linux/sound/usb/pcm.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/init.h>
#include <linux/slab.h>
#include <linux/bitrev.h>
#include <linux/ratelimit.h>
#include <linux/usb.h>
#include <linux/usb/audio.h>
#include <linux/usb/audio-v2.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include "usbaudio.h"
#include "card.h"
#include "quirks.h"
#include "debug.h"
#include "endpoint.h"
#include "helper.h"
#include "pcm.h"
#include "clock.h"
#include "power.h"
#define SUBSTREAM_FLAG_DATA_EP_STARTED 0
#define SUBSTREAM_FLAG_SYNC_EP_STARTED 1
/* return the estimated delay based on USB frame counters */
snd_pcm_uframes_t snd_usb_pcm_delay(struct snd_usb_substream *subs,
unsigned int rate)
{
int current_frame_number;
int frame_diff;
int est_delay;
if (!subs->last_delay)
return 0; /* short path */
current_frame_number = usb_get_current_frame_number(subs->dev);
/*
* HCD implementations use different widths, use lower 8 bits.
* The delay will be managed up to 256ms, which is more than
* enough
*/
frame_diff = (current_frame_number - subs->last_frame_number) & 0xff;
/* Approximation based on number of samples per USB frame (ms),
some truncation for 44.1 but the estimate is good enough */
est_delay = frame_diff * rate / 1000;
if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
est_delay = subs->last_delay - est_delay;
else
est_delay = subs->last_delay + est_delay;
if (est_delay < 0)
est_delay = 0;
return est_delay;
}
/*
* return the current pcm pointer. just based on the hwptr_done value.
*/
static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_usb_substream *subs;
unsigned int hwptr_done;
subs = (struct snd_usb_substream *)substream->runtime->private_data;
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 17:09:00 +03:00
if (atomic_read(&subs->stream->chip->shutdown))
return SNDRV_PCM_POS_XRUN;
spin_lock(&subs->lock);
hwptr_done = subs->hwptr_done;
substream->runtime->delay = snd_usb_pcm_delay(subs,
substream->runtime->rate);
spin_unlock(&subs->lock);
return hwptr_done / (substream->runtime->frame_bits >> 3);
}
/*
* find a matching audio format
*/
static struct audioformat *find_format(struct snd_usb_substream *subs)
{
struct audioformat *fp;
struct audioformat *found = NULL;
int cur_attr = 0, attr;
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!(fp->formats & pcm_format_to_bits(subs->pcm_format)))
continue;
if (fp->channels != subs->channels)
continue;
if (subs->cur_rate < fp->rate_min ||
subs->cur_rate > fp->rate_max)
continue;
if (! (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)) {
unsigned int i;
for (i = 0; i < fp->nr_rates; i++)
if (fp->rate_table[i] == subs->cur_rate)
break;
if (i >= fp->nr_rates)
continue;
}
attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE;
if (! found) {
found = fp;
cur_attr = attr;
continue;
}
/* avoid async out and adaptive in if the other method
* supports the same format.
* this is a workaround for the case like
* M-audio audiophile USB.
*/
if (attr != cur_attr) {
if ((attr == USB_ENDPOINT_SYNC_ASYNC &&
subs->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
(attr == USB_ENDPOINT_SYNC_ADAPTIVE &&
subs->direction == SNDRV_PCM_STREAM_CAPTURE))
continue;
if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC &&
subs->direction == SNDRV_PCM_STREAM_PLAYBACK) ||
(cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE &&
subs->direction == SNDRV_PCM_STREAM_CAPTURE)) {
found = fp;
cur_attr = attr;
continue;
}
}
/* find the format with the largest max. packet size */
if (fp->maxpacksize > found->maxpacksize) {
found = fp;
cur_attr = attr;
}
}
return found;
}
static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
struct usb_host_interface *alts,
struct audioformat *fmt)
{
struct usb_device *dev = chip->dev;
unsigned int ep;
unsigned char data[1];
int err;
if (get_iface_desc(alts)->bNumEndpoints < 1)
return -EINVAL;
ep = get_endpoint(alts, 0)->bEndpointAddress;
data[0] = 1;
if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep,
data, sizeof(data))) < 0) {
usb_audio_err(chip, "%d:%d: cannot set enable PITCH\n",
iface, ep);
return err;
}
return 0;
}
static int init_pitch_v2(struct snd_usb_audio *chip, int iface,
struct usb_host_interface *alts,
struct audioformat *fmt)
{
struct usb_device *dev = chip->dev;
unsigned char data[1];
int err;
data[0] = 1;
if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
UAC2_EP_CS_PITCH << 8, 0,
data, sizeof(data))) < 0) {
usb_audio_err(chip, "%d:%d: cannot set enable PITCH (v2)\n",
iface, fmt->altsetting);
return err;
}
return 0;
}
/*
* initialize the pitch control and sample rate
*/
int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
struct usb_host_interface *alts,
struct audioformat *fmt)
{
/* if endpoint doesn't have pitch control, bail out */
if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL))
return 0;
switch (fmt->protocol) {
case UAC_VERSION_1:
default:
return init_pitch_v1(chip, iface, alts, fmt);
case UAC_VERSION_2:
return init_pitch_v2(chip, iface, alts, fmt);
}
}
static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep)
{
int err;
if (!subs->data_endpoint)
return -EINVAL;
if (!test_and_set_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags)) {
struct snd_usb_endpoint *ep = subs->data_endpoint;
dev_dbg(&subs->dev->dev, "Starting data EP @%p\n", ep);
ep->data_subs = subs;
err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags);
return err;
}
}
if (subs->sync_endpoint &&
!test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) {
struct snd_usb_endpoint *ep = subs->sync_endpoint;
if (subs->data_endpoint->iface != subs->sync_endpoint->iface ||
subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) {
err = usb_set_interface(subs->dev,
subs->sync_endpoint->iface,
subs->sync_endpoint->altsetting);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
dev_err(&subs->dev->dev,
"%d:%d: cannot set interface (%d)\n",
subs->sync_endpoint->iface,
subs->sync_endpoint->altsetting, err);
return -EIO;
}
}
dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep);
ep->sync_slave = subs->data_endpoint;
err = snd_usb_endpoint_start(ep, can_sleep);
if (err < 0) {
clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags);
return err;
}
}
return 0;
}
static void stop_endpoints(struct snd_usb_substream *subs, bool wait)
{
if (test_and_clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags))
snd_usb_endpoint_stop(subs->sync_endpoint);
if (test_and_clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags))
snd_usb_endpoint_stop(subs->data_endpoint);
if (wait) {
snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint);
snd_usb_endpoint_sync_pending_stop(subs->data_endpoint);
}
}
static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
unsigned int altsetting,
struct usb_host_interface **alts,
