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Commit e83bff72 authored by Ajay Agarwal's avatar Ajay Agarwal
Browse files

usb: gadget: split out audio core



Abstract the peripheral side ALSA sound card code into a
component that can be called by various functions, so the
various flavors can be split apart and selectively reused.

Visible changes:
 - add uac_params structure to pass audio paramteres for
   g_audio_setup
 - make ALSA sound card's name configurable
 - add [in/out]_ep_maxpsize
 - allocate snd_uac_chip structure during g_audio_setup
 - add u_audio_[start/stop]_[capture/playback] functions

This change differs from opensource patch in that it does not
modify the UAC2 driver to use u_audio helper file for ALSA
sound card support.

Change-Id: Ia67ec02ed442a67bc359efc2655aa9ba49b79368
Signed-off-by: default avatarRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: default avatarFelipe Balbi <felipe.balbi@linux.intel.com>
Git-commit: eb9fecb9e69b0be8c267c55b0bb52a08e8fb6bee
Git-repo: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git


Signed-off-by: default avatarAjay Agarwal <ajaya@codeaurora.org>
parent a1f2b053
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+3 −0
Original line number Diff line number Diff line
@@ -151,6 +151,9 @@ config USB_U_SERIAL
config USB_U_ETHER
	tristate

config USB_U_AUDIO
	tristate

config USB_F_SERIAL
	tristate

+1 −0
Original line number Diff line number Diff line
@@ -32,6 +32,7 @@ usb_f_mass_storage-y := f_mass_storage.o storage_common.o
obj-$(CONFIG_USB_F_MASS_STORAGE)+= usb_f_mass_storage.o
usb_f_fs-y			:= f_fs.o
obj-$(CONFIG_USB_F_FS)		+= usb_f_fs.o
obj-$(CONFIG_USB_U_AUDIO)	+= u_audio.o
usb_f_uac1-y			:= f_uac1.o u_uac1.o
obj-$(CONFIG_USB_F_UAC1)	+= usb_f_uac1.o
usb_f_uac2-y			:= f_uac2.o
+662 −0
Original line number Diff line number Diff line
/*
 * u_audio.c -- interface to USB gadget "ALSA sound card" utilities
 *
 * Copyright (C) 2016
 * Author: Ruslan Bilovol <ruslan.bilovol@gmail.com>
 *
 * Sound card implementation was cut-and-pasted with changes
 * from f_uac2.c and has:
 *    Copyright (C) 2011
 *    Yadwinder Singh (yadi.brar01@gmail.com)
 *    Jaswinder Singh (jaswinder.singh@linaro.org)
 *
 * 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.
 */

#include <linux/module.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>

#include "u_audio.h"

#define BUFF_SIZE_MAX	(PAGE_SIZE * 16)
#define PRD_SIZE_MAX	PAGE_SIZE
#define MIN_PERIODS	4

struct uac_req {
	struct uac_rtd_params *pp; /* parent param */
	struct usb_request *req;
};

/* Runtime data params for one stream */
struct uac_rtd_params {
	struct snd_uac_chip *uac; /* parent chip */
	bool ep_enabled; /* if the ep is enabled */
	/* Size of the ring buffer */
	size_t dma_bytes;
	unsigned char *dma_area;

	struct snd_pcm_substream *ss;

	/* Ring buffer */
	ssize_t hw_ptr;

	void *rbuf;

	size_t period_size;

	unsigned max_psize;	/* MaxPacketSize of endpoint */
	struct uac_req *ureq;

	spinlock_t lock;
};

struct snd_uac_chip {
	struct g_audio *audio_dev;

	struct uac_rtd_params p_prm;
	struct uac_rtd_params c_prm;

	struct snd_card *card;
	struct snd_pcm *pcm;

	/* timekeeping for the playback endpoint */
	unsigned int p_interval;
	unsigned int p_residue;

