// SPDX-License-Identifier: GPL-2.0+
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/*
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* u_audio.c -- interface to USB gadget "ALSA sound card" utilities
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*
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* Copyright (C) 2016
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* Author: Ruslan Bilovol <ruslan.bilovol@gmail.com>
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*
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* Sound card implementation was cut-and-pasted with changes
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* from f_uac2.c and has:
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* Copyright (C) 2011
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* Yadwinder Singh (yadi.brar01@gmail.com)
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* Jaswinder Singh (jaswinder.singh@linaro.org)
|
*/
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#include <linux/module.h>
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#include <linux/usb/audio.h>
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#include <sound/control.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include "u_audio.h"
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#define BUFF_SIZE_MAX (PAGE_SIZE * 16)
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#define PRD_SIZE_MAX PAGE_SIZE
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#define MIN_PERIODS 4
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|
#define CLK_PPM_GROUP_SIZE 20
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static struct class *audio_class;
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struct uac_req {
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struct uac_rtd_params *pp; /* parent param */
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struct usb_request *req;
|
};
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/* Runtime data params for one stream */
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struct uac_rtd_params {
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struct snd_uac_chip *uac; /* parent chip */
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bool ep_enabled; /* if the ep is enabled */
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struct snd_pcm_substream *ss;
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/* Ring buffer */
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ssize_t hw_ptr;
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void *rbuf;
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unsigned max_psize; /* MaxPacketSize of endpoint */
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struct uac_req *ureq;
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spinlock_t lock;
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};
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struct snd_uac_chip {
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struct g_audio *audio_dev;
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struct uac_rtd_params p_prm;
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struct uac_rtd_params c_prm;
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struct snd_card *card;
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struct snd_pcm *pcm;
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/* timekeeping for the playback endpoint */
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unsigned int p_interval;
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unsigned int p_residue;
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/* pre-calculated values for playback iso completion */
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unsigned int p_pktsize;
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unsigned int p_pktsize_residue;
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unsigned int p_framesize;
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};
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static const struct snd_pcm_hardware uac_pcm_hardware = {
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.info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER
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| SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID
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| SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME,
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.rates = SNDRV_PCM_RATE_CONTINUOUS,
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.periods_max = BUFF_SIZE_MAX / PRD_SIZE_MAX,
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.buffer_bytes_max = BUFF_SIZE_MAX,
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.period_bytes_max = PRD_SIZE_MAX,
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.periods_min = MIN_PERIODS,
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};
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static void u_audio_iso_complete(struct usb_ep *ep, struct usb_request *req)
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{
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unsigned pending;
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unsigned long flags, flags2;
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unsigned int hw_ptr;
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int status = req->status;
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struct uac_req *ur = req->context;
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struct snd_pcm_substream *substream;
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struct snd_pcm_runtime *runtime;
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struct uac_rtd_params *prm = ur->pp;
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struct snd_uac_chip *uac = prm->uac;
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/* i/f shutting down */
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if (!prm->ep_enabled) {
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usb_ep_free_request(ep, req);
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return;
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}
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if (req->status == -ESHUTDOWN)
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return;
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/*
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* We can't really do much about bad xfers.
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* Afterall, the ISOCH xfers could fail legitimately.
|
*/
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if (status)
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pr_debug("%s: iso_complete status(%d) %d/%d\n",
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__func__, status, req->actual, req->length);
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substream = prm->ss;
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/* Do nothing if ALSA isn't active */
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if (!substream)
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goto exit;
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snd_pcm_stream_lock_irqsave(substream, flags2);
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runtime = substream->runtime;
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if (!runtime || !snd_pcm_running(substream)) {
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snd_pcm_stream_unlock_irqrestore(substream, flags2);
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goto exit;
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}
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spin_lock_irqsave(&prm->lock, flags);
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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/*
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* For each IN packet, take the quotient of the current data
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* rate and the endpoint's interval as the base packet size.
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* If there is a residue from this division, add it to the
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* residue accumulator.