unsigned int *ep)
{
struct usb_interface *iface;
struct usb_interface_descriptor *altsd;
struct usb_endpoint_descriptor *epd;
iface = usb_ifnum_to_if(dev, ifnum);
if (!iface || iface->num_altsetting < altsetting + 1)
return -ENOENT;
*alts = &iface->altsetting[altsetting];
altsd = get_iface_desc(*alts);
if (altsd->bAlternateSetting != altsetting ||
altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC ||
(altsd->bInterfaceSubClass != 2 &&
altsd->bInterfaceProtocol != 2 ) ||
altsd->bNumEndpoints < 1)
return -ENOENT;
epd = get_endpoint(*alts, 0);
if (!usb_endpoint_is_isoc_in(epd) ||
(epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
USB_ENDPOINT_USAGE_IMPLICIT_FB)
return -ENOENT;
*ep = epd->bEndpointAddress;
return 0;
}
static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
struct usb_device *dev,
struct usb_interface_descriptor *altsd,
unsigned int attr)
{
struct usb_host_interface *alts;
struct usb_interface *iface;
unsigned int ep;
/* Implicit feedback sync EPs consumers are always playback EPs */
if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
return 0;
switch (subs->stream->chip->usb_id) {
case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
ep = 0x81;
iface = usb_ifnum_to_if(dev, 3);
if (!iface || iface->num_altsetting == 0)
return -EINVAL;
alts = &iface->altsetting[1];
goto add_sync_ep;
break;
case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
case USB_ID(0x0763, 0x2081):
ep = 0x81;
iface = usb_ifnum_to_if(dev, 2);
if (!iface || iface->num_altsetting == 0)
return -EINVAL;
alts = &iface->altsetting[1];
goto add_sync_ep;
}
if (attr == USB_ENDPOINT_SYNC_ASYNC &&
altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
altsd->bInterfaceProtocol == 2 &&
altsd->bNumEndpoints == 1 &&
USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
altsd->bAlternateSetting,
&alts, &ep) >= 0) {
goto add_sync_ep;
}
/* No quirk */
return 0;
add_sync_ep:
subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, ep, !subs->direction,
SND_USB_ENDPOINT_TYPE_DATA);
if (!subs->sync_endpoint)
return -EINVAL;
subs->data_endpoint->sync_master = subs->sync_endpoint;
return 0;
}
static int set_sync_endpoint(struct snd_usb_substream *subs,
struct audioformat *fmt,
struct usb_device *dev,
struct usb_host_interface *alts,
struct usb_interface_descriptor *altsd)
{
int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
unsigned int ep, attr;
bool implicit_fb;
int err;
/* we need a sync pipe in async OUT or adaptive IN mode */
/* check the number of EP, since some devices have broken
* descriptors which fool us. if it has only one EP,
* assume it as adaptive-out or sync-in.
*/
attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
ALSA: usb: fix corrupted pointers due to interface setting change When a transition occurs between alternate settings that do not use the same synchronization method, the substream pointers were not reset. This prevents audio from being played during the second transition. Identified and tested with M-Audio Transit device (0763:2006 Midiman M-Audio Transit) Details of the issue: First playback to adaptive endpoint: $ aplay -Dhw:1,0 ~/24_96.wav Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo [ 3169.297556] usb 1-2: setting usb interface 1:1 [ 3169.297568] usb 1-2: Creating new playback data endpoint #3 [ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0 [ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000 first playback to asynchronous endpoint: $ aplay -Dhw:1,0 ~/16_48.wav Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo [ 3204.520251] usb 1-2: setting usb interface 1:3 [ 3204.520264] usb 1-2: Creating new playback data endpoint #3 [ 3204.520272] usb 1-2: Creating new capture sync endpoint #83 [ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0 [ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0 [ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000 [ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000 second playback to adaptive endpoint: no audio and error on terminal: $ aplay -Dhw:1,0 ~/24_96.wav Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes, Rate 96000 Hz, Stereo aplay: pcm_write:1939: write error: Input/output error [ 3239.483589] usb 1-2: setting usb interface 1:1 [ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000 [ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0 [ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0 This last line shows that a sync endpoint is used when it shouldn't. The sync endpoint is no longer valid and the pointers are corrupted Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-15 01:19:42 +03:00
if ((is_playback && (attr != USB_ENDPOINT_SYNC_ASYNC)) ||
(!is_playback && (attr != USB_ENDPOINT_SYNC_ADAPTIVE))) {
/*
* In these modes the notion of sync_endpoint is irrelevant.
* Reset pointers to avoid using stale data from previously
* used settings, e.g. when configuration and endpoints were
* changed
*/
subs->sync_endpoint = NULL;
subs->data_endpoint->sync_master = NULL;
}
err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr);
if (err < 0)
return err;
if (altsd->bNumEndpoints < 2)
return 0;
if ((is_playback && (attr == USB_ENDPOINT_SYNC_SYNC ||
attr == USB_ENDPOINT_SYNC_ADAPTIVE)) ||
(!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE))
return 0;
/*
* In case of illegal SYNC_NONE for OUT endpoint, we keep going to see
* if we don't find a sync endpoint, as on M-Audio Transit. In case of
* error fall back to SYNC mode and don't create sync endpoint
*/
/* check sync-pipe endpoint */
/* ... and check descriptor size before accessing bSynchAddress
because there is a version of the SB Audigy 2 NX firmware lacking
the audio fields in the endpoint descriptors */
if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
(get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
get_endpoint(alts, 1)->bSynchAddress != 0)) {
dev_err(&dev->dev,
"%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
fmt->iface, fmt->altsetting,
get_endpoint(alts, 1)->bmAttributes,
get_endpoint(alts, 1)->bLength,
get_endpoint(alts, 1)->bSynchAddress);
if (is_playback && attr == USB_ENDPOINT_SYNC_NONE)
return 0;
return -EINVAL;
}
ep = get_endpoint(alts, 1)->bEndpointAddress;
if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
(!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
dev_err(&dev->dev,
"%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
fmt->iface, fmt->altsetting,
is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
if (is_playback && attr == USB_ENDPOINT_SYNC_NONE)
return 0;
return -EINVAL;
}
implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
== USB_ENDPOINT_USAGE_IMPLICIT_FB;
subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, ep, !subs->direction,
implicit_fb ?
