From 8dbec761cf858ea65900c8a094599857208d8c3a Mon Sep 17 00:00:00 2001
From: "N.N." <matju@users.sourceforge.net>
Date: Tue, 5 Jan 2010 22:49:36 +0000
Subject: svn path=/trunk/; revision=12907

---
 desiredata/src/s_audio_alsa.c | 554 ------------------------------------------
 1 file changed, 554 deletions(-)
 delete mode 100644 desiredata/src/s_audio_alsa.c

(limited to 'desiredata/src/s_audio_alsa.c')

diff --git a/desiredata/src/s_audio_alsa.c b/desiredata/src/s_audio_alsa.c
deleted file mode 100644
index b812fba0..00000000
--- a/desiredata/src/s_audio_alsa.c
+++ /dev/null
@@ -1,554 +0,0 @@
-/* Copyright (c) 1997-2003 Guenter Geiger, Miller Puckette, Larry Troxler, Winfried Ritsch, Karl MacMillan, and others.
-* For information on usage and redistribution, and for a DISCLAIMER OF ALL
-* WARRANTIES, see the file, "LICENSE.txt," in this distribution.  */
-
-/* this file inputs and outputs audio using the ALSA API available on linux. */
-
-/* support for ALSA pcmv2 api by Karl MacMillan<karlmac@peabody.jhu.edu> */ 
-/* support for ALSA MMAP noninterleaved by Winfried Ritsch, IEM */
-
-#include <alsa/asoundlib.h>
-
-#include "desire.h"
-using namespace desire;
-#include <errno.h>
-#include <stdio.h>
-#include <unistd.h>
-#include <stdlib.h>
-#include <string.h>
-#include <sys/types.h>
-#include <sys/time.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <sched.h>
-#include <sys/mman.h>
-#include "s_audio_alsa.h"
-
-/* Defines */
-#define DEBUG(x) x
-#define DEBUG2(x) {x;}
-
-/* needed for alsa 0.9 compatibility: */
-#if (SND_LIB_MAJOR < 1)
-#define ALSAAPI9
-#endif
-
-//static void alsa_close_audio();
-static void alsa_checkiosync();
-static void alsa_numbertoname(int iodev, char *devname, int nchar);
-static int alsa_jittermax;
-static void alsa_close_audio();
-#define ALSA_DEFJITTERMAX 3
-
-    /* don't assume we can turn all 31 bits when doing float-to-fix; 
-    otherwise some audio drivers (e.g. Midiman/ALSA) wrap around. */
-#define FMAX 0x7ffff000
-#define CLIP32(x) (((x)>FMAX)?FMAX:((x) < -FMAX)?-FMAX:(x))
-
-static char *alsa_snd_buf;
-static int alsa_snd_bufsize;
-static int alsa_buf_samps;
-static snd_pcm_status_t *alsa_status; 
-static int alsa_usemmap;
-t_alsa alsai,alsao;
-
-static void check_error(int err, const char *why) {if (err<0) error("%s: %s", why, snd_strerror(err));}
-
-static int alsaio_canmmap(t_alsa_dev *dev) {
-    snd_pcm_hw_params_t *hw_params;
-    int err1, err2;
-    snd_pcm_hw_params_alloca(&hw_params);
-    err1 = snd_pcm_hw_params_any(dev->a_handle, hw_params);
-    if (err1 < 0) {
-      check_error(err1,"Broken configuration: no configurations available"); 
-      return 0;
-    }
-    err1 =     snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
-    if (err1 < 0) {
-        err2 = snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
-    } else err2 = -1;
-#if 0
-    error("err 1 %d (%s), err2 %d (%s)", err1, snd_strerror(err1), err2, snd_strerror(err2));
-#endif
-    return err1<0 && err2>=0;
-}
-
-static int alsaio_setup(t_alsa_dev *dev, int out, int *channels, int *rate, int nfrags, int frag_size) {
-    int bufsizeforthis, err;
-    snd_pcm_hw_params_t* hw_params;
-    unsigned int tmp_uint;
-    snd_pcm_uframes_t tmp_snd_pcm_uframes;
-    if (sys_verbose) {
-	if (out) post("configuring sound output...");
-        else     post("configuring sound input...");
-    }
-    /* set hardware parameters... */
-    snd_pcm_hw_params_alloca(&hw_params);
