Commit e89bb428 authored by Ruslan Bilovol's avatar Ruslan Bilovol Committed by Greg Kroah-Hartman
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usb: gadget: u_audio: add real feedback implementation



This adds interface between userspace and feedback endpoint to report real
feedback frequency to the Host.

Current implementation adds new userspace interface ALSA mixer control
"Capture Pitch 1000000" (similar to aloop driver's "PCM Rate Shift 100000"
mixer control)

Value in PPM is chosen to have correction value agnostic of the actual HW
rate, which the application is not necessarily dealing with, while still
retaining a good enough precision to allow smooth clock correction on the
playback side, if necessary.

Similar to sound/usb/endpoint.c, a slow down is allowed up to 25%. This
has no impact on the required bandwidth. Speedup correction has an impact
on the bandwidth reserved for the isochronous endpoint. The default
allowed speedup is 500ppm. This seems to be more than enough but, if
necessary, this is configurable through a module parameter. The reserved
bandwidth is rounded up to the next packet size.

Usage of this new control is easy to implement in existing userspace tools
like alsaloop from alsa-utils.

Signed-off-by: default avatarRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: default avatarJerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20210603220104.1216001-4-jbrunet@baylibre.com


Signed-off-by: default avatarGreg Kroah-Hartman <gregkh@linuxfoundation.org>
parent 40c73b30
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+1 −0
Original line number Diff line number Diff line
@@ -9,6 +9,7 @@ Description:
		c_srate    capture sampling rate
		c_ssize    capture sample size (bytes)
		c_sync     capture synchronization type (async/adaptive)
		fb_max     maximum extra bandwidth in async mode
		p_chmask   playback channel mask
		p_srate    playback sampling rate
		p_ssize    playback sample size (bytes)
+1 −0
Original line number Diff line number Diff line
@@ -729,6 +729,7 @@ The uac2 function provides these attributes in its function directory:
	c_srate		capture sampling rate
	c_ssize		capture sample size (bytes)
	c_sync		capture synchronization type (async/adaptive)
	fb_max          maximum extra bandwidth in async mode
	p_chmask	playback channel mask
	p_srate		playback sampling rate
	p_ssize		playback sample size (bytes)
+8 −1
Original line number Diff line number Diff line
@@ -584,8 +584,11 @@ static int set_ep_max_packet_size(const struct f_uac2_opts *uac2_opts,
		ssize = uac2_opts->c_ssize;
	}

	if (!is_playback && (uac2_opts->c_sync == USB_ENDPOINT_SYNC_ASYNC))
		srate = srate * (1000 + uac2_opts->fb_max) / 1000;

	max_size_bw = num_channels(chmask) * ssize *
		((srate / (factor / (1 << (ep_desc->bInterval - 1)))) + 1);
		DIV_ROUND_UP(srate, factor / (1 << (ep_desc->bInterval - 1)));
	ep_desc->wMaxPacketSize = cpu_to_le16(min_t(u16, max_size_bw,
						    max_size_ep));

@@ -957,6 +960,7 @@ afunc_bind(struct usb_configuration *cfg, struct usb_function *fn)
	agdev->params.c_srate = uac2_opts->c_srate;
	agdev->params.c_ssize = uac2_opts->c_ssize;
	agdev->params.req_number = uac2_opts->req_number;
	agdev->params.fb_max = uac2_opts->fb_max;
	ret = g_audio_setup(agdev, "UAC2 PCM", "UAC2_Gadget");
	if (ret)
		goto err_free_descs;
@@ -1329,6 +1333,7 @@ UAC2_ATTRIBUTE(c_srate);
UAC2_ATTRIBUTE_SYNC(c_sync);
UAC2_ATTRIBUTE(c_ssize);
UAC2_ATTRIBUTE(req_number);
UAC2_ATTRIBUTE(fb_max);

static struct configfs_attribute *f_uac2_attrs[] = {
	&f_uac2_opts_attr_p_chmask,
@@ -1339,6 +1344,7 @@ static struct configfs_attribute *f_uac2_attrs[] = {
	&f_uac2_opts_attr_c_ssize,
	&f_uac2_opts_attr_c_sync,
	&f_uac2_opts_attr_req_number,
	&f_uac2_opts_attr_fb_max,
	NULL,
};