SND_USB_ENDPOINT_TYPE_DATA :
SND_USB_ENDPOINT_TYPE_SYNC);
if (!subs->sync_endpoint) {
if (is_playback && attr == USB_ENDPOINT_SYNC_NONE)
return 0;
return -EINVAL;
}
subs->data_endpoint->sync_master = subs->sync_endpoint;
return 0;
}
/*
* find a matching format and set up the interface
*/
static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
{
struct usb_device *dev = subs->dev;
struct usb_host_interface *alts;
struct usb_interface_descriptor *altsd;
struct usb_interface *iface;
int err;
iface = usb_ifnum_to_if(dev, fmt->iface);
if (WARN_ON(!iface))
return -EINVAL;
alts = &iface->altsetting[fmt->altset_idx];
altsd = get_iface_desc(alts);
if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
return -EINVAL;
if (fmt == subs->cur_audiofmt)
return 0;
/* close the old interface */
if (subs->interface >= 0 && subs->interface != fmt->iface) {
err = usb_set_interface(subs->dev, subs->interface, 0);
if (err < 0) {
dev_err(&dev->dev,
"%d:%d: return to setting 0 failed (%d)\n",
fmt->iface, fmt->altsetting, err);
return -EIO;
}
subs->interface = -1;
subs->altset_idx = 0;
}
/* set interface */
if (subs->interface != fmt->iface ||
subs->altset_idx != fmt->altset_idx) {
err = snd_usb_select_mode_quirk(subs, fmt);
if (err < 0)
return -EIO;
err = usb_set_interface(dev, fmt->iface, fmt->altsetting);
if (err < 0) {
dev_err(&dev->dev,
"%d:%d: usb_set_interface failed (%d)\n",
fmt->iface, fmt->altsetting, err);
return -EIO;
}
dev_dbg(&dev->dev, "setting usb interface %d:%d\n",
fmt->iface, fmt->altsetting);
subs->interface = fmt->iface;
subs->altset_idx = fmt->altset_idx;
snd_usb_set_interface_quirk(dev);
}
subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
alts, fmt->endpoint, subs->direction,
SND_USB_ENDPOINT_TYPE_DATA);
if (!subs->data_endpoint)
return -EINVAL;
err = set_sync_endpoint(subs, fmt, dev, alts, altsd);
if (err < 0)
return err;
err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt);
if (err < 0)
return err;
subs->cur_audiofmt = fmt;
snd_usb_set_format_quirk(subs, fmt);
return 0;
}
/*
* Return the score of matching two audioformats.
* Veto the audioformat if:
* - It has no channels for some reason.
* - Requested PCM format is not supported.
* - Requested sample rate is not supported.
*/
static int match_endpoint_audioformats(struct snd_usb_substream *subs,
struct audioformat *fp,
struct audioformat *match, int rate,
snd_pcm_format_t pcm_format)
{
int i;
int score = 0;
if (fp->channels < 1) {
dev_dbg(&subs->dev->dev,
"%s: (fmt @%p) no channels\n", __func__, fp);
return 0;
}
if (!(fp->formats & pcm_format_to_bits(pcm_format))) {
dev_dbg(&subs->dev->dev,
"%s: (fmt @%p) no match for format %d\n", __func__,
fp, pcm_format);
return 0;
}
for (i = 0; i < fp->nr_rates; i++) {
if (fp->rate_table[i] == rate) {
score++;
break;
}
}
if (!score) {
dev_dbg(&subs->dev->dev,
"%s: (fmt @%p) no match for rate %d\n", __func__,
fp, rate);
return 0;
}
if (fp->channels == match->channels)
score++;
dev_dbg(&subs->dev->dev,
"%s: (fmt @%p) score %d\n", __func__, fp, score);
return score;
}
/*
* Configure the sync ep using the rate and pcm format of the data ep.
*/
static int configure_sync_endpoint(struct snd_usb_substream *subs)
{
int ret;
struct audioformat *fp;
struct audioformat *sync_fp = NULL;
int cur_score = 0;
int sync_period_bytes = subs->period_bytes;
struct snd_usb_substream *sync_subs =
&subs->stream->substream[subs->direction ^ 1];
if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA ||
!subs->stream)
return snd_usb_endpoint_set_params(subs->sync_endpoint,
subs->pcm_format,
subs->channels,
subs->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 23:51:58 +04:00
0, 0,
subs->cur_rate,
subs->cur_audiofmt,
NULL);
/* Try to find the best matching audioformat. */
list_for_each_entry(fp, &sync_subs->fmt_list, list) {
int score = match_endpoint_audioformats(subs,
fp, subs->cur_audiofmt,
subs->cur_rate, subs->pcm_format);
if (score > cur_score) {
sync_fp = fp;
cur_score = score;
}
}
if (unlikely(sync_fp == NULL)) {
dev_err(&subs->dev->dev,
"%s: no valid audioformat for sync ep %x found\n",
__func__, sync_subs->ep_num);
return -EINVAL;
}
/*
* Recalculate the period bytes if channel number differ between
* data and sync ep audioformat.
*/
if (sync_fp->channels != subs->channels) {
sync_period_bytes = (subs->period_bytes / subs->channels) *
sync_fp->channels;
dev_dbg(&subs->dev->dev,
"%s: adjusted sync ep period bytes (%d -> %d)\n",
__func__, subs->period_bytes, sync_period_bytes);
}
ret = snd_usb_endpoint_set_params(subs->sync_endpoint,
subs->pcm_format,
sync_fp->channels,
sync_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 23:51:58 +04:00
0, 0,
subs->cur_rate,
sync_fp,
NULL);
return ret;
}
/*
* configure endpoint params
*
* called during initial setup and upon resume
*/
static int configure_endpoint(struct snd_usb_substream *subs)
{
int ret;
/* format changed */
stop_endpoints(subs, true);
ret = snd_usb_endpoint_set_params(subs->data_endpoint,
subs->pcm_format,
subs->channels,
subs->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 23:51:58 +04:00
subs->period_frames,
subs->buffer_periods,
subs->cur_rate,
subs->cur_audiofmt,
subs->sync_endpoint);
if (ret < 0)
return ret;
if (subs->sync_endpoint)
ret = configure_sync_endpoint(subs);
return ret;
}
/*
* hw_params callback
*
* allocate a buffer and set the given audio format.
