Use the new set_blksz() interface to set the block size.

This simplifies the code and allows any block size multiple of 1ms to
be used when play and recording number of channels are not the same.

ok mpi@

1 files changed, 34 insertions, 46 deletions

diff --git a/sys/dev/usb/uaudio.c b/sys/dev/usb/uaudio.c
index ac32f857523..a950308ba8e 100644
--- a/sys/dev/usb/uaudio.c
+++ b/sys/dev/usb/uaudio.c
@@ -1,4 +1,4 @@
-/*	$OpenBSD: uaudio.c,v 1.145 2019/08/25 09:16:04 miko Exp $	*/
+/*	$OpenBSD: uaudio.c,v 1.146 2019/09/05 05:38:40 ratchov Exp $	*/
 /*
  * Copyright (c) 2018 Alexandre Ratchov <alex@caoua.org>
  *
@@ -403,7 +403,8 @@ int uaudio_open(void *, int);
 void uaudio_close(void *);
 int uaudio_set_params(void *, int, int, struct audio_params *,
     struct audio_params *);
-int uaudio_round_blocksize(void *, int);
+unsigned int uaudio_set_blksz(void *, int,
+    struct audio_params *, struct audio_params *, unsigned int);
 int uaudio_trigger_output(void *, void *, void *, int,
     void (*)(void *), void *, struct audio_params *);
 int uaudio_trigger_input(void *, void *, void *, int,
@@ -457,7 +458,7 @@ struct audio_hw_if uaudio_hw_if = {
 	uaudio_open,		/* open */
 	uaudio_close,		/* close */
 	uaudio_set_params,	/* set_params */
-	uaudio_round_blocksize,	/* round_blocksize */
+	NULL,			/* round_blocksize */
 	NULL,			/* commit_settings */
 	NULL,			/* init_output */
 	NULL,			/* init_input */
@@ -477,7 +478,8 @@ struct audio_hw_if uaudio_hw_if = {
 	uaudio_trigger_output,	/* trigger_output */
 	uaudio_trigger_input,	/* trigger_input */
 	uaudio_copy_output,	/* copy_output */
-	uaudio_underrun		/* underrun */
+	uaudio_underrun,	/* underrun */
+	uaudio_set_blksz	/* set_blksz */
 };
 
 /*
@@ -3962,57 +3964,43 @@ uaudio_set_params(void *self, int setmode, int usemode,
 	return 0;
 }
 
-int
-uaudio_round_blocksize(void *self, int blksz)
+unsigned int
+uaudio_set_blksz(void *self, int mode,
+    struct audio_params *p, struct audio_params *r, unsigned int blksz)
 {
 	struct uaudio_softc *sc = self;
-	struct uaudio_alt *a;
-	unsigned int rbpf, pbpf;
-	unsigned int blksz_max;
+	unsigned int fps, fps_min;
+	unsigned int blksz_max, blksz_min;
 
 	/*
-	 * XXX: We don't know if we're called for the play or record
-	 * direction, so we can't calculate maximum blksz. This would
-	 * require a change in the audio(9) interface. Meanwhile, we
-	 * use the direction with the greatest sample size; it gives
-	 * the correct result: indeed, if we return:
-	 *
-	 *	blksz_max = max(pbpf, rbpf) * nsamp_max
-	 *
-	 * in turn the audio(4) layer will use:
-	 *
-	 *	min(blksz_max / pbpf, blksz_max / rbpf)
-	 *
-	 * which is exactly nsamp_max.
+	 * minimum block size is two transfers, see uaudio_stream_open()
 	 */
-
-	if (sc->mode & AUMODE_PLAY) {
-		a = sc->params->palt;
-		pbpf = a->bps * a->nch;
-	} else
-		pbpf = 1;
-
-	if (sc->mode & AUMODE_RECORD) {
-		a = sc->params->ralt;
-		rbpf = a->bps * a->nch;
-	} else
-		rbpf = 1;
+	fps_min = sc->ufps;
+	if (mode & AUMODE_PLAY) {
+		fps = sc->params->palt->fps;
+		if (fps_min > fps)
+			fps_min = fps;
+	}
+	if (mode & AUMODE_RECORD) {
+		fps = sc->params->ralt->fps;
+		if (fps_min > fps)
+			fps_min = fps;
+	}
+	blksz_min = (sc->rate * 2 + fps_min - 1) / fps_min;
 
 	/*
-	 * Limit the block size to (slightly more than):
-	 *
-	 *	sc->host_nframes / UAUDIO_NXFERS_MIN
-	 *
-	 * (micro-)frames of audio. Transfers are slightly larger than
-	 * the audio block size (few bytes to make the "safe" block
-	 * size plus one extra millisecond). We reserve an extra 15%
-	 * for that.
+	 * max block size is only limited by the number of frames the
+	 * host can schedule
 	 */
-	blksz_max = (pbpf > rbpf ? pbpf : rbpf) *
-	    sc->rate * (sc->host_nframes / UAUDIO_NXFERS_MIN) / sc->ufps *
-	    85 / 100;
+	blksz_max = sc->rate * (sc->host_nframes / UAUDIO_NXFERS_MIN) /
+	    sc->ufps * 85 / 100;
+
+	if (blksz > blksz_max)
+		blksz = blksz_max;
+	else if (blksz < blksz_min)
+		blksz = blksz_min;
 
-	return blksz < blksz_max ? blksz : blksz_max;
+	return blksz;
 }
 
 int