linux/sound/usb/endpoint.c

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/*
* 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 the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
#include <linux/gfp.h>
#include <linux/init.h>
#include <linux/ratelimit.h>
#include <linux/usb.h>
#include <linux/usb/audio.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include "usbaudio.h"
#include "helper.h"
#include "card.h"
#include "endpoint.h"
#include "pcm.h"
#include "quirks.h"
#define EP_FLAG_RUNNING 1
#define EP_FLAG_STOPPING 2
/*
* snd_usb_endpoint is a model that abstracts everything related to an
* USB endpoint and its streaming.
*
* There are functions to activate and deactivate the streaming URBs and
* optional callbacks to let the pcm logic handle the actual content of the
* packets for playback and record. Thus, the bus streaming and the audio
* handlers are fully decoupled.
*
* There are two different types of endpoints in audio applications.
*
* SND_USB_ENDPOINT_TYPE_DATA handles full audio data payload for both
* inbound and outbound traffic.
*
* SND_USB_ENDPOINT_TYPE_SYNC endpoints are for inbound traffic only and
* expect the payload to carry Q10.14 / Q16.16 formatted sync information
* (3 or 4 bytes).
*
* Each endpoint has to be configured prior to being used by calling
* snd_usb_endpoint_set_params().
*
* The model incorporates a reference counting, so that multiple users
* can call snd_usb_endpoint_start() and snd_usb_endpoint_stop(), and
* only the first user will effectively start the URBs, and only the last
* one to stop it will tear the URBs down again.
*/
/*
* convert a sampling rate into our full speed format (fs/1000 in Q16.16)
* this will overflow at approx 524 kHz
*/
static inline unsigned get_usb_full_speed_rate(unsigned int rate)
{
return ((rate << 13) + 62) / 125;
}
/*
* convert a sampling rate into USB high speed format (fs/8000 in Q16.16)
* this will overflow at approx 4 MHz
*/
static inline unsigned get_usb_high_speed_rate(unsigned int rate)
{
return ((rate << 10) + 62) / 125;
}
/*
* release a urb data
*/
static void release_urb_ctx(struct snd_urb_ctx *u)
{
if (u->buffer_size)
usb_free_coherent(u->ep->chip->dev, u->buffer_size,
u->urb->transfer_buffer,
u->urb->transfer_dma);
usb_free_urb(u->urb);
u->urb = NULL;
}
static const char *usb_error_string(int err)
{
switch (err) {
case -ENODEV:
return "no device";
case -ENOENT:
return "endpoint not enabled";
case -EPIPE:
return "endpoint stalled";
case -ENOSPC:
return "not enough bandwidth";
case -ESHUTDOWN:
return "device disabled";
case -EHOSTUNREACH:
return "device suspended";
case -EINVAL:
case -EAGAIN:
case -EFBIG:
case -EMSGSIZE:
return "internal error";
default:
return "unknown error";
}
}
/**
* snd_usb_endpoint_implicit_feedback_sink: Report endpoint usage type
*
* @ep: The snd_usb_endpoint
*
* Determine whether an endpoint is driven by an implicit feedback
* data endpoint source.
*/
int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep)
{
return ep->sync_master &&
ep->sync_master->type == SND_USB_ENDPOINT_TYPE_DATA &&
ep->type == SND_USB_ENDPOINT_TYPE_DATA &&
usb_pipeout(ep->pipe);
}
/*
* For streaming based on information derived from sync endpoints,
* prepare_outbound_urb_sizes() will call next_packet_size() to
* determine the number of samples to be sent in the next packet.
*
* For implicit feedback, next_packet_size() is unused.
*/
int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep)
{
unsigned long flags;
int ret;
if (ep->fill_max)
return ep->maxframesize;
spin_lock_irqsave(&ep->lock, flags);
ep->phase = (ep->phase & 0xffff)
+ (ep->freqm << ep->datainterval);
ret = min(ep->phase >> 16, ep->maxframesize);
spin_unlock_irqrestore(&ep->lock, flags);
return ret;
}
static void retire_outbound_urb(struct snd_usb_endpoint *ep,
struct snd_urb_ctx *urb_ctx)
{
if (ep->retire_data_urb)
ep->retire_data_urb(ep->data_subs, urb_ctx->urb);
}
static void retire_inbound_urb(struct snd_usb_endpoint *ep,
struct snd_urb_ctx *urb_ctx)
{
struct urb *urb = urb_ctx->urb;
if (unlikely(ep->skip_packets > 0)) {
ep->skip_packets--;
return;
}
if (ep->sync_slave)
snd_usb_handle_sync_urb(ep->sync_slave, ep, urb);
if (ep->retire_data_urb)
ep->retire_data_urb(ep->data_subs, urb);
}
/*
* Prepare a PLAYBACK urb for submission to the bus.