*
* so far we use a physically linear buffer although packetize transfer
* doesn't need a continuous area.
* if sg buffer is supported on the later version of alsa, we'll follow
* that.
*/
static int snd_usb_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct snd_usb_substream *subs = substream->runtime->private_data;
struct audioformat *fmt;
int ret;
ret = snd_pcm_lib_alloc_vmalloc_buffer(substream,
params_buffer_bytes(hw_params));
if (ret < 0)
return ret;
subs->pcm_format = params_format(hw_params);
subs->period_bytes = params_period_bytes(hw_params);
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 23:51:58 +04:00
subs->period_frames = params_period_size(hw_params);
subs->buffer_periods = params_periods(hw_params);
subs->channels = params_channels(hw_params);
subs->cur_rate = params_rate(hw_params);
fmt = find_format(subs);
if (!fmt) {
dev_dbg(&subs->dev->dev,
"cannot set format: format = %#x, rate = %d, channels = %d\n",
subs->pcm_format, subs->cur_rate, subs->channels);
return -EINVAL;
}
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 17:09:00 +03:00
ret = snd_usb_lock_shutdown(subs->stream->chip);
if (ret < 0)
return ret;
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 17:09:00 +03:00
ret = set_format(subs, fmt);
snd_usb_unlock_shutdown(subs->stream->chip);
if (ret < 0)
return ret;
subs->interface = fmt->iface;
subs->altset_idx = fmt->altset_idx;
subs->need_setup_ep = true;
return 0;
}
/*
* hw_free callback
*
* reset the audio format and release the buffer
*/
static int snd_usb_hw_free(struct snd_pcm_substream *substream)
{
struct snd_usb_substream *subs = substream->runtime->private_data;
subs->cur_audiofmt = NULL;
subs->cur_rate = 0;
subs->period_bytes = 0;
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 17:09:00 +03:00
if (!snd_usb_lock_shutdown(subs->stream->chip)) {
stop_endpoints(subs, true);
snd_usb_endpoint_deactivate(subs->sync_endpoint);
snd_usb_endpoint_deactivate(subs->data_endpoint);
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 17:09:00 +03:00
snd_usb_unlock_shutdown(subs->stream->chip);
}
return snd_pcm_lib_free_vmalloc_buffer(substream);
}
/*
* prepare callback
*
* only a few subtle things...
*/
static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_usb_substream *subs = runtime->private_data;
struct usb_host_interface *alts;
struct usb_interface *iface;
int ret;
if (! subs->cur_audiofmt) {
dev_err(&subs->dev->dev, "no format is specified!\n");
return -ENXIO;
}
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 17:09:00 +03:00
ret = snd_usb_lock_shutdown(subs->stream->chip);
if (ret < 0)
return ret;
if (snd_BUG_ON(!subs->data_endpoint)) {
ret = -EIO;
goto unlock;
}
snd_usb_endpoint_sync_pending_stop(subs->sync_endpoint);
snd_usb_endpoint_sync_pending_stop(subs->data_endpoint);
ret = set_format(subs, subs->cur_audiofmt);
if (ret < 0)
goto unlock;
if (subs->need_setup_ep) {
iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface);
alts = &iface->altsetting[subs->cur_audiofmt->altset_idx];
ret = snd_usb_init_sample_rate(subs->stream->chip,
subs->cur_audiofmt->iface,
alts,
subs->cur_audiofmt,
subs->cur_rate);
if (ret < 0)
goto unlock;
ret = configure_endpoint(subs);
if (ret < 0)
goto unlock;
subs->need_setup_ep = false;
}
/* some unit conversions in runtime */
subs->data_endpoint->maxframesize =
bytes_to_frames(runtime, subs->data_endpoint->maxpacksize);
subs->data_endpoint->curframesize =
bytes_to_frames(runtime, subs->data_endpoint->curpacksize);
/* reset the pointer */
subs->hwptr_done = 0;
subs->transfer_done = 0;
subs->last_delay = 0;
subs->last_frame_number = 0;
runtime->delay = 0;
/* for playback, submit the URBs now; otherwise, the first hwptr_done
* updates for all URBs would happen at the same time when starting */
if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK)
ret = start_endpoints(subs, true);
unlock:
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 17:09:00 +03:00
snd_usb_unlock_shutdown(subs->stream->chip);
return ret;
}
static struct snd_pcm_hardware snd_usb_hardware =
{
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_PAUSE,
.buffer_bytes_max = 1024 * 1024,
.period_bytes_min = 64,
.period_bytes_max = 512 * 1024,
.periods_min = 2,
.periods_max = 1024,
};
static int hw_check_valid_format(struct snd_usb_substream *subs,
struct snd_pcm_hw_params *params,
struct audioformat *fp)
{
struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME);
struct snd_mask check_fmts;
unsigned int ptime;
/* check the format */
snd_mask_none(&check_fmts);
check_fmts.bits[0] = (u32)fp->formats;
check_fmts.bits[1] = (u32)(fp->formats >> 32);
snd_mask_intersect(&check_fmts, fmts);
if (snd_mask_empty(&check_fmts)) {
hwc_debug(" > check: no supported format %d\n", fp->format);
return 0;
}
/* check the channels */
if (fp->channels < ct->min || fp->channels > ct->max) {
hwc_debug(" > check: no valid channels %d (%d/%d)\n", fp->channels, ct->min, ct->max);
return 0;
}
/* check the rate is within the range */
if (fp->rate_min > it->max || (fp->rate_min == it->max && it->openmax)) {
hwc_debug(" > check: rate_min %d > max %d\n", fp->rate_min, it->max);
return 0;
}
if (fp->rate_max < it->min || (fp->rate_max == it->min && it->openmin)) {
hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min);
return 0;
}
/* check whether the period time is >= the data packet interval */
if (subs->speed != USB_SPEED_FULL) {
ptime = 125 * (1 << fp->datainterval);
if (ptime > pt->max || (ptime == pt->max && pt->openmax)) {
hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max);
return 0;
}
}
return 1;
}
static int hw_rule_rate(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
struct audioformat *fp;
struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
unsigned int rmin, rmax;
int changed;
hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max);
changed = 0;
rmin = rmax = 0;
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!hw_check_valid_format(subs, params, fp))
continue;
if (changed++) {
if (rmin > fp->rate_min)
rmin = fp->rate_min;
if (rmax < fp->rate_max)
rmax = fp->rate_max;
} else {
rmin = fp->rate_min;
rmax = fp->rate_max;
}
}
if (!changed) {
hwc_debug(" --> get empty\n");
it->empty = 1;
return -EINVAL;
}
changed = 0;
if (it->min < rmin) {
it->min = rmin;
it->openmin = 0;
changed = 1;
}
if (it->max > rmax) {
it->max = rmax;
it->openmax = 0;
changed = 1;
}
if (snd_interval_checkempty(it)) {
it->empty = 1;
return -EINVAL;
}
hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed);
return changed;
}
static int hw_rule_channels(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
struct audioformat *fp;
struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
unsigned int rmin, rmax;
int changed;
hwc_debug("hw_rule_channels: (%d,%d)\n", it->min, it->max);
changed = 0;
rmin = rmax = 0;
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!hw_check_valid_format(subs, params, fp))
continue;
if (changed++) {
if (rmin > fp->channels)
rmin = fp->channels;