-    /* get the default params */
-    err = snd_pcm_hw_params_any(dev->a_handle, hw_params);
-    check_error(err, "snd_pcm_hw_params_any");
-    /* try to set interleaved access */
-    err = snd_pcm_hw_params_set_access(dev->a_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
-    if (err < 0) return -1;
-    check_error(err, "snd_pcm_hw_params_set_access");
-    /* Try to set 32 bit format first */
-    err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params, SND_PCM_FORMAT_S32);
-    if (err<0) {
-        error("PD-ALSA: 32 bit format not available - using 16");
-        err = snd_pcm_hw_params_set_format(dev->a_handle, hw_params,SND_PCM_FORMAT_S16);
-        check_error(err, "snd_pcm_hw_params_set_format");
-        dev->a_sampwidth = 2;
-    } else dev->a_sampwidth = 4;
-    if (sys_verbose) post("Sample width set to %d bytes", dev->a_sampwidth);
-    /* set the subformat */
-    err = snd_pcm_hw_params_set_subformat(dev->a_handle, hw_params, SND_PCM_SUBFORMAT_STD);
-    check_error(err, "snd_pcm_hw_params_set_subformat");
-    /* set the number of channels */
-    tmp_uint = *channels;
-    err = snd_pcm_hw_params_set_channels_min(dev->a_handle, hw_params, &tmp_uint);
-    check_error(err, "snd_pcm_hw_params_set_channels");
-    if (tmp_uint != (unsigned)*channels) post("ALSA: set input channels to %d", tmp_uint);
-    *channels = tmp_uint;
-    dev->a_channels = *channels;
-    /* set the sampling rate */
-    err = snd_pcm_hw_params_set_rate_min(dev->a_handle, hw_params, (unsigned int *)rate, 0);
-    check_error(err, "snd_pcm_hw_params_set_rate_min (input)");
-#if 0
-    err = snd_pcm_hw_params_get_rate(hw_params, &subunitdir);
-    post("input sample rate %d", err);
-#endif
-    /* set the period - ie frag size */
-    /* LATER try this to get a recommended period size...
-       right now, it trips an assertion failure in ALSA lib */
-#ifdef ALSAAPI9
-    err = snd_pcm_hw_params_set_period_size_near(dev->a_handle, hw_params, (snd_pcm_uframes_t)frag_size, 0);
-#else
-    tmp_snd_pcm_uframes = frag_size;
-    err = snd_pcm_hw_params_set_period_size_near(dev->a_handle, hw_params, &tmp_snd_pcm_uframes, 0);
-#endif
-    check_error(err, "snd_pcm_hw_params_set_period_size_near (input)");
-    /* set the number of periods - ie numfrags */
-#ifdef ALSAAPI9
-    err = snd_pcm_hw_params_set_periods_near(dev->a_handle, hw_params, nfrags, 0);
-#else
-    tmp_uint = nfrags;
-    err = snd_pcm_hw_params_set_periods_near(dev->a_handle, hw_params, &tmp_uint, 0);
-#endif
-    check_error(err, "snd_pcm_hw_params_set_periods_near (input)");
-    /* set the buffer size */
-#ifdef ALSAAPI9
-    err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle, hw_params, nfrags * frag_size);
-#else
-    tmp_snd_pcm_uframes = nfrags * frag_size;
-    err = snd_pcm_hw_params_set_buffer_size_near(dev->a_handle, hw_params, &tmp_snd_pcm_uframes);
-#endif
-    check_error(err, "snd_pcm_hw_params_set_buffer_size_near (input)");
-    err = snd_pcm_hw_params(dev->a_handle, hw_params);
-    check_error(err, "snd_pcm_hw_params (input)");
-    /* set up the buffer */
-    bufsizeforthis = sys_dacblocksize * dev->a_sampwidth * *channels;
-    if (alsa_snd_buf) {
-        if (alsa_snd_bufsize < bufsizeforthis) {
-            if (!(alsa_snd_buf = (char *)realloc(alsa_snd_buf, bufsizeforthis))) {error("out of memory"); return 0;}
-            memset(alsa_snd_buf, 0, bufsizeforthis);
-            alsa_snd_bufsize = bufsizeforthis;
