From: Linus Torvalds <torvalds@linux-foundation.org>
Date: Thu, 6 Jul 2017 17:56:51 +0000 (-0700)
Subject: Merge tag 'sound-4.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai... 
X-Git-Url: http://git.lede-project.org./?a=commitdiff_plain;h=920f2ecdf6c3b3526f60fbd38c68597953cad3ee;p=openwrt%2Fstaging%2Fblogic.git

Merge tag 'sound-4.13-rc1' of git://git./linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This development cycle resulted in a fair amount of changes in both
  core and driver sides. The most significant change in ALSA core is
  about PCM. Also the support of of-graph card and the new DAPM widget
  for DSP are noteworthy changes in ASoC core. And there're lots of
  small changes splat over the tree, as you can see in diffstat.

  Below are a few highlights:

  ALSA core:
   - Removal of set_fs() hackery from PCM core stuff, and the code
     reorganization / optimization thereafter
   - Improved support of PCM ack ops, and a new ABI for improved
     control/status mmap handling
   - Lots of constifications in various codes

  ASoC core:
   - The support of of-graph card, which may work as a better generic
     device for a replacement of simple-card
   - New widget types intended mainly for use with DSPs

  ASoC drivers:
   - New drivers for Allwinner V3s SoCs
   - Ensonic ES8316 codec support
   - More Intel SKL and KBL works
   - More device support for Intel SST Atom (mostly for cheap tablets
     and 2-in-1 devices)
   - Support for Rockchip PDM controllers
   - Support for STM32 I2S and S/PDIF controllers
   - Support for ZTE AUD96P22 codecs

  HD-audio:
   - Support of new Realtek codecs (ALC215/ALC285/ALC289), more quirks
     for HP and Dell machines
   - A few more fixes for i915 component binding"

* tag 'sound-4.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (418 commits)
  ALSA: hda - Fix unbalance of i915 module refcount
  ASoC: Intel: Skylake: Remove driver debugfs exit
  ASoC: Intel: Skylake: explicitly add the headers sst-dsp.h
  ALSA: hda/realtek - Remove GPIO_MASK
  ALSA: hda/realtek - Fix typo of pincfg for Dell quirk
  ALSA: pcm: add a documentation for tracepoints
  ALSA: atmel: ac97c: fix error return code in atmel_ac97c_probe()
  ALSA: x86: fix error return code in hdmi_lpe_audio_probe()
  ASoC: Intel: Skylake: Add support to read firmware registers
  ASoC: Intel: Skylake: Add sram address to sst_addr structure
  ASoC: Intel: Skylake: Debugfs facility to dump module config
  ASoC: Intel: Skylake: Add debugfs support
  ASoC: fix semicolon.cocci warnings
  ASoC: rt5645: Add quirk override by module option
  ASoC: rsnd: make arrays path and cmd_case static const
  ASoC: audio-graph-card: add widgets and routing for external amplifier support
  ASoC: audio-graph-card: update bindings for amplifier support
  ASoC: rt5665: calibration should be done before jack detection
  ASoC: rsnd: constify dev_pm_ops structures.
  ASoC: nau8825: change crosstalk-bypass property to bool type
  ...
---