if (rmax < fp->channels)
rmax = fp->channels;
} else {
rmin = fp->channels;
rmax = fp->channels;
}
}
if (!changed) {
hwc_debug(" --> get empty\n");
it->empty = 1;
return -EINVAL;
}
changed = 0;
if (it->min < rmin) {
it->min = rmin;
it->openmin = 0;
changed = 1;
}
if (it->max > rmax) {
it->max = rmax;
it->openmax = 0;
changed = 1;
}
if (snd_interval_checkempty(it)) {
it->empty = 1;
return -EINVAL;
}
hwc_debug(" --> (%d, %d) (changed = %d)\n", it->min, it->max, changed);
return changed;
}
static int hw_rule_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
struct audioformat *fp;
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
u64 fbits;
u32 oldbits[2];
int changed;
hwc_debug("hw_rule_format: %x:%x\n", fmt->bits[0], fmt->bits[1]);
fbits = 0;
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!hw_check_valid_format(subs, params, fp))
continue;
fbits |= fp->formats;
}
oldbits[0] = fmt->bits[0];
oldbits[1] = fmt->bits[1];
fmt->bits[0] &= (u32)fbits;
fmt->bits[1] &= (u32)(fbits >> 32);
if (!fmt->bits[0] && !fmt->bits[1]) {
hwc_debug(" --> get empty\n");
return -EINVAL;
}
changed = (oldbits[0] != fmt->bits[0] || oldbits[1] != fmt->bits[1]);
hwc_debug(" --> %x:%x (changed = %d)\n", fmt->bits[0], fmt->bits[1], changed);
return changed;
}
static int hw_rule_period_time(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_usb_substream *subs = rule->private;
struct audioformat *fp;
struct snd_interval *it;
unsigned char min_datainterval;
unsigned int pmin;
int changed;
it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME);
hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max);
min_datainterval = 0xff;
list_for_each_entry(fp, &subs->fmt_list, list) {
if (!hw_check_valid_format(subs, params, fp))
continue;
min_datainterval = min(min_datainterval, fp->datainterval);
}
if (min_datainterval == 0xff) {
hwc_debug(" --> get empty\n");
it->empty = 1;
return -EINVAL;
}
pmin = 125 * (1 << min_datainterval);
changed = 0;
if (it->min < pmin) {
it->min = pmin;
it->openmin = 0;
changed = 1;
}
if (snd_interval_checkempty(it)) {
it->empty = 1;
return -EINVAL;
}
hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed);
return changed;
}
/*
* If the device supports unusual bit rates, does the request meet these?
*/
static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime,
struct snd_usb_substream *subs)
{
struct audioformat *fp;
int *rate_list;
int count = 0, needs_knot = 0;
int err;
kfree(subs->rate_list.list);
subs->rate_list.list = NULL;
list_for_each_entry(fp, &subs->fmt_list, list) {
if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS)
return 0;
count += fp->nr_rates;
if (fp->rates & SNDRV_PCM_RATE_KNOT)
needs_knot = 1;
}
if (!needs_knot)
return 0;
subs->rate_list.list = rate_list =
kmalloc(sizeof(int) * count, GFP_KERNEL);
if (!subs->rate_list.list)
return -ENOMEM;
subs->rate_list.count = count;
subs->rate_list.mask = 0;
count = 0;
list_for_each_entry(fp, &subs->fmt_list, list) {
int i;
for (i = 0; i < fp->nr_rates; i++)
rate_list[count++] = fp->rate_table[i];
}
err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
&subs->rate_list);
if (err < 0)
return err;
return 0;
}
/*
* set up the runtime hardware information.
*/
static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs)
{
struct audioformat *fp;
unsigned int pt, ptmin;
int param_period_time_if_needed;
int err;
runtime->hw.formats = subs->formats;
runtime->hw.rate_min = 0x7fffffff;
runtime->hw.rate_max = 0;
runtime->hw.channels_min = 256;
runtime->hw.channels_max = 0;
runtime->hw.rates = 0;
ptmin = UINT_MAX;
/* check min/max rates and channels */
list_for_each_entry(fp, &subs->fmt_list, list) {
runtime->hw.rates |= fp->rates;
if (runtime->hw.rate_min > fp->rate_min)
runtime->hw.rate_min = fp->rate_min;
if (runtime->hw.rate_max < fp->rate_max)
runtime->hw.rate_max = fp->rate_max;
if (runtime->hw.channels_min > fp->channels)
runtime->hw.channels_min = fp->channels;
if (runtime->hw.channels_max < fp->channels)
runtime->hw.channels_max = fp->channels;
if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) {
/* FIXME: there might be more than one audio formats... */
runtime->hw.period_bytes_min = runtime->hw.period_bytes_max =
fp->frame_size;
}
pt = 125 * (1 << fp->datainterval);
ptmin = min(ptmin, pt);
}
err = snd_usb_autoresume(subs->stream->chip);
if (err < 0)
return err;
param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME;
if (subs->speed == USB_SPEED_FULL)
/* full speed devices have fixed data packet interval */
ptmin = 1000;
if (ptmin == 1000)
/* if period time doesn't go below 1 ms, no rules needed */
param_period_time_if_needed = -1;
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME,
ptmin, UINT_MAX);
if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
hw_rule_rate, subs,
SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_HW_PARAM_CHANNELS,
param_period_time_if_needed,
-1)) < 0)
goto rep_err;
if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_channels, subs,
SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_HW_PARAM_RATE,
param_period_time_if_needed,
-1)) < 0)
goto rep_err;
if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_format, subs,
SNDRV_PCM_HW_PARAM_RATE,
SNDRV_PCM_HW_PARAM_CHANNELS,
param_period_time_if_needed,
-1)) < 0)
goto rep_err;
if (param_period_time_if_needed >= 0) {
err = snd_pcm_hw_rule_add(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_TIME,
hw_rule_period_time, subs,
SNDRV_PCM_HW_PARAM_FORMAT,
SNDRV_PCM_HW_PARAM_CHANNELS,
SNDRV_PCM_HW_PARAM_RATE,
-1);
if (err < 0)
goto rep_err;
}
if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0)
goto rep_err;
return 0;
rep_err:
snd_usb_autosuspend(subs->stream->chip);
return err;
}
static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction)
{
struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_usb_substream *subs = &as->substream[direction];
subs->interface = -1;
subs->altset_idx = 0;
runtime->hw = snd_usb_hardware;
runtime->private_data = subs;
subs->pcm_substream = substream;
/* runtime PM is also done there */
/* initialize DSD/DOP context */
subs->dsd_dop.byte_idx = 0;
subs->dsd_dop.channel = 0;
subs->dsd_dop.marker = 1;
return setup_hw_info(runtime, subs);
}
static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction)
{
struct snd_usb_stream *as = snd_pcm_substream_chip(substream);
struct snd_usb_substream *subs = &as->substream[direction];
stop_endpoints(subs, true);
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 17:09:00 +03:00
if (subs->interface >= 0 &&
!snd_usb_lock_shutdown(subs->stream->chip)) {
usb_set_interface(subs->dev, subs->interface, 0);
subs->interface = -1;
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 17:09:00 +03:00
snd_usb_unlock_shutdown(subs->stream->chip);
}
subs->pcm_substream = NULL;
snd_usb_autosuspend(subs->stream->chip);
return 0;
}
/* Since a URB can handle only a single linear buffer, we must use double
* buffering when the data to be transferred overflows the buffer boundary.
