From: Eduardo Valentin <eduardo.valentin@xxxxxxxxxxx>
Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.
Signed-off-by: Eduardo Valentin <eduardo.valentin@xxxxxxxxxxx>
---
sound/arm/omap/omap-alsa.c | 163 +++++++++++++++++++++++---------------------
1 files changed, 86 insertions(+), 77 deletions(-)
diff --git a/sound/arm/omap/omap-alsa.c b/sound/arm/omap/omap-alsa.c
index 8d0f394..056ae92 100644
--- a/sound/arm/omap/omap-alsa.c
+++ b/sound/arm/omap/omap-alsa.c
@@ -1,6 +1,6 @@
/*
* sound/arm/omap-alsa.c
- *
+ *
* Alsa Driver for OMAP
*
* Copyright (C) 2005 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
@@ -9,7 +9,7 @@
*
* Copyright (C) 2006 Mika Laitio <lamikr@xxxxxxxxx>
*
- * Based on sa11xx-uda1341.c,
+ * Based on sa11xx-uda1341.c,
* Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@xxxxxxxxx>
*
* This program is free software; you can redistribute it and/or modify it
@@ -34,10 +34,10 @@
*
* History:
*
- * 2005-07-29 INdT Kernel Team - Alsa driver for omap osk. Creation of new
+ * 2005-07-29 INdT Kernel Team - Alsa driver for omap osk. Creation of new
* file omap-aic23.c
- *
- * 2005-12-18 Dirk Behme - Added L/R Channel Interchange fix as proposed
+ *
+ * 2005-12-18 Dirk Behme - Added L/R Channel Interchange fix as proposed
* by Ajaya Babu
*
*/
@@ -53,14 +53,18 @@
#include <asm/arch/omap-alsa.h>
#include "omap-alsa-dma.h"
-MODULE_AUTHOR("Mika Laitio, Daniel Petrini, David Cohen, Anderson Briglia - INdT");
+MODULE_AUTHOR("Mika Laitio");
+MODULE_AUTHOR("Daniel Petrini");
+MODULE_AUTHOR("David Cohen");
+MODULE_AUTHOR("Anderson Briglia");
+
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("OMAP driver for ALSA");
MODULE_ALIAS("omap_alsa_mcbsp.1");
-static char *id = NULL;
-static struct snd_card_omap_codec *alsa_codec = NULL;
-static struct omap_alsa_codec_config *alsa_codec_config = NULL;
+static char *id;
+static struct snd_card_omap_codec *alsa_codec;
+static struct omap_alsa_codec_config *alsa_codec_config;
/*
* HW interface start and stop helper functions
@@ -101,10 +105,9 @@ static void omap_alsa_audio_init(struct snd_card_omap_codec *omap_alsa)
audio_ifc_stop;
}
-/*
- * DMA functions
+/*
+ * DMA functions
* Depends on omap-alsa-dma.c functions and (omap) dma.c
- *
*/
static int audio_dma_request(struct audio_stream *s,
void (*callback) (void *))
@@ -152,7 +155,8 @@ static u_int audio_get_dma_pos(struct audio_stream *s)
spin_unlock_irqrestore(&s->dma_lock, flags);
/* Now, the position related to the end of that period */
- offset = bytes_to_frames(runtime, s->offset) - bytes_to_frames(runtime, count);
+ offset = bytes_to_frames(runtime, s->offset) -
+ bytes_to_frames(runtime, count);
if (offset >= runtime->buffer_size)
offset = 0;
@@ -191,29 +195,28 @@ static void audio_process_dma(struct audio_stream *s)
unsigned int dma_size;
unsigned int offset;
int ret;
-
+
ADEBUG();
runtime = substream->runtime;
if (s->active) {
dma_size = frames_to_bytes(runtime, runtime->period_size);
offset = dma_size * s->period;
- snd_assert(dma_size <= DMA_BUF_SIZE,);
+ snd_assert(dma_size <= DMA_BUF_SIZE, return);
/*
* On omap1510 based devices, we need to call the stop_dma
* before calling the start_dma or we will not receive the
* irq from DMA after the first transfered/played buffer.
* (invocation of callback_omap_alsa_sound_dma() method).
