OpenCloudOS-Kernel/sound/pci/echoaudio/echoaudio.c

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// SPDX-License-Identifier: GPL-2.0-only
/*
* ALSA driver for Echoaudio soundcards.
* Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
* Copyright (C) 2020 Mark Hills <mark@xwax.org>
*/
#include <linux/module.h>
MODULE_AUTHOR("Giuliano Pochini <pochini@shiny.it>");
MODULE_LICENSE("GPL v2");
MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver");
MODULE_DEVICE_TABLE(pci, snd_echo_ids);
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for " ECHOCARD_NAME " soundcard.");
module_param_array(id, charp, NULL, 0444);
MODULE_PARM_DESC(id, "ID string for " ECHOCARD_NAME " soundcard.");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard.");
static const unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999};
static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1);
static int get_firmware(const struct firmware **fw_entry,
struct echoaudio *chip, const short fw_index)
{
int err;
char name[30];
#ifdef CONFIG_PM_SLEEP
if (chip->fw_cache[fw_index]) {
dev_dbg(chip->card->dev,
"firmware requested: %s is cached\n",
card_fw[fw_index].data);
*fw_entry = chip->fw_cache[fw_index];
return 0;
}
#endif
dev_dbg(chip->card->dev,
"firmware requested: %s\n", card_fw[fw_index].data);
snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data);
err = request_firmware(fw_entry, name, &chip->pci->dev);
if (err < 0)
dev_err(chip->card->dev,
"get_firmware(): Firmware not available (%d)\n", err);
#ifdef CONFIG_PM_SLEEP
else
chip->fw_cache[fw_index] = *fw_entry;
#endif
return err;
}
static void free_firmware(const struct firmware *fw_entry,
struct echoaudio *chip)
{
#ifdef CONFIG_PM_SLEEP
dev_dbg(chip->card->dev, "firmware not released (kept in cache)\n");
#else
release_firmware(fw_entry);
#endif
}
static void free_firmware_cache(struct echoaudio *chip)
{
#ifdef CONFIG_PM_SLEEP
int i;
for (i = 0; i < 8 ; i++)
if (chip->fw_cache[i]) {
release_firmware(chip->fw_cache[i]);
dev_dbg(chip->card->dev, "release_firmware(%d)\n", i);
}
#endif
}
/******************************************************************************
PCM interface
******************************************************************************/
static void audiopipe_free(struct snd_pcm_runtime *runtime)
{
struct audiopipe *pipe = runtime->private_data;
if (pipe->sgpage.area)
snd_dma_free_pages(&pipe->sgpage);
kfree(pipe);
}
static int hw_rule_capture_format_by_channels(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *c = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_mask fmt;
snd_mask_any(&fmt);
#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
/* >=2 channels cannot be S32_BE */
if (c->min == 2) {
fmt.bits[0] &= ~SNDRV_PCM_FMTBIT_S32_BE;
return snd_mask_refine(f, &fmt);
}
#endif
/* > 2 channels cannot be U8 and S32_BE */
if (c->min > 2) {
fmt.bits[0] &= ~(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_BE);
return snd_mask_refine(f, &fmt);
}
/* Mono is ok with any format */
return 0;
}
static int hw_rule_capture_channels_by_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *c = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_interval ch;
snd_interval_any(&ch);
/* S32_BE is mono (and stereo) only */
if (f->bits[0] == SNDRV_PCM_FMTBIT_S32_BE) {
ch.min = 1;
#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
ch.max = 2;
#else
ch.max = 1;
#endif
ch.integer = 1;
return snd_interval_refine(c, &ch);
}
/* U8 can be only mono or stereo */
if (f->bits[0] == SNDRV_PCM_FMTBIT_U8) {
ch.min = 1;
ch.max = 2;
ch.integer = 1;
return snd_interval_refine(c, &ch);
}
/* S16_LE, S24_3LE and S32_LE support any number of channels. */
return 0;
}
static int hw_rule_playback_format_by_channels(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *c = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_mask fmt;
u64 fmask;
snd_mask_any(&fmt);
fmask = fmt.bits[0] + ((u64)fmt.bits[1] << 32);
/* >2 channels must be S16_LE, S24_3LE or S32_LE */
if (c->min > 2) {
fmask &= SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE;
/* 1 channel must be S32_BE or S32_LE */
} else if (c->max == 1)
fmask &= SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE;
#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
/* 2 channels cannot be S32_BE */
else if (c->min == 2 && c->max == 2)
fmask &= ~SNDRV_PCM_FMTBIT_S32_BE;
#endif
else
return 0;
fmt.bits[0] &= (u32)fmask;
fmt.bits[1] &= (u32)(fmask >> 32);
return snd_mask_refine(f, &fmt);
}
static int hw_rule_playback_channels_by_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *c = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_interval ch;
u64 fmask;
snd_interval_any(&ch);
ch.integer = 1;
fmask = f->bits[0] + ((u64)f->bits[1] << 32);
/* S32_BE is mono (and stereo) only */
if (fmask == SNDRV_PCM_FMTBIT_S32_BE) {
ch.min = 1;
#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
ch.max = 2;
#else
ch.max = 1;
#endif
/* U8 is stereo only */
} else if (fmask == SNDRV_PCM_FMTBIT_U8)
ch.min = ch.max = 2;
/* S16_LE and S24_3LE must be at least stereo */
else if (!(fmask & ~(SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_3LE)))
ch.min = 2;
else
return 0;
return snd_interval_refine(c, &ch);
}
/* Since the sample rate is a global setting, do allow the user to change the
sample rate only if there is only one pcm device open. */
static int hw_rule_sample_rate(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct echoaudio *chip = rule->private;
struct snd_interval fixed;
int err;
mutex_lock(&chip->mode_mutex);
if (chip->can_set_rate) {
err = 0;
} else {
snd_interval_any(&fixed);
fixed.min = fixed.max = chip->sample_rate;
err = snd_interval_refine(rate, &fixed);
}
mutex_unlock(&chip->mode_mutex);
return err;
}
static int pcm_open(struct snd_pcm_substream *substream,
signed char max_channels)
{
struct echoaudio *chip;
struct snd_pcm_runtime *runtime;
struct audiopipe *pipe;
int err, i;
if (max_channels <= 0)
return -EAGAIN;
chip = snd_pcm_substream_chip(substream);
runtime = substream->runtime;
pipe = kzalloc(sizeof(struct audiopipe), GFP_KERNEL);
if (!pipe)
return -ENOMEM;
pipe->index = -1; /* Not configured yet */
/* Set up hw capabilities and contraints */
memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware));
dev_dbg(chip->card->dev, "max_channels=%d\n", max_channels);
pipe->constr.list = channels_list;
pipe->constr.mask = 0;
for (i = 0; channels_list[i] <= max_channels; i++);
pipe->constr.count = i;
if (pipe->hw.channels_max > max_channels)
pipe->hw.channels_max = max_channels;
if (chip->digital_mode == DIGITAL_MODE_ADAT) {
pipe->hw.rate_max = 48000;
pipe->hw.rates &= SNDRV_PCM_RATE_8000_48000;
}
runtime->hw = pipe->hw;
runtime->private_data = pipe;
runtime->private_free = audiopipe_free;
snd_pcm_set_sync(substream);
/* Only mono and any even number of channels are allowed */
err = snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
&pipe->constr);
if (err < 0)
return err;
/* All periods should have the same size */
err = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (err < 0)
return err;
/* The hw accesses memory in chunks 32 frames long and they should be
