/* 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 */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "QF/cvar.h" #include "QF/qargs.h" #include "QF/sys.h" #include "snd_internal.h" typedef struct { const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t offset; snd_pcm_uframes_t nframes; } alsa_pkt_t; static int snd_inited; static int snd_blocked = 0; static snd_pcm_uframes_t buffer_size; static void *alsa_handle; static snd_pcm_t *pcm; static snd_async_handler_t *async_handler; static int snd_bits; static cvar_t snd_bits_cvar = { .name = "snd_bits", .description = "sound sample depth. 0 is system default", .default_value = "0", .flags = CVAR_ROM, .value = { .type = &cexpr_int, .value = &snd_bits }, }; static char *snd_device; static cvar_t snd_device_cvar = { .name = "snd_device", .description = "sound device. \"\" is system default", .default_value = "", .flags = CVAR_ROM, .value = { .type = 0, .value = &snd_device }, }; static int snd_rate; static cvar_t snd_rate_cvar = { .name = "snd_rate", .description = "sound playback rate. 0 is system default", .default_value = "0", .flags = CVAR_ROM, .value = { .type = &cexpr_int, .value = &snd_rate }, }; static int snd_stereo; static cvar_t snd_stereo_cvar = { .name = "snd_stereo", .description = "sound stereo output", .default_value = "1", .flags = CVAR_ROM, .value = { .type = &cexpr_int, .value = &snd_stereo }, }; //FIXME xfer probably should not be touching this (such data should probably //come through snd_t) static snd_output_data_t plugin_info_snd_output_data; #define QF_ALSA_NEED(ret, func, params) \ static ret (*qf##func) params; #include "alsa_funcs_list.h" #undef QF_ALSA_NEED static bool load_libasound (void) { if (!(alsa_handle = dlopen ("libasound.so.2", RTLD_GLOBAL | RTLD_NOW))) { Sys_Printf ("Couldn't load libasound.so.2: %s\n", dlerror ()); return false; } #define QF_ALSA_NEED(ret, func, params) \ if (!(qf##func = dlsym (alsa_handle, #func))) { \ Sys_Printf ("Couldn't load ALSA function %s\n", #func); \ dlclose (alsa_handle); \ alsa_handle = 0; \ return false; \ } #include "alsa_funcs_list.h" #undef QF_ALSA_NEED return true; } #define snd_pcm_hw_params_sizeof qfsnd_pcm_hw_params_sizeof #define snd_pcm_sw_params_sizeof qfsnd_pcm_sw_params_sizeof static void SNDDMA_Init_Cvars (void) { Cvar_Register (&snd_stereo_cvar, 0, 0); Cvar_Register (&snd_rate_cvar, 0, 0); Cvar_Register (&snd_device_cvar, 0, 0); Cvar_Register (&snd_bits_cvar, 0, 0); } static __attribute__((const)) snd_pcm_uframes_t round_buffer_size (snd_pcm_uframes_t sz) { snd_pcm_uframes_t mask = ~0; while (sz & mask) { sz &= mask; mask <<= 1; } return sz; } static inline int clamp_16 (int val) { if (val > 0x7fff) val = 0x7fff; else if (val < -0x8000) val = -0x8000; return val; } static inline int clamp_8 (int val) { if (val > 0x7f) val = 0x7f; else if (val < -0x80) val = -0x80; return val; } static void alsa_ni_xfer (snd_t *snd, portable_samplepair_t *paintbuffer, int count, float volume) { const snd_pcm_channel_area_t *areas; int out_idx, out_max; float *p; areas = snd->xfer_data; p = (float *) paintbuffer; out_max = snd->frames - 1; out_idx = *plugin_info_snd_output_data.paintedtime; while (out_idx > out_max) out_idx -= out_max + 1; if (snd->samplebits == 16) { short *out_0 = (short *) areas[0].addr; short *out_1 = (short *) areas[1].addr; if (snd->channels == 2) { while (count--) { out_0[out_idx] = clamp_16 ((*p++ * volume) * 0x8000); out_1[out_idx] = clamp_16 ((*p++ * volume) * 0x8000); if (out_idx++ > out_max) out_idx = 0; } } else { while (count--) { out_0[out_idx] = clamp_16 ((*p++ * volume) * 0x8000); p++; // skip right channel if (out_idx++ > out_max) out_idx = 0; } } } else if (snd->samplebits == 8) { byte *out_0 = (byte *) areas[0].addr; byte *out_1 = (byte *) areas[1].addr; if (snd->channels == 2) { while (count--) { out_0[out_idx] = clamp_8 ((*p++ * volume) * 0x80); out_1[out_idx] = clamp_8 ((*p++ * volume) * 0x80); if (out_idx++ > out_max) out_idx = 0; } } else { while (count--) { out_0[out_idx] = clamp_8 ((*p++ * volume) * 0x8000); p++; // skip right channel if (out_idx++ > out_max) out_idx = 0; } } } } static void alsa_xfer (snd_t *snd, portable_samplepair_t *paintbuffer, int count, float volume) { int out_idx, out_max, step, val; float *p; alsa_pkt_t *packet = snd->xfer_data;; p = (float *) paintbuffer; count *= snd->channels; out_max = (snd->frames * snd->channels) - 1; out_idx = snd->paintedtime * snd->channels; while (out_idx > out_max) out_idx -= out_max + 1; step = 3 - snd->channels; if (snd->samplebits == 16) { short *out = (short *) packet->areas[0].addr; while (count--) { val = (*p * volume) * 0x8000; p += step; if (val > 0x7fff) val = 0x7fff; else if (val < -0x8000) val = -0x8000; out[out_idx++] = val; if (out_idx > out_max) out_idx = 0; } } else if (snd->samplebits == 8) { unsigned char *out = (unsigned char *) packet->areas[0].addr; while (count--) { val = (*p * volume) * 128; p += step; if (val > 0x7f) val = 0x7f; else if (val < -0x80) val = -0x80; out[out_idx++] = val + 0x80; if (out_idx > out_max) out_idx = 0; } } } static int alsa_recover (snd_pcm_t *pcm, int err) { if (err == -EPIPE) { Sys_Printf ("snd_alsa: xrun\n"); // xrun if ((err = qfsnd_pcm_prepare (pcm)) < 0) { Sys_MaskPrintf (SYS_snd, "snd_alsa: recover from xrun failed: %s\n", qfsnd_strerror (err)); return err; } return 0; } else if (err == -ESTRPIPE) { Sys_Printf ("snd_alsa: suspend\n"); // suspend while ((err = qfsnd_pcm_resume(pcm)) == -EAGAIN) { usleep (20 * 1000); } if (err < 0 && (err = qfsnd_pcm_prepare (pcm)) < 0) { Sys_MaskPrintf (SYS_snd, "snd_alsa: recover from suspend failed: %s\n", qfsnd_strerror (err)); return err; } return 0; } return err; } static int alsa_process (snd_pcm_t *pcm, snd_t *snd) { alsa_pkt_t packet; int res; int ret = 1; snd_pcm_uframes_t size = snd->submission_chunk; snd->xfer_data = &packet; while (size > 0) { packet.nframes = size; if ((res = qfsnd_pcm_mmap_begin (pcm, &packet.areas, &packet.offset, &packet.nframes)) < 0) { if ((res = alsa_recover (pcm, -EPIPE)) < 0) { Sys_Printf ("snd_alsa: XRUN recovery failed: %s\n", qfsnd_strerror (res)); snd->xfer_data = 0; return res; } ret = 0; } snd->buffer = packet.areas[0].addr; snd->paint_channels (snd, snd->paintedtime + packet.nframes); if ((res = qfsnd_pcm_mmap_commit (pcm, packet.offset, packet.nframes)) < 0 || (snd_pcm_uframes_t) res != packet.nframes) { if ((res = alsa_recover (pcm, res >= 0 ? -EPIPE : res)) < 0) { Sys_Printf ("snd_alsa: XRUN recovery