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-rw-r--r--sound/soc/codecs/tlv320aic23.c2
-rw-r--r--sound/soc/codecs/wm8350.c4
-rw-r--r--sound/soc/codecs/wm8940.c2
-rw-r--r--sound/soc/imx/mxc-ssi.c8
-rw-r--r--sound/soc/omap/Kconfig13
-rw-r--r--sound/soc/omap/omap-pcm.c8
-rw-r--r--sound/soc/omap/omap3evm.c2
-rw-r--r--sound/soc/omap/omap3pandora.c3
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c17
-rw-r--r--sound/soc/s3c24xx/s3c64xx-i2s.c2
-rw-r--r--sound/soc/soc-core.c11
-rw-r--r--sound/soc/soc-dapm.c7
12 files changed, 54 insertions, 25 deletions
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 0b8dcb5cd72..6b24d8bb02b 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -265,8 +265,8 @@ static const int bosr_usb_divisor_table[] = {
#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
static const unsigned short sr_valid_mask[] = {
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
- LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
LOWER_GROUP, /* Usb, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
UPPER_GROUP, /* Usb, bosr - 1*/
};
/*
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3ff0373dff8..593d5b9c9f0 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -579,7 +579,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
- WM8350_LEFT_INPUT_VOLUME, 14, 1, 0),
+ WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
};
/* Right Input Mixer */
@@ -589,7 +589,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
- WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0),
+ WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
};
/* Left Mic Mixer */
diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c
index da97aae475a..1ef2454c520 100644
--- a/sound/soc/codecs/wm8940.c
+++ b/sound/soc/codecs/wm8940.c
@@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940,
codec->reg_cache = &wm8940->reg_cache;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, control);
- if (ret == 0) {
+ if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c
index 3806ff2c0cd..ccdefe60e75 100644
--- a/sound/soc/imx/mxc-ssi.c
+++ b/sound/soc/imx/mxc-ssi.c
@@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai,
break;
}
- /* sync */
- if (!(fmt & SND_SOC_DAIFMT_ASYNC))
- scr |= SSI_SCR_SYN;
-
- /* tdm - only for stereo atm */
- if (fmt & SND_SOC_DAIFMT_TDM)
- scr |= SSI_SCR_NET;
-
if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) {
SSI1_STCR = stcr;
SSI1_SRCR = srcr;
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee9839be8..653a362425d 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -21,7 +21,18 @@ config SND_OMAP_SOC_AMS_DELTA
select SND_OMAP_SOC_MCBSP
select SND_SOC_CX20442
help
- Say Y if you want to add support for SoC audio on Amstrad Delta.
+ Say Y if you want to add support for SoC audio device connected to
+ a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
+
+ Note that in order to get those devices fully supported, you have to
+ build the kernel with standard serial port driver included and
+ configured for at least 4 ports. Then, from userspace, you must load
+ a line discipline #19 on the modem (ttyS3) serial line. The simplest
+ way to achieve this is to install util-linux-ng and use the included
+ ldattach utility. This can be started automatically from udev,