32-bytes-aligned. It's not a requirement, but it seems that IRQs are
generated with a resolution of 32 frames. Thus we need the following */
err = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32);
if (err < 0)
return err;
err = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 32);
if (err < 0)
return err;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
hw_rule_sample_rate, chip,
SNDRV_PCM_HW_PARAM_RATE, -1);
if (err < 0)
return err;
/* Allocate a page for the scatter-gather list */
err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
&chip->pci->dev,
PAGE_SIZE, &pipe->sgpage);
if (err < 0) {
dev_err(chip->card->dev, "s-g list allocation failed\n");
return err;
}
/*
* Sole ownership required to set the rate
*/
dev_dbg(chip->card->dev, "pcm_open opencount=%d can_set_rate=%d, rate_set=%d",
chip->opencount, chip->can_set_rate, chip->rate_set);
chip->opencount++;
if (chip->opencount > 1 && chip->rate_set)
chip->can_set_rate = 0;
return 0;
}
static int pcm_analog_in_open(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
int err;
err = pcm_open(substream,
num_analog_busses_in(chip) - substream->number);
if (err < 0)
return err;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_capture_channels_by_format, NULL,
SNDRV_PCM_HW_PARAM_FORMAT, -1);
if (err < 0)
return err;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_capture_format_by_channels, NULL,
SNDRV_PCM_HW_PARAM_CHANNELS, -1);
if (err < 0)
return err;
return 0;
}
static int pcm_analog_out_open(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
int max_channels, err;
#ifdef ECHOCARD_HAS_VMIXER
max_channels = num_pipes_out(chip);
#else
max_channels = num_analog_busses_out(chip);
#endif
err = pcm_open(substream, max_channels - substream->number);
if (err < 0)
return err;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_playback_channels_by_format,
NULL,
SNDRV_PCM_HW_PARAM_FORMAT, -1);
if (err < 0)
return err;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_playback_format_by_channels,
NULL,
SNDRV_PCM_HW_PARAM_CHANNELS, -1);
if (err < 0)
return err;
return 0;
}
#ifdef ECHOCARD_HAS_DIGITAL_IO
static int pcm_digital_in_open(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
int err, max_channels;
max_channels = num_digital_busses_in(chip) - substream->number;
mutex_lock(&chip->mode_mutex);
if (chip->digital_mode == DIGITAL_MODE_ADAT)
err = pcm_open(substream, max_channels);
else /* If the card has ADAT, subtract the 6 channels
* that S/PDIF doesn't have
*/
err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
if (err < 0)
goto din_exit;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_capture_channels_by_format, NULL,
SNDRV_PCM_HW_PARAM_FORMAT, -1);
if (err < 0)
goto din_exit;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_capture_format_by_channels, NULL,
SNDRV_PCM_HW_PARAM_CHANNELS, -1);
if (err < 0)
goto din_exit;
din_exit:
mutex_unlock(&chip->mode_mutex);
return err;
}
#ifndef ECHOCARD_HAS_VMIXER /* See the note in snd_echo_new_pcm() */
static int pcm_digital_out_open(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
int err, max_channels;
max_channels = num_digital_busses_out(chip) - substream->number;
mutex_lock(&chip->mode_mutex);
if (chip->digital_mode == DIGITAL_MODE_ADAT)
err = pcm_open(substream, max_channels);
else /* If the card has ADAT, subtract the 6 channels
* that S/PDIF doesn't have
*/
err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
if (err < 0)
goto dout_exit;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_playback_channels_by_format,
NULL, SNDRV_PCM_HW_PARAM_FORMAT,
-1);
if (err < 0)
goto dout_exit;
err = snd_pcm_hw_rule_add(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_playback_format_by_channels,
NULL, SNDRV_PCM_HW_PARAM_CHANNELS,
-1);
if (err < 0)
goto dout_exit;
dout_exit:
mutex_unlock(&chip->mode_mutex);
return err;
}
#endif /* !ECHOCARD_HAS_VMIXER */
#endif /* ECHOCARD_HAS_DIGITAL_IO */
static int pcm_close(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
/* Nothing to do here. Audio is already off and pipe will be
* freed by its callback
*/
mutex_lock(&chip->mode_mutex);
dev_dbg(chip->card->dev, "pcm_open opencount=%d can_set_rate=%d, rate_set=%d",
chip->opencount, chip->can_set_rate, chip->rate_set);
chip->opencount--;
switch (chip->opencount) {
case 1:
chip->can_set_rate = 1;
break;
case 0:
chip->rate_set = 0;
break;
}
mutex_unlock(&chip->mode_mutex);
return 0;
}
/* Channel allocation and scatter-gather list setup */
static int init_engine(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params,
int pipe_index, int interleave)
{
struct echoaudio *chip;
int err, per, rest, page, edge, offs;
struct audiopipe *pipe;
chip = snd_pcm_substream_chip(substream);
pipe = (struct audiopipe *) substream->runtime->private_data;
/* Sets up che hardware. If it's already initialized, reset and
* redo with the new parameters
*/
spin_lock_irq(&chip->lock);
if (pipe->index >= 0) {
dev_dbg(chip->card->dev, "hwp_ie free(%d)\n", pipe->index);
err = free_pipes(chip, pipe);
snd_BUG_ON(err);
chip->substream[pipe->index] = NULL;
}
err = allocate_pipes(chip, pipe, pipe_index, interleave);
if (err < 0) {
spin_unlock_irq(&chip->lock);
dev_err(chip->card->dev, "allocate_pipes(%d) err=%d\n",
pipe_index, err);
return err;
}
spin_unlock_irq(&chip->lock);
dev_dbg(chip->card->dev, "allocate_pipes()=%d\n", pipe_index);
dev_dbg(chip->card->dev,
"pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n",
params_buffer_bytes(hw_params), params_periods(hw_params),
params_period_bytes(hw_params));
sglist_init(chip, pipe);
edge = PAGE_SIZE;
for (offs = page = per = 0; offs < params_buffer_bytes(hw_params);
per++) {
rest = params_period_bytes(hw_params);
if (offs + rest > params_buffer_bytes(hw_params))
rest = params_buffer_bytes(hw_params) - offs;
while (rest) {
dma_addr_t addr;
addr = snd_pcm_sgbuf_get_addr(substream, offs);
if (rest <= edge - offs) {
sglist_add_mapping(chip, pipe, addr, rest);
sglist_add_irq(chip, pipe);
offs += rest;
rest = 0;
} else {
sglist_add_mapping(chip, pipe, addr,
edge - offs);
rest -= edge - offs;
offs = edge;
}
if (offs == edge) {
edge += PAGE_SIZE;
page++;
}
}
}
/* Close the ring buffer */
sglist_wrap(chip, pipe);
/* This stuff is used by the irq handler, so it must be
* initialized before chip->substream
*/
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
pipe->last_period = 0;
pipe->last_counter = 0;
pipe->position = 0;
smp_wmb();
chip->substream[pipe_index] = substream;
chip->rate_set = 1;
spin_lock_irq(&chip->lock);
set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den);
spin_unlock_irq(&chip->lock);
return 0;
}
static int pcm_analog_in_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
return init_engine(substream, hw_params, px_analog_in(chip) +
substream->number, params_channels(hw_params));
}
static int pcm_analog_out_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
return init_engine(substream, hw_params, substream->number,
params_channels(hw_params));
}
#ifdef ECHOCARD_HAS_DIGITAL_IO
static int pcm_digital_in_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
return init_engine(substream, hw_params, px_digital_in(chip) +
