X-Git-Url: http://de.git.xonotic.org/?p=xonotic%2Fdarkplaces.git;a=blobdiff_plain;f=snd_alsa.c;h=3f4292c84e7f73edf2a2f49a7f36707c0980a374;hp=67a9ed60c78726ccf73f95c3d0bb7d7d7135a5c9;hb=09f884dd87cbb5b3da9e261d8c0841cdb9365c5b;hpb=5bdc0879026939f551a3ff217064732d59731be2 diff --git a/snd_alsa.c b/snd_alsa.c index 67a9ed60..3f4292c8 100644 --- a/snd_alsa.c +++ b/snd_alsa.c @@ -1,10 +1,5 @@ /* - 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 @@ -26,311 +21,504 @@ */ -#include +// ALSA module, used by Linux #include "quakedef.h" -static int snd_inited; -static snd_pcm_uframes_t buffer_size; +#include + +#include "snd_main.h" + -static const char *pcmname = NULL; -static snd_pcm_t *pcm; +#define NB_PERIODS 4 -qboolean SNDDMA_Init (void) +static snd_pcm_t* pcm_handle = NULL; +static snd_pcm_sframes_t expected_delay = 0; +static unsigned int alsasoundtime; + +static snd_seq_t* seq_handle = NULL; + +/* +==================== +SndSys_Init + +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) { - 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) + 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; +// COMMANDLINEOPTION: Linux ALSA Sound: -sndseqin : selects which sequencer port to use for input, by default no sequencer port is used (MIDI note events from that port get mapped to MIDINOTE keys that can be bound) + 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 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 selects which pcm device to use, 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; + } + + // pick a buffer size that is a power of 2 (by masking off low bits) + buffer_size = i = (int)(requested->speed * 0.15f); + while (buffer_size & (buffer_size-1)) + buffer_size &= (buffer_size-1); + // then check if it is the nearest power of 2 and bump it up if not + if (i - buffer_size >= buffer_size >> 1) + buffer_size *= 2; + + 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", + // pick a period size near the buffer_size we got from ALSA + snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size); + 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 = 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_insane.integer && vid_activewindow) + Con_DPrintf ("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_insane.integer && 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; + } + } + } }