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*/
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req->length = uac->p_pktsize;
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uac->p_residue += uac->p_pktsize_residue;
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/*
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* Whenever there are more bytes in the accumulator than we
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* need to add one more sample frame, increase this packet's
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* size and decrease the accumulator.
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*/
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if (uac->p_residue / uac->p_interval >= uac->p_framesize) {
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req->length += uac->p_framesize;
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uac->p_residue -= uac->p_framesize *
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uac->p_interval;
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}
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req->actual = req->length;
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}
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hw_ptr = prm->hw_ptr;
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spin_unlock_irqrestore(&prm->lock, flags);
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/* Pack USB load in ALSA ring buffer */
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pending = runtime->dma_bytes - hw_ptr;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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if (unlikely(pending < req->actual)) {
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memcpy(req->buf, runtime->dma_area + hw_ptr, pending);
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memcpy(req->buf + pending, runtime->dma_area,
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req->actual - pending);
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} else {
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memcpy(req->buf, runtime->dma_area + hw_ptr,
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req->actual);
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}
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} else {
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if (unlikely(pending < req->actual)) {
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memcpy(runtime->dma_area + hw_ptr, req->buf, pending);
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memcpy(runtime->dma_area, req->buf + pending,
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req->actual - pending);
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} else {
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memcpy(runtime->dma_area + hw_ptr, req->buf,
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req->actual);
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}
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}
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spin_lock_irqsave(&prm->lock, flags);
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/* update hw_ptr after data is copied to memory */
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prm->hw_ptr = (hw_ptr + req->actual) % runtime->dma_bytes;
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hw_ptr = prm->hw_ptr;
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spin_unlock_irqrestore(&prm->lock, flags);
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snd_pcm_stream_unlock_irqrestore(substream, flags2);
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if ((hw_ptr % snd_pcm_lib_period_bytes(substream)) < req->actual)
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snd_pcm_period_elapsed(substream);
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exit:
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if (usb_ep_queue(ep, req, GFP_ATOMIC))
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dev_err(uac->card->dev, "%d Error!\n", __LINE__);
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}
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static int uac_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
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{
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struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
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struct uac_rtd_params *prm;
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struct g_audio *audio_dev;
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struct uac_params *params;
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unsigned long flags;
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int err = 0;
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audio_dev = uac->audio_dev;
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params = &audio_dev->params;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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prm = &uac->p_prm;
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else
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prm = &uac->c_prm;
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spin_lock_irqsave(&prm->lock, flags);
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/* Reset */
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prm->hw_ptr = 0;
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switch (cmd) {
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case SNDRV_PCM_TRIGGER_START:
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case SNDRV_PCM_TRIGGER_RESUME:
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prm->ss = substream;
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break;
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case SNDRV_PCM_TRIGGER_STOP:
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case SNDRV_PCM_TRIGGER_SUSPEND:
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prm->ss = NULL;
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break;
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default:
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err = -EINVAL;
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}
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spin_unlock_irqrestore(&prm->lock, flags);
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/* Clear buffer after Play stops */
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && !prm->ss)
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memset(prm->rbuf, 0, prm->max_psize * params->req_number);
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return err;
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}
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static snd_pcm_uframes_t uac_pcm_pointer(struct snd_pcm_substream *substream)
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{
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struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
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struct uac_rtd_params *prm;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
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prm = &uac->p_prm;
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else
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prm = &uac->c_prm;
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return bytes_to_frames(substream->runtime, prm->hw_ptr);
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}
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static int uac_pcm_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *hw_params)
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{
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return snd_pcm_lib_malloc_pages(substream,
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params_buffer_bytes(hw_params));
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}
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static int uac_pcm_hw_free(struct snd_pcm_substream *substream)
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{
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return snd_pcm_lib_free_pages(substream);
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}
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static int uac_pcm_open(struct snd_pcm_substream *substream)
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{
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struct snd_uac_chip *uac = snd_pcm_substream_chip(substream);
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct g_audio *audio_dev = uac->audio_dev;
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struct uac_params *params;
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int p_ssize, c_ssize;
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int p_srate, c_srate;
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int p_chmask, c_chmask;
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params = &audio_dev->params;
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p_ssize = params->p_ssize;
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c_ssize = params->c_ssize;
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p_srate = params->p_srate_active;