failed: %s\n", qfsnd_strerror (res)); snd->xfer_data = 0; return res; } ret = 0; } size -= packet.nframes; } snd->xfer_data = 0; return ret; } static void alsa_callback (snd_async_handler_t *handler) { snd_pcm_t *pcm = qfsnd_async_handler_get_pcm (handler); snd_t *snd = qfsnd_async_handler_get_callback_private (handler); int res; int avail; int first = 0; while (1) { snd_pcm_state_t state = qfsnd_pcm_state (pcm); if (state == SND_PCM_STATE_XRUN) { if ((res = alsa_recover (pcm, -EPIPE)) < 0) { Sys_Printf ("snd_alsa: XRUN recovery failed: %s\n", qfsnd_strerror (res)); //FIXME disable/restart sound return; } } else if (state == SND_PCM_STATE_SUSPENDED) { if ((res = alsa_recover (pcm, -EPIPE)) < 0) { Sys_Printf ("snd_alsa: suspend recovery failed: %s\n", qfsnd_strerror (res)); //FIXME disable/restart sound return; } } if ((avail = qfsnd_pcm_avail_update (pcm)) < 0) { if ((res = alsa_recover (pcm, -EPIPE)) < 0) { Sys_Printf ("snd_alsa: avail update failed: %s\n", qfsnd_strerror (res)); //FIXME disable/restart sound return; } first = 1; continue; } if (avail < snd->submission_chunk) { if (first) { first = 0; if ((res = qfsnd_pcm_start (pcm)) < 0) { Sys_Printf ("snd_alsa: start failed: %s\n", qfsnd_strerror (res)); return; } continue; } break; } if ((res = alsa_process (pcm, snd))) { if (res < 0) { //FIXME disable/restart sound return; } break; } first = 1; } } static int alsa_open_playback (snd_t *snd, const char *device) { if (!*device) { device = "default"; } Sys_Printf ("Using PCM %s.\n", device); int res = qfsnd_pcm_open (&pcm, device, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); if (res < 0) { Sys_Printf ("snd_alsa: audio open error: %s\n", qfsnd_strerror (res)); return 0; } return 1; } static int alsa_playback_set_mmap (snd_t *snd, snd_pcm_hw_params_t *hw) { int res; res = qfsnd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_INTERLEAVED); if (res == 0) { snd->xfer = alsa_xfer; return 1; } Sys_MaskPrintf (SYS_snd, "snd_alsa: Failure to set interleaved PCM " "access. (%d) %s\n", res, qfsnd_strerror (res)); res = qfsnd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_NONINTERLEAVED); if (res == 0) { snd->xfer = alsa_ni_xfer; return 1; } Sys_MaskPrintf (SYS_snd, "snd_alsa: Failure to set noninterleaved PCM " "access. (%d) %s\n", res, qfsnd_strerror (res)); Sys_Printf ("snd_alsa: could not set mmap access\n"); return 0; } static int alsa_playback_set_bps (snd_t *snd, snd_pcm_hw_params_t *hw) { int res; int bps = 0; if (snd_bits == 16) { bps = SND_PCM_FORMAT_S16_LE; snd->samplebits = 16; } else if (snd_bits == 8) { bps = SND_PCM_FORMAT_U8; snd->samplebits = 8; } else if (snd_bits) { Sys_Printf ("snd_alsa: invalid sample bits: %d\n", bps); return 0; } if (bps) { if ((res = qfsnd_pcm_hw_params_set_format (pcm, hw, bps)) == 0) { return 1; } } else { bps = SND_PCM_FORMAT_S16_LE; if ((res = qfsnd_pcm_hw_params_set_format (pcm, hw, bps)) == 0) { snd->samplebits = 16; return 1; } bps = SND_PCM_FORMAT_U8; if ((res = qfsnd_pcm_hw_params_set_format (pcm, hw, bps)) == 0) { snd->samplebits = 8; return 1; } Sys_Printf ("snd_alsa: no usable formats. %s\n", qfsnd_strerror (res)); } snd->samplebits = -1; Sys_Printf ("snd_alsa: desired format not supported\n"); return 0; } static int