	/* pre-calculated values for playback iso completion */
	unsigned int p_pktsize;
	unsigned int p_pktsize_residue;
	unsigned int p_framesize;
};

static struct snd_pcm_hardware uac_pcm_hardware = {
	.info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER
		 | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID
		 | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME,
	.rates = SNDRV_PCM_RATE_CONTINUOUS,
	.periods_max = BUFF_SIZE_MAX / PRD_SIZE_MAX,
	.buffer_bytes_max = BUFF_SIZE_MAX,
	.period_bytes_max = PRD_SIZE_MAX,
	.periods_min = MIN_PERIODS,
};

static void u_audio_iso_complete(struct usb_ep *ep, struct usb_request *req)
{
	unsigned pending;
	unsigned long flags;
	unsigned int hw_ptr;
	bool update_alsa = false;
	int status = req->status;
	struct uac_req *ur = req->context;
	struct snd_pcm_substream *substream;
	struct uac_rtd_params *prm = ur->pp;
	struct snd_uac_chip *uac = prm->uac;

	/* i/f shutting down */
	if (!prm->ep_enabled || req->status == -ESHUTDOWN)
		return;

	/*
	 * We can't really do much about bad xfers.
	 * Afterall, the ISOCH xfers could fail legitimately.
	 */
	if (status)
		pr_debug("%s: iso_complete status(%d) %d/%d\n",
			__func__, status, req->actual, req->length);

	substream = prm->ss;

	/* Do nothing if ALSA isn't active */
	if (!substream)
		goto exit;

	spin_lock_irqsave(&prm->lock, flags);

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		/*
		 * For each IN packet, take the quotient of the current data
		 * rate and the endpoint's interval as the base packet size.
		 * If there is a residue from this division, add it to the
		 * residue accumulator.
		 */
		req->length = uac->p_pktsize;
		uac->p_residue += uac->p_pktsize_residue;

		/*
		 * Whenever there are more bytes in the accumulator than we
		 * need to add one more sample frame, increase this packet's
		 * size and decrease the accumulator.
		 */
		if (uac->p_residue / uac->p_interval >= uac->p_framesize) {
			req->length += uac->p_framesize;
			uac->p_residue -= uac->p_framesize *
					   uac->p_interval;
		}

		req->actual = req->length;
	}

	pending = prm->hw_ptr % prm->period_size;
	pending += req->actual;
	if (pending >= prm->period_size)
		update_alsa = true;

	hw_ptr = prm->hw_ptr;
	prm->hw_ptr = (prm->hw_ptr + req->actual) % prm->dma_bytes;

	spin_unlock_irqrestore(&prm->lock, flags);

	/* Pack USB load in ALSA ring buffer */
	pending = prm->dma_bytes - hw_ptr;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		if (unlikely(pending < req->actual)) {
			memcpy(req->buf, prm->dma_area + hw_ptr, pending);
			memcpy(req->buf + pending, prm->dma_area,
			       req->actual - pending);
		} else {
			memcpy(req->buf, prm->dma_area + hw_ptr, req->actual);
		}
	} else {
		if (unlikely(pending < req->actual)) {
			memcpy(prm->dma_area + hw_ptr, req->buf, pending);
			memcpy(prm->dma_area, req->buf + pending,
			       req->actual - pending);
		} else {
			memcpy(prm->dma_area + hw_ptr, req->buf, req->actual);
		}
	}

exit:
	if (usb_ep_queue(ep, req, GFP_ATOMIC))
		dev_err(uac->card->dev, "%d Error!\n", __LINE__);

	if (update_alsa)
		snd_pcm_period_elapsed(substream);
}

static int uac_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
	struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
	struct uac_rtd_params *prm;
	struct g_audio *audio_dev;
	struct uac_params *params;
	unsigned long flags;
	int err = 0;

	audio_dev = uac->audio_dev;
	params = &audio_dev->params;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		prm = &uac->p_prm;
	else
		prm = &uac->c_prm;

	spin_lock_irqsave(&prm->lock, flags);

