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authorAlexander Tsoy <alexander@tsoy.me>2020-04-24 05:24:48 +0300
committerTakashi Iwai <tiwai@suse.de>2020-04-24 08:25:24 +0200
commitf0bd62b64016508938df9babe47f65c2c727d25c (patch)
tree4f7be7dcbf03a436e603b8245ad1deddcbec8dc5
parent10635d2d2a4c95044217fe8ea6e5f5451fd07ae8 (diff)
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ALSA: usb-audio: Improve frames size computation
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@tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai@suse.de>
-rw-r--r--sound/usb/card.h4
-rw-r--r--sound/usb/endpoint.c43
-rw-r--r--sound/usb/endpoint.h1
-rw-r--r--sound/usb/pcm.c2
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);