alsa_playback_set_channels (snd_t *snd, snd_pcm_hw_params_t *hw) { int res; int channels = 1; if (snd_stereo) { channels = 2; } if ((res = qfsnd_pcm_hw_params_set_channels (pcm, hw, channels)) == 0) { snd->channels = channels; return 1; } Sys_Printf ("snd_alsa: desired channels not supported\n"); return 0; } static int alsa_playback_set_rate (snd_t *snd, snd_pcm_hw_params_t *hw) { int res; unsigned rate = 0; static int default_rates[] = { 48000, 44100, 22050, 11025, 0 }; if (snd_rate) { rate = snd_rate; } if (rate) { if ((res = qfsnd_pcm_hw_params_set_rate (pcm, hw, rate, 0)) == 0) { snd->speed = rate; return 1; } Sys_Printf ("snd_alsa: desired rate %i not supported. %s\n", rate, qfsnd_strerror (res)); } else { // use default rate int dir = 0; for (int *def_rate = default_rates; *def_rate; def_rate++) { rate = *def_rate; res = qfsnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, &dir); if (res == 0) { snd->speed = rate; return 1; } } Sys_Printf ("snd_alsa: no usable rate\n"); } return 0; } static int alsa_playback_set_period_size (snd_t *snd, snd_pcm_hw_params_t *hw) { int res; snd_pcm_uframes_t period_size; // works out to about 5.5ms (5.3 for 48k, 5.8 for 44k1) but consistent for // different sample rates give or take rounding period_size = 64 * (snd->speed / 11025); res = qfsnd_pcm_hw_params_set_period_size_near (pcm, hw, &period_size, 0); if (res == 0) { // don't mix less than this in frames: res = qfsnd_pcm_hw_params_get_period_size (hw, &period_size, 0); if (res == 0) { snd->submission_chunk = period_size; return 1; } Sys_Printf ("snd_alsa: unable to get period size. %s\n", qfsnd_strerror (res)); } else { Sys_Printf ("snd_alsa: unable to set period size near %i. %s\n", (int) period_size, qfsnd_strerror (res)); } return 0; } static int SNDDMA_Init (snd_t *snd) { int res; const char *device = snd_device; snd_pcm_hw_params_t *hw; snd_pcm_sw_params_t *sw; if (!load_libasound ()) return false; snd_pcm_hw_params_alloca (&hw); snd_pcm_sw_params_alloca (&sw); while (1) { if (!alsa_open_playback (snd, device)) { return 0; } if ((res = qfsnd_pcm_hw_params_any (pcm, hw)) < 0) { Sys_Printf ("snd_alsa: error setting hw_params_any. %s\n", qfsnd_strerror (res)); goto error; } if (alsa_playback_set_mmap (snd, hw)) { break; } if (*device) { goto error; } qfsnd_pcm_close (pcm); device = "plughw"; } if (!alsa_playback_set_bps (snd, hw)) { goto error; } if (!alsa_playback_set_channels (snd, hw)) { goto error; } if (!alsa_playback_set_rate (snd, hw)) { goto error; } if (!alsa_playback_set_period_size (snd, hw)) { goto error; } if ((res = qfsnd_pcm_hw_params (pcm, hw)) < 0) { Sys_Printf ("snd_alsa: unable to install hw params: %s\n", qfsnd_strerror (res)); goto error; } if ((res = qfsnd_pcm_sw_params_current (pcm, sw)) < 0) { Sys_Printf ("snd_alsa: unable to determine current sw params. %s\n", qfsnd_strerror (res)); goto error; } if ((res = qfsnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U)) < 0) { Sys_Printf ("snd_alsa: unable to set playback threshold. %s\n", qfsnd_strerror (res)); goto error; } if ((res = qfsnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U)) < 0) { Sys_Printf ("snd_alsa: unable to set playback stop threshold. %s\n", qfsnd_strerror (res)); goto error; } if ((res = qfsnd_pcm_sw_params (pcm, sw)) < 0) { Sys_Printf ("snd_alsa: unable to install sw params. %s\n", qfsnd_strerror (res)); goto error; } snd->framepos = 0; if ((res = qfsnd_pcm_hw_params_get_buffer_size (hw, &buffer_size)) < 0) { Sys_Printf ("snd_alsa: unable to get buffer size. %s\n", qfsnd_strerror (res)); goto error; } if (buffer_size != round_buffer_size (buffer_size)) { Sys_Printf ("snd_alsa: WARNING: non-power of 2 buffer size. sound may be unsatisfactory\n"); Sys_Printf ("recommend using either the plughw, or hw devices or adjusting dmix\n"); Sys_Printf ("to have a power of 2 buffer size\n"); } if ((res = qfsnd_async_add_pcm_handler (&async_handler, pcm, alsa_callback, snd)) < 0) { Sys_Printf ("snd_alsa: unable to register async handler: %s", qfsnd_strerror (res)); goto error; } snd->frames = buffer_size; snd->threaded = 1;//XXX FIXME double check whether it's always true // send the first period to fill the buffer // also sets snd->buffer if (alsa_process (pcm, snd) < 0) { goto error; } qfsnd_pcm_start (pcm); Sys_Printf ("%5d channels %sinterleaved\n", snd->channels, snd->xfer ? "non-" : ""); Sys_Printf ("%5d samples (%.1fms)\n", snd->frames, 1000.0 * snd->frames / snd->speed); Sys_Printf ("%5d samplepos\n", snd->framepos); Sys_Printf ("%5d samplebits\n", snd->samplebits); Sys_Printf ("%5d submission_chunk (%.1fms)\n", snd->submission_chunk, 1000.0 * snd->submission_chunk / snd->speed); Sys_Printf ("%5d speed\n", snd->speed); Sys_Printf ("0x%lx dma buffer\n", (long) snd->buffer); snd_inited = 1; return 1; error: qfsnd_pcm_close (pcm); snd->channels = 0; snd->frames = 0; snd->samplebits = 0; snd->submission_chunk = 0; snd->speed = 0; return 0; } static void SNDDMA_shutdown (snd_t *snd) { if (snd_inited) { qfsnd_async_del_handler (async_handler); async_handler = 0; qfsnd_pcm_close (pcm); snd_inited = 0; } } static void SNDDMA_BlockSound (snd_t *snd) { if (snd_inited && ++snd_blocked == 1) qfsnd_pcm_pause (pcm, 1); } static void SNDDMA_UnblockSound (snd_t *snd) { if (!snd_inited || !snd_blocked) return; if (!--snd_blocked) qfsnd_pcm_pause (pcm, 0); } static general_data_t plugin_info_general_data = { }; static general_funcs_t plugin_info_general_funcs = { .init = SNDDMA_Init_Cvars, .shutdown = NULL, }; static snd_output_data_t plugin_info_snd_output_data = { .model = som_pull, }; static snd_output_funcs_t plugin_info_snd_output_funcs = { .init = SNDDMA_Init, .shutdown = SNDDMA_shutdown, .block_sound = SNDDMA_BlockSound, .unblock_sound = SNDDMA_UnblockSound, }; static plugin_data_t plugin_info_data = { .general = &plugin_info_general_data, .snd_output = &plugin_info_snd_output_data, }; static plugin_funcs_t plugin_info_funcs = { .general = &plugin_info_general_funcs, .snd_output = &plugin_info_snd_output_funcs, }; static plugin_t plugin_info = { .type = qfp_snd_output, .api_version = QFPLUGIN_VERSION, .plugin_version = "0.1", .description = "ALSA digital output", .copyright = "Copyright (C) 1996-1997 id Software, Inc.\n" "Copyright (C) 1999,2000,2001 contributors of the QuakeForge " "project\n" "Please see the file \"AUTHORS\" for a list of contributors", .functions = &plugin_info_funcs, .data = &plugin_info_data, }; PLUGIN_INFO(snd_output, alsa) { return &plugin_info; }