	/* Reset */
	prm->hw_ptr = 0;

	switch (cmd) {
	case SNDRV_PCM_TRIGGER_START:
	case SNDRV_PCM_TRIGGER_RESUME:
		prm->ss = substream;
		break;
	case SNDRV_PCM_TRIGGER_STOP:
	case SNDRV_PCM_TRIGGER_SUSPEND:
		prm->ss = NULL;
		break;
	default:
		err = -EINVAL;
	}

	spin_unlock_irqrestore(&prm->lock, flags);

	/* Clear buffer after Play stops */
	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !prm->ss)
		memset(prm->rbuf, 0, prm->max_psize * params->req_number);

	return err;
}

static snd_pcm_uframes_t uac_pcm_pointer(struct snd_pcm_substream *substream)
{
	struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
	struct uac_rtd_params *prm;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		prm = &uac->p_prm;
	else
		prm = &uac->c_prm;

	return bytes_to_frames(substream->runtime, prm->hw_ptr);
}

static int uac_pcm_hw_params(struct snd_pcm_substream *substream,
			       struct snd_pcm_hw_params *hw_params)
{
	struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
	struct uac_rtd_params *prm;
	int err;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		prm = &uac->p_prm;
	else
		prm = &uac->c_prm;

	err = snd_pcm_lib_malloc_pages(substream,
					params_buffer_bytes(hw_params));
	if (err >= 0) {
		prm->dma_bytes = substream->runtime->dma_bytes;
		prm->dma_area = substream->runtime->dma_area;
		prm->period_size = params_period_bytes(hw_params);
	}

	return err;
}

static int uac_pcm_hw_free(struct snd_pcm_substream *substream)
{
	struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
	struct uac_rtd_params *prm;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		prm = &uac->p_prm;
	else
		prm = &uac->c_prm;

	prm->dma_area = NULL;
	prm->dma_bytes = 0;
	prm->period_size = 0;

	return snd_pcm_lib_free_pages(substream);
}

static int uac_pcm_open(struct snd_pcm_substream *substream)
{
	struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
	struct snd_pcm_runtime *runtime = substream->runtime;
	struct g_audio *audio_dev;
	struct uac_params *params;
	int p_ssize, c_ssize;
	int p_srate, c_srate;
	int p_chmask, c_chmask;

	audio_dev = uac->audio_dev;
	params = &audio_dev->params;
	p_ssize = params->p_ssize;
	c_ssize = params->c_ssize;
	p_srate = params->p_srate;
	c_srate = params->c_srate;
	p_chmask = params->p_chmask;
	c_chmask = params->c_chmask;
	uac->p_residue = 0;

	runtime->hw = uac_pcm_hardware;

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		spin_lock_init(&uac->p_prm.lock);
		runtime->hw.rate_min = p_srate;
		switch (p_ssize) {
		case 3:
			runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_3LE;
			break;
		case 4:
			runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
			break;
		default:
			runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
			break;
		}
		runtime->hw.channels_min = num_channels(p_chmask);
		runtime->hw.period_bytes_min = 2 * uac->p_prm.max_psize
						/ runtime->hw.periods_min;
	} else {
		spin_lock_init(&uac->c_prm.lock);
		runtime->hw.rate_min = c_srate;
		switch (c_ssize) {
		case 3:
			runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_3LE;
			break;
		case 4:
			runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
			break;
		default:
			runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
			break;
		}
		runtime->hw.channels_min = num_channels(c_chmask);
		runtime->hw.period_bytes_min = 2 * uac->c_prm.max_psize
						/ runtime->hw.periods_min;
	}

	runtime->hw.rate_max = runtime->hw.rate_min;
	runtime->hw.channels_max = runtime->hw.channels_min;