*/
static void prepare_outbound_urb(struct snd_usb_endpoint *ep,
struct snd_urb_ctx *ctx)
{
int i;
struct urb *urb = ctx->urb;
unsigned char *cp = urb->transfer_buffer;
urb->dev = ep->chip->dev; /* we need to set this at each time */
switch (ep->type) {
case SND_USB_ENDPOINT_TYPE_DATA:
if (ep->prepare_data_urb) {
ep->prepare_data_urb(ep->data_subs, urb);
} else {
/* no data provider, so send silence */
unsigned int offs = 0;
for (i = 0; i < ctx->packets; ++i) {
int counts;
if (ctx->packet_size[i])
counts = ctx->packet_size[i];
else
counts = snd_usb_endpoint_next_packet_size(ep);
urb->iso_frame_desc[i].offset = offs * ep->stride;
urb->iso_frame_desc[i].length = counts * ep->stride;
offs += counts;
}
urb->number_of_packets = ctx->packets;
urb->transfer_buffer_length = offs * ep->stride;
memset(urb->transfer_buffer, ep->silence_value,
offs * ep->stride);
}
break;
case SND_USB_ENDPOINT_TYPE_SYNC:
if (snd_usb_get_speed(ep->chip->dev) >= USB_SPEED_HIGH) {
/*
* fill the length and offset of each urb descriptor.
* the fixed 12.13 frequency is passed as 16.16 through the pipe.
*/
urb->iso_frame_desc[0].length = 4;
urb->iso_frame_desc[0].offset = 0;
cp[0] = ep->freqn;
cp[1] = ep->freqn >> 8;
cp[2] = ep->freqn >> 16;
cp[3] = ep->freqn >> 24;
} else {
/*
* fill the length and offset of each urb descriptor.
* the fixed 10.14 frequency is passed through the pipe.
*/
urb->iso_frame_desc[0].length = 3;
urb->iso_frame_desc[0].offset = 0;
cp[0] = ep->freqn >> 2;
cp[1] = ep->freqn >> 10;
cp[2] = ep->freqn >> 18;
}
break;
}
}
/*
* Prepare a CAPTURE or SYNC urb for submission to the bus.
*/
static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep,
struct snd_urb_ctx *urb_ctx)
{
int i, offs;
struct urb *urb = urb_ctx->urb;
urb->dev = ep->chip->dev; /* we need to set this at each time */
switch (ep->type) {
case SND_USB_ENDPOINT_TYPE_DATA:
offs = 0;
for (i = 0; i < urb_ctx->packets; i++) {
urb->iso_frame_desc[i].offset = offs;
urb->iso_frame_desc[i].length = ep->curpacksize;
offs += ep->curpacksize;
}
urb->transfer_buffer_length = offs;
urb->number_of_packets = urb_ctx->packets;
break;
case SND_USB_ENDPOINT_TYPE_SYNC:
urb->iso_frame_desc[0].length = min(4u, ep->syncmaxsize);
urb->iso_frame_desc[0].offset = 0;
break;
}
}
/*
* Send output urbs that have been prepared previously. URBs are dequeued
* from ep->ready_playback_urbs and in case there there aren't any available
* or there are no packets that have been prepared, this function does
* nothing.
*
* The reason why the functionality of sending and preparing URBs is separated
* is that host controllers don't guarantee the order in which they return
* inbound and outbound packets to their submitters.
*
* This function is only used for implicit feedback endpoints. For endpoints
* driven by dedicated sync endpoints, URBs are immediately re-submitted
* from their completion handler.
*/
static void queue_pending_output_urbs(struct snd_usb_endpoint *ep)
{
while (test_bit(EP_FLAG_RUNNING, &ep->flags)) {
unsigned long flags;
struct snd_usb_packet_info *uninitialized_var(packet);
struct snd_urb_ctx *ctx = NULL;
struct urb *urb;
int err, i;
spin_lock_irqsave(&ep->lock, flags);
if (ep->next_packet_read_pos != ep->next_packet_write_pos) {
packet = ep->next_packet + ep->next_packet_read_pos;
ep->next_packet_read_pos++;
ep->next_packet_read_pos %= MAX_URBS;
/* take URB out of FIFO */
if (!list_empty(&ep->ready_playback_urbs))
ctx = list_first_entry(&ep->ready_playback_urbs,
struct snd_urb_ctx, ready_list);
}
spin_unlock_irqrestore(&ep->lock, flags);
if (ctx == NULL)
return;
list_del_init(&ctx->ready_list);
urb = ctx->urb;
/* copy over the length information */
for (i = 0; i < packet->packets; i++)
ctx->packet_size[i] = packet->packet_size[i];
/* call the data handler to fill in playback data */
prepare_outbound_urb(ep, ctx);
err = usb_submit_urb(ctx->urb, GFP_ATOMIC);
if (err < 0)
usb_audio_err(ep->chip,
"Unable to submit urb #%d: %d (urb %p)\n",
ctx->index, err, ctx->urb);
else
set_bit(ctx->index, &ep->active_mask);
}
}
/*
* complete callback for urbs
*/
static void snd_complete_urb(struct urb *urb)
{
struct snd_urb_ctx *ctx = urb->context;
struct snd_usb_endpoint *ep = ctx->ep;
struct snd_pcm_substream *substream;
unsigned long flags;
int err;
if (unlikely(urb->status == -ENOENT || /* unlinked */
urb->status == -ENODEV || /* device removed */
urb->status == -ECONNRESET || /* unlinked */
