platform: alsa support
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@ -27,6 +27,7 @@ GNU General Public License for more details.
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#define SOUND_NULL 0
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#define SOUND_SDL 1
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#define SOUND_OPENSLES 2
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#define SOUND_ALSA 3
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// crash handler (XASH_CRASHHANDLER)
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#define CRASHHANDLER_NULL 0
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@ -81,7 +81,7 @@ SETUP BACKENDS DEFINITIONS
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#endif
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#ifndef XASH_SOUND
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#define XASH_SOUND SOUND_NULL
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#define XASH_SOUND SOUND_ALSA
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#endif
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#define XASH_USE_EVDEV
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#endif // android case
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@ -0,0 +1,295 @@
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/*
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s_alsa.c - alsa sound hardware output
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Copyright (C) 2019 mittorn
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This program is free software: you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation, either version 3 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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*/
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#include "platform/platform.h"
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#if XASH_SOUND == SOUND_ALSA
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#include <alsa/asoundlib.h>
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#include "sound.h"
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#define BUFFER_SAMPLES 4096
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#define SUBMISSION_CHUNK BUFFER_SAMPLES / 2
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static struct s_alsa_t
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{
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snd_pcm_t *pcm_handle;
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snd_pcm_hw_params_t *hw_params;
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struct sndinfo * si;
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int period_size;
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} s_alsa;
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/*
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==================
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SNDDMA_Init
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Initialize ALSA pcm device, and bind it to sndinfo.
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==================
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*/
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qboolean SNDDMA_Init( void )
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{
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int err, dir = 0;
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unsigned int r;
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snd_pcm_uframes_t p = 0;
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string device = "default";
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Sys_GetParmFromCmdLine( "-alsadev", device );
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if( ( err = snd_pcm_open( &s_alsa.pcm_handle, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) ) < 0)
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{
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Con_Printf( "ALSA: cannot open device %s(%s)\n", device, snd_strerror( err ) );
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return false;
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}
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if( ( err = snd_pcm_hw_params_malloc( &s_alsa.hw_params )) < 0 )
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{
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Con_Printf( "ALSA: cannot allocate hw params(%s)\n", snd_strerror( err ) );
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snd_pcm_close(s_alsa.pcm_handle);
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return false;
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}
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if( ( err = snd_pcm_hw_params_any( s_alsa.pcm_handle, s_alsa.hw_params ) ) < 0 )
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{
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Con_Printf( "ALSA: cannot init hw params(%s)\n", snd_strerror( err ) );
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snd_pcm_hw_params_free( s_alsa.hw_params );
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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if( ( err = snd_pcm_hw_params_set_access( s_alsa.pcm_handle, s_alsa.hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ) ) < 0 )
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{
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Con_Printf( "ALSA: cannot set access(%s)\n", snd_strerror( err ) );
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snd_pcm_hw_params_free( s_alsa.hw_params );
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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if( ( err = snd_pcm_hw_params_set_format( s_alsa.pcm_handle, s_alsa.hw_params, SND_PCM_FORMAT_S16 ) ) < 0 )
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{
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Con_Printf("ALSA: 16 bit not supported\n");
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snd_pcm_hw_params_free( s_alsa.hw_params );
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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dma.format.speed = SOUND_DMA_SPEED;
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r = dma.format.speed;
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if( ( err = snd_pcm_hw_params_set_rate_near( s_alsa.pcm_handle, s_alsa.hw_params, &r, &dir ) ) < 0 )
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{
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Con_Printf( "ALSA: cannot set rate %d(%s)\n", r, snd_strerror( err ) );
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snd_pcm_hw_params_free(s_alsa.hw_params);
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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else
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{
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// rate succeeded, but is perhaps slightly different
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if( dir != 0 )
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{
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Con_Printf( "ALSA: rate %d not supported, using %d\n", SOUND_DMA_SPEED, r );
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dma.format.speed = r;
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dir = 0;
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}
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}
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dma.format.channels = 2;
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if( ( err = snd_pcm_hw_params_set_channels(s_alsa.pcm_handle, s_alsa.hw_params, dma.format.channels ) ) < 0 )
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{
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Con_Printf( "ALSA: cannot set channels %d(%s)\n", 2, snd_strerror( err ) );
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snd_pcm_hw_params_free( s_alsa.hw_params );
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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p = BUFFER_SAMPLES / dma.format.channels;
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if( ( err = snd_pcm_hw_params_set_period_size_near( s_alsa.pcm_handle, s_alsa.hw_params, &p, &dir ) ) < 0 )
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{
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Con_Printf("ALSA: cannot set period size (%s)\n", snd_strerror(err));
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snd_pcm_hw_params_free(s_alsa.hw_params);
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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else
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{
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// period succeeded, but is perhaps slightly different
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if( dir != 0 )
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{
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snd_pcm_hw_params_get_period_size(s_alsa.hw_params, &p, NULL);
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Con_Printf( "ALSA: period %d not supported, using %lu\n", ( BUFFER_SAMPLES / dma.format.channels ), p );
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dir = 0;
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}
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}
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// set params
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if( ( err = snd_pcm_hw_params( s_alsa.pcm_handle, s_alsa.hw_params ) ) < 0 )
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{
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Con_Printf( "ALSA: cannot set params(%s)\n", snd_strerror( err ) );
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snd_pcm_hw_params_free(s_alsa.hw_params);
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snd_pcm_close( s_alsa.pcm_handle );
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return false;
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}
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// request period size again
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snd_pcm_hw_params_get_period_size( s_alsa.hw_params, &p, NULL );
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s_alsa.period_size = p;
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Con_Printf( "ALSA: period size %lu\n", p );
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dma.buffer = Z_Malloc( BUFFER_SAMPLES * 2 ); //allocate pcm frame buffer
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dma.samplepos = 0;
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dma.samples = BUFFER_SAMPLES;
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dma.format.width = 2;
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dma.initialized = 1;
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snd_pcm_prepare( s_alsa.pcm_handle );
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return true;
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}
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/*
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==============
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SNDDMA_GetDMAPos
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return the current sample position (in mono samples read)
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inside the recirculating dma buffer, so the mixing code will know
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how many sample are required to fill it up.