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c_srate = params->c_srate_active;
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p_chmask = params->p_chmask;
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c_chmask = params->c_chmask;
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uac->p_residue = 0;
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runtime->hw = uac_pcm_hardware;
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if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
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spin_lock_init(&uac->p_prm.lock);
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runtime->hw.rate_min = p_srate;
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switch (p_ssize) {
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case 3:
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runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_3LE;
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break;
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case 4:
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runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
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break;
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default:
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runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
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break;
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}
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runtime->hw.channels_min = num_channels(p_chmask);
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runtime->hw.period_bytes_min = 2 * uac->p_prm.max_psize
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/ runtime->hw.periods_min;
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} else {
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spin_lock_init(&uac->c_prm.lock);
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runtime->hw.rate_min = c_srate;
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switch (c_ssize) {
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case 3:
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runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_3LE;
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break;
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case 4:
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runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE;
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break;
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default:
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runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
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break;
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}
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runtime->hw.channels_min = num_channels(c_chmask);
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runtime->hw.period_bytes_min = 2 * uac->c_prm.max_psize
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/ runtime->hw.periods_min;
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}
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runtime->hw.rate_max = runtime->hw.rate_min;
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runtime->hw.channels_max = runtime->hw.channels_min;
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snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
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return 0;
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}
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static int uac_pcm_rate_info(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_info *uinfo)
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{
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uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
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uinfo->count = 1;
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uinfo->value.integer.min = 0;
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uinfo->value.integer.max = 324000;
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return 0;
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}
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static int uac_pcm_rate_get(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_uac_chip *uac = snd_kcontrol_chip(kcontrol);
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struct g_audio *audio_dev = uac->audio_dev;
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struct uac_params *params = &audio_dev->params;
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if (kcontrol->private_value == SNDRV_PCM_STREAM_CAPTURE)
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ucontrol->value.integer.value[0] = params->c_srate_active;
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else if (kcontrol->private_value == SNDRV_PCM_STREAM_PLAYBACK)
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ucontrol->value.integer.value[0] = params->p_srate_active;
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else
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return -EINVAL;
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return 0;
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}
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static struct snd_kcontrol_new uac_pcm_controls[] = {
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{
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.iface = SNDRV_CTL_ELEM_IFACE_PCM,
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.name = "Capture Rate",
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.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
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.info = uac_pcm_rate_info,
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.get = uac_pcm_rate_get,
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.private_value = SNDRV_PCM_STREAM_CAPTURE,
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},
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{
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.iface = SNDRV_CTL_ELEM_IFACE_PCM,
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.name = "Playback Rate",
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.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
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.info = uac_pcm_rate_info,
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.get = uac_pcm_rate_get,
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.private_value = SNDRV_PCM_STREAM_PLAYBACK,
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},
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};
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/* ALSA cries without these function pointers */
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static int uac_pcm_null(struct snd_pcm_substream *substream)
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{
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return 0;
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}
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static const struct snd_pcm_ops uac_pcm_ops = {
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.open = uac_pcm_open,
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.close = uac_pcm_null,
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.ioctl = snd_pcm_lib_ioctl,
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.hw_params = uac_pcm_hw_params,
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.hw_free = uac_pcm_hw_free,
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.trigger = uac_pcm_trigger,
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.pointer = uac_pcm_pointer,
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.prepare = uac_pcm_null,
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};
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static inline void free_ep(struct uac_rtd_params *prm, struct usb_ep *ep)
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{
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struct snd_uac_chip *uac = prm->uac;
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struct g_audio *audio_dev;
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struct uac_params *params;
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int i;
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if (!prm->ep_enabled)
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return;
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audio_dev = uac->audio_dev;
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params = &audio_dev->params;
|
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for (i = 0; i < params->req_number; i++) {
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if (prm->ureq[i].req) {
|
if (usb_ep_dequeue(ep, prm->ureq[i].req))
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usb_ep_free_request(ep, prm->ureq[i].req);
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/*
|
* If usb_ep_dequeue() cannot successfully dequeue the
|
* request, the request will be freed by the completion
|
* callback.