ALSA: usb-audio: Avoid nested autoresume calls After the recent fix of runtime PM for USB-audio driver, we got a lockdep warning like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc8+ #61 Not tainted --------------------------------------------- pulseaudio/980 is trying to acquire lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] but task is already holding lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] This comes from snd_usb_autoresume() invoking down_read() and it's used in a nested way. Although it's basically safe, per se (as these are read locks), it's better to reduce such spurious warnings. The read lock is needed to guarantee the execution of "shutdown" (cleanup at disconnection) task after all concurrent tasks are finished. This can be implemented in another better way. Also, the current check of chip->in_pm isn't good enough for protecting the racy execution of multiple auto-resumes. This patch rewrites the logic of snd_usb_autoresume() & co; namely, - The recursive call of autopm is avoided by the new refcount, chip->active. The chip->in_pm flag is removed accordingly. - Instead of rwsem, another refcount, chip->usage_count, is introduced for tracking the period to delay the shutdown procedure. At the last clear of this refcount, wake_up() to the shutdown waiter is called. - The shutdown flag is replaced with shutdown atomic count; this is for reducing the lock. - Two new helpers are introduced to simplify the management of these refcounts; snd_usb_lock_shutdown() increases the usage_count, checks the shutdown state, and does autoresume. snd_usb_unlock_shutdown() does the opposite. Most of mixer and other codes just need this, and simply returns an error if it receives an error from lock. Fixes: 9003ebb13f61 ('ALSA: usb-audio: Fix runtime PM unbalance') Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-25 14:09:00 +00:00
urb->status == -ESHUTDOWN)) /* device disabled */
goto exit_clear;
/* device disconnected */
if (unlikely(atomic_read(&ep->chip->shutdown)))
goto exit_clear;
if (usb_pipeout(ep->pipe)) {
retire_outbound_urb(ep, ctx);
/* can be stopped during retire callback */
if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
goto exit_clear;
if (snd_usb_endpoint_implicit_feedback_sink(ep)) {
spin_lock_irqsave(&ep->lock, flags);
list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
spin_unlock_irqrestore(&ep->lock, flags);
queue_pending_output_urbs(ep);
goto exit_clear;
}
prepare_outbound_urb(ep, ctx);
} else {
retire_inbound_urb(ep, ctx);
/* can be stopped during retire callback */
if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags)))
goto exit_clear;
prepare_inbound_urb(ep, ctx);
}
err = usb_submit_urb(urb, GFP_ATOMIC);
if (err == 0)
return;
usb_audio_err(ep->chip, "cannot submit urb (err = %d)\n", err);
if (ep->data_subs && ep->data_subs->pcm_substream) {
substream = ep->data_subs->pcm_substream;
snd_pcm_stop_xrun(substream);
}
exit_clear:
clear_bit(ctx->index, &ep->active_mask);
}
/**
* snd_usb_add_endpoint: Add an endpoint to an USB audio chip
*
* @chip: The chip
* @alts: The USB host interface
* @ep_num: The number of the endpoint to use
* @direction: SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE
* @type: SND_USB_ENDPOINT_TYPE_DATA or SND_USB_ENDPOINT_TYPE_SYNC
*
* If the requested endpoint has not been added to the given chip before,
* a new instance is created. Otherwise, a pointer to the previoulsy
* created instance is returned. In case of any error, NULL is returned.
*
* New endpoints will be added to chip->ep_list and must be freed by
* calling snd_usb_endpoint_free().
*/
struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
struct usb_host_interface *alts,
int ep_num, int direction, int type)
{
struct snd_usb_endpoint *ep;
int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
if (WARN_ON(!alts))
return NULL;
mutex_lock(&chip->mutex);
list_for_each_entry(ep, &chip->ep_list, list) {
if (ep->ep_num == ep_num &&
ep->iface == alts->desc.bInterfaceNumber &&
ep->altsetting == alts->desc.bAlternateSetting) {
usb_audio_dbg(ep->chip,
"Re-using EP %x in iface %d,%d @%p\n",
ep_num, ep->iface, ep->altsetting, ep);
goto __exit_unlock;
}
}
usb_audio_dbg(chip, "Creating new %s %s endpoint #%x\n",
is_playback ? "playback" : "capture",
type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync",
ep_num);
ep = kzalloc(sizeof(*ep), GFP_KERNEL);
if (!ep)
goto __exit_unlock;
ep->chip = chip;
spin_lock_init(&ep->lock);
ep->type = type;
ep->ep_num = ep_num;
ep->iface = alts->desc.bInterfaceNumber;
ep->altsetting = alts->desc.bAlternateSetting;
INIT_LIST_HEAD(&ep->ready_playback_urbs);
ep_num &= USB_ENDPOINT_NUMBER_MASK;
if (is_playback)
ep->pipe = usb_sndisocpipe(chip->dev, ep_num);
else
ep->pipe = usb_rcvisocpipe(chip->dev, ep_num);
if (type == SND_USB_ENDPOINT_TYPE_SYNC) {
if (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
get_endpoint(alts, 1)->bRefresh >= 1 &&
get_endpoint(alts, 1)->bRefresh <= 9)
ep->syncinterval = get_endpoint(alts, 1)->bRefresh;
else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL)
ep->syncinterval = 1;
else if (get_endpoint(alts, 1)->bInterval >= 1 &&
get_endpoint(alts, 1)->bInterval <= 16)
ep->syncinterval = get_endpoint(alts, 1)->bInterval - 1;
else
ep->syncinterval = 3;
ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize);
if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ &&
ep->syncmaxsize == 4)
ep->udh01_fb_quirk = 1;
}
list_add_tail(&ep->list, &chip->ep_list);
__exit_unlock:
mutex_unlock(&chip->mutex);
return ep;
}
/*
* wait until all urbs are processed.