-        }
-    } else {
-        if (!(alsa_snd_buf = (char *)malloc(bufsizeforthis))) {error("out of memory"); return 0;}
-        memset(alsa_snd_buf, 0, bufsizeforthis);
-        alsa_snd_bufsize = bufsizeforthis;
-    }
-    return 1;
-}
-
-/* return 0 on success */
-static int alsa_open_audio(
-int  naudioindev, int * audioindev, int  nchindev, int * chindev,
-int naudiooutdev, int *audiooutdev, int nchoutdev, int *choutdev, int rate, int dummy) {
-    int err, inchans = 0, outchans = 0;
-    char devname[512];
-    int frag_size = (sys_blocksize ? sys_blocksize : ALSA_DEFFRAGSIZE);
-    int nfrags, i;
-    nfrags = int(sys_schedadvance * (float)rate / (1e6 * frag_size));
-    /* save our belief as to ALSA's buffer size for later */
-    alsa_buf_samps = nfrags * frag_size;
-    alsai.ndev = alsao.ndev = 0;
-    alsa_jittermax = ALSA_DEFJITTERMAX;
-    if (sys_verbose) post("audio buffer set to %d", (int)(0.001 * sys_schedadvance));
-    for (int i=0; i<naudioindev; i++) {
-        alsa_numbertoname(audioindev[i], devname, 512);
-        err = snd_pcm_open(&alsai.dev[alsai.ndev].a_handle, devname, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
-        check_error(err, "snd_pcm_open (input)");
-        if (err<0) continue;
-        alsai.dev[alsai.ndev].a_devno = audioindev[i];
-        snd_pcm_nonblock(alsai.dev[alsai.ndev].a_handle, 1);
-        if (sys_verbose) post("opened input device name %s", devname);
-        alsai.ndev++;
-    }
-    for (int i=0; i<naudiooutdev; i++) {
-        alsa_numbertoname(audiooutdev[i], devname, 512);
-        err = snd_pcm_open(&alsao.dev[alsao.ndev].a_handle, devname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-        check_error(err, "snd_pcm_open (output)");
-        if (err<0) continue;
-        alsao.dev[alsao.ndev].a_devno = audiooutdev[i];
-        snd_pcm_nonblock(alsao.dev[alsao.ndev].a_handle, 1);
-        alsao.ndev++;
-    }
-    if (!alsai.ndev && !alsao.ndev) goto blewit;
-    /* If all the open devices support mmap_noninterleaved, let's call Wini's code in s_audio_alsamm.c */
-    alsa_usemmap = 1;
-    for (int i=0; i<alsai.ndev; i++) if (!alsaio_canmmap(&alsai.dev[i])) alsa_usemmap = 0;
-    for (int i=0; i<alsao.ndev; i++) if (!alsaio_canmmap(&alsao.dev[i])) alsa_usemmap = 0;
-    if (alsa_usemmap) {
-        post("using mmap audio interface");
-        if (alsamm_open_audio(rate)) goto blewit; else return 0;
-    }
-    for (int i=0; i<alsai.ndev; i++) {
-        int channels = chindev[i];
-        if (alsaio_setup(&alsai.dev[i], 0, &channels, &rate, nfrags, frag_size) < 0) goto blewit;
-        inchans += channels;
-    }
-    for (int i=0; i<alsao.ndev; i++) {
-        int channels = choutdev[i];
-        if (alsaio_setup(&alsao.dev[i], 1, &channels, &rate, nfrags, frag_size) < 0) goto blewit;
-        outchans += channels;
-    }
-    if (!inchans && !outchans) goto blewit;
-    for (int i=0; i<alsai.ndev; i++) snd_pcm_prepare(alsai.dev[i].a_handle);
-    for (int i=0; i<alsao.ndev; i++) snd_pcm_prepare(alsao.dev[i].a_handle);
-    /* if duplex we can link the channels so they start together; however j is not used, so wtf */
-    for (int i=0; i<alsai.ndev; i++) {
-        //for (int j=0; j<alsao.ndev; j++) {
-            if (alsai.dev[i].a_devno == alsao.dev[i].a_devno) {
-                snd_pcm_link(alsai.dev[i].a_handle,alsao.dev[i].a_handle);
-            }
-        //}
-    }
-    /* allocate the status variables */