* To avoid inconsistencies when updating hwptr_done, we use double buffering
* for all URBs.
*/
static void retire_capture_urb(struct snd_usb_substream *subs,
struct urb *urb)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
unsigned int stride, frames, bytes, oldptr;
int i, period_elapsed = 0;
unsigned long flags;
unsigned char *cp;
int current_frame_number;
/* read frame number here, update pointer in critical section */
current_frame_number = usb_get_current_frame_number(subs->dev);
stride = runtime->frame_bits >> 3;
for (i = 0; i < urb->number_of_packets; i++) {
cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset + subs->pkt_offset_adj;
if (urb->iso_frame_desc[i].status && printk_ratelimit()) {
dev_dbg(&subs->dev->dev, "frame %d active: %d\n",
i, urb->iso_frame_desc[i].status);
// continue;
}
bytes = urb->iso_frame_desc[i].actual_length;
frames = bytes / stride;
if (!subs->txfr_quirk)
bytes = frames * stride;
if (bytes % (runtime->sample_bits >> 3) != 0) {
int oldbytes = bytes;
bytes = frames * stride;
dev_warn(&subs->dev->dev,
"Corrected urb data len. %d->%d\n",
oldbytes, bytes);
}
/* update the current pointer */
spin_lock_irqsave(&subs->lock, flags);
oldptr = subs->hwptr_done;
subs->hwptr_done += bytes;
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
frames = (bytes + (oldptr % stride)) / stride;
subs->transfer_done += frames;
if (subs->transfer_done >= runtime->period_size) {
subs->transfer_done -= runtime->period_size;
period_elapsed = 1;
}
/* capture delay is by construction limited to one URB,
* reset delays here
*/
runtime->delay = subs->last_delay = 0;
/* realign last_frame_number */
subs->last_frame_number = current_frame_number;
subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
spin_unlock_irqrestore(&subs->lock, flags);
/* copy a data chunk */
if (oldptr + bytes > runtime->buffer_size * stride) {
unsigned int bytes1 =
runtime->buffer_size * stride - oldptr;
memcpy(runtime->dma_area + oldptr, cp, bytes1);
memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1);
} else {
memcpy(runtime->dma_area + oldptr, cp, bytes);
}
}
if (period_elapsed)
snd_pcm_period_elapsed(subs->pcm_substream);
}
static inline void fill_playback_urb_dsd_dop(struct snd_usb_substream *subs,
struct urb *urb, unsigned int bytes)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
unsigned int stride = runtime->frame_bits >> 3;
unsigned int dst_idx = 0;
unsigned int src_idx = subs->hwptr_done;
unsigned int wrap = runtime->buffer_size * stride;
u8 *dst = urb->transfer_buffer;
u8 *src = runtime->dma_area;
u8 marker[] = { 0x05, 0xfa };
/*
* The DSP DOP format defines a way to transport DSD samples over
* normal PCM data endpoints. It requires stuffing of marker bytes
* (0x05 and 0xfa, alternating per sample frame), and then expects
* 2 additional bytes of actual payload. The whole frame is stored
* LSB.
*
* Hence, for a stereo transport, the buffer layout looks like this,
* where L refers to left channel samples and R to right.
*
* L1 L2 0x05 R1 R2 0x05 L3 L4 0xfa R3 R4 0xfa
* L5 L6 0x05 R5 R6 0x05 L7 L8 0xfa R7 R8 0xfa
* .....