	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);

	return 0;
}

/* ALSA cries without these function pointers */
static int uac_pcm_null(struct snd_pcm_substream *substream)
{
	return 0;
}

static struct snd_pcm_ops uac_pcm_ops = {
	.open = uac_pcm_open,
	.close = uac_pcm_null,
	.ioctl = snd_pcm_lib_ioctl,
	.hw_params = uac_pcm_hw_params,
	.hw_free = uac_pcm_hw_free,
	.trigger = uac_pcm_trigger,
	.pointer = uac_pcm_pointer,
	.prepare = uac_pcm_null,
};

static inline void free_ep(struct uac_rtd_params *prm, struct usb_ep *ep)
{
	struct snd_uac_chip *uac = prm->uac;
	struct g_audio *audio_dev;
	struct uac_params *params;
	int i;

	if (!prm->ep_enabled)
		return;

	prm->ep_enabled = false;

	audio_dev = uac->audio_dev;
	params = &audio_dev->params;

	for (i = 0; i < params->req_number; i++) {
		if (prm->ureq[i].req) {
			usb_ep_dequeue(ep, prm->ureq[i].req);
			usb_ep_free_request(ep, prm->ureq[i].req);
			prm->ureq[i].req = NULL;
		}
	}

	if (usb_ep_disable(ep))
		dev_err(uac->card->dev, "%s:%d Error!\n", __func__, __LINE__);
}


int u_audio_start_capture(struct g_audio *audio_dev)
{
	struct snd_uac_chip *uac = audio_dev->uac;
	struct usb_gadget *gadget = audio_dev->gadget;
	struct device *dev = &gadget->dev;
	struct usb_request *req;
	struct usb_ep *ep;
	struct uac_rtd_params *prm;
	struct uac_params *params = &audio_dev->params;
	int req_len, i;

	ep = audio_dev->out_ep;
	prm = &uac->c_prm;
	config_ep_by_speed(gadget, &audio_dev->func, ep);
	req_len = prm->max_psize;

	prm->ep_enabled = true;
	usb_ep_enable(ep);

	for (i = 0; i < params->req_number; i++) {
		if (!prm->ureq[i].req) {
			req = usb_ep_alloc_request(ep, GFP_ATOMIC);
			if (req == NULL)
				return -ENOMEM;

			prm->ureq[i].req = req;
			prm->ureq[i].pp = prm;

			req->zero = 0;
			req->context = &prm->ureq[i];
			req->length = req_len;
			req->complete = u_audio_iso_complete;
			req->buf = prm->rbuf + i * prm->max_psize;
		}

		if (usb_ep_queue(ep, prm->ureq[i].req, GFP_ATOMIC))
			dev_err(dev, "%s:%d Error!\n", __func__, __LINE__);
	}

	return 0;
}
EXPORT_SYMBOL_GPL(u_audio_start_capture);

void u_audio_stop_capture(struct g_audio *audio_dev)
{
	struct snd_uac_chip *uac = audio_dev->uac;

	free_ep(&uac->c_prm, audio_dev->out_ep);
}
EXPORT_SYMBOL_GPL(u_audio_stop_capture);

int u_audio_start_playback(struct g_audio *audio_dev)
{
	struct snd_uac_chip *uac = audio_dev->uac;
	struct usb_gadget *gadget = audio_dev->gadget;
	struct device *dev = &gadget->dev;
	struct usb_request *req;
	struct usb_ep *ep;
	struct uac_rtd_params *prm;
	struct uac_params *params = &audio_dev->params;
	unsigned int factor, rate;
	const struct usb_endpoint_descriptor *ep_desc;
	int req_len, i;

	ep = audio_dev->in_ep;
	prm = &uac->p_prm;
	config_ep_by_speed(gadget, &audio_dev->func, ep);

	ep_desc = ep->desc;

	/* pre-calculate the playback endpoint's interval */
	if (gadget->speed == USB_SPEED_FULL)
		factor = 1000;
	else
		factor = 8000;