*/
- if (cpu_is_omap1510()) {
+ if (cpu_is_omap1510())
omap_stop_alsa_sound_dma(s);
- }
+
ret = omap_start_alsa_sound_dma(s,
(dma_addr_t)runtime->dma_area + offset,
dma_size);
if (ret) {
- printk(KERN_ERR
- "audio_process_dma: cannot queue DMA buffer (%i)\n",
- ret);
+ printk(KERN_ERR "audio_process_dma: cannot"
+ " queue DMA buffer (%i)\n", ret);
return;
}
@@ -224,15 +227,15 @@ static void audio_process_dma(struct audio_stream *s)
}
}
-/*
+/*
* This is called when dma IRQ occurs at the end of each transmited block
*/
void callback_omap_alsa_sound_dma(void *data)
{
struct audio_stream *s = data;
-
+
ADEBUG();
- /*
+ /*
* If we are getting a callback for an active stream then we inform
* the PCM middle layer we've finished a period
*/
@@ -240,25 +243,25 @@ void callback_omap_alsa_sound_dma(void *data)
snd_pcm_period_elapsed(s->stream);
spin_lock(&s->dma_lock);
- if (s->periods > 0)
+ if (s->periods > 0)
s->periods--;
-
+
audio_process_dma(s);
spin_unlock(&s->dma_lock);
}
-/*
+/*
* Alsa section
* PCM settings and callbacks
*/
-static int snd_omap_alsa_trigger(struct snd_pcm_substream * substream, int cmd)
+static int snd_omap_alsa_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct snd_card_omap_codec *chip =
snd_pcm_substream_chip(substream);
int stream_id = substream->pstr->stream;
struct audio_stream *s = &chip->s[stream_id];
int err = 0;
-
+
ADEBUG();
/* note local interrupts are already disabled in the midlevel code */
spin_lock(&s->dma_lock);
@@ -277,16 +280,16 @@ static int snd_omap_alsa_trigger(struct snd_pcm_substream * substream, int cmd)
break;
}
spin_unlock(&s->dma_lock);
-
+
return err;
}
-static int snd_omap_alsa_prepare(struct snd_pcm_substream * substream)
+static int snd_omap_alsa_prepare(struct snd_pcm_substream *substream)
{
struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
struct audio_stream *s = &chip->s[substream->pstr->stream];
-
+
ADEBUG();
/* set requested samplerate */
alsa_codec_config->codec_set_samplerate(runtime->rate);
@@ -298,63 +301,64 @@ static int snd_omap_alsa_prepare(struct snd_pcm_substream * substream)
return 0;
}
-static snd_pcm_uframes_t snd_omap_alsa_pointer(struct snd_pcm_substream *substream)
+static snd_pcm_uframes_t
+snd_omap_alsa_pointer(struct snd_pcm_substream *substream)
{
struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
- ADEBUG();
+ ADEBUG();
return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
}
-static int snd_card_omap_alsa_open(struct snd_pcm_substream * substream)
+static int snd_card_omap_alsa_open(struct snd_pcm_substream *substream)
{
struct snd_card_omap_codec *chip =
snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
int stream_id = substream->pstr->stream;
int err;
-
+
ADEBUG();
chip->s[stream_id].stream = substream;
alsa_codec_config->codec_clock_on();
- if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
+ if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
runtime->hw = *(alsa_codec_config->snd_omap_alsa_playback);
- else
+ else
runtime->hw = *(alsa_codec_config->snd_omap_alsa_capture);
-
- if ((err = snd_pcm_hw_constraint_integer(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+
+ err = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (err < 0)
return err;
-
- if ((err = snd_pcm_hw_constraint_list(runtime,
- 0,
- SNDRV_PCM_HW_PARAM_RATE,
- alsa_codec_config->hw_constraints_rates)) < 0)
+
+ err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ alsa_codec_config->hw_constraints_rates);
+ if (err < 0)
return err;
-
+
return 0;
}
-static int snd_card_omap_alsa_close(struct snd_pcm_substream * substream)
+static int snd_card_omap_alsa_close(struct snd_pcm_substream *substream)
{
struct snd_card_omap_codec *chip = snd_pcm_substream_chip(substream);
-
+
ADEBUG();
alsa_codec_config->codec_clock_off();
chip->s[substream->pstr->stream].stream = NULL;
-
+
return 0;
}
/* HW params & free */
-static int snd_omap_alsa_hw_params(struct snd_pcm_substream * substream,
- struct snd_pcm_hw_params * hw_params)
+static int snd_omap_alsa_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
{
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
}
-static int snd_omap_alsa_hw_free(struct snd_pcm_substream * substream)
+static int snd_omap_alsa_hw_free(struct snd_pcm_substream *substream)
{
return snd_pcm_lib_free_pages(substream);
}