*/
static int wait_clear_urbs(struct snd_usb_endpoint *ep)
{
unsigned long end_time = jiffies + msecs_to_jiffies(1000);
int alive;
do {
alive = bitmap_weight(&ep->active_mask, ep->nurbs);
if (!alive)
break;
schedule_timeout_uninterruptible(1);
} while (time_before(jiffies, end_time));
if (alive)
usb_audio_err(ep->chip,
"timeout: still %d active urbs on EP #%x\n",
alive, ep->ep_num);
clear_bit(EP_FLAG_STOPPING, &ep->flags);
return 0;
}
/* sync the pending stop operation;
* this function itself doesn't trigger the stop operation
*/
void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep)
{
if (ep && test_bit(EP_FLAG_STOPPING, &ep->flags))
wait_clear_urbs(ep);
}
/*
* unlink active urbs.
*/
static int deactivate_urbs(struct snd_usb_endpoint *ep, bool force)
{
unsigned int i;
ALSA: usb-audio: Avoid nested autoresume calls After the recent fix of runtime PM for USB-audio driver, we got a lockdep warning like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc8+ #61 Not tainted --------------------------------------------- pulseaudio/980 is trying to acquire lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] but task is already holding lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] This comes from snd_usb_autoresume() invoking down_read() and it's used in a nested way. Although it's basically safe, per se (as these are read locks), it's better to reduce such spurious warnings. The read lock is needed to guarantee the execution of "shutdown" (cleanup at disconnection) task after all concurrent tasks are finished. This can be implemented in another better way. Also, the current check of chip->in_pm isn't good enough for protecting the racy execution of multiple auto-resumes. This patch rewrites the logic of snd_usb_autoresume() & co; namely, - The recursive call of autopm is avoided by the new refcount, chip->active. The chip->in_pm flag is removed accordingly. - Instead of rwsem, another refcount, chip->usage_count, is introduced for tracking the period to delay the shutdown procedure. At the last clear of this refcount, wake_up() to the shutdown waiter is called. - The shutdown flag is replaced with shutdown atomic count; this is for reducing the lock. - Two new helpers are introduced to simplify the management of these refcounts; snd_usb_lock_shutdown() increases the usage_count, checks the shutdown state, and does autoresume. snd_usb_unlock_shutdown() does the opposite. Most of mixer and other codes just need this, and simply returns an error if it receives an error from lock. Fixes: 9003ebb13f61 ('ALSA: usb-audio: Fix runtime PM unbalance') Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-25 14:09:00 +00:00
if (!force && atomic_read(&ep->chip->shutdown)) /* to be sure... */
return -EBADFD;
clear_bit(EP_FLAG_RUNNING, &ep->flags);
INIT_LIST_HEAD(&ep->ready_playback_urbs);
ep->next_packet_read_pos = 0;
ep->next_packet_write_pos = 0;
for (i = 0; i < ep->nurbs; i++) {
if (test_bit(i, &ep->active_mask)) {
if (!test_and_set_bit(i, &ep->unlink_mask)) {
struct urb *u = ep->urb[i].urb;
usb_unlink_urb(u);
}
}
}
return 0;
}
/*
* release an endpoint's urbs
*/
static void release_urbs(struct snd_usb_endpoint *ep, int force)
{
int i;
/* route incoming urbs to nirvana */
ep->retire_data_urb = NULL;
ep->prepare_data_urb = NULL;
/* stop urbs */
deactivate_urbs(ep, force);
wait_clear_urbs(ep);
for (i = 0; i < ep->nurbs; i++)
release_urb_ctx(&ep->urb[i]);
if (ep->syncbuf)
usb_free_coherent(ep->chip->dev, SYNC_URBS * 4,
ep->syncbuf, ep->sync_dma);
ep->syncbuf = NULL;
ep->nurbs = 0;
}
/*
* configure a data endpoint
*/
static int data_ep_set_params(struct snd_usb_endpoint *ep,
snd_pcm_format_t pcm_format,
unsigned int channels,
unsigned int period_bytes,
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
unsigned int frames_per_period,
unsigned int periods_per_buffer,
struct audioformat *fmt,
struct snd_usb_endpoint *sync_ep)
{
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb;
unsigned int max_packs_per_period, urbs_per_period, urb_packs;
unsigned int max_urbs, i;
int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels;
if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) {
/*
* When operating in DSD DOP mode, the size of a sample frame
* in hardware differs from the actual physical format width
* because we need to make room for the DOP markers.