920f2ecdf6c3b3526f60fbd38c68597953cad3ee
diff --cc drivers/usb/gadget/function/u_uac1_legacy.c
index 8aa76b4dc117,000000000000..fa4684a1c54c
mode 100644,000000..100644
--- a/drivers/usb/gadget/function/u_uac1_legacy.c
+++ b/drivers/usb/gadget/function/u_uac1_legacy.c
@@@ -1,315 -1,0 +1,310 @@@
 +/*
 + * u_uac1.c -- ALSA audio utilities for Gadget stack
 + *
 + * Copyright (C) 2008 Bryan Wu <cooloney@kernel.org>
 + * Copyright (C) 2008 Analog Devices, Inc
 + *
 + * Enter bugs at http://blackfin.uclinux.org/
 + *
 + * Licensed under the GPL-2 or later.
 + */
 +
 +#include <linux/kernel.h>
 +#include <linux/module.h>
 +#include <linux/slab.h>
 +#include <linux/device.h>
 +#include <linux/delay.h>
 +#include <linux/ctype.h>
 +#include <linux/random.h>
 +#include <linux/syscalls.h>
 +
 +#include "u_uac1_legacy.h"
 +
 +/*
 + * This component encapsulates the ALSA devices for USB audio gadget
 + */
 +
 +/*-------------------------------------------------------------------------*/
 +
 +/**
 + * Some ALSA internal helper functions
 + */
 +static int snd_interval_refine_set(struct snd_interval *i, unsigned int val)
 +{
 +	struct snd_interval t;
 +	t.empty = 0;
 +	t.min = t.max = val;
 +	t.openmin = t.openmax = 0;
 +	t.integer = 1;
 +	return snd_interval_refine(i, &t);
 +}
 +
 +static int _snd_pcm_hw_param_set(struct snd_pcm_hw_params *params,
 +				 snd_pcm_hw_param_t var, unsigned int val,
 +				 int dir)
 +{
 +	int changed;
 +	if (hw_is_mask(var)) {
 +		struct snd_mask *m = hw_param_mask(params, var);
 +		if (val == 0 && dir < 0) {
 +			changed = -EINVAL;
 +			snd_mask_none(m);
 +		} else {
 +			if (dir > 0)
 +				val++;
 +			else if (dir < 0)
 +				val--;
 +			changed = snd_mask_refine_set(
 +					hw_param_mask(params, var), val);
 +		}
 +	} else if (hw_is_interval(var)) {
 +		struct snd_interval *i = hw_param_interval(params, var);
 +		if (val == 0 && dir < 0) {
 +			changed = -EINVAL;
 +			snd_interval_none(i);
 +		} else if (dir == 0)
 +			changed = snd_interval_refine_set(i, val);
 +		else {
 +			struct snd_interval t;
 +			t.openmin = 1;
 +			t.openmax = 1;
 +			t.empty = 0;
 +			t.integer = 0;
 +			if (dir < 0) {
 +				t.min = val - 1;
 +				t.max = val;
 +			} else {
 +				t.min = val;
 +				t.max = val+1;
 +			}
 +			changed = snd_interval_refine(i, &t);
 +		}
 +	} else
 +		return -EINVAL;
 +	if (changed) {
 +		params->cmask |= 1 << var;
 +		params->rmask |= 1 << var;
 +	}
 +	return changed;
 +}
 +/*-------------------------------------------------------------------------*/
 +
 +/**
 + * Set default hardware params
 + */
 +static int playback_default_hw_params(struct gaudio_snd_dev *snd)
 +{
 +	struct snd_pcm_substream *substream = snd->substream;
 +	struct snd_pcm_hw_params *params;
 +	snd_pcm_sframes_t result;
 +
 +       /*
 +	* SNDRV_PCM_ACCESS_RW_INTERLEAVED,
 +	* SNDRV_PCM_FORMAT_S16_LE
 +	* CHANNELS: 2
 +	* RATE: 48000
 +	*/
 +	snd->access = SNDRV_PCM_ACCESS_RW_INTERLEAVED;
 +	snd->format = SNDRV_PCM_FORMAT_S16_LE;
 +	snd->channels = 2;
 +	snd->rate = 48000;
 +
 +	params = kzalloc(sizeof(*params), GFP_KERNEL);
 +	if (!params)
 +		return -ENOMEM;
 +
 +	_snd_pcm_hw_params_any(params);
 +	_snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_ACCESS,
 +			snd->access, 0);
 +	_snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_FORMAT,
 +			snd->format, 0);
 +	_snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_CHANNELS,
 +			snd->channels, 0);
 +	_snd_pcm_hw_param_set(params, SNDRV_PCM_HW_PARAM_RATE,
 +			snd->rate, 0);
 +
 +	snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL);
 +	snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, params);
 +
 +	result = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_PREPARE, NULL);
 +	if (result < 0) {
 +		ERROR(snd->card,
 +			"Preparing sound card failed: %d\n", (int)result);
 +		kfree(params);
 +		return result;
 +	}
 +
 +	/* Store the hardware parameters */
 +	snd->access = params_access(params);
 +	snd->format = params_format(params);
 +	snd->channels = params_channels(params);
 +	snd->rate = params_rate(params);
 +
 +	kfree(params);
 +
 +	INFO(snd->card,
 +		"Hardware params: access %x, format %x, channels %d, rate %d\n",
 +		snd->access, snd->format, snd->channels, snd->rate);