|
*/
|
|
prm->ureq[i].req = NULL;
|
}
|
}
|
|
prm->ep_enabled = false;
|
|
if (usb_ep_disable(ep))
|
dev_err(uac->card->dev, "%s:%d Error!\n", __func__, __LINE__);
|
}
|
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static struct snd_kcontrol *u_audio_get_ctl(struct g_audio *audio_dev,
|
const char *name)
|
{
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struct snd_ctl_elem_id elem_id;
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|
memset(&elem_id, 0, sizeof(elem_id));
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elem_id.iface = SNDRV_CTL_ELEM_IFACE_PCM;
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strlcpy(elem_id.name, name, sizeof(elem_id.name));
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return snd_ctl_find_id(audio_dev->uac->card, &elem_id);
|
}
|
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int u_audio_set_capture_srate(struct g_audio *audio_dev, int srate)
|
{
|
struct snd_kcontrol *ctl = u_audio_get_ctl(audio_dev, "Capture Rate");
|
struct uac_params *params = &audio_dev->params;
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int i;
|
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for (i = 0; i < UAC_MAX_RATES; i++) {
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if (params->c_srate[i] == srate) {
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audio_dev->usb_state[SET_SAMPLE_RATE_OUT] = true;
|
schedule_work(&audio_dev->work);
|
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params->c_srate_active = srate;
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snd_ctl_notify(audio_dev->uac->card,
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SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id);
|
return 0;
|
}
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if (params->c_srate[i] == 0)
|
break;
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}
|
|
return -EINVAL;
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}
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EXPORT_SYMBOL_GPL(u_audio_set_capture_srate);
|
|
static void u_audio_set_playback_pktsize(struct g_audio *audio_dev, int srate)
|
{
|
struct uac_params *params = &audio_dev->params;
|
struct snd_uac_chip *uac = audio_dev->uac;
|
struct usb_gadget *gadget = audio_dev->gadget;
|
const struct usb_endpoint_descriptor *ep_desc;
|
struct uac_rtd_params *prm;
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unsigned int factor;
|
|
prm = &uac->p_prm;
|
/* set srate before starting playback, epin is not configured */
|
if (!prm->ep_enabled)
|
return;
|
|
ep_desc = audio_dev->in_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);
|
uac->p_interval = factor / (1 << (ep_desc->bInterval - 1));
|
uac->p_pktsize = min_t(unsigned int,
|
uac->p_framesize *
|
(params->p_srate_active / uac->p_interval),
|
prm->max_psize);
|
|
if (uac->p_pktsize < prm->max_psize)
|
uac->p_pktsize_residue = uac->p_framesize *
|
(params->p_srate_active % uac->p_interval);
|
else
|
uac->p_pktsize_residue = 0;
|
}
|
|
int u_audio_set_playback_srate(struct g_audio *audio_dev, int srate)
|
{
|
struct snd_kcontrol *ctl = u_audio_get_ctl(audio_dev, "Playback Rate");
|
struct uac_params *params = &audio_dev->params;
|
int i;
|
|
for (i = 0; i < UAC_MAX_RATES; i++) {
|
if (params->p_srate[i] == srate) {
|
audio_dev->usb_state[SET_SAMPLE_RATE_IN] = true;
|
schedule_work(&audio_dev->work);
|
|
params->p_srate_active = srate;
|
u_audio_set_playback_pktsize(audio_dev, srate);
|
snd_ctl_notify(audio_dev->uac->card,
|
SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id);
|
return 0;
|
}
|
if (params->p_srate[i] == 0)
|
break;
|
}
|
|
return -EINVAL;
|
}
|
EXPORT_SYMBOL_GPL(u_audio_set_playback_srate);
|
|
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;
|
|
/*
|
* For better compatibility on some PC Hosts which
|
* failed to send SetInterface(AltSet=0) to stop
|
* capture last time. It needs to stop capture
|
* prior to start capture next time.