	/* pre-compute some values for iso_complete() */
	uac->p_framesize = params->p_ssize *
			    num_channels(params->p_chmask);
	rate = params->p_srate * uac->p_framesize;
	uac->p_interval = factor / (1 << (ep_desc->bInterval - 1));
	uac->p_pktsize = min_t(unsigned int, rate / uac->p_interval,
				prm->max_psize);

	if (uac->p_pktsize < prm->max_psize)
		uac->p_pktsize_residue = rate % uac->p_interval;
	else
		uac->p_pktsize_residue = 0;

	req_len = uac->p_pktsize;
	uac->p_residue = 0;

	prm->ep_enabled = true;
	usb_ep_enable(ep);

	for (i = 0; i < params->req_number; i++) {
		if (!prm->ureq[i].req) {
			req = usb_ep_alloc_request(ep, GFP_ATOMIC);
			if (req == NULL)
				return -ENOMEM;

			prm->ureq[i].req = req;
			prm->ureq[i].pp = prm;

			req->zero = 0;
			req->context = &prm->ureq[i];
			req->length = req_len;
			req->complete = u_audio_iso_complete;
			req->buf = prm->rbuf + i * prm->max_psize;
		}

		if (usb_ep_queue(ep, prm->ureq[i].req, GFP_ATOMIC))
			dev_err(dev, "%s:%d Error!\n", __func__, __LINE__);
	}

	return 0;
}
EXPORT_SYMBOL_GPL(u_audio_start_playback);

void u_audio_stop_playback(struct g_audio *audio_dev)
{
	struct snd_uac_chip *uac = audio_dev->uac;

	free_ep(&uac->p_prm, audio_dev->in_ep);
}
EXPORT_SYMBOL_GPL(u_audio_stop_playback);

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 uac_params *params;
	int p_chmask, c_chmask;
	int err;

	if (!g_audio)
		return -EINVAL;

	uac = kzalloc(sizeof(*uac), GFP_KERNEL);
	if (!uac)
		return -ENOMEM;
	g_audio->uac = uac;
	uac->audio_dev = g_audio;

	params = &g_audio->params;
	p_chmask = params->p_chmask;
	c_chmask = params->c_chmask;

	if (c_chmask) {
		struct uac_rtd_params *prm = &uac->c_prm;

		uac->c_prm.uac = uac;
		prm->max_psize = g_audio->out_ep_maxpsize;

		prm->ureq = kcalloc(params->req_number, sizeof(struct uac_req),
				GFP_KERNEL);
		if (!prm->ureq) {
			err = -ENOMEM;
			goto fail;
		}

		prm->rbuf = kcalloc(params->req_number, prm->max_psize,
				GFP_KERNEL);
		if (!prm->rbuf) {
			prm->max_psize = 0;
			err = -ENOMEM;
			goto fail;
		}
	}

	if (p_chmask) {
		struct uac_rtd_params *prm = &uac->p_prm;

		uac->p_prm.uac = uac;
		prm->max_psize = g_audio->in_ep_maxpsize;

		prm->ureq = kcalloc(params->req_number, sizeof(struct uac_req),
				GFP_KERNEL);
		if (!prm->ureq) {
			err = -ENOMEM;
			goto fail;
		}

		prm->rbuf = kcalloc(params->req_number, prm->max_psize,
				GFP_KERNEL);
		if (!prm->rbuf) {
			prm->max_psize = 0;
			err = -ENOMEM;
			goto fail;
		}
	}

	/* Choose any slot, with no id */
	err = snd_card_new(&g_audio->gadget->dev,
			-1, NULL, THIS_MODULE, 0, &card);
	if (err < 0)
		goto fail;

	uac->card = card;

	/*
	 * Create first PCM device
	 * Create a substream only for non-zero channel streams
	 */
	err = snd_pcm_new(uac->card, pcm_name, 0,
			       p_chmask ? 1 : 0, c_chmask ? 1 : 0, &pcm);
	if (err < 0)
		goto snd_fail;

	strcpy(pcm->name, pcm_name);
	pcm->private_data = uac;
	uac->pcm = pcm;

