[PATCH 1/2] ALSA: usb-audio: Improve frames size computation

[Alexander Tsoy]

For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.

But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).

This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.

Signed-off-by: Alexander Tsoy <alexander at tsoy.me>
---
 sound/usb/card.h     |  4 ++++
 sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++-----
 sound/usb/endpoint.h |  1 +
 sound/usb/pcm.c      |  2 ++
 4 files changed, 45 insertions(+), 5 deletions(-)

diff --git a/sound/usb/card.h b/sound/usb/card.h
index 395403a2d33f..820e564656ed 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -84,6 +84,10 @@ struct snd_usb_endpoint {
 	dma_addr_t sync_dma;		/* DMA address of syncbuf */
 
 	unsigned int pipe;		/* the data i/o pipe */
+	unsigned int framesize[2];	/* small/large frame sizes in samples */
+	unsigned int sample_rem;	/* remainder from division fs/fps */
+	unsigned int sample_accum;	/* sample accumulator */
+	unsigned int fps;		/* frames per second */
 	unsigned int freqn;		/* nominal sampling rate in fs/fps in Q16.16 format */
 	unsigned int freqm;		/* momentary sampling rate in fs/fps in Q16.16 format */
 	int	   freqshift;		/* how much to shift the feedback value to get Q16.16 */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 4a9a2f6ef5a4..d8dc7cb56d43 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -124,12 +124,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep)
 
 /*
  * For streaming based on information derived from sync endpoints,
- * prepare_outbound_urb_sizes() will call next_packet_size() to
+ * prepare_outbound_urb_sizes() will call slave_next_packet_size() to
  * determine the number of samples to be sent in the next packet.
  *
- * For implicit feedback, next_packet_size() is unused.
+ * For implicit feedback, slave_next_packet_size() is unused.
  */
-int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
+int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep)
 {
 	unsigned long flags;
 	int ret;
@@ -146,6 +146,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
 	return ret;
 }
 
+/*
+ * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes()
+ * will call next_packet_size() to determine the number of samples to be
+ * sent in the next packet.
+ */
+int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
+{
+	int ret;
+
+	if (ep->fill_max)
+		return ep->maxframesize;
+
+	ep->sample_accum += ep->sample_rem;
+	if (ep->sample_accum >= ep->fps) {
+		ep->sample_accum -= ep->fps;
+		ret = ep->framesize[1];
+	} else {
+		ret = ep->framesize[0];
+	}
+
+	return ret;
+}
+
 static void retire_outbound_urb(struct snd_usb_endpoint *ep,
 				struct snd_urb_ctx *urb_ctx)
 {
@@ -190,6 +213,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep,
 
 		if (ctx->packet_size[i])
 			counts = ctx->packet_size[i];
+		else if (ep->sync_master)
+			counts = snd_usb_endpoint_slave_next_packet_size(ep);
 		else
 			counts = snd_usb_endpoint_next_packet_size(ep);
 
@@ -874,10 +899,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
 	ep->maxpacksize = fmt->maxpacksize;
 	ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX);
 
-	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL)
+	if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) {
 		ep->freqn = get_usb_full_speed_rate(rate);
-	else
+		ep->fps = 1000;
+	} else {
 		ep->freqn = get_usb_high_speed_rate(rate);
+		ep->fps = 8000;
+	}
+
+	ep->sample_rem = rate % ep->fps;
+	ep->framesize[0] = rate / ep->fps;
+	ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps;
 
 	/* calculate the frequency in 16.16 format */
 	ep->freqm = ep->freqn;
@@ -936,6 +968,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep)
 	ep->active_mask = 0;
 	ep->unlink_mask = 0;
 	ep->phase = 0;
+	ep->sample_accum = 0;
 
 	snd_usb_endpoint_start_quirk(ep);
 
diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h
index 63a39d4fa8d8..d23fa0a8c11b 100644
--- a/sound/usb/endpoint.h
+++ b/sound/usb/endpoint.h
@@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_free(struct snd_usb_endpoint *ep);
 
 int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep);
+int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep);
 int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep);
 
 void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a4e4064f9aee..b50965ab3b3a 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1579,6 +1579,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs,
 	for (i = 0; i < ctx->packets; i++) {
 		if (ctx->packet_size[i])
 			counts = ctx->packet_size[i];
+		else if (ep->sync_master)
+			counts = snd_usb_endpoint_slave_next_packet_size(ep);
 		else
 			counts = snd_usb_endpoint_next_packet_size(ep);
 
-- 
2.26.2