-    if (!alsa_status) {
-        err = snd_pcm_status_malloc(&alsa_status);
-        check_error(err, "snd_pcm_status_malloc");
-    }
-    /* fill the buffer with silence */
-    memset(alsa_snd_buf, 0, alsa_snd_bufsize);
-    if (outchans) {
-        i = (frag_size * nfrags)/sys_dacblocksize + 1;
-        while (i--) {
-            for (int i=0; i<alsao.ndev; i++)
-                snd_pcm_writei(alsao.dev[i].a_handle, alsa_snd_buf, sys_dacblocksize);
-        }
-    } else if (inchans) {
-        for (int i=0; i<alsai.ndev; i++)
-            if ((err = snd_pcm_start(alsai.dev[i].a_handle)) < 0) check_error(err, "input start failed");
-    }
-    return 0;
-blewit:
-    sys_inchannels = 0;
-    sys_outchannels = 0;
-    alsa_close_audio();
-    return 1;
-}
-
-static void alsa_close_audio() {
-    int err;
-    if (alsa_usemmap) {alsamm_close_audio(); return;}
-    for (int i=0; i<alsai.ndev; i++) {
-        err = snd_pcm_close(alsai.dev[i].a_handle);
-        check_error(err, "snd_pcm_close (input)");
-    }
-    for (int i=0; i<alsao.ndev; i++) {
-        err = snd_pcm_close(alsao.dev[i].a_handle);
-        check_error(err, "snd_pcm_close (output)");
-    }
-    alsai.ndev = alsao.ndev = 0;
-}
-
-int alsa_send_dacs() {
-#ifdef DEBUG_ALSA_XFER
-    static int xferno = 0;
-    static int callno = 0;
-#endif
-    static double timenow;
-    double timelast;
-    t_sample *fp1, *fp2;
-    int j, k, iodev, result, ch;
-    int chansintogo, chansouttogo;
-    unsigned int transfersize;
-    if (alsa_usemmap) return alsamm_send_dacs();
-    if (!alsai.ndev && !alsao.ndev) return SENDDACS_NO;
-    chansintogo = sys_inchannels;
-    chansouttogo = sys_outchannels;
-    transfersize = sys_dacblocksize;
-    timelast = timenow;
-    timenow = sys_getrealtime();
-#ifdef DEBUG_ALSA_XFER
-    if (timenow - timelast > 0.050) post("(%d)", int(1000 * (timenow - timelast)));
-    callno++;
-#endif
-    alsa_checkiosync();     /* check I/O are in sync and data not late */
-    for (int i=0; i<alsai.ndev; i++) {
-        snd_pcm_status(alsai.dev[i].a_handle, alsa_status);
-        if (snd_pcm_status_get_avail(alsa_status) < transfersize) return SENDDACS_NO;
-    }
-    for (int i=0; i<alsao.ndev; i++) {
-        snd_pcm_status(alsao.dev[i].a_handle, alsa_status);
-        if (snd_pcm_status_get_avail(alsa_status) < transfersize) return SENDDACS_NO;
-    }
-    /* do output */
-    fp1 = sys_soundout; ch = 0;
-    for (int iodev=0; iodev<alsao.ndev; iodev++) {
-        int thisdevchans = alsao.dev[iodev].a_channels;
-        int chans = min(chansouttogo,thisdevchans);
-        chansouttogo -= chans;
-	int i;
-        if (alsao.dev[iodev].a_sampwidth == 4) {
-            for (i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
-	        fp2 = fp1;
-                for (int j = ch, k = sys_dacblocksize; k--; j += thisdevchans, fp2++) {
-                    float s1 = *fp2 * INT32_MAX;
-                    ((t_alsa_sample32 *)alsa_snd_buf)[j] = CLIP32(int(s1));
-		}
-            }
-            for (; i<thisdevchans; i++, ch++)
-                for (int j = ch, k = sys_dacblocksize; k--; j += thisdevchans) ((t_alsa_sample32 *)alsa_snd_buf)[j] = 0;
-        } else {
-            for (i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
-	        fp2=fp1;
-                for (int j=ch, k=sys_dacblocksize; k--; j += thisdevchans, fp2++) {
-                    int s = int(*fp2 * 32767.);
-                    if (s > 32767) s = 32767; else if (s < -32767) s = -32767;