@@ -1378,6 +1384,7 @@ static struct usb_function_instance *afunc_alloc_inst(void)
	opts->c_ssize = UAC2_DEF_CSSIZE;
	opts->c_sync = UAC2_DEF_CSYNC;
	opts->req_number = UAC2_DEF_REQ_NUM;
	opts->fb_max = UAC2_DEF_FB_MAX;
	return &opts->func_inst;
}

+115 −9
Original line number Diff line number Diff line
@@ -16,6 +16,7 @@
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/control.h>

#include "u_audio.h"

@@ -35,12 +36,12 @@ struct uac_rtd_params {

	void *rbuf;

	unsigned int pitch;	/* Stream pitch ratio to 1000000 */
	unsigned int max_psize;	/* MaxPacketSize of endpoint */

	struct usb_request **reqs;

	struct usb_request *req_fback; /* Feedback endpoint request */
	unsigned int ffback; /* Real frequency reported by feedback endpoint */
	bool fb_ep_enabled; /* if the ep is enabled */
};

@@ -75,18 +76,29 @@ static const struct snd_pcm_hardware uac_pcm_hardware = {
};

static void u_audio_set_fback_frequency(enum usb_device_speed speed,
					unsigned int freq, void *buf)
					unsigned long long freq,
					unsigned int pitch,
					void *buf)
{
	u32 ff = 0;

	/*
	 * Because the pitch base is 1000000, the final divider here
	 * will be 1000 * 1000000 = 1953125 << 9
	 *
	 * Instead of dealing with big numbers lets fold this 9 left shift
	 */

	if (speed == USB_SPEED_FULL) {
		/*
		 * Full-speed feedback endpoints report frequency
		 * in samples/microframe
		 * in samples/frame
		 * Format is encoded in Q10.10 left-justified in the 24 bits,
		 * so that it has a Q10.14 format.
		 *
		 * ff = (freq << 14) / 1000
		 */
		ff = DIV_ROUND_UP((freq << 14), 1000);
		freq <<= 5;
	} else {
		/*
		 * High-speed feedback endpoints report frequency
@@ -94,9 +106,14 @@ static void u_audio_set_fback_frequency(enum usb_device_speed speed,
		 * Format is encoded in Q12.13 fitted into four bytes so that
		 * the binary point is located between the second and the third
		 * byte fromat (that is Q16.16)
		 *
		 * ff = (freq << 16) / 8000
		 */
		ff = DIV_ROUND_UP((freq << 13), 1000);
		freq <<= 4;
	}

	ff = DIV_ROUND_CLOSEST_ULL((freq * pitch), 1953125);

	*(__le32 *)buf = cpu_to_le32(ff);
}

@@ -209,8 +226,8 @@ static void u_audio_iso_fback_complete(struct usb_ep *ep,
	struct uac_rtd_params *prm = req->context;
	struct snd_uac_chip *uac = prm->uac;
	struct g_audio *audio_dev = uac->audio_dev;
	struct uac_params *params = &audio_dev->params;
	int status = req->status;
	unsigned long flags;

	/* i/f shutting down */
	if (!prm->fb_ep_enabled || req->status == -ESHUTDOWN)
@@ -225,7 +242,8 @@ static void u_audio_iso_fback_complete(struct usb_ep *ep,
			__func__, status, req->actual, req->length);

	u_audio_set_fback_frequency(audio_dev->gadget->speed,
				    prm->ffback, req->buf);
				    params->c_srate, prm->pitch,
				    req->buf);

	if (usb_ep_queue(ep, req, GFP_ATOMIC))
		dev_err(uac->card->dev, "%d Error!\n", __LINE__);
@@ -480,9 +498,10 @@ int u_audio_start_capture(struct g_audio *audio_dev)
	 * Always start with original frequency since its deviation can't
	 * be meauserd at start of playback
	 */
	prm->ffback = params->c_srate;
	prm->pitch = 1000000;
	u_audio_set_fback_frequency(audio_dev->gadget->speed,
				    prm->ffback, req_fback->buf);
				    params->c_srate, prm->pitch,
				    req_fback->buf);

	if (usb_ep_queue(ep_fback, req_fback, GFP_ATOMIC))
		dev_err(dev, "%s:%d Error!\n", __func__, __LINE__);
@@ -578,12 +597,82 @@ void u_audio_stop_playback(struct g_audio *audio_dev)
}
EXPORT_SYMBOL_GPL(u_audio_stop_playback);