|
*/
|
if (audio_dev->stream_state[STATE_OUT])
|
u_audio_stop_capture(audio_dev);
|
|
audio_dev->usb_state[SET_INTERFACE_OUT] = true;
|
audio_dev->stream_state[STATE_OUT] = true;
|
schedule_work(&audio_dev->work);
|
|
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);
|
|
audio_dev->usb_state[SET_INTERFACE_OUT] = true;
|
audio_dev->stream_state[STATE_OUT] = false;
|
schedule_work(&audio_dev->work);
|
}
|
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 = audio_dev->device;
|
struct usb_request *req;
|
struct usb_ep *ep;
|
struct uac_rtd_params *prm;
|
struct uac_params *params = &audio_dev->params;
|
int req_len, i;
|
|
/*
|
* For better compatibility on some PC Hosts which
|
* failed to send SetInterface(AltSet=0) to stop
|
* playback last time. It needs to stop playback
|
* prior to start playback next time.
|
*/
|
if (audio_dev->stream_state[STATE_IN])
|
u_audio_stop_playback(audio_dev);
|
|
audio_dev->usb_state[SET_INTERFACE_IN] = true;
|
audio_dev->stream_state[STATE_IN] = true;
|
schedule_work(&audio_dev->work);
|
|
dev_dbg(dev, "start playback with rate %d\n", params->p_srate_active);
|
ep = audio_dev->in_ep;
|
prm = &uac->p_prm;
|
config_ep_by_speed(gadget, &audio_dev->func, ep);
|
|
prm->ep_enabled = true;
|
usb_ep_enable(ep);
|
|
u_audio_set_playback_pktsize(audio_dev, params->p_srate_active);
|
req_len = uac->p_pktsize;
|
uac->p_residue = 0;
|
|
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);
|
|
audio_dev->usb_state[SET_INTERFACE_IN] = true;
|
audio_dev->stream_state[STATE_IN] = false;
|
schedule_work(&audio_dev->work);
|
}
|
EXPORT_SYMBOL_GPL(u_audio_stop_playback);
|
|
int u_audio_fu_set_cmd(struct usb_audio_control *con, u8 cmd, int value)
|
{
|
struct g_audio *audio_dev = (struct g_audio *)con->context;
|
struct uac_params *params = &audio_dev->params;
|
|
switch (cmd) {
|
case UAC_SET_CUR:
|
if (!strncmp(con->name, "Capture Mute", 12)) {
|
params->c_mute = value;
|
audio_dev->usb_state[SET_MUTE_OUT] = true;
|
} else if (!strncmp(con->name, "Capture Volume", 14)) {
|
params->c_volume = value;
|
audio_dev->usb_state[SET_VOLUME_OUT] = true;
|
} else if (!strncmp(con->name, "Playback Mute", 13)) {
|
params->p_mute = value;
|
audio_dev->usb_state[SET_MUTE_IN] = true;
|
} else if (!strncmp(con->name, "Playback Volume", 15)) {
|
params->p_volume = value;
|
audio_dev->usb_state[SET_VOLUME_IN] = true;
|
}
|
break;
|
case UAC_SET_RES:
|
/* fall through */
|
default:
|
return 0;
|
}
|
|
con->data[cmd] = value;
|
schedule_work(&audio_dev->work);
|
|
return 0;
|
}
|
EXPORT_SYMBOL_GPL(u_audio_fu_set_cmd);
|
|
int u_audio_fu_get_cmd(struct usb_audio_control *con, u8 cmd)
|
{
|
struct g_audio *audio_dev = (struct g_audio *)con->context;
|
|
dev_dbg(audio_dev->device, "GET_CMD con %s cmd %d data %d\n",
|
con->name, cmd, (int16_t)con->data[cmd]);
|
return con->data[cmd];
|
}
|
EXPORT_SYMBOL_GPL(u_audio_fu_get_cmd);
|
|
static void g_audio_work(struct work_struct *data)
|
{
|
struct g_audio *audio = container_of(data, struct g_audio, work);
|
struct uac_params *params = &audio->params;
|
char *uac_event[4] = { NULL, NULL, NULL, NULL };
|
char str[19];
|
signed short volume;
|
int i;
|
|
for (i = 0; i < SET_USB_STATE_MAX; i++) {
|
if (!audio->usb_state[i])
|
continue;
|
|
switch (i) {
|
case SET_INTERFACE_OUT:
|
uac_event[0] = "USB_STATE=SET_INTERFACE";
|
uac_event[1] = "STREAM_DIRECTION=OUT";
|
uac_event[2] = audio->stream_state[STATE_OUT] ?