substream->number, params_channels(hw_params));
}
#ifndef ECHOCARD_HAS_VMIXER /* See the note in snd_echo_new_pcm() */
static int pcm_digital_out_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
return init_engine(substream, hw_params, px_digital_out(chip) +
substream->number, params_channels(hw_params));
}
#endif /* !ECHOCARD_HAS_VMIXER */
#endif /* ECHOCARD_HAS_DIGITAL_IO */
static int pcm_hw_free(struct snd_pcm_substream *substream)
{
struct echoaudio *chip;
struct audiopipe *pipe;
chip = snd_pcm_substream_chip(substream);
pipe = (struct audiopipe *) substream->runtime->private_data;
spin_lock_irq(&chip->lock);
if (pipe->index >= 0) {
dev_dbg(chip->card->dev, "pcm_hw_free(%d)\n", pipe->index);
free_pipes(chip, pipe);
chip->substream[pipe->index] = NULL;
pipe->index = -1;
}
spin_unlock_irq(&chip->lock);
return 0;
}
static int pcm_prepare(struct snd_pcm_substream *substream)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct audioformat format;
int pipe_index = ((struct audiopipe *)runtime->private_data)->index;
dev_dbg(chip->card->dev, "Prepare rate=%d format=%d channels=%d\n",
runtime->rate, runtime->format, runtime->channels);
format.interleave = runtime->channels;
format.data_are_bigendian = 0;
format.mono_to_stereo = 0;
switch (runtime->format) {
case SNDRV_PCM_FORMAT_U8:
format.bits_per_sample = 8;
break;
case SNDRV_PCM_FORMAT_S16_LE:
format.bits_per_sample = 16;
break;
case SNDRV_PCM_FORMAT_S24_3LE:
format.bits_per_sample = 24;
break;
case SNDRV_PCM_FORMAT_S32_BE:
format.data_are_bigendian = 1;
fallthrough;
case SNDRV_PCM_FORMAT_S32_LE:
format.bits_per_sample = 32;
break;
default:
dev_err(chip->card->dev,
"Prepare error: unsupported format %d\n",
runtime->format);
return -EINVAL;
}
if (snd_BUG_ON(pipe_index >= px_num(chip)))
return -EINVAL;
/*
* We passed checks we can do independently; now take
* exclusive control
*/
spin_lock_irq(&chip->lock);
if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index))) {
spin_unlock_irq(&chip->lock);
return -EINVAL;
}
set_audio_format(chip, pipe_index, &format);
spin_unlock_irq(&chip->lock);
return 0;
}
static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct echoaudio *chip = snd_pcm_substream_chip(substream);
struct audiopipe *pipe;
int i, err;
u32 channelmask = 0;
struct snd_pcm_substream *s;
snd_pcm_group_for_each_entry(s, substream) {
for (i = 0; i < DSP_MAXPIPES; i++) {
if (s == chip->substream[i]) {
channelmask |= 1 << i;
snd_pcm_trigger_done(s, substream);
}
}
}
spin_lock(&chip->lock);
switch (cmd) {
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
for (i = 0; i < DSP_MAXPIPES; i++) {
if (channelmask & (1 << i)) {
pipe = chip->substream[i]->runtime->private_data;
switch (pipe->state) {
case PIPE_STATE_STOPPED:
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
pipe->last_period = 0;
pipe->last_counter = 0;
pipe->position = 0;
*pipe->dma_counter = 0;
fallthrough;
case PIPE_STATE_PAUSED:
pipe->state = PIPE_STATE_STARTED;
break;
case PIPE_STATE_STARTED:
break;
}
}
}
err = start_transport(chip, channelmask,
chip->pipe_cyclic_mask);
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_STOP:
for (i = 0; i < DSP_MAXPIPES; i++) {
if (channelmask & (1 << i)) {
pipe = chip->substream[i]->runtime->private_data;
pipe->state = PIPE_STATE_STOPPED;
}
}
err = stop_transport(chip, channelmask);
break;
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
for (i = 0; i < DSP_MAXPIPES; i++) {
if (channelmask & (1 << i)) {
pipe = chip->substream[i]->runtime->private_data;
pipe->state = PIPE_STATE_PAUSED;
}
}
err = pause_transport(chip, channelmask);
break;
default:
err = -EINVAL;
}
spin_unlock(&chip->lock);
return err;
}
static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct audiopipe *pipe = runtime->private_data;
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
u32 counter, step;
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
/*
* IRQ handling runs concurrently. Do not share tracking of
* counter with it, which would race or require locking
*/
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
counter = le32_to_cpu(*pipe->dma_counter); /* presumed atomic */
step = counter - pipe->last_counter; /* handles wrapping */
pipe->last_counter = counter;
/* counter doesn't neccessarily wrap on a multiple of
* buffer_size, so can't derive the position; must
* accumulate */
pipe->position += step;
pipe->position %= frames_to_bytes(runtime, runtime->buffer_size); /* wrap */
return bytes_to_frames(runtime, pipe->position);
}
/* pcm *_ops structures */
static const struct snd_pcm_ops analog_playback_ops = {
.open = pcm_analog_out_open,
.close = pcm_close,
.hw_params = pcm_analog_out_hw_params,
.hw_free = pcm_hw_free,
.prepare = pcm_prepare,
.trigger = pcm_trigger,
.pointer = pcm_pointer,
};
static const struct snd_pcm_ops analog_capture_ops = {
.open = pcm_analog_in_open,
.close = pcm_close,
.hw_params = pcm_analog_in_hw_params,
.hw_free = pcm_hw_free,
.prepare = pcm_prepare,
.trigger = pcm_trigger,
.pointer = pcm_pointer,
};
#ifdef ECHOCARD_HAS_DIGITAL_IO
#ifndef ECHOCARD_HAS_VMIXER
static const struct snd_pcm_ops digital_playback_ops = {
.open = pcm_digital_out_open,
.close = pcm_close,
.hw_params = pcm_digital_out_hw_params,
.hw_free = pcm_hw_free,
.prepare = pcm_prepare,
.trigger = pcm_trigger,
.pointer = pcm_pointer,
};
#endif /* !ECHOCARD_HAS_VMIXER */
static const struct snd_pcm_ops digital_capture_ops = {
.open = pcm_digital_in_open,
.close = pcm_close,
.hw_params = pcm_digital_in_hw_params,
.hw_free = pcm_hw_free,
.prepare = pcm_prepare,
.trigger = pcm_trigger,
.pointer = pcm_pointer,
};
#endif /* ECHOCARD_HAS_DIGITAL_IO */
/* Preallocate memory only for the first substream because it's the most
* used one
*/
static void snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev)
{
struct snd_pcm_substream *ss;
int stream;
for (stream = 0; stream < 2; stream++)
for (ss = pcm->streams[stream].substream; ss; ss = ss->next)
snd_pcm_set_managed_buffer(ss, SNDRV_DMA_TYPE_DEV_SG,
dev,
ss->number ? 0 : 128<<10,
256<<10);
}
/*<--snd_echo_probe() */
static int snd_echo_new_pcm(struct echoaudio *chip)
{
struct snd_pcm *pcm;
int err;
#ifdef ECHOCARD_HAS_VMIXER
/* This card has a Vmixer, that is there is no direct mapping from PCM
streams to physical outputs. The user can mix the streams as he wishes
via control interface and it's possible to send any stream to any
output, thus it makes no sense to keep analog and digital outputs
separated */
/* PCM#0 Virtual outputs and analog inputs */
err = snd_pcm_new(chip->card, "PCM", 0, num_pipes_out(chip),
num_analog_busses_in(chip), &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
chip->analog_pcm = pcm;
strcpy(pcm->name, chip->card->shortname);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
snd_echo_preallocate_pages(pcm, &chip->pci->dev);
#ifdef ECHOCARD_HAS_DIGITAL_IO
/* PCM#1 Digital inputs, no outputs */
err = snd_pcm_new(chip->card, "Digital PCM", 1, 0,
num_digital_busses_in(chip), &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
chip->digital_pcm = pcm;
strcpy(pcm->name, chip->card->shortname);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