*
*/
while (bytes--) {
if (++subs->dsd_dop.byte_idx == 3) {
/* frame boundary? */
dst[dst_idx++] = marker[subs->dsd_dop.marker];
src_idx += 2;
subs->dsd_dop.byte_idx = 0;
if (++subs->dsd_dop.channel % runtime->channels == 0) {
/* alternate the marker */
subs->dsd_dop.marker++;
subs->dsd_dop.marker %= ARRAY_SIZE(marker);
subs->dsd_dop.channel = 0;
}
} else {
/* stuff the DSD payload */
int idx = (src_idx + subs->dsd_dop.byte_idx - 1) % wrap;
if (subs->cur_audiofmt->dsd_bitrev)
dst[dst_idx++] = bitrev8(src[idx]);
else
dst[dst_idx++] = src[idx];
subs->hwptr_done++;
}
}
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
}
static void copy_to_urb(struct snd_usb_substream *subs, struct urb *urb,
int offset, int stride, unsigned int bytes)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
if (subs->hwptr_done + bytes > runtime->buffer_size * stride) {
/* err, the transferred area goes over buffer boundary. */
unsigned int bytes1 =
runtime->buffer_size * stride - subs->hwptr_done;
memcpy(urb->transfer_buffer + offset,
runtime->dma_area + subs->hwptr_done, bytes1);
memcpy(urb->transfer_buffer + offset + bytes1,
runtime->dma_area, bytes - bytes1);
} else {
memcpy(urb->transfer_buffer + offset,
runtime->dma_area + subs->hwptr_done, bytes);
}
subs->hwptr_done += bytes;
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
}
ALSA: USB-audio: Add quirk for Zoom R16/24 playback The Zoom R16/24 have a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. (Curiously, capture data does not contain this and requires no quirk.) The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). In order to make the quirk as unintrusive as possible, for pcm.c:prepare_playback_urb(), the isochronous packet descriptors are initially set up in the same way no matter if the quirk is enabled or not. Once it is time to actually copy the data into the outgoing packet buffer (together with the added length descriptors) the isochronous descriptors are adjusted in order take the increased payload length into account. For endpoint.c:prepare_silent_urb() it makes more sense to modify the actual function, partly because the function is less complex to start with and partly because it is not as time-critical as prepare_playback_urb() (whose bulk is run with interrupts disabled), so the (minute) additional time spent in the non-quirk case is motivated by the simplicity of having a single function for all cases. The quirk is controlled by the new tx_length_quirk member in struct snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c and endpoint.c from quirks.c in a similar manner to the txfr_quirk member in the same structs. In contrast to txfr_quirk however, the quirk is enabled directly in quirks.c:create_standard_audio_quirk() by checking the USB ID in that function. Another option would be to introduce a new QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk very plain to see in the quirk table, but it was felt that the additional code needed to implement it this way would just make the implementation more complex with no real gain. Tested with a Zoom R16, both by doing capture and playback separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together using Ardour, as well as Audacity and Qtractor together with jackd. The R24 is reportedly compatible with the R16 when used as an audio interface. Both devices share the same USB ID and have the same number of inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the patch. Regression tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Tested-by: Panu Matilainen <pmatilai@laiskiainen.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19 09:52:53 +03:00
static unsigned int copy_to_urb_quirk(struct snd_usb_substream *subs,
struct urb *urb, int stride,
unsigned int bytes)
{
__le32 packet_length;
int i;
/* Put __le32 length descriptor at start of each packet. */
for (i = 0; i < urb->number_of_packets; i++) {
unsigned int length = urb->iso_frame_desc[i].length;
unsigned int offset = urb->iso_frame_desc[i].offset;
packet_length = cpu_to_le32(length);
offset += i * sizeof(packet_length);
urb->iso_frame_desc[i].offset = offset;
urb->iso_frame_desc[i].length += sizeof(packet_length);
memcpy(urb->transfer_buffer + offset,
&packet_length, sizeof(packet_length));
copy_to_urb(subs, urb, offset + sizeof(packet_length),
stride, length);
}
/* Adjust transfer size accordingly. */
bytes += urb->number_of_packets * sizeof(packet_length);
return bytes;
}
static void prepare_playback_urb(struct snd_usb_substream *subs,
struct urb *urb)
{
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
struct snd_usb_endpoint *ep = subs->data_endpoint;
struct snd_urb_ctx *ctx = urb->context;
unsigned int counts, frames, bytes;
int i, stride, period_elapsed = 0;
unsigned long flags;
stride = runtime->frame_bits >> 3;
frames = 0;
urb->number_of_packets = 0;
spin_lock_irqsave(&subs->lock, flags);
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 23:51:58 +04:00
subs->frame_limit += ep->max_urb_frames;
for (i = 0; i < ctx->packets; i++) {
if (ctx->packet_size[i])
counts = ctx->packet_size[i];
else
counts = snd_usb_endpoint_next_packet_size(ep);
/* set up descriptor */
urb->iso_frame_desc[i].offset = frames * ep->stride;
urb->iso_frame_desc[i].length = counts * ep->stride;
frames += counts;
urb->number_of_packets++;
subs->transfer_done += counts;
if (subs->transfer_done >= runtime->period_size) {
subs->transfer_done -= runtime->period_size;
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 23:51:58 +04:00
subs->frame_limit = 0;
period_elapsed = 1;
if (subs->fmt_type == UAC_FORMAT_TYPE_II) {
if (subs->transfer_done > 0) {
/* FIXME: fill-max mode is not
* supported yet */
frames -= subs->transfer_done;
counts -= subs->transfer_done;
urb->iso_frame_desc[i].length =
counts * ep->stride;
subs->transfer_done = 0;
}
i++;
if (i < ctx->packets) {
/* add a transfer delimiter */
urb->iso_frame_desc[i].offset =
frames * ep->stride;
urb->iso_frame_desc[i].length = 0;
urb->number_of_packets++;
}
break;
}
}
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 23:51:58 +04:00
/* finish at the period boundary or after enough frames */
if ((period_elapsed ||
subs->transfer_done >= subs->frame_limit) &&
!snd_usb_endpoint_implicit_feedback_sink(ep))
break;
}
bytes = frames * ep->stride;
if (unlikely(subs->pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE &&
subs->cur_audiofmt->dsd_dop)) {
fill_playback_urb_dsd_dop(subs, urb, bytes);
} else if (unlikely(subs->pcm_format == SNDRV_PCM_FORMAT_DSD_U8 &&
subs->cur_audiofmt->dsd_bitrev)) {
/* bit-reverse the bytes */
u8 *buf = urb->transfer_buffer;
for (i = 0; i < bytes; i++) {
int idx = (subs->hwptr_done + i)
% (runtime->buffer_size * stride);
buf[i] = bitrev8(runtime->dma_area[idx]);
}
subs->hwptr_done += bytes;
if (subs->hwptr_done >= runtime->buffer_size * stride)
subs->hwptr_done -= runtime->buffer_size * stride;
} else {
/* usual PCM */