 +
 +	return 0;
 +}
 +
 +/**
 + * Playback audio buffer data by ALSA PCM device
 + */
 +size_t u_audio_playback(struct gaudio *card, void *buf, size_t count)
 +{
 +	struct gaudio_snd_dev	*snd = &card->playback;
 +	struct snd_pcm_substream *substream = snd->substream;
 +	struct snd_pcm_runtime *runtime = substream->runtime;
- 	mm_segment_t old_fs;
 +	ssize_t result;
 +	snd_pcm_sframes_t frames;
 +
 +try_again:
 +	if (runtime->status->state == SNDRV_PCM_STATE_XRUN ||
 +		runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
 +		result = snd_pcm_kernel_ioctl(substream,
 +				SNDRV_PCM_IOCTL_PREPARE, NULL);
 +		if (result < 0) {
 +			ERROR(card, "Preparing sound card failed: %d\n",
 +					(int)result);
 +			return result;
 +		}
 +	}
 +
 +	frames = bytes_to_frames(runtime, count);
- 	old_fs = get_fs();
- 	set_fs(KERNEL_DS);
- 	result = snd_pcm_lib_write(snd->substream, (void __user *)buf, frames);
++	result = snd_pcm_kernel_write(snd->substream, buf, frames);
 +	if (result != frames) {
 +		ERROR(card, "Playback error: %d\n", (int)result);
- 		set_fs(old_fs);
 +		goto try_again;
 +	}
- 	set_fs(old_fs);
 +
 +	return 0;
 +}
 +
 +int u_audio_get_playback_channels(struct gaudio *card)
 +{
 +	return card->playback.channels;
 +}
 +
 +int u_audio_get_playback_rate(struct gaudio *card)
 +{
 +	return card->playback.rate;
 +}
 +
 +/**
 + * Open ALSA PCM and control device files
 + * Initial the PCM or control device
 + */
 +static int gaudio_open_snd_dev(struct gaudio *card)
 +{
 +	struct snd_pcm_file *pcm_file;
 +	struct gaudio_snd_dev *snd;
 +	struct f_uac1_legacy_opts *opts;
 +	char *fn_play, *fn_cap, *fn_cntl;
 +
 +	opts = container_of(card->func.fi, struct f_uac1_legacy_opts,
 +			    func_inst);
 +	fn_play = opts->fn_play;
 +	fn_cap = opts->fn_cap;
 +	fn_cntl = opts->fn_cntl;
 +
 +	/* Open control device */
 +	snd = &card->control;
 +	snd->filp = filp_open(fn_cntl, O_RDWR, 0);
 +	if (IS_ERR(snd->filp)) {
 +		int ret = PTR_ERR(snd->filp);
 +		ERROR(card, "unable to open sound control device file: %s\n",
 +				fn_cntl);
 +		snd->filp = NULL;
 +		return ret;
 +	}
 +	snd->card = card;
 +
 +	/* Open PCM playback device and setup substream */
 +	snd = &card->playback;
 +	snd->filp = filp_open(fn_play, O_WRONLY, 0);
 +	if (IS_ERR(snd->filp)) {
 +		int ret = PTR_ERR(snd->filp);
 +
 +		ERROR(card, "No such PCM playback device: %s\n", fn_play);
 +		snd->filp = NULL;
 +		return ret;
 +	}
 +	pcm_file = snd->filp->private_data;
 +	snd->substream = pcm_file->substream;
 +	snd->card = card;
 +	playback_default_hw_params(snd);
 +
 +	/* Open PCM capture device and setup substream */
 +	snd = &card->capture;
 +	snd->filp = filp_open(fn_cap, O_RDONLY, 0);
 +	if (IS_ERR(snd->filp)) {
 +		ERROR(card, "No such PCM capture device: %s\n", fn_cap);
 +		snd->substream = NULL;
 +		snd->card = NULL;
 +		snd->filp = NULL;
 +	} else {
 +		pcm_file = snd->filp->private_data;
 +		snd->substream = pcm_file->substream;
 +		snd->card = card;
 +	}
 +
 +	return 0;
 +}
 +
 +/**
 + * Close ALSA PCM and control device files
 + */
 +static int gaudio_close_snd_dev(struct gaudio *gau)
 +{
 +	struct gaudio_snd_dev	*snd;
 +
 +	/* Close control device */
 +	snd = &gau->control;
 +	if (snd->filp)
 +		filp_close(snd->filp, NULL);
 +
 +	/* Close PCM playback device and setup substream */
 +	snd = &gau->playback;
 +	if (snd->filp)
 +		filp_close(snd->filp, NULL);
 +
 +	/* Close PCM capture device and setup substream */
 +	snd = &gau->capture;
 +	if (snd->filp)
 +		filp_close(snd->filp, NULL);
 +
 +	return 0;
 +}
 +
 +/**
 + * gaudio_setup - setup ALSA interface and preparing for USB transfer
 + *
 + * This sets up PCM, mixer or MIDI ALSA devices fore USB gadget using.
 + *
 + * Returns negative errno, or zero on success
 + */
 +int gaudio_setup(struct gaudio *card)
 +{
 +	int	ret;
 +
 +	ret = gaudio_open_snd_dev(card);
 +	if (ret)
 +		ERROR(card, "we need at least one control device\n");
 +
 +	return ret;
 +
 +}
 +
 +/**
 + * gaudio_cleanup - remove ALSA device interface
 + *
 + * This is called to free all resources allocated by @gaudio_setup().
 + */
 +void gaudio_cleanup(struct gaudio *the_card)
 +{
 +	if (the_card)
 +		gaudio_close_snd_dev(the_card);
 +}
 +