*/
frame_bits += channels << 3;
}
ep->datainterval = fmt->datainterval;
ep->stride = frame_bits >> 3;
ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0;
/* assume max. frequency is 25% higher than nominal */
ep->freqmax = ep->freqn + (ep->freqn >> 2);
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
/* but wMaxPacketSize might reduce this */
if (ep->maxpacksize && ep->maxpacksize < maxsize) {
/* whatever fits into a max. size packet */
maxsize = ep->maxpacksize;
ep->freqmax = (maxsize / (frame_bits >> 3))
<< (16 - ep->datainterval);
}
if (ep->fill_max)
ep->curpacksize = ep->maxpacksize;
else
ep->curpacksize = maxsize;
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) {
packs_per_ms = 8 >> ep->datainterval;
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
max_packs_per_urb = MAX_PACKS_HS;
} else {
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
packs_per_ms = 1;
max_packs_per_urb = MAX_PACKS;
}
if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep))
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
max_packs_per_urb = min(max_packs_per_urb,
1U << sync_ep->syncinterval);
max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval);
/*
* Capture endpoints need to use small URBs because there's no way
* to tell in advance where the next period will end, and we don't
* want the next URB to complete much after the period ends.
*
* Playback endpoints with implicit sync much use the same parameters
* as their corresponding capture endpoint.
*/
if (usb_pipein(ep->pipe) ||
snd_usb_endpoint_implicit_feedback_sink(ep)) {
urb_packs = packs_per_ms;
/*
* Wireless devices can poll at a max rate of once per 4ms.
* For dataintervals less than 5, increase the packet count to
* allow the host controller to use bursting to fill in the
* gaps.
*/
if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) {
int interval = ep->datainterval;
while (interval < 5) {
urb_packs <<= 1;
++interval;
}
}
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
/* make capture URBs <= 1 ms and smaller than a period */
urb_packs = min(max_packs_per_urb, urb_packs);
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
while (urb_packs > 1 && urb_packs * maxsize >= period_bytes)
urb_packs >>= 1;
ep->nurbs = MAX_URBS;
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
/*
* Playback endpoints without implicit sync are adjusted so that
* a period fits as evenly as possible in the smallest number of
* URBs. The total number of URBs is adjusted to the size of the
* ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits.
*/
} else {
/* determine how small a packet can be */
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
minsize = (ep->freqn >> (16 - ep->datainterval)) *
(frame_bits >> 3);
/* with sync from device, assume it can be 12% lower */
if (sync_ep)
minsize -= minsize >> 3;
minsize = max(minsize, 1u);
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
/* how many packets will contain an entire ALSA period? */
max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize);
/* how many URBs will contain a period? */
urbs_per_period = DIV_ROUND_UP(max_packs_per_period,
max_packs_per_urb);
/* how many packets are needed in each URB? */
urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period);
/* limit the number of frames in a single URB */
ep->max_urb_frames = DIV_ROUND_UP(frames_per_period,
urbs_per_period);
/* try to use enough URBs to contain an entire ALSA buffer */
max_urbs = min((unsigned) MAX_URBS,
MAX_QUEUE * packs_per_ms / urb_packs);
ep->nurbs = min(max_urbs, urbs_per_period * periods_per_buffer);
}
/* allocate and initialize data urbs */
for (i = 0; i < ep->nurbs; i++) {
struct snd_urb_ctx *u = &ep->urb[i];
u->index = i;
u->ep = ep;
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
u->packets = urb_packs;
u->buffer_size = maxsize * u->packets;
if (fmt->fmt_type == UAC_FORMAT_TYPE_II)
u->packets++; /* for transfer delimiter */
u->urb = usb_alloc_urb(u->packets, GFP_KERNEL);
if (!u->urb)
goto out_of_memory;
u->urb->transfer_buffer =
usb_alloc_coherent(ep->chip->dev, u->buffer_size,
GFP_KERNEL, &u->urb->transfer_dma);
if (!u->urb->transfer_buffer)
goto out_of_memory;
u->urb->pipe = ep->pipe;
u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
u->urb->interval = 1 << ep->datainterval;
u->urb->context = u;
u->urb->complete = snd_complete_urb;
INIT_LIST_HEAD(&u->ready_list);
}
return 0;
out_of_memory:
release_urbs(ep, 0);
return -ENOMEM;
}
/*
* configure a sync endpoint
*/
static int sync_ep_set_params(struct snd_usb_endpoint *ep)
{
int i;
ep->syncbuf = usb_alloc_coherent(ep->chip->dev, SYNC_URBS * 4,
GFP_KERNEL, &ep->sync_dma);
if (!ep->syncbuf)
return -ENOMEM;
for (i = 0; i < SYNC_URBS; i++) {
struct snd_urb_ctx *u = &ep->urb[i];
u->index = i;
u->ep = ep;
u->packets = 1;
u->urb = usb_alloc_urb(1, GFP_KERNEL);
if (!u->urb)
goto out_of_memory;
u->urb->transfer_buffer = ep->syncbuf + i * 4;
u->urb->transfer_dma = ep->sync_dma + i * 4;
u->urb->transfer_buffer_length = 4;
u->urb->pipe = ep->pipe;
u->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
u->urb->number_of_packets = 1;
u->urb->interval = 1 << ep->syncinterval;
u->urb->context = u;
u->urb->complete = snd_complete_urb;
}
ep->nurbs = SYNC_URBS;
return 0;
out_of_memory:
release_urbs(ep, 0);
return -ENOMEM;
}
/**
* snd_usb_endpoint_set_params: configure an snd_usb_endpoint
*
* @ep: the snd_usb_endpoint to configure
* @pcm_format: the audio fomat.