ALSA: USB-audio: Add quirk for Zoom R16/24 playback The Zoom R16/24 have a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. (Curiously, capture data does not contain this and requires no quirk.) The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). In order to make the quirk as unintrusive as possible, for pcm.c:prepare_playback_urb(), the isochronous packet descriptors are initially set up in the same way no matter if the quirk is enabled or not. Once it is time to actually copy the data into the outgoing packet buffer (together with the added length descriptors) the isochronous descriptors are adjusted in order take the increased payload length into account. For endpoint.c:prepare_silent_urb() it makes more sense to modify the actual function, partly because the function is less complex to start with and partly because it is not as time-critical as prepare_playback_urb() (whose bulk is run with interrupts disabled), so the (minute) additional time spent in the non-quirk case is motivated by the simplicity of having a single function for all cases. The quirk is controlled by the new tx_length_quirk member in struct snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c and endpoint.c from quirks.c in a similar manner to the txfr_quirk member in the same structs. In contrast to txfr_quirk however, the quirk is enabled directly in quirks.c:create_standard_audio_quirk() by checking the USB ID in that function. Another option would be to introduce a new QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk very plain to see in the quirk table, but it was felt that the additional code needed to implement it this way would just make the implementation more complex with no real gain. Tested with a Zoom R16, both by doing capture and playback separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together using Ardour, as well as Audacity and Qtractor together with jackd. The R24 is reportedly compatible with the R16 when used as an audio interface. Both devices share the same USB ID and have the same number of inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the patch. Regression tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes. Signed-off-by: Ricard Wanderlof <ricardw@axis.com> Tested-by: Panu Matilainen <pmatilai@laiskiainen.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19 09:52:53 +03:00
if (!subs->tx_length_quirk)
copy_to_urb(subs, urb, 0, stride, bytes);
else
bytes = copy_to_urb_quirk(subs, urb, stride, bytes);
/* bytes is now amount of outgoing data */
}
/* update delay with exact number of samples queued */
runtime->delay = subs->last_delay;
runtime->delay += frames;
subs->last_delay = runtime->delay;
/* realign last_frame_number */
subs->last_frame_number = usb_get_current_frame_number(subs->dev);
subs->last_frame_number &= 0xFF; /* keep 8 LSBs */
if (subs->trigger_tstamp_pending_update) {
/* this is the first actual URB submitted,
* update trigger timestamp to reflect actual start time
*/
snd_pcm_gettime(runtime, &runtime->trigger_tstamp);
subs->trigger_tstamp_pending_update = false;
}
spin_unlock_irqrestore(&subs->lock, flags);
urb->transfer_buffer_length = bytes;
if (period_elapsed)
snd_pcm_period_elapsed(subs->pcm_substream);
}
/*
* process after playback data complete
* - decrease the delay count again
*/
static void retire_playback_urb(struct snd_usb_substream *subs,
struct urb *urb)
{
unsigned long flags;
struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime;
struct snd_usb_endpoint *ep = subs->data_endpoint;
int processed = urb->transfer_buffer_length / ep->stride;
int est_delay;
/* ignore the delay accounting when procssed=0 is given, i.e.
* silent payloads are procssed before handling the actual data
*/
if (!processed)
return;
spin_lock_irqsave(&subs->lock, flags);
if (!subs->last_delay)
goto out; /* short path */
est_delay = snd_usb_pcm_delay(subs, runtime->rate);
/* update delay with exact number of samples played */
if (processed > subs->last_delay)
subs->last_delay = 0;
else
subs->last_delay -= processed;
runtime->delay = subs->last_delay;
/*
* Report when delay estimate is off by more than 2ms.
* The error should be lower than 2ms since the estimate relies
* on two reads of a counter updated every ms.
*/
if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2)
dev_dbg_ratelimited(&subs->dev->dev,
"delay: estimated %d, actual %d\n",
est_delay, subs->last_delay);
if (!subs->running) {
/* update last_frame_number for delay counting here since
* prepare_playback_urb won't be called during pause
*/
subs->last_frame_number =
usb_get_current_frame_number(subs->dev) & 0xff;
}
out:
spin_unlock_irqrestore(&subs->lock, flags);
}
static int snd_usb_playback_open(struct snd_pcm_substream *substream)
{
return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK);
}
static int snd_usb_playback_close(struct snd_pcm_substream *substream)
{
return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK);
}
static int snd_usb_capture_open(struct snd_pcm_substream *substream)
{
return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE);
}
static int snd_usb_capture_close(struct snd_pcm_substream *substream)
{
return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE);
}
static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream,
int cmd)
{
struct snd_usb_substream *subs = substream->runtime->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
subs->trigger_tstamp_pending_update = true;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
subs->data_endpoint->prepare_data_urb = prepare_playback_urb;
subs->data_endpoint->retire_data_urb = retire_playback_urb;
subs->running = 1;
return 0;
case SNDRV_PCM_TRIGGER_STOP:
stop_endpoints(subs, false);
subs->running = 0;
return 0;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
subs->data_endpoint->prepare_data_urb = NULL;
/* keep retire_data_urb for delay calculation */
subs->data_endpoint->retire_data_urb = retire_playback_urb;
subs->running = 0;
return 0;
}
return -EINVAL;
}
static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream,
int cmd)
{
int err;
struct snd_usb_substream *subs = substream->runtime->private_data;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
err = start_endpoints(subs, false);
if (err < 0)
return err;
subs->data_endpoint->retire_data_urb = retire_capture_urb;
subs->running = 1;
return 0;
case SNDRV_PCM_TRIGGER_STOP:
stop_endpoints(subs, false);
subs->running = 0;
return 0;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
subs->data_endpoint->retire_data_urb = NULL;
subs->running = 0;
return 0;
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
subs->data_endpoint->retire_data_urb = retire_capture_urb;
subs->running = 1;
return 0;
}
return -EINVAL;
}
static struct snd_pcm_ops snd_usb_playback_ops = {
.open = snd_usb_playback_open,
.close = snd_usb_playback_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_usb_hw_params,
.hw_free = snd_usb_hw_free,
.prepare = snd_usb_pcm_prepare,
.trigger = snd_usb_substream_playback_trigger,
.pointer = snd_usb_pcm_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
.mmap = snd_pcm_lib_mmap_vmalloc,
};
static struct snd_pcm_ops snd_usb_capture_ops = {
.open = snd_usb_capture_open,
.close = snd_usb_capture_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_usb_hw_params,
.hw_free = snd_usb_hw_free,
.prepare = snd_usb_pcm_prepare,
.trigger = snd_usb_substream_capture_trigger,
.pointer = snd_usb_pcm_pointer,
.page = snd_pcm_lib_get_vmalloc_page,
.mmap = snd_pcm_lib_mmap_vmalloc,
};
void snd_usb_set_pcm_ops(struct snd_pcm *pcm, int stream)
{
snd_pcm_set_ops(pcm, stream,
stream == SNDRV_PCM_STREAM_PLAYBACK ?
&snd_usb_playback_ops : &snd_usb_capture_ops);
}