snd_echo_preallocate_pages(pcm, &chip->pci->dev);
#endif /* ECHOCARD_HAS_DIGITAL_IO */
#else /* ECHOCARD_HAS_VMIXER */
/* The card can manage substreams formed by analog and digital channels
at the same time, but I prefer to keep analog and digital channels
separated, because that mixed thing is confusing and useless. So we
register two PCM devices: */
/* PCM#0 Analog i/o */
err = snd_pcm_new(chip->card, "Analog PCM", 0,
num_analog_busses_out(chip),
num_analog_busses_in(chip), &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
chip->analog_pcm = pcm;
strcpy(pcm->name, chip->card->shortname);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
snd_echo_preallocate_pages(pcm, &chip->pci->dev);
#ifdef ECHOCARD_HAS_DIGITAL_IO
/* PCM#1 Digital i/o */
err = snd_pcm_new(chip->card, "Digital PCM", 1,
num_digital_busses_out(chip),
num_digital_busses_in(chip), &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
chip->digital_pcm = pcm;
strcpy(pcm->name, chip->card->shortname);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &digital_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
snd_echo_preallocate_pages(pcm, &chip->pci->dev);
#endif /* ECHOCARD_HAS_DIGITAL_IO */
#endif /* ECHOCARD_HAS_VMIXER */
return 0;
}
/******************************************************************************
Control interface
******************************************************************************/
#if !defined(ECHOCARD_HAS_VMIXER) || defined(ECHOCARD_HAS_LINE_OUT_GAIN)
/******************* PCM output volume *******************/
static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = num_busses_out(chip);
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = ECHOGAIN_MAXOUT;
return 0;
}
static int snd_echo_output_gain_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c;
chip = snd_kcontrol_chip(kcontrol);
for (c = 0; c < num_busses_out(chip); c++)
ucontrol->value.integer.value[c] = chip->output_gain[c];
return 0;
}
static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c, changed, gain;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
spin_lock_irq(&chip->lock);
for (c = 0; c < num_busses_out(chip); c++) {
gain = ucontrol->value.integer.value[c];
/* Ignore out of range values */
if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
continue;
if (chip->output_gain[c] != gain) {
set_output_gain(chip, c, gain);
changed = 1;
}
}
if (changed)
update_output_line_level(chip);
spin_unlock_irq(&chip->lock);
return changed;
}
#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
/* On the Mia this one controls the line-out volume */
static const struct snd_kcontrol_new snd_echo_line_output_gain = {
.name = "Line Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
SNDRV_CTL_ELEM_ACCESS_TLV_READ,
.info = snd_echo_output_gain_info,
.get = snd_echo_output_gain_get,
.put = snd_echo_output_gain_put,
.tlv = {.p = db_scale_output_gain},
};
#else
static const struct snd_kcontrol_new snd_echo_pcm_output_gain = {
.name = "PCM Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
.info = snd_echo_output_gain_info,
.get = snd_echo_output_gain_get,
.put = snd_echo_output_gain_put,
.tlv = {.p = db_scale_output_gain},
};
#endif
#endif /* !ECHOCARD_HAS_VMIXER || ECHOCARD_HAS_LINE_OUT_GAIN */
#ifdef ECHOCARD_HAS_INPUT_GAIN
/******************* Analog input volume *******************/
static int snd_echo_input_gain_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = num_analog_busses_in(chip);
uinfo->value.integer.min = ECHOGAIN_MININP;
uinfo->value.integer.max = ECHOGAIN_MAXINP;
return 0;
}
static int snd_echo_input_gain_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c;
chip = snd_kcontrol_chip(kcontrol);
for (c = 0; c < num_analog_busses_in(chip); c++)
ucontrol->value.integer.value[c] = chip->input_gain[c];
return 0;
}
static int snd_echo_input_gain_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c, gain, changed;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
spin_lock_irq(&chip->lock);
for (c = 0; c < num_analog_busses_in(chip); c++) {
gain = ucontrol->value.integer.value[c];
/* Ignore out of range values */
if (gain < ECHOGAIN_MININP || gain > ECHOGAIN_MAXINP)
continue;
if (chip->input_gain[c] != gain) {
set_input_gain(chip, c, gain);
changed = 1;
}
}
if (changed)
update_input_line_level(chip);
spin_unlock_irq(&chip->lock);
return changed;
}
static const DECLARE_TLV_DB_SCALE(db_scale_input_gain, -2500, 50, 0);
static const struct snd_kcontrol_new snd_echo_line_input_gain = {
.name = "Line Capture Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
.info = snd_echo_input_gain_info,
.get = snd_echo_input_gain_get,
.put = snd_echo_input_gain_put,
.tlv = {.p = db_scale_input_gain},
};
#endif /* ECHOCARD_HAS_INPUT_GAIN */
#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
/************ Analog output nominal level (+4dBu / -10dBV) ***************/
static int snd_echo_output_nominal_info (struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = num_analog_busses_out(chip);
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int snd_echo_output_nominal_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c;
chip = snd_kcontrol_chip(kcontrol);
for (c = 0; c < num_analog_busses_out(chip); c++)
ucontrol->value.integer.value[c] = chip->nominal_level[c];
return 0;
}
static int snd_echo_output_nominal_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c, changed;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
spin_lock_irq(&chip->lock);
for (c = 0; c < num_analog_busses_out(chip); c++) {
if (chip->nominal_level[c] != ucontrol->value.integer.value[c]) {
set_nominal_level(chip, c,
ucontrol->value.integer.value[c]);
changed = 1;
}
}
if (changed)
update_output_line_level(chip);
spin_unlock_irq(&chip->lock);
return changed;
}
static const struct snd_kcontrol_new snd_echo_output_nominal_level = {
.name = "Line Playback Switch (-10dBV)",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.info = snd_echo_output_nominal_info,
.get = snd_echo_output_nominal_get,
.put = snd_echo_output_nominal_put,
};
#endif /* ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL */
#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
/*************** Analog input nominal level (+4dBu / -10dBV) ***************/
static int snd_echo_input_nominal_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = num_analog_busses_in(chip);
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int snd_echo_input_nominal_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c;
chip = snd_kcontrol_chip(kcontrol);
for (c = 0; c < num_analog_busses_in(chip); c++)
ucontrol->value.integer.value[c] =
chip->nominal_level[bx_analog_in(chip) + c];
return 0;
}
static int snd_echo_input_nominal_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int c, changed;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
spin_lock_irq(&chip->lock);
for (c = 0; c < num_analog_busses_in(chip); c++) {
if (chip->nominal_level[bx_analog_in(chip) + c] !=
ucontrol->value.integer.value[c]) {
set_nominal_level(chip, bx_analog_in(chip) + c,
ucontrol->value.integer.value[c]);
changed = 1;
}
}
if (changed)
update_output_line_level(chip); /* "Output" is not a mistake
* here.