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===============
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*/
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int SNDDMA_GetDMAPos( void )
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{
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return dma.samplepos;
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}
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/*
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==============
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SNDDMA_GetSoundtime
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update global soundtime
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===============
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*/
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int SNDDMA_GetSoundtime( void )
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{
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static int buffers, oldsamplepos;
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int samplepos, fullsamples;
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fullsamples = dma.samples / 2;
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// it is possible to miscount buffers
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// if it has wrapped twice between
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// calls to S_Update. Oh well.
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samplepos = SNDDMA_GetDMAPos( );
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if( samplepos < oldsamplepos )
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{
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buffers++; // buffer wrapped
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if( paintedtime > 0x40000000 )
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{
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// time to chop things off to avoid 32 bit limits
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buffers = 0;
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paintedtime = fullsamples;
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S_StopAllSounds( true );
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}
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}
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oldsamplepos = samplepos;
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return ( buffers * fullsamples + samplepos / 2 );
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}
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/*
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==============
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SNDDMA_Shutdown
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Closes the ALSA pcm device and frees the dma buffer.
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===============
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*/
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void SNDDMA_Shutdown(void)
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{
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Con_Printf( "Shutting down audio.\n" );
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dma.initialized = false;
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if( dma.buffer )
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{
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snd_pcm_drop( s_alsa.pcm_handle );
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snd_pcm_close( s_alsa.pcm_handle );
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Mem_Free( dma.buffer );
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dma.buffer = NULL;
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}
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}
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/*
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==============
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SNDDMA_Submit
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Send sound to device if buffer isn't really the dma buffer
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===============
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*/
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void SNDDMA_Submit( void )
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{
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int avail = snd_pcm_avail_update( s_alsa.pcm_handle );
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while( avail >= s_alsa.period_size )
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{
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int size = dma.samples << 1;
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int pos = dma.samplepos << 1;
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unsigned long len = s_alsa.period_size;
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int wrapped = pos + len - size;
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if( wrapped < 0 )
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{
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snd_pcm_writei( s_alsa.pcm_handle, dma.buffer + pos, len / 4 );
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dma.samplepos += len >> 1;
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}
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else
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{
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int remaining = size - pos;
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snd_pcm_writei( s_alsa.pcm_handle, dma.buffer + pos, remaining / 4 );
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snd_pcm_writei( s_alsa.pcm_handle, dma.buffer, wrapped / 4 );
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dma.samplepos = wrapped >> 1;
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}
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avail = snd_pcm_avail_update( s_alsa.pcm_handle );
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}
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}
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/*
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==============
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SNDDMA_BeginPainting
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Callback provided by the engine in case we need it. We don't.
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==============
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*/
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void SNDDMA_BeginPainting(void)
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{
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}
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/*
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===========
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SNDDMA_Activate
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Called when the main window gains or loses focus.
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The window have been destroyed and recreated
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between a deactivate and an activate.
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===========
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*/
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void SNDDMA_Activate( qboolean active )
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{
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snd_pcm_pause( s_alsa.pcm_handle, active );
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}
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#endif
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@ -28,7 +28,7 @@ def configure(conf):
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conf.check_cc(lib = i)
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elif conf.options.FBDEV_SW:
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conf.define('XASH_FBDEV', 1)
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conf.env.FBDEV
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conf.check_cc(lib = 'asound')
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else:
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conf.load('sdl2')
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if not conf.env.HAVE_SDL2:
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@ -57,7 +57,7 @@ def build(bld):
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# basic build: dedicated only, no dependencies
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if bld.env.DEST_OS != 'win32':
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libs += [ 'DL' , 'M', 'PTHREAD' ]
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libs += [ 'DL' , 'M' , 'RT', 'PTHREAD']
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source += bld.path.ant_glob(['platform/posix/*.c'])
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else:
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libs += ['USER32', 'SHELL32', 'GDI32', 'ADVAPI32', 'DBGHELP', 'PSAPI', 'WS2_32' ]
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@ -82,9 +82,12 @@ def build(bld):
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'client/*.c',
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'client/vgui/*.c',
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'client/avi/*.c'])
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else:
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if bld.env.DEST_OS == 'linux':
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libs.append('RT')
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if bld.env.DEST_OS == 'linux' and not bld.env.HAVE_SDL2:
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libs.append('RT')
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if not bld.env.DEDICATED:
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libs.append('ASOUND')
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# HACK: public headers must be put before common, so we don't get wrong mathlib included
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includes = ['common', 'server', 'client', 'client/vgui', '.', '../public', '../common', '../pm_shared' ]
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