",
+ snd_strerror (err));
+ return 0;
+ }
+ }
+
+ if (expected_delay < delay)
+ {
+ Con_DPrintf ("SndSys_GetSoundTime: expected_delay(%ld) < delay(%ld)\n",
+ expected_delay, delay);
+ timediff = 0;
+ }
+ else
+ timediff = expected_delay - delay;
+ expected_delay = delay;
+
+ alsasoundtime += (unsigned int)timediff;
+
+ return alsasoundtime;
+}
+
+
+/*
+====================
+SndSys_LockRenderBuffer
+
+Get the exclusive lock on "snd_renderbuffer"
+====================
+*/
+qboolean SndSys_LockRenderBuffer (void)
+{
+ // Nothing to do
+ return true;
+}
+
+
+/*
+====================
+SndSys_UnlockRenderBuffer
+
+Release the exclusive lock on "snd_renderbuffer"
+====================
+*/
+void SndSys_UnlockRenderBuffer (void)
{
- return shm->buffer;
+ // Nothing to do
}
-void S_UnlockBuffer(void)
+/*
+====================
+SndSys_SendKeyEvents
+
+Send keyboard events originating from the sound system (e.g. MIDI)
+====================
+*/
+void SndSys_SendKeyEvents(void)
{
+ snd_seq_event_t *event;
+ if(!seq_handle)
+ return;
+ for(;;)
+ {
+ if(snd_seq_event_input(seq_handle, &event) <= 0)
+ break;
+ if(event)
+ {
+ switch(event->type)
+ {
+ case SND_SEQ_EVENT_NOTEON:
+ if(event->data.note.velocity)
+ {
+ Key_Event(K_MIDINOTE0 + event->data.note.note, 0, true);
+ break;
+ }
+ case SND_SEQ_EVENT_NOTEOFF:
+ Key_Event(K_MIDINOTE0 + event->data.note.note, 0, false);
+ break;
+ }
+ }
+ }
}
projects / xonotic / darkplaces.git / blobdiff