*/
spin_unlock_irq(&chip->lock);
return changed;
}
static const struct snd_kcontrol_new snd_echo_intput_nominal_level = {
.name = "Line Capture Switch (-10dBV)",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.info = snd_echo_input_nominal_info,
.get = snd_echo_input_nominal_get,
.put = snd_echo_input_nominal_put,
};
#endif /* ECHOCARD_HAS_INPUT_NOMINAL_LEVEL */
#ifdef ECHOCARD_HAS_MONITOR
/******************* Monitor mixer *******************/
static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members" This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark@xwax.org> Reported-by: S. Christian Collins <s.chriscollins@gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable@vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-09-26 08:11:49 +08:00
uinfo->count = 1;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = ECHOGAIN_MAXOUT;
return 0;
}
static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
unsigned int out = ucontrol->id.index / num_busses_in(chip);
unsigned int in = ucontrol->id.index % num_busses_in(chip);
if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS)
return -EINVAL;
ucontrol->value.integer.value[0] = chip->monitor_gain[out][in];
return 0;
}
static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int changed, gain;
unsigned int out, in;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
out = ucontrol->id.index / num_busses_in(chip);
in = ucontrol->id.index % num_busses_in(chip);
if (out >= ECHO_MAXAUDIOOUTPUTS || in >= ECHO_MAXAUDIOINPUTS)
return -EINVAL;
gain = ucontrol->value.integer.value[0];
if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
return -EINVAL;
if (chip->monitor_gain[out][in] != gain) {
spin_lock_irq(&chip->lock);
set_monitor_gain(chip, out, in, gain);
update_output_line_level(chip);
spin_unlock_irq(&chip->lock);
changed = 1;
}
return changed;
}
static struct snd_kcontrol_new snd_echo_monitor_mixer = {
.name = "Monitor Mixer Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
.info = snd_echo_mixer_info,
.get = snd_echo_mixer_get,
.put = snd_echo_mixer_put,
.tlv = {.p = db_scale_output_gain},
};
#endif /* ECHOCARD_HAS_MONITOR */
#ifdef ECHOCARD_HAS_VMIXER
/******************* Vmixer *******************/
static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members" This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark@xwax.org> Reported-by: S. Christian Collins <s.chriscollins@gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable@vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-09-26 08:11:49 +08:00
uinfo->count = 1;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = ECHOGAIN_MAXOUT;
return 0;
}
static int snd_echo_vmixer_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] =
chip->vmixer_gain[ucontrol->id.index / num_pipes_out(chip)]
[ucontrol->id.index % num_pipes_out(chip)];
return 0;
}
static int snd_echo_vmixer_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int gain, changed;
short vch, out;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
out = ucontrol->id.index / num_pipes_out(chip);
vch = ucontrol->id.index % num_pipes_out(chip);
gain = ucontrol->value.integer.value[0];
if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
return -EINVAL;
if (chip->vmixer_gain[out][vch] != ucontrol->value.integer.value[0]) {
spin_lock_irq(&chip->lock);
set_vmixer_gain(chip, out, vch, ucontrol->value.integer.value[0]);
update_vmixer_level(chip);
spin_unlock_irq(&chip->lock);
changed = 1;
}
return changed;
}
static struct snd_kcontrol_new snd_echo_vmixer = {
.name = "VMixer Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
.info = snd_echo_vmixer_info,
.get = snd_echo_vmixer_get,
.put = snd_echo_vmixer_put,
.tlv = {.p = db_scale_output_gain},
};
#endif /* ECHOCARD_HAS_VMIXER */
#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
/******************* Digital mode switch *******************/
static int snd_echo_digital_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const names[4] = {
"S/PDIF Coaxial", "S/PDIF Optical", "ADAT Optical",
"S/PDIF Cdrom"
};
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
return snd_ctl_enum_info(uinfo, 1, chip->num_digital_modes, names);
}
static int snd_echo_digital_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int i, mode;
chip = snd_kcontrol_chip(kcontrol);
mode = chip->digital_mode;
for (i = chip->num_digital_modes - 1; i >= 0; i--)
if (mode == chip->digital_mode_list[i]) {
ucontrol->value.enumerated.item[0] = i;
break;
}
return 0;
}
static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int changed;
unsigned short emode, dmode;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
emode = ucontrol->value.enumerated.item[0];
if (emode >= chip->num_digital_modes)
return -EINVAL;
dmode = chip->digital_mode_list[emode];
if (dmode != chip->digital_mode) {
/* mode_mutex is required to make this operation atomic wrt
pcm_digital_*_open() and set_input_clock() functions. */
mutex_lock(&chip->mode_mutex);
/* Do not allow the user to change the digital mode when a pcm
device is open because it also changes the number of channels
and the allowed sample rates */
if (chip->opencount) {
changed = -EAGAIN;
} else {
changed = set_digital_mode(chip, dmode);
/* If we had to change the clock source, report it */
if (changed > 0 && chip->clock_src_ctl) {
snd_ctl_notify(chip->card,
SNDRV_CTL_EVENT_MASK_VALUE,
&chip->clock_src_ctl->id);
dev_dbg(chip->card->dev,
"SDM() =%d\n", changed);
}
if (changed >= 0)
changed = 1; /* No errors */
}
mutex_unlock(&chip->mode_mutex);
}
return changed;
}
static const struct snd_kcontrol_new snd_echo_digital_mode_switch = {
.name = "Digital mode Switch",
.iface = SNDRV_CTL_ELEM_IFACE_CARD,
.info = snd_echo_digital_mode_info,
.get = snd_echo_digital_mode_get,
.put = snd_echo_digital_mode_put,
};
#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
#ifdef ECHOCARD_HAS_DIGITAL_IO
/******************* S/PDIF mode switch *******************/
static int snd_echo_spdif_mode_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const names[2] = {"Consumer", "Professional"};
return snd_ctl_enum_info(uinfo, 1, 2, names);
}
static int snd_echo_spdif_mode_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = !!chip->professional_spdif;
return 0;
}
static int snd_echo_spdif_mode_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int mode;
chip = snd_kcontrol_chip(kcontrol);
mode = !!ucontrol->value.enumerated.item[0];
if (mode != chip->professional_spdif) {
spin_lock_irq(&chip->lock);
set_professional_spdif(chip, mode);
spin_unlock_irq(&chip->lock);
return 1;
}
return 0;
}
static const struct snd_kcontrol_new snd_echo_spdif_mode_switch = {
.name = "S/PDIF mode Switch",
.iface = SNDRV_CTL_ELEM_IFACE_CARD,
.info = snd_echo_spdif_mode_info,
.get = snd_echo_spdif_mode_get,
.put = snd_echo_spdif_mode_put,
};
#endif /* ECHOCARD_HAS_DIGITAL_IO */
#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
/******************* Select input clock source *******************/
static int snd_echo_clock_source_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static const char * const names[8] = {
"Internal", "Word", "Super", "S/PDIF", "ADAT", "ESync",
"ESync96", "MTC"
};
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
return snd_ctl_enum_info(uinfo, 1, chip->num_clock_sources, names);
}
static int snd_echo_clock_source_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int i, clock;
chip = snd_kcontrol_chip(kcontrol);
clock = chip->input_clock;
for (i = 0; i < chip->num_clock_sources; i++)
if (clock == chip->clock_source_list[i])
ucontrol->value.enumerated.item[0] = i;
return 0;
}
static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int changed;
unsigned int eclock, dclock;
changed = 0;
chip = snd_kcontrol_chip(kcontrol);
eclock = ucontrol->value.enumerated.item[0];
if (eclock >= chip->input_clock_types)
return -EINVAL;
dclock = chip->clock_source_list[eclock];
if (chip->input_clock != dclock) {
mutex_lock(&chip->mode_mutex);
spin_lock_irq(&chip->lock);
changed = set_input_clock(chip, dclock);
if (!changed)
changed = 1; /* no errors */
spin_unlock_irq(&chip->lock);
mutex_unlock(&chip->mode_mutex);
}
if (changed < 0)
dev_dbg(chip->card->dev,
"seticlk val%d err 0x%x\n", dclock, changed);
return changed;
}
static const struct snd_kcontrol_new snd_echo_clock_source_switch = {
.name = "Sample Clock Source",
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.info = snd_echo_clock_source_info,