|
"STREAM_STATE=ON" : "STREAM_STATE=OFF";
|
break;
|
case SET_INTERFACE_IN:
|
uac_event[0] = "USB_STATE=SET_INTERFACE";
|
uac_event[1] = "STREAM_DIRECTION=IN";
|
uac_event[2] = audio->stream_state[STATE_IN] ?
|
"STREAM_STATE=ON" : "STREAM_STATE=OFF";
|
break;
|
case SET_SAMPLE_RATE_OUT:
|
uac_event[0] = "USB_STATE=SET_SAMPLE_RATE";
|
uac_event[1] = "STREAM_DIRECTION=OUT";
|
snprintf(str, sizeof(str), "SAMPLE_RATE=%d",
|
params->c_srate_active);
|
uac_event[2] = str;
|
break;
|
case SET_SAMPLE_RATE_IN:
|
uac_event[0] = "USB_STATE=SET_SAMPLE_RATE";
|
uac_event[1] = "STREAM_DIRECTION=IN";
|
snprintf(str, sizeof(str), "SAMPLE_RATE=%d",
|
params->p_srate_active);
|
uac_event[2] = str;
|
break;
|
case SET_MUTE_OUT:
|
uac_event[0] = "USB_STATE=SET_MUTE";
|
uac_event[1] = "STREAM_DIRECTION=OUT";
|
snprintf(str, sizeof(str), "MUTE=%d", params->c_mute);
|
uac_event[2] = str;
|
break;
|
case SET_MUTE_IN:
|
uac_event[0] = "USB_STATE=SET_MUTE";
|
uac_event[1] = "STREAM_DIRECTION=IN";
|
snprintf(str, sizeof(str), "MUTE=%d", params->p_mute);
|
uac_event[2] = str;
|
break;
|
case SET_VOLUME_OUT:
|
uac_event[0] = "USB_STATE=SET_VOLUME";
|
uac_event[1] = "STREAM_DIRECTION=OUT";
|
volume = (signed short)params->c_volume;
|
volume /= UAC_VOLUME_RES;
|
snprintf(str, sizeof(str), "VOLUME=%d%%", volume + 50);
|
uac_event[2] = str;
|
break;
|
case SET_VOLUME_IN:
|
uac_event[0] = "USB_STATE=SET_VOLUME";
|
uac_event[1] = "STREAM_DIRECTION=IN";
|
volume = (signed short)params->p_volume;
|
volume /= UAC_VOLUME_RES;
|
snprintf(str, sizeof(str), "VOLUME=%d%%", volume + 50);
|
uac_event[2] = str;
|
break;
|
case SET_AUDIO_CLK:
|
uac_event[0] = "USB_STATE=SET_AUDIO_CLK";
|
snprintf(str, sizeof(str), "PPM=%d", params->ppm);
|
uac_event[1] = str;
|
default:
|
break;
|
}
|
|
audio->usb_state[i] = false;
|
kobject_uevent_env(&audio->device->kobj, KOBJ_CHANGE,
|
uac_event);
|
dev_dbg(audio->device, "%s: sent uac uevent %s %s %s\n",
|
__func__, uac_event[0], uac_event[1], uac_event[2]);
|
}
|
}
|
|
static void ppm_calculate_work(struct work_struct *data)
|
{
|
struct g_audio *g_audio = container_of(data, struct g_audio,
|
ppm_work.work);
|
struct usb_gadget *gadget = g_audio->gadget;
|
uint32_t frame_number, fn_msec, clk_msec;
|
struct frame_number_data *fn = g_audio->fn;
|
uint64_t time_now, time_msec_tmp;
|
int32_t ppm;
|
static int32_t ppms[CLK_PPM_GROUP_SIZE];
|
static int32_t ppm_sum;
|