-                    ((t_alsa_sample16 *)alsa_snd_buf)[j] = s;
-		}
-            }
-            for (; i < thisdevchans; i++, ch++)
-                for (int j = ch, k = sys_dacblocksize; k--; j += thisdevchans) ((t_alsa_sample16 *)alsa_snd_buf)[j] = 0;
-        }
-        result = snd_pcm_writei(alsao.dev[iodev].a_handle, alsa_snd_buf, transfersize);
-        if (result != (int)transfersize) {
-    #ifdef DEBUG_ALSA_XFER
-            if (result >= 0 || errno == EAGAIN) post("ALSA: write returned %d of %d", result, transfersize);
-            else error("ALSA: write: %s", snd_strerror(errno));
-            post("inputcount %d, outputcount %d, outbufsize %d",
-                    inputcount, outputcount, (ALSA_EXTRABUFFER + sys_advance_samples) * alsao.dev[iodev].a_sampwidth * outchannels);
-    #endif
-            sys_log_error(ERR_DACSLEPT);
-            return SENDDACS_NO;
-        }
-        /* zero out the output buffer */
-        memset(sys_soundout, 0, sys_dacblocksize * sizeof(*sys_soundout) * sys_outchannels);
-        if (sys_getrealtime() - timenow > 0.002) {
-    #ifdef DEBUG_ALSA_XFER
-            post("output %d took %d msec", callno, int(1000 * (timenow - timelast)));
-    #endif
-            timenow = sys_getrealtime();
-            sys_log_error(ERR_DACSLEPT);
-        }
-    }
-    /* do input */
-    for (iodev = 0, fp1 = sys_soundin, ch = 0; iodev < alsai.ndev; iodev++) {
-        int thisdevchans = alsai.dev[iodev].a_channels;
-        int chans = (chansintogo < thisdevchans ? chansintogo : thisdevchans);
-        chansouttogo -= chans;
-        result = snd_pcm_readi(alsai.dev[iodev].a_handle, alsa_snd_buf, transfersize);
-        if (result < (int)transfersize) {
-#ifdef DEBUG_ALSA_XFER
-            if (result<0) error("snd_pcm_read %d %d: %s", callno, xferno, snd_strerror(errno));
-            else post("snd_pcm_read %d %d returned only %d", callno, xferno, result);
-            post("inputcount %d, outputcount %d, inbufsize %d",
-                    inputcount, outputcount, (ALSA_EXTRABUFFER + sys_advance_samples) * alsai.dev[iodev].a_sampwidth * inchannels);
-#endif
-            sys_log_error(ERR_ADCSLEPT);
-            return SENDDACS_NO;
-        }
-        if (alsai.dev[iodev].a_sampwidth == 4) {
-            for (int i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
-                for (j = ch, k = sys_dacblocksize, fp2 = fp1; k--; j += thisdevchans, fp2++)
-                    *fp2 = (float) ((t_alsa_sample32 *)alsa_snd_buf)[j] * (1./ INT32_MAX);
-            }
-        } else {
-            for (int i=0; i<chans; i++, ch++, fp1 += sys_dacblocksize) {
-                for (j = ch, k = sys_dacblocksize, fp2 = fp1; k--; j += thisdevchans, fp2++)
-                    *fp2 = (float) ((t_alsa_sample16 *)alsa_snd_buf)[j] * 3.051850e-05;
-            }
-        }
-    }
-#ifdef DEBUG_ALSA_XFER
-    xferno++;
-#endif
-    if (sys_getrealtime() - timenow > 0.002) {
-#ifdef DEBUG_ALSA_XFER
-        post("routine took %d msec", int(1000 * (sys_getrealtime() - timenow)));
-#endif
-        sys_log_error(ERR_ADCSLEPT);
-    }
-    return SENDDACS_YES;
-}
-
-void alsa_printstate() {
-    int result, i=0;
-    snd_pcm_sframes_t indelay, outdelay;
-    if (sys_audioapi != API_ALSA) {
-        error("restart-audio: implemented for ALSA only.");
-        return;
-    }
-    if (sys_inchannels) {
-        result = snd_pcm_delay(alsai.dev[i].a_handle, &indelay);