.get = snd_echo_clock_source_get,
.put = snd_echo_clock_source_put,
};
#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
#ifdef ECHOCARD_HAS_PHANTOM_POWER
/******************* Phantom power switch *******************/
#define snd_echo_phantom_power_info snd_ctl_boolean_mono_info
static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] = chip->phantom_power;
return 0;
}
static int snd_echo_phantom_power_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
int power, changed = 0;
power = !!ucontrol->value.integer.value[0];
if (chip->phantom_power != power) {
spin_lock_irq(&chip->lock);
changed = set_phantom_power(chip, power);
spin_unlock_irq(&chip->lock);
if (changed == 0)
changed = 1; /* no errors */
}
return changed;
}
static const struct snd_kcontrol_new snd_echo_phantom_power_switch = {
.name = "Phantom power Switch",
.iface = SNDRV_CTL_ELEM_IFACE_CARD,
.info = snd_echo_phantom_power_info,
.get = snd_echo_phantom_power_get,
.put = snd_echo_phantom_power_put,
};
#endif /* ECHOCARD_HAS_PHANTOM_POWER */
#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
/******************* Digital input automute switch *******************/
#define snd_echo_automute_info snd_ctl_boolean_mono_info
static int snd_echo_automute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] = chip->digital_in_automute;
return 0;
}
static int snd_echo_automute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
int automute, changed = 0;
automute = !!ucontrol->value.integer.value[0];
if (chip->digital_in_automute != automute) {
spin_lock_irq(&chip->lock);
changed = set_input_auto_mute(chip, automute);
spin_unlock_irq(&chip->lock);
if (changed == 0)
changed = 1; /* no errors */
}
return changed;
}
static const struct snd_kcontrol_new snd_echo_automute_switch = {
.name = "Digital Capture Switch (automute)",
.iface = SNDRV_CTL_ELEM_IFACE_CARD,
.info = snd_echo_automute_info,
.get = snd_echo_automute_get,
.put = snd_echo_automute_put,
};
#endif /* ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE */
/******************* VU-meters switch *******************/
#define snd_echo_vumeters_switch_info snd_ctl_boolean_mono_info
static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
spin_lock_irq(&chip->lock);
set_meters_on(chip, ucontrol->value.integer.value[0]);
spin_unlock_irq(&chip->lock);
return 1;
}
static const struct snd_kcontrol_new snd_echo_vumeters_switch = {
.name = "VU-meters Switch",
.iface = SNDRV_CTL_ELEM_IFACE_CARD,
.access = SNDRV_CTL_ELEM_ACCESS_WRITE,
.info = snd_echo_vumeters_switch_info,
.put = snd_echo_vumeters_switch_put,
};
/***** Read VU-meters (input, output, analog and digital together) *****/
static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members" This reverts commit 275353bb684e to fix a regression which can abort 'alsactl' program in alsa-utils due to assertion in alsa-lib. alsactl: control.c:2513: snd_ctl_elem_value_get_integer: Assertion `idx < sizeof(obj->value.integer.value) / sizeof(obj->value.integer.value[0])' failed. alsactl: control.c:2976: snd_ctl_elem_value_get_integer: Assertion `idx < ARRAY_SIZE(obj->value.integer.value)' failed. This commit is a band-aid. In a point of usage of ALSA control interface, the drivers still bring an issue that they prevent userspace applications to have a consistent way to parse each levels of the dimension information via ALSA control interface. Let me investigate this issue. Current implementation of the drivers have three control element sets with dimension information: * 'Monitor Mixer Volume' (type: integer) * 'VMixer Volume' (type: integer) * 'VU-meters' (type: boolean) Although the number of elements named as 'Monitor Mixer Volume' differs depending on drivers in this group, it can be calculated by macros defined by each driver (= (BX_NUM - BX_ANALOG_IN) * BX_ANALOG_IN). Each of the elements has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * (BX_NUM - BX_ANALOG_IN)). For these elements, userspace applications are expected to handle the dimension information so that all of the elements construct a matrix where the number of rows and columns are represented by the dimension information. The same way is applied to elements named as 'VMixer Volume'. The number of these elements can also be calculated by macros defined by each drivers (= PX_ANALOG_IN * BX_ANALOG_IN). Each of the element has one member for value and has dimension information with 2 levels (= BX_ANALOG_IN * PX_ANALOG_IN). All of the elements construct a matrix with the dimension information. An element named as 'VU-meters' gets a different way in a point of dimension information. The element includes 96 members for value. The element has dimension information with 3 levels (= 3 or 2 * 16 * 2). For this element, userspace applications are expected to handle the dimension information so that all of the members for value construct a matrix where the number of rows and columns are represented by the dimension information. This is different from the way for the former. As a summary, the drivers were not designed to produce a consistent way to parse the dimension information. This makes it hard for general userspace applications such as amixer to parse the information by a consistent way, and actually no userspace applications except for 'echomixer' utilize the dimension information. Additionally, no drivers excluding this group use the information. The reverted commit was written based on the latter way. A commit 860c1994a70a ('ALSA: control: add dimension validator for userspace elements') is written based on the latter way, too. The patch should be reconsider too in the same time to re-define a consistent way to parse the dimension information. Reported-by: Mark Hills <mark@xwax.org> Reported-by: S. Christian Collins <s.chriscollins@gmail.com> Fixes: 275353bb684e ('ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members') Cc: <stable@vger.kernel.org> # v4.8+ Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-09-26 08:11:49 +08:00
uinfo->count = 96;
uinfo->value.integer.min = ECHOGAIN_MINOUT;
uinfo->value.integer.max = 0;
return 0;
}
static int snd_echo_vumeters_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
chip = snd_kcontrol_chip(kcontrol);
get_audio_meters(chip, ucontrol->value.integer.value);
return 0;
}
static const struct snd_kcontrol_new snd_echo_vumeters = {
.name = "VU-meters",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.access = SNDRV_CTL_ELEM_ACCESS_READ |
SNDRV_CTL_ELEM_ACCESS_VOLATILE |
SNDRV_CTL_ELEM_ACCESS_TLV_READ,
.info = snd_echo_vumeters_info,
.get = snd_echo_vumeters_get,
.tlv = {.p = db_scale_output_gain},
};
/*** Channels info - it exports informations about the number of channels ***/
static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 6;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1 << ECHO_CLOCK_NUMBER;
return 0;
}
static int snd_echo_channels_info_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct echoaudio *chip;
int detected, clocks, bit, src;
chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] = num_busses_in(chip);
ucontrol->value.integer.value[1] = num_analog_busses_in(chip);
ucontrol->value.integer.value[2] = num_busses_out(chip);
ucontrol->value.integer.value[3] = num_analog_busses_out(chip);
ucontrol->value.integer.value[4] = num_pipes_out(chip);
/* Compute the bitmask of the currently valid input clocks */
detected = detect_input_clocks(chip);
clocks = 0;
src = chip->num_clock_sources - 1;
for (bit = ECHO_CLOCK_NUMBER - 1; bit >= 0; bit--)
if (detected & (1 << bit))
for (; src >= 0; src--)
if (bit == chip->clock_source_list[src]) {
clocks |= 1 << src;
break;
}
ucontrol->value.integer.value[5] = clocks;
return 0;
}
static const struct snd_kcontrol_new snd_echo_channels_info = {
.name = "Channels info",
.iface = SNDRV_CTL_ELEM_IFACE_HWDEP,
.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
.info = snd_echo_channels_info_info,
.get = snd_echo_channels_info_get,
};
/******************************************************************************
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
IRQ Handling
******************************************************************************/
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
/* Check if a period has elapsed since last interrupt
*
* Don't make any updates to state; PCM core handles this with the
* correct locks.