@@ -384,17 +388,17 @@ static struct snd_pcm_ops snd_card_omap_alsa_capture_ops = {
/*
* Alsa init and exit section
- *
* Inits pcm alsa structures, allocate the alsa buffer, suspend, resume
*/
-static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa,
+static int __init snd_card_omap_alsa_pcm(struct snd_card_omap_codec *omap_alsa,
int device)
{
struct snd_pcm *pcm;
int err;
-
+
ADEBUG();
- if ((err = snd_pcm_new(omap_alsa->card, "OMAP PCM", device, 1, 1, &pcm)) < 0)
+ err = snd_pcm_new(omap_alsa->card, "OMAP PCM", device, 1, 1, &pcm);
+ if (err < 0)
return err;
/* sets up initial buffer with continuous allocation */
@@ -434,11 +438,12 @@ int snd_omap_alsa_suspend(struct platform_device *pdev, pm_message_t state)
{
struct snd_card_omap_codec *chip;
struct snd_card *card = platform_get_drvdata(pdev);
-
+
if (card->power_state != SNDRV_CTL_POWER_D3hot) {
chip = card->private_data;
if (chip->card->power_state != SNDRV_CTL_POWER_D3hot) {
- snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot);
+ snd_power_change_state(chip->card,
+ SNDRV_CTL_POWER_D3hot);
snd_pcm_suspend_all(chip->pcm);
/* Mutes and turn clock off */
alsa_codec_config->codec_clock_off();
@@ -453,7 +458,7 @@ int snd_omap_alsa_resume(struct platform_device *pdev)
struct snd_card_omap_codec *chip;
struct snd_card *card = platform_get_drvdata(pdev);
- if (card->power_state != SNDRV_CTL_POWER_D0) {
+ if (card->power_state != SNDRV_CTL_POWER_D0) {
chip = card->private_data;
if (chip->card->power_state != SNDRV_CTL_POWER_D0) {
snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0);
@@ -466,11 +471,11 @@ int snd_omap_alsa_resume(struct platform_device *pdev)
#endif /* CONFIG_PM */
-void snd_omap_alsa_free(struct snd_card * card)
+void snd_omap_alsa_free(struct snd_card *card)
{
struct snd_card_omap_codec *chip = card->private_data;
ADEBUG();
-
+
/*
* Turn off codec after it is done.
* Can't do it immediately, since it may still have
@@ -487,27 +492,28 @@ void snd_omap_alsa_free(struct snd_card * card)
/* module init & exit */
-/*
+/*
* Inits alsa soudcard structure.
* Called by the probe method in codec after function pointers has been set.
*/
-int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_codec_config *config)
+int snd_omap_alsa_post_probe(struct platform_device *pdev,
+ struct omap_alsa_codec_config *config)
{
int err = 0;
int def_rate;
struct snd_card *card;
-
+
ADEBUG();
alsa_codec_config = config;
alsa_codec_config->codec_clock_setup();
- alsa_codec_config->codec_clock_on();
+ alsa_codec_config->codec_clock_on();
omap_mcbsp_request(AUDIO_MCBSP);
omap_mcbsp_stop(AUDIO_MCBSP);
omap_mcbsp_config(AUDIO_MCBSP, alsa_codec_config->mcbsp_regs_alsa);
omap_mcbsp_start(AUDIO_MCBSP);
-
+
if (alsa_codec_config && alsa_codec_config->codec_configure_dev)
alsa_codec_config->codec_configure_dev();
@@ -526,18 +532,20 @@ int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_code
card->private_free = snd_omap_alsa_free;
alsa_codec->card = card;
- def_rate = alsa_codec_config->get_default_samplerate();
+ def_rate = alsa_codec_config->get_default_samplerate();
alsa_codec->samplerate = def_rate;
spin_lock_init(&alsa_codec->s[0].dma_lock);
spin_lock_init(&alsa_codec->s[1].dma_lock);
/* mixer */
- if ((err = snd_omap_mixer(alsa_codec)) < 0)
+ err = snd_omap_mixer(alsa_codec);
+ if (err < 0)
goto nodev3;
/* PCM */
- if ((err = snd_card_omap_alsa_pcm(alsa_codec, 0)) < 0)
+ err = snd_card_omap_alsa_pcm(alsa_codec, 0);
+ if (err < 0)
goto nodev3;
strcpy(card->driver, "OMAP_ALSA");
@@ -546,16 +554,17 @@ int snd_omap_alsa_post_probe(struct platform_device *pdev, struct omap_alsa_code
snd_omap_init_mixer();
snd_card_set_dev(card, &pdev->dev);
-
- if ((err = snd_card_register(card)) == 0) {
+
+ err = snd_card_register(card);
+ if (err == 0) {
printk(KERN_INFO "audio support initialized\n");
platform_set_drvdata(pdev, card);
return 0;
}
-
+
nodev3:
- kfree(alsa_codec);
-nodev2:
+ kfree(alsa_codec);
+nodev2:
snd_card_free(card);
nodev1:
omap_mcbsp_stop(AUDIO_MCBSP);
@@ -568,14 +577,14 @@ int snd_omap_alsa_remove(struct platform_device *pdev)
{
struct snd_card *card = platform_get_drvdata(pdev);
struct snd_card_omap_codec *chip = card->private_data;
-
+
snd_card_free(card);
alsa_codec = NULL;
card->private_data = NULL;
kfree(chip);
-
+
platform_set_drvdata(pdev, NULL);
-
+
return 0;
}
--
1.5.5-rc3.GIT
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