static int u_audio_pitch_info(struct snd_kcontrol *kcontrol,
				   struct snd_ctl_elem_info *uinfo)
{
	struct uac_rtd_params *prm = snd_kcontrol_chip(kcontrol);
	struct snd_uac_chip *uac = prm->uac;
	struct g_audio *audio_dev = uac->audio_dev;
	struct uac_params *params = &audio_dev->params;
	unsigned int pitch_min, pitch_max;

	pitch_min = (1000 - FBACK_SLOW_MAX) * 1000;
	pitch_max = (1000 + params->fb_max) * 1000;

	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
	uinfo->count = 1;
	uinfo->value.integer.min = pitch_min;
	uinfo->value.integer.max = pitch_max;
	uinfo->value.integer.step = 1;
	return 0;
}

static int u_audio_pitch_get(struct snd_kcontrol *kcontrol,
				   struct snd_ctl_elem_value *ucontrol)
{
	struct uac_rtd_params *prm = snd_kcontrol_chip(kcontrol);

	ucontrol->value.integer.value[0] = prm->pitch;

	return 0;
}

static int u_audio_pitch_put(struct snd_kcontrol *kcontrol,
				  struct snd_ctl_elem_value *ucontrol)
{
	struct uac_rtd_params *prm = snd_kcontrol_chip(kcontrol);
	struct snd_uac_chip *uac = prm->uac;
	struct g_audio *audio_dev = uac->audio_dev;
	struct uac_params *params = &audio_dev->params;
	unsigned int val;
	unsigned int pitch_min, pitch_max;
	int change = 0;

	pitch_min = (1000 - FBACK_SLOW_MAX) * 1000;
	pitch_max = (1000 + params->fb_max) * 1000;

	val = ucontrol->value.integer.value[0];

	if (val < pitch_min)
		val = pitch_min;
	if (val > pitch_max)
		val = pitch_max;

	if (prm->pitch != val) {
		prm->pitch = val;
		change = 1;
	}

	return change;
}

static const struct snd_kcontrol_new u_audio_controls[]  = {
{
	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
	.name =         "Capture Pitch 1000000",
	.info =         u_audio_pitch_info,
	.get =          u_audio_pitch_get,
	.put =          u_audio_pitch_put,
},
};

int g_audio_setup(struct g_audio *g_audio, const char *pcm_name,
					const char *card_name)
{
	struct snd_uac_chip *uac;
	struct snd_card *card;
	struct snd_pcm *pcm;
	struct snd_kcontrol *kctl;
	struct uac_params *params;
	int p_chmask, c_chmask;
	int err;
@@ -671,6 +760,23 @@ int g_audio_setup(struct g_audio *g_audio, const char *pcm_name,
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &uac_pcm_ops);
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &uac_pcm_ops);

	if (c_chmask && g_audio->in_ep_fback) {
		strscpy(card->mixername, card_name, sizeof(card->driver));

		kctl = snd_ctl_new1(&u_audio_controls[0], &uac->c_prm);
		if (!kctl) {
			err = -ENOMEM;
			goto snd_fail;
		}

		kctl->id.device = pcm->device;
		kctl->id.subdevice = 0;

		err = snd_ctl_add(card, kctl);
		if (err < 0)
			goto snd_fail;
	}

	strscpy(card->driver, card_name, sizeof(card->driver));
	strscpy(card->shortname, card_name, sizeof(card->shortname));
	sprintf(card->longname, "%s %i", card_name, card->dev->id);
+9 −0
Original line number Diff line number Diff line
@@ -11,6 +11,14 @@

#include <linux/usb/composite.h>

/*
 * Same maximum frequency deviation on the slower side as in
 * sound/usb/endpoint.c. Value is expressed in per-mil deviation.
 * The maximum deviation on the faster side will be provided as
 * parameter, as it impacts the endpoint required bandwidth.
 */
#define FBACK_SLOW_MAX	250

struct uac_params {
	/* playback */
	int p_chmask;	/* channel mask */
@@ -23,6 +31,7 @@ struct uac_params {
	int c_ssize;	/* sample size */

	int req_number; /* number of preallocated requests */
	int fb_max;	/* upper frequency drift feedback limit per-mil */
};

struct g_audio {
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