*
* \return true if a period has elapsed, otherwise false
*/
static bool period_has_elapsed(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct audiopipe *pipe = runtime->private_data;
u32 counter, step;
size_t period_bytes;
if (pipe->state != PIPE_STATE_STARTED)
return false;
period_bytes = frames_to_bytes(runtime, runtime->period_size);
counter = le32_to_cpu(*pipe->dma_counter); /* presumed atomic */
step = counter - pipe->last_period; /* handles wrapping */
step -= step % period_bytes; /* acknowledge whole periods only */
if (step == 0)
return false; /* haven't advanced a whole period yet */
pipe->last_period += step; /* used exclusively by us */
return true;
}
IRQ: Maintain regs pointer globally rather than passing to IRQ handlers Maintain a per-CPU global "struct pt_regs *" variable which can be used instead of passing regs around manually through all ~1800 interrupt handlers in the Linux kernel. The regs pointer is used in few places, but it potentially costs both stack space and code to pass it around. On the FRV arch, removing the regs parameter from all the genirq function results in a 20% speed up of the IRQ exit path (ie: from leaving timer_interrupt() to leaving do_IRQ()). Where appropriate, an arch may override the generic storage facility and do something different with the variable. On FRV, for instance, the address is maintained in GR28 at all times inside the kernel as part of general exception handling. Having looked over the code, it appears that the parameter may be handed down through up to twenty or so layers of functions. Consider a USB character device attached to a USB hub, attached to a USB controller that posts its interrupts through a cascaded auxiliary interrupt controller. A character device driver may want to pass regs to the sysrq handler through the input layer which adds another few layers of parameter passing. I've build this code with allyesconfig for x86_64 and i386. I've runtested the main part of the code on FRV and i386, though I can't test most of the drivers. I've also done partial conversion for powerpc and MIPS - these at least compile with minimal configurations. This will affect all archs. Mostly the changes should be relatively easy. Take do_IRQ(), store the regs pointer at the beginning, saving the old one: struct pt_regs *old_regs = set_irq_regs(regs); And put the old one back at the end: set_irq_regs(old_regs); Don't pass regs through to generic_handle_irq() or __do_IRQ(). In timer_interrupt(), this sort of change will be necessary: - update_process_times(user_mode(regs)); - profile_tick(CPU_PROFILING, regs); + update_process_times(user_mode(get_irq_regs())); + profile_tick(CPU_PROFILING); I'd like to move update_process_times()'s use of get_irq_regs() into itself, except that i386, alone of the archs, uses something other than user_mode(). Some notes on the interrupt handling in the drivers: (*) input_dev() is now gone entirely. The regs pointer is no longer stored in the input_dev struct. (*) finish_unlinks() in drivers/usb/host/ohci-q.c needs checking. It does something different depending on whether it's been supplied with a regs pointer or not. (*) Various IRQ handler function pointers have been moved to type irq_handler_t. Signed-Off-By: David Howells <dhowells@redhat.com> (cherry picked from 1b16e7ac850969f38b375e511e3fa2f474a33867 commit)
2006-10-05 21:55:46 +08:00
static irqreturn_t snd_echo_interrupt(int irq, void *dev_id)
{
struct echoaudio *chip = dev_id;
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
int ss, st;
spin_lock(&chip->lock);
st = service_irq(chip);
if (st < 0) {
spin_unlock(&chip->lock);
return IRQ_NONE;
}
/* The hardware doesn't tell us which substream caused the irq,
thus we have to check all running substreams. */
for (ss = 0; ss < DSP_MAXPIPES; ss++) {
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
struct snd_pcm_substream *substream;
substream = chip->substream[ss];
ALSA: echoaudio: Address bugs in the interrupt handling Distorted audio appears occasionally, affecting either playback or capture and requiring the affected substream to be closed by all applications and re-opened. The best way I have found to reproduce the bug is to use dmix in combination with Chromium, which opens the audio device multiple times in threads. Anecdotally, the problems appear to have increased with faster CPUs. I ruled out 32-bit counter wrapping; it often happens much earlier. Since applying this patch I have not had problems, where previously they would occur several times a day. The patch targets the following issues: * Check for progress using the counter from the hardware, not after it has been truncated to the buffer. This is a clean way to address a possible bug where if a whole ringbuffer advances between interrupts, it goes unnoticed. * Move last_period state from chip to pipe This more logically belongs as part of pipe, and code is reasier to read if it is "counter position last time a period elapsed". Now the code has no references to period count. A period is just when the regular counter crosses a threshold. This increases readability and reduces scope for bugs. * Treat period notification and buffer advance independently: This helps to clarify what is the responsibility of the interrupt handler, and what is pcm_pointer(). Removing shared state between these operations means race conditions are fixed without introducing locks. Synchronisation is only around the read of pipe->dma_counter. There may be cache line contention around "struct audiopipe" but I did not have cause to profile this. Pay attention to be robust where dma_counter wrapping is not a multiple of period_size or buffer_size. This is a revised patch based on feedback from Takashi and Giuliano. Signed-off-by: Mark Hills <mark@xwax.org> Link: https://lore.kernel.org/r/20200708101848.3457-5-mark@xwax.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-07-08 18:18:48 +08:00
if (substream && period_has_elapsed(substream)) {
spin_unlock(&chip->lock);
snd_pcm_period_elapsed(substream);
spin_lock(&chip->lock);
}
}
spin_unlock(&chip->lock);
#ifdef ECHOCARD_HAS_MIDI
if (st > 0 && chip->midi_in) {
snd_rawmidi_receive(chip->midi_in, chip->midi_buffer, st);
dev_dbg(chip->card->dev, "rawmidi_iread=%d\n", st);
}
#endif
return IRQ_HANDLED;
}
/******************************************************************************
Module construction / destruction
******************************************************************************/
static void snd_echo_free(struct snd_card *card)
{
struct echoaudio *chip = card->private_data;
if (chip->comm_page)
rest_in_peace(chip);
if (chip->irq >= 0)
free_irq(chip->irq, chip);
/* release chip data */
free_firmware_cache(chip);
}
/* <--snd_echo_probe() */
static int snd_echo_create(struct snd_card *card,
struct pci_dev *pci)
{
struct echoaudio *chip = card->private_data;
int err;
size_t sz;
pci_write_config_byte(pci, PCI_LATENCY_TIMER, 0xC0);
err = pcim_enable_device(pci);
if (err < 0)
return err;
pci_set_master(pci);
/* Allocate chip if needed */
spin_lock_init(&chip->lock);
chip->card = card;
chip->pci = pci;
chip->irq = -1;
chip->opencount = 0;
mutex_init(&chip->mode_mutex);
chip->can_set_rate = 1;
/* PCI resource allocation */
err = pci_request_regions(pci, ECHOCARD_NAME);
if (err < 0)
return err;
chip->dsp_registers_phys = pci_resource_start(pci, 0);
sz = pci_resource_len(pci, 0);
if (sz > PAGE_SIZE)
sz = PAGE_SIZE; /* We map only the required part */
chip->dsp_registers = devm_ioremap(&pci->dev, chip->dsp_registers_phys, sz);
if (!chip->dsp_registers) {
dev_err(chip->card->dev, "ioremap failed\n");
return -ENOMEM;
}
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