* @channels: the number of audio channels.
* @period_bytes: the number of bytes in one alsa period.
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
* @period_frames: the number of frames in one alsa period.
* @buffer_periods: the number of periods in one alsa buffer.
* @rate: the frame rate.
* @fmt: the USB audio format information
* @sync_ep: the sync endpoint to use, if any
*
* Determine the number of URBs to be used on this endpoint.
* An endpoint must be configured before it can be started.
* An endpoint that is already running can not be reconfigured.
*/
int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep,
snd_pcm_format_t pcm_format,
unsigned int channels,
unsigned int period_bytes,
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
unsigned int period_frames,
unsigned int buffer_periods,
unsigned int rate,
struct audioformat *fmt,
struct snd_usb_endpoint *sync_ep)
{
int err;
if (ep->use_count != 0) {
usb_audio_warn(ep->chip,
"Unable to change format on ep #%x: already in use\n",
ep->ep_num);
return -EBUSY;
}
/* release old buffers, if any */
release_urbs(ep, 0);
ep->datainterval = fmt->datainterval;
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)
ep->freqn = get_usb_full_speed_rate(rate);
else
ep->freqn = get_usb_high_speed_rate(rate);
/* calculate the frequency in 16.16 format */
ep->freqm = ep->freqn;
ep->freqshift = INT_MIN;
ep->phase = 0;
switch (ep->type) {
case SND_USB_ENDPOINT_TYPE_DATA:
err = data_ep_set_params(ep, pcm_format, channels,
ALSA: improve buffer size computations for USB PCM audio This patch changes the way URBs are allocated and their sizes are determined for PCM playback in the snd-usb-audio driver. Currently the driver allocates too few URBs for endpoints that don't use implicit sync, making underruns more likely to occur. This may be a holdover from before I/O delays could be measured accurately; in any case, it is no longer necessary. The patch allocates as many URBs as possible, subject to four limitations: The total number of URBs for the endpoint is not allowed to exceed MAX_URBS (which the patch increases from 8 to 12). The total number of packets per URB is not allowed to exceed MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is decreased from 20 to 6. The total duration of queued data is not allowed to exceed MAX_QUEUE, which is decreased from 24 ms to 18 ms. The total number of ALSA frames in the output queue is not allowed to exceed the ALSA buffer size. The last requirement is the hardest to implement. Currently the number of URBs needed to fill a buffer cannot be determined in advance, because a buffer contains a fixed number of frames whereas the number of frames in an URB varies to match shifts in the device's clock rate. To solve this problem, the patch changes the logic for deciding how many packets an URB should contain. Rather than using as many as possible without exceeding an ALSA period boundary, now the driver uses only as many packets as needed to transfer a predetermined number of frames. As a result, unless the device's clock has an exceedingly variable rate, the number of URBs making up each period (and hence each buffer) will remain constant. The overall effect of the patch is that playback works better in low-latency settings. The user can still specify values for frames/period and periods/buffer that exceed the capabilities of the hardware, of course. But for values that are within those capabilities, the performance will be improved. For example, testing shows that a high-speed device can handle 32 frames/period and 3 periods/buffer at 48 KHz, whereas the current driver starts to get glitchy at 64 frames/period and 2 periods/buffer. A side effect of these changes is that the "nrpacks" module parameter is no longer used. The patch removes it. Signed-off-by: Alan Stern <stern@rowland.harvard.edu> CC: Clemens Ladisch <clemens@ladisch.de> Tested-by: Daniel Mack <zonque@gmail.com> Tested-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-24 19:51:58 +00:00
period_bytes, period_frames,
buffer_periods, fmt, sync_ep);
break;
case SND_USB_ENDPOINT_TYPE_SYNC:
err = sync_ep_set_params(ep);
break;
default:
err = -EINVAL;
}
usb_audio_dbg(ep->chip,
"Setting params for ep #%x (type %d, %d urbs), ret=%d\n",
ep->ep_num, ep->type, ep->nurbs, err);
return err;
}
/**
* snd_usb_endpoint_start: start an snd_usb_endpoint
*
* @ep: the endpoint to start
* @can_sleep: flag indicating whether the operation is executed in
* non-atomic context
*
* A call to this function will increment the use count of the endpoint.
* In case it is not already running, the URBs for this endpoint will be
* submitted. Otherwise, this function does nothing.
*
* Must be balanced to calls of snd_usb_endpoint_stop().
*
* Returns an error if the URB submission failed, 0 in all other cases.
*/
int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep)
{
int err;
unsigned int i;
ALSA: usb-audio: Avoid nested autoresume calls After the recent fix of runtime PM for USB-audio driver, we got a lockdep warning like: ============================================= [ INFO: possible recursive locking detected ] 4.2.0-rc8+ #61 Not tainted --------------------------------------------- pulseaudio/980 is trying to acquire lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] but task is already holding lock: (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio] This comes from snd_usb_autoresume() invoking down_read() and it's used in a nested way. Although it's basically safe, per se (as these are read locks), it's better to reduce such spurious warnings. The read lock is needed to guarantee the execution of "shutdown" (cleanup at disconnection) task after all concurrent tasks are finished. This can be implemented in another better way. Also, the current check of chip->in_pm isn't good enough for protecting the racy execution of multiple auto-resumes. This patch rewrites the logic of snd_usb_autoresume() & co; namely, - The recursive call of autopm is avoided by the new refcount, chip->active. The chip->in_pm flag is removed accordingly. - Instead of rwsem, another refcount, chip->usage_count, is introduced for tracking the period to delay the shutdown procedure. At the last clear of this refcount, wake_up() to the shutdown waiter is called. - The shutdown flag is replaced with shutdown atomic count; this is for reducing the lock. - Two new helpers are introduced to simplify the management of these refcounts; snd_usb_lock_shutdown() increases the usage_count, checks the shutdown state, and does autoresume. snd_usb_unlock_shutdown() does the opposite. Most of mixer and other codes just need this, and simply returns an error if it receives an error from lock. Fixes: 9003ebb13f61 ('ALSA: usb-audio: Fix runtime PM unbalance') Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-25 14:09:00 +00:00
if (atomic_read(&ep->chip->shutdown))
return -EBADFD;
/* already running? */
if (++ep->use_count != 1)
return 0;
/* just to be sure */
deactivate_urbs(ep, false);
if (can_sleep)
wait_clear_urbs(ep);
ep->active_mask = 0;
ep->unlink_mask = 0;
ep->phase = 0;
snd_usb_endpoint_start_quirk(ep);
/*
* If this endpoint has a data endpoint as implicit feedback source,
* don't start the urbs here. Instead, mark them all as available,
* wait for the record urbs to return and queue the playback urbs
* from that context.
*/
set_bit(EP_FLAG_RUNNING, &ep->flags);
if (snd_usb_endpoint_implicit_feedback_sink(ep)) {
for (i = 0; i < ep->nurbs; i++) {
struct snd_urb_ctx *ctx = ep->urb + i;
list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs);
}
return 0;
}
for (i = 0; i < ep->nurbs; i++) {
struct urb *urb = ep->urb[i].urb;
if (snd_BUG_ON(!urb))
goto __error;
if (usb_pipeout(ep->pipe)) {
prepare_outbound_urb(ep, urb->context);
} else {
prepare_inbound_urb(ep, urb->context);
}
err = usb_submit_urb(urb, GFP_ATOMIC);
if (err < 0) {
usb_audio_err(ep->chip,
"cannot submit urb %d, error %d: %s\n",
i, err, usb_error_string(err));
goto __error;
}
set_bit(i, &ep->active_mask);
}
return 0;
__error:
clear_bit(EP_FLAG_RUNNING, &ep->flags);
ep->use_count--;
deactivate_urbs(ep, false);
return -EPIPE;
}
/**
* snd_usb_endpoint_stop: stop an snd_usb_endpoint
*
* @ep: the endpoint to stop (may be NULL)
*
* A call to this function will decrement the use count of the endpoint.
* In case the last user has requested the endpoint stop, the URBs will
* actually be deactivated.
*
* Must be balanced to calls of snd_usb_endpoint_start().
*
* The caller needs to synchronize the pending stop operation via
* snd_usb_endpoint_sync_pending_stop().
*/
void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep)
{
if (!ep)
return;
if (snd_BUG_ON(ep->use_count == 0))
return;
if (--ep->use_count == 0) {
deactivate_urbs(ep, false);
ep->data_subs = NULL;
ep->sync_slave = NULL;
ep->retire_data_urb = NULL;
ep->prepare_data_urb = NULL;
set_bit(EP_FLAG_STOPPING, &ep->flags);
}
}
/**
* snd_usb_endpoint_deactivate: deactivate an snd_usb_endpoint
*
* @ep: the endpoint to deactivate
*
* If the endpoint is not currently in use, this functions will
* deactivate its associated URBs.
*
* In case of any active users, this functions does nothing.
*/
void snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep)
{
if (!ep)
return;
if (ep->use_count != 0)
return;
deactivate_urbs(ep, true);
wait_clear_urbs(ep);
}
/**
* snd_usb_endpoint_release: Tear down an snd_usb_endpoint
*
* @ep: the endpoint to release
*
* This function does not care for the endpoint's use count but will tear
* down all the streaming URBs immediately.
*/
void snd_usb_endpoint_release(struct snd_usb_endpoint *ep)
{
release_urbs(ep, 1);
}
/**
* snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint
*
* @ep: the endpoint to free
*
* This free all resources of the given ep.
*/
void snd_usb_endpoint_free(struct snd_usb_endpoint *ep)
{
kfree(ep);
}
/**
* snd_usb_handle_sync_urb: parse an USB sync packet
*
* @ep: the endpoint to handle the packet
* @sender: the sending endpoint
* @urb: the received packet
*
* This function is called from the context of an endpoint that received
* the packet and is used to let another endpoint object handle the payload.
*/
void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep,
struct snd_usb_endpoint *sender,
const struct urb *urb)
{
int shift;
unsigned int f;
unsigned long flags;
snd_BUG_ON(ep == sender);
/*
* In case the endpoint is operating in implicit feedback mode, prepare
* a new outbound URB that has the same layout as the received packet
* and add it to the list of pending urbs. queue_pending_output_urbs()
* will take care of them later.
*/
if (snd_usb_endpoint_implicit_feedback_sink(ep) &&
ep->use_count != 0) {
/* implicit feedback case */
int i, bytes = 0;
struct snd_urb_ctx *in_ctx;
struct snd_usb_packet_info *out_packet;
in_ctx = urb->context;
/* Count overall packet size */
for (i = 0; i < in_ctx->packets; i++)
if (urb->iso_frame_desc[i].status == 0)
bytes += urb->iso_frame_desc[i].actual_length;
/*
* skip empty packets. At least M-Audio's Fast Track Ultra stops
* streaming once it received a 0-byte OUT URB
*/
if (bytes == 0)
return;
spin_lock_irqsave(&ep->lock, flags);
out_packet = ep->next_packet + ep->next_packet_write_pos;
/*
* Iterate through the inbound packet and prepare the lengths
* for the output packet. The OUT packet we are about to send
* will have the same amount of payload bytes per stride as the
* IN packet we just received. Since the actual size is scaled
* by the stride, use the sender stride to calculate the length
* in case the number of channels differ between the implicitly
* fed-back endpoint and the synchronizing endpoint.
*/
out_packet->packets = in_ctx->packets;
for (i = 0; i < in_ctx->packets; i++) {
if (urb->iso_frame_desc[i].status == 0)
out_packet->packet_size[i] =
urb->iso_frame_desc[i].actual_length / sender->stride;
else
out_packet->packet_size[i] = 0;
}
ep->next_packet_write_pos++;
ep->next_packet_write_pos %= MAX_URBS;
spin_unlock_irqrestore(&ep->lock, flags);
queue_pending_output_urbs(ep);
return;
}
/*
* process after playback sync complete
*
* Full speed devices report feedback values in 10.14 format as samples
* per frame, high speed devices in 16.16 format as samples per
* microframe.
*
* Because the Audio Class 1 spec was written before USB 2.0, many high
* speed devices use a wrong interpretation, some others use an
* entirely different format.
*
* Therefore, we cannot predict what format any particular device uses
* and must detect it automatically.
*/
if (urb->iso_frame_desc[0].status != 0 ||
urb->iso_frame_desc[0].actual_length < 3)
return;
f = le32_to_cpup(urb->transfer_buffer);
if (urb->iso_frame_desc[0].actual_length == 3)
f &= 0x00ffffff;
else
f &= 0x0fffffff;
if (f == 0)
return;
if (unlikely(sender->udh01_fb_quirk)) {
/*
* The TEAC UD-H01 firmware sometimes changes the feedback value
* by +/- 0x1.0000.
*/
if (f < ep->freqn - 0x8000)
f += 0x10000;
else if (f > ep->freqn + 0x8000)
f -= 0x10000;
} else if (unlikely(ep->freqshift == INT_MIN)) {
/*
* The first time we see a feedback value, determine its format
* by shifting it left or right until it matches the nominal
* frequency value. This assumes that the feedback does not
* differ from the nominal value more than +50% or -25%.
*/
shift = 0;
while (f < ep->freqn - ep->freqn / 4) {
f <<= 1;
shift++;
}
while (f > ep->freqn + ep->freqn / 2) {
f >>= 1;
shift--;
}
ep->freqshift = shift;
} else if (ep->freqshift >= 0)
f <<= ep->freqshift;
else
f >>= -ep->freqshift;
if (likely(f >= ep->freqn - ep->freqn / 8 && f <= ep->freqmax)) {
/*
* If the frequency looks valid, set it.
* This value is referred to in prepare_playback_urb().
*/
spin_lock_irqsave(&ep->lock, flags);
ep->freqm = f;
spin_unlock_irqrestore(&ep->lock, flags);
} else {
/*
* Out of range; maybe the shift value is wrong.
* Reset it so that we autodetect again the next time.
*/
ep->freqshift = INT_MIN;
}
}