	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &uac_pcm_ops);
	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &uac_pcm_ops);

	strcpy(card->driver, card_name);
	strcpy(card->shortname, card_name);
	sprintf(card->longname, "%s %i", card_name, card->dev->id);

	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
		snd_dma_continuous_data(GFP_KERNEL), 0, BUFF_SIZE_MAX);

	err = snd_card_register(card);

	if (!err)
		return 0;

snd_fail:
	snd_card_free(card);
fail:
	kfree(uac->p_prm.ureq);
	kfree(uac->c_prm.ureq);
	kfree(uac->p_prm.rbuf);
	kfree(uac->c_prm.rbuf);
	kfree(uac);

	return err;
}
EXPORT_SYMBOL_GPL(g_audio_setup);

void g_audio_cleanup(struct g_audio *g_audio)
{
	struct snd_uac_chip *uac;
	struct snd_card *card;

	if (!g_audio || !g_audio->uac)
		return;

	uac = g_audio->uac;
	card = uac->card;
	if (card)
		snd_card_free(card);

	kfree(uac->p_prm.ureq);
	kfree(uac->c_prm.ureq);
	kfree(uac->p_prm.rbuf);
	kfree(uac->c_prm.rbuf);
	kfree(uac);
}
EXPORT_SYMBOL_GPL(g_audio_cleanup);

MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("USB gadget \"ALSA sound card\" utilities");
MODULE_AUTHOR("Ruslan Bilovol");
+95 −0
Original line number Diff line number Diff line
/*
 * u_audio.h -- interface to USB gadget "ALSA sound card" utilities
 *
 * Copyright (C) 2016
 * Author: Ruslan Bilovol <ruslan.bilovol@gmail.com>
 *
 * 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.
 *
 */

#ifndef __U_AUDIO_H
#define __U_AUDIO_H

#include <linux/usb/composite.h>

struct uac_params {
	/* playback */
	int p_chmask;	/* channel mask */
	int p_srate;	/* rate in Hz */
	int p_ssize;	/* sample size */

	/* capture */
	int c_chmask;	/* channel mask */
	int c_srate;	/* rate in Hz */
	int c_ssize;	/* sample size */

	int req_number; /* number of preallocated requests */
};

struct g_audio {
	struct usb_function func;
	struct usb_gadget *gadget;

	struct usb_ep *in_ep;
	struct usb_ep *out_ep;

	/* Max packet size for all in_ep possible speeds */
	unsigned int in_ep_maxpsize;
	/* Max packet size for all out_ep possible speeds */
	unsigned int out_ep_maxpsize;

	/* The ALSA Sound Card it represents on the USB-Client side */
	struct snd_uac_chip *uac;

	struct uac_params params;
};

static inline struct g_audio *func_to_g_audio(struct usb_function *f)
{
	return container_of(f, struct g_audio, func);
}

static inline uint num_channels(uint chanmask)
{
	uint num = 0;

	while (chanmask) {
		num += (chanmask & 1);
		chanmask >>= 1;
	}

	return num;
}

/*
 * g_audio_setup - initialize one virtual ALSA sound card
 * @g_audio: struct with filled params, in_ep_maxpsize, out_ep_maxpsize
 * @pcm_name: the id string for a PCM instance of this sound card
 * @card_name: name of this soundcard
 *
 * This sets up the single virtual ALSA sound card that may be exported by a
 * gadget driver using this framework.
 *
 * Context: may sleep
 *
 * Returns zero on success, or a negative error on failure.
 */
int g_audio_setup(struct g_audio *g_audio, const char *pcm_name,
					const char *card_name);
void g_audio_cleanup(struct g_audio *g_audio);

int u_audio_start_capture(struct g_audio *g_audio);
void u_audio_stop_capture(struct g_audio *g_audio);
int u_audio_start_playback(struct g_audio *g_audio);
void u_audio_stop_playback(struct g_audio *g_audio);

#endif /* __U_AUDIO_H */