KBUILD_MODNAME, chip)) {
dev_err(chip->card->dev, "cannot grab irq\n");
return -EBUSY;
}
chip->irq = pci->irq;
card->sync_irq = chip->irq;
dev_dbg(card->dev, "pci=%p irq=%d subdev=%04x Init hardware...\n",
chip->pci, chip->irq, chip->pci->subsystem_device);
card->private_free = snd_echo_free;
/* Create the DSP comm page - this is the area of memory used for most
of the communication with the DSP, which accesses it via bus mastering */
chip->commpage_dma_buf =
snd_devm_alloc_pages(&pci->dev, SNDRV_DMA_TYPE_DEV,
sizeof(struct comm_page));
if (!chip->commpage_dma_buf)
return -ENOMEM;
chip->comm_page_phys = chip->commpage_dma_buf->addr;
chip->comm_page = (struct comm_page *)chip->commpage_dma_buf->area;
err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
if (err >= 0)
err = set_mixer_defaults(chip);
if (err < 0) {
dev_err(card->dev, "init_hw err=%d\n", err);
return err;
}
return 0;
}
/* constructor */
static int __snd_echo_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
static int dev;
struct snd_card *card;
struct echoaudio *chip;
char *dsp;
__maybe_unused int i;
int err;
if (dev >= SNDRV_CARDS)
return -ENODEV;
if (!enable[dev]) {
dev++;
return -ENOENT;
}
i = 0;
err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE,
sizeof(*chip), &card);
if (err < 0)
return err;
chip = card->private_data;
err = snd_echo_create(card, pci);
if (err < 0)
return err;
strcpy(card->driver, "Echo_" ECHOCARD_NAME);
strcpy(card->shortname, chip->card_name);
dsp = "56301";
if (pci_id->device == 0x3410)
dsp = "56361";
sprintf(card->longname, "%s rev.%d (DSP%s) at 0x%lx irq %i",
card->shortname, pci_id->subdevice & 0x000f, dsp,
chip->dsp_registers_phys, chip->irq);
err = snd_echo_new_pcm(chip);
if (err < 0) {
dev_err(chip->card->dev, "new pcm error %d\n", err);
return err;
}
#ifdef ECHOCARD_HAS_MIDI
if (chip->has_midi) { /* Some Mia's do not have midi */
err = snd_echo_midi_create(card, chip);
if (err < 0) {
dev_err(chip->card->dev, "new midi error %d\n", err);
return err;
}
}
#endif
#ifdef ECHOCARD_HAS_VMIXER
snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip));
if (err < 0)
return err;
#ifdef ECHOCARD_HAS_LINE_OUT_GAIN
err = snd_ctl_add(chip->card,
snd_ctl_new1(&snd_echo_line_output_gain, chip));
if (err < 0)
return err;
#endif
#else /* ECHOCARD_HAS_VMIXER */
err = snd_ctl_add(chip->card,
snd_ctl_new1(&snd_echo_pcm_output_gain, chip));
if (err < 0)
return err;
#endif /* ECHOCARD_HAS_VMIXER */
#ifdef ECHOCARD_HAS_INPUT_GAIN
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip));
if (err < 0)
return err;
#endif
#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
if (!chip->hasnt_input_nominal_level) {
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_intput_nominal_level, chip));
if (err < 0)
return err;
}
#endif
#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_output_nominal_level, chip));
if (err < 0)
return err;
#endif
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters_switch, chip));
if (err < 0)
return err;
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters, chip));
if (err < 0)
return err;
#ifdef ECHOCARD_HAS_MONITOR
snd_echo_monitor_mixer.count = num_busses_in(chip) * num_busses_out(chip);
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_monitor_mixer, chip));
if (err < 0)
return err;
#endif
#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_automute_switch, chip));
if (err < 0)
return err;
#endif
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_channels_info, chip));
if (err < 0)
return err;
#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
/* Creates a list of available digital modes */
chip->num_digital_modes = 0;
for (i = 0; i < 6; i++)
if (chip->digital_modes & (1 << i))
chip->digital_mode_list[chip->num_digital_modes++] = i;
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_digital_mode_switch, chip));
if (err < 0)
return err;
#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
/* Creates a list of available clock sources */
chip->num_clock_sources = 0;
for (i = 0; i < 10; i++)
if (chip->input_clock_types & (1 << i))
chip->clock_source_list[chip->num_clock_sources++] = i;
if (chip->num_clock_sources > 1) {
chip->clock_src_ctl = snd_ctl_new1(&snd_echo_clock_source_switch, chip);
err = snd_ctl_add(chip->card, chip->clock_src_ctl);
if (err < 0)
return err;
}
#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
#ifdef ECHOCARD_HAS_DIGITAL_IO
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_spdif_mode_switch, chip));
if (err < 0)
return err;
#endif
#ifdef ECHOCARD_HAS_PHANTOM_POWER
if (chip->has_phantom_power) {
err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_phantom_power_switch, chip));
if (err < 0)
return err;
}
#endif
err = snd_card_register(card);
if (err < 0)
return err;
dev_info(card->dev, "Card registered: %s\n", card->longname);
pci_set_drvdata(pci, chip);
dev++;
return 0;
}
static int snd_echo_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
return snd_card_free_on_error(&pci->dev, __snd_echo_probe(pci, pci_id));
}
#if defined(CONFIG_PM_SLEEP)
static int snd_echo_suspend(struct device *dev)
{
struct echoaudio *chip = dev_get_drvdata(dev);
#ifdef ECHOCARD_HAS_MIDI
/* This call can sleep */
if (chip->midi_out)
snd_echo_midi_output_trigger(chip->midi_out, 0);
#endif
spin_lock_irq(&chip->lock);
if (wait_handshake(chip)) {
spin_unlock_irq(&chip->lock);
return -EIO;
}
clear_handshake(chip);
if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) {
spin_unlock_irq(&chip->lock);
return -EIO;
}
spin_unlock_irq(&chip->lock);
chip->dsp_code = NULL;
free_irq(chip->irq, chip);
chip->irq = -1;
chip->card->sync_irq = -1;
return 0;
}
static int snd_echo_resume(struct device *dev)
{
struct pci_dev *pci = to_pci_dev(dev);
struct echoaudio *chip = dev_get_drvdata(dev);
struct comm_page *commpage, *commpage_bak;
u32 pipe_alloc_mask;
int err;
commpage = chip->comm_page;
commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL);
if (commpage_bak == NULL)
return -ENOMEM;
err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
if (err < 0) {
kfree(commpage_bak);
dev_err(dev, "resume init_hw err=%d\n", err);
return err;
}
/* Temporarily set chip->pipe_alloc_mask=0 otherwise
* restore_dsp_settings() fails.
*/
pipe_alloc_mask = chip->pipe_alloc_mask;
chip->pipe_alloc_mask = 0;
err = restore_dsp_rettings(chip);
chip->pipe_alloc_mask = pipe_alloc_mask;
if (err < 0) {
kfree(commpage_bak);
return err;
}
memcpy(&commpage->audio_format, &commpage_bak->audio_format,
sizeof(commpage->audio_format));
memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr,
sizeof(commpage->sglist_addr));
memcpy(&commpage->midi_output, &commpage_bak->midi_output,
sizeof(commpage->midi_output));
kfree(commpage_bak);
if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED,
KBUILD_MODNAME, chip)) {
dev_err(chip->card->dev, "cannot grab irq\n");
return -EBUSY;
}
chip->irq = pci->irq;
chip->card->sync_irq = chip->irq;
dev_dbg(dev, "resume irq=%d\n", chip->irq);
#ifdef ECHOCARD_HAS_MIDI
if (chip->midi_input_enabled)
enable_midi_input(chip, true);
if (chip->midi_out)
snd_echo_midi_output_trigger(chip->midi_out, 1);
#endif
return 0;
}
static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume);
#define SND_ECHO_PM_OPS &snd_echo_pm
#else
#define SND_ECHO_PM_OPS NULL
#endif /* CONFIG_PM_SLEEP */
/******************************************************************************
Everything starts and ends here
******************************************************************************/
/* pci_driver definition */
static struct pci_driver echo_driver = {
.name = KBUILD_MODNAME,
.id_table = snd_echo_ids,
.probe = snd_echo_probe,
.driver = {
.pm = SND_ECHO_PM_OPS,
},
};
module_pci_driver(echo_driver);