int32_t cnt = fn->second % CLK_PPM_GROUP_SIZE;
|
|
time_now = ktime_get_raw();
|
frame_number = gadget->ops->get_frame(gadget);
|
|
if (g_audio->fn->time_last &&
|
time_now - g_audio->fn->time_last > 1500000000ULL)
|
dev_warn(g_audio->device, "PPM work scheduled too slow!\n");
|
|
g_audio->fn->time_last = time_now;
|
|
/*
|
* If usb is disconnected, the controller will not receive the
|
* SoF signal and frame number will be invalid. Because we can't
|
* get accurate time of disconnect and whether the gadget will be
|
* plugged into the same host next time or not. We must clear all
|
* statistics.
|
*/
|
if (gadget->state != USB_STATE_CONFIGURED) {
|
memset(g_audio->fn, 0, sizeof(*g_audio->fn));
|
dev_dbg(g_audio->device, "Disconnect. frame number is cleared\n");
|
goto out;
|
}
|
|
/* Fist statistic to record begin frame number and system time */
|
if (!g_audio->fn->second++) {
|
g_audio->fn->time_begin = g_audio->fn->time_last;
|
g_audio->fn->fn_begin = frame_number;
|
g_audio->fn->fn_last = frame_number;
|
goto out;
|
}
|
|
/*
|
* For DWC3 Controller, only 13 bits is used to store frame(micro)
|
* number. In other words, the frame number will overflow at most
|
* 2.047 seconds. We add another registor fn_overflow the record
|
* total frame number.
|
*/
|
if (frame_number <= g_audio->fn->fn_last)
|
g_audio->fn->fn_overflow++;
|
g_audio->fn->fn_last = frame_number;
|
|
if (!g_audio->fn->fn_overflow)
|
goto out;
|
|
/* The lower 3 bits represent micro number frame, we don't need it */
|
fn_msec = (((fn->fn_overflow - 1) << 14) +
|
(BIT(14) + fn->fn_last - fn->fn_begin) + BIT(2)) >> 3;
|
time_msec_tmp = fn->time_last - fn->time_begin + 500000ULL;
|
do_div(time_msec_tmp, 1000000U);
|
clk_msec = (uint32_t)time_msec_tmp;
|
|
/*
|
* According to the definition of ppm:
|
* host_clk = (1 + ppm / 1000000) * gadget_clk
|
* we can get:
|
* ppm = (host_clk - gadget_clk) * 1000000 / gadget_clk
|
*/
|
ppm = (fn_msec > clk_msec) ?
|
(fn_msec - clk_msec) * 1000000L / clk_msec :
|
-((clk_msec - fn_msec) * 1000000L / clk_msec);
|
|
ppm_sum = ppm_sum - ppms[cnt] + ppm;
|
ppms[cnt] = ppm;
|
|
dev_dbg(g_audio->device,
|
"frame %u msec %u ppm_calc %d ppm_avage(%d) %d\n",
|
fn_msec, clk_msec, ppm, CLK_PPM_GROUP_SIZE,
|
ppm_sum / CLK_PPM_GROUP_SIZE);
|
|
/*
|
* We calculate the average of ppm over a period of time. If the
|
* latest frame number is too far from the average, no event will
|
* be sent.