-        if (result<0) error("snd_pcm_delay 1 failed"); else post( "in delay %d", int( indelay));
-    }
-    if (sys_outchannels) {
-        result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
-        if (result<0) error("snd_pcm_delay 2 failed"); else post("out delay %d", int(outdelay));
-    }
-    post("sum %ld (%ld mod 64)", indelay + outdelay, (indelay+outdelay)%64);
-    post("buf samples %d", alsa_buf_samps);
-}
-
-
-void alsa_putzeros(int iodev, int n) {
-    memset(alsa_snd_buf, 0, alsao.dev[iodev].a_sampwidth * sys_dacblocksize * alsao.dev[iodev].a_channels);
-    for (int i=0; i<n; i++) snd_pcm_writei(alsao.dev[iodev].a_handle, alsa_snd_buf, sys_dacblocksize);
-}
-
-void alsa_getzeros(int iodev, int n) {
-    for (int i=0; i<n; i++) snd_pcm_readi(alsai.dev[iodev].a_handle, alsa_snd_buf, sys_dacblocksize);
-}
-
-/* call this only if both input and output are open */
-static void alsa_checkiosync() {
-    int result, giveup = 1000, alreadylogged = 0;
-    snd_pcm_sframes_t minphase, maxphase, thisphase, outdelay;
-    while (1) {
-        if (giveup-- <= 0) {post("tried but couldn't sync A/D/A"); alsa_jittermax += 1; return;}
-        minphase = 0x7fffffff;
-        maxphase = -0x7fffffff;
-        for (int i=0; i<alsao.ndev; i++) {
-            result     = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
-            if (result < 0) {
-                snd_pcm_prepare(alsao.dev[i].a_handle);
-                result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
-            }
-            if (result<0) {
-                error("output snd_pcm_delay failed: %s", snd_strerror(result));
-                if (snd_pcm_status(alsao.dev[i].a_handle, alsa_status)<0) error("output snd_pcm_status failed");
-                else post("astate %d", snd_pcm_status_get_state(alsa_status));
-                return;
-            }
-            thisphase = alsa_buf_samps - outdelay;
-            if (thisphase < minphase) minphase = thisphase;
-            if (thisphase > maxphase) maxphase = thisphase;
-            if (outdelay < 0)
-                sys_log_error(ERR_DATALATE), alreadylogged = 1;
-        }
-        for (int i=0; i<alsai.ndev; i++) {
-            result =     snd_pcm_delay(alsai.dev[i].a_handle, &thisphase);
-            if (result < 0) {
-                snd_pcm_prepare(alsai.dev[i].a_handle);
-                result = snd_pcm_delay(alsai.dev[i].a_handle, &thisphase);
-            }
-            if (result < 0) {
-                error("output snd_pcm_delay failed: %s", snd_strerror(result));
-                if (snd_pcm_status(alsao.dev[i].a_handle, alsa_status) < 0) error("output snd_pcm_status failed");
-                else post("astate %d", snd_pcm_status_get_state(alsa_status));
-                return;
-            }
-            if (thisphase < minphase) minphase = thisphase;
-            if (thisphase > maxphase) maxphase = thisphase;
-        }
-        /* the "correct" position is for all the phases to be exactly equal;
-           but since we only make corrections sys_dacblocksize samples at a time,
-           we just ask that the spread be not more than 3/4 of a block.  */
-        if (maxphase <= minphase + (alsa_jittermax * (sys_dacblocksize / 4))) break;
-        if (!alreadylogged) sys_log_error(ERR_RESYNC), alreadylogged = 1;
-        for (int i=0; i<alsao.ndev; i++) {
-            result = snd_pcm_delay(alsao.dev[i].a_handle, &outdelay);
-            if (result < 0) break;
-            thisphase = alsa_buf_samps - outdelay;
-            if (thisphase > minphase + sys_dacblocksize) {
-                alsa_putzeros(i,1);
-#if DEBUGSYNC
-                post("putz %d %d", (int)thisphase, (int)minphase);
-#endif
-            }
-        }
-        for (int i=0; i<alsai.ndev; i++) {
-            result = snd_pcm_delay(alsai.dev[i].a_handle, &thisphase);
-            if (result < 0) break;
-            if (thisphase > minphase + sys_dacblocksize) {
-                alsa_getzeros(i, 1);
-#if DEBUGSYNC
-                post("getz %d %d", (int)thisphase, (int)minphase);
-#endif
-            }
-        }
-    }
-#if DEBUGSYNC
-    if (alreadylogged) post("done");
-#endif
-}
-
-static int   alsa_nnames = 0;
-static char **alsa_names = 0;
-
-void alsa_adddev(char *name) {
-    if (alsa_nnames) alsa_names = (char **)t_resizebytes(alsa_names, alsa_nnames*sizeof(char *), (alsa_nnames+1)*sizeof(char *));
-    else alsa_names = (char **)t_getbytes(sizeof(char *));
-    alsa_names[alsa_nnames] = gensym(name)->s_name;
-    alsa_nnames++;
-}
-
-static void alsa_numbertoname(int devno, char *devname, int nchar) {
-    int ndev = 0, cardno = -1;
-    while (!snd_card_next(&cardno) && cardno >= 0) ndev++;
-    if (devno < 2*ndev) {
-        if (devno & 1) snprintf(devname, nchar, "plughw:%d", devno/2);
-        else snprintf(devname, nchar, "hw:%d", devno/2);
-    } else if (devno <2*ndev + alsa_nnames)
-        snprintf(devname, nchar, "%s", alsa_names[devno - 2*ndev]);
-    else snprintf(devname, nchar, "???");
-}
-
-/* For each hardware card found, we list two devices, the "hard" and
-   "plug" one.  The card scan is derived from portaudio code. */
-static void alsa_getdevs(char *indevlist, int *nindevs, char *outdevlist, int *noutdevs, int *canmulti, int maxndev, int devdescsize) {
-    int ndev = 0, cardno = -1, i, j;
-    *canmulti = 2;  /* supports multiple devices */
-    while (!snd_card_next(&cardno) && cardno >= 0) {
-        snd_ctl_t *ctl;
-        snd_ctl_card_info_t *info;
-        char devname[80];
-        char *desc;
-        if (2*ndev + 2 > maxndev) break;
-        /* apparently, "cardno" is just a counter; but check that here */
-        if (ndev != cardno) post("oops: ALSA cards not reported in order?");
-        sprintf(devname, "hw:%d", cardno);
-        /* post("try %s..", devname); */
-        if (snd_ctl_open(&ctl, devname, 0) >= 0) {
-            snd_ctl_card_info_malloc(&info);
-            snd_ctl_card_info(ctl, info);
-            desc = strdup(snd_ctl_card_info_get_name(info));
-            snd_ctl_card_info_free(info);
-        } else {
-            error("ALSA card scan error");
-            desc = strdup("???");
-        }
-        sprintf(indevlist  +  2*ndev    * devdescsize, "%s (hardware)", desc);
-        sprintf(indevlist  + (2*ndev+1) * devdescsize, "%s (plug-in)",  desc);
-        sprintf(outdevlist +  2*ndev    * devdescsize, "%s (hardware)", desc);
-        sprintf(outdevlist + (2*ndev+1) * devdescsize, "%s (plug-in)",  desc);
-        ndev++;
-        free(desc);
-    }
-    for (i=0, j=2*ndev; i<alsa_nnames; i++, j++) {
-        if (j >= maxndev) break;
-        snprintf(indevlist + j * devdescsize, devdescsize, "%s", alsa_names[i]);
-    }
-    *nindevs = *noutdevs = j;
-}
-
-struct t_audioapi alsa_api = {
-	alsa_open_audio,
-	alsa_close_audio,
-	alsa_send_dacs,
-	alsa_getdevs,
-};
-- 
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