|
*/
|
if (abs(ppm_sum / CLK_PPM_GROUP_SIZE - ppm) < 3) {
|
ppm = ppm_sum > 0 ?
|
(ppm_sum + CLK_PPM_GROUP_SIZE / 2) / CLK_PPM_GROUP_SIZE :
|
(ppm_sum - CLK_PPM_GROUP_SIZE / 2) / CLK_PPM_GROUP_SIZE;
|
if (ppm != g_audio->params.ppm) {
|
g_audio->params.ppm = ppm;
|
g_audio->usb_state[SET_AUDIO_CLK] = true;
|
schedule_work(&g_audio->work);
|
}
|
}
|
|
out:
|
schedule_delayed_work(&g_audio->ppm_work, 1 * HZ);
|
}
|
|
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;
|
int i;
|
|
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;
|
|
g_audio->fn = kzalloc(sizeof(*g_audio->fn), GFP_KERNEL);
|
if (!g_audio->fn) {
|
err = -ENOMEM;
|
goto fail;
|
}
|
|
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;
|
|
strlcpy(pcm->name, pcm_name, sizeof(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);
|
|
strlcpy(card->driver, card_name, sizeof(card->driver));
|
strlcpy(card->shortname, card_name, sizeof(card->shortname));
|
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);
|
|
/* Add controls */
|
for (i = 0; i < ARRAY_SIZE(uac_pcm_controls); i++) {
|
err = snd_ctl_add(card,
|
snd_ctl_new1(&uac_pcm_controls[i], uac));
|
if (err < 0)
|
goto snd_fail;
|
}
|
|
err = snd_card_register(card);
|
if (err < 0)
|
goto snd_fail;
|
|
g_audio->device = device_create(audio_class, NULL, MKDEV(0, 0), NULL,
|
"%s", g_audio->uac->card->longname);
|
if (IS_ERR(g_audio->device)) {
|
err = PTR_ERR(g_audio->device);
|
goto snd_fail;
|
}
|
|
INIT_WORK(&g_audio->work, g_audio_work);
|
INIT_DELAYED_WORK(&g_audio->ppm_work, ppm_calculate_work);
|
ppm_calculate_work(&g_audio->ppm_work.work);
|
|
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);
|
kfree(g_audio->fn);
|
|
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;
|
|
cancel_work_sync(&g_audio->work);
|
cancel_delayed_work_sync(&g_audio->ppm_work);
|
device_destroy(g_audio->device->class, g_audio->device->devt);
|
g_audio->device = NULL;
|
|
uac = g_audio->uac;
|
card = uac->card;
|
if (card)
|
snd_card_free(card);
|
|
free_ep(&uac->c_prm, g_audio->out_ep);
|
free_ep(&uac->p_prm, g_audio->in_ep);
|
|
kfree(uac->p_prm.ureq);
|
kfree(uac->c_prm.ureq);
|
kfree(uac->p_prm.rbuf);
|
kfree(uac->c_prm.rbuf);
|
kfree(uac);
|
kfree(g_audio->fn);
|
}
|
EXPORT_SYMBOL_GPL(g_audio_cleanup);
|
|
static int __init u_audio_init(void)
|
{
|
int err = 0;
|
|
audio_class = class_create(THIS_MODULE, "u_audio");
|
if (IS_ERR(audio_class)) {
|
err = PTR_ERR(audio_class);
|
audio_class = NULL;
|
}
|
|
return err;
|
}
|
module_init(u_audio_init);
|
|
static void __exit u_audio_exit(void)
|
{
|
if (audio_class)
|
class_destroy(audio_class);
|
}
|
module_exit(u_audio_exit);
|
|
MODULE_LICENSE("GPL");
|
MODULE_DESCRIPTION("USB gadget \"ALSA sound card\" utilities");
|
MODULE_AUTHOR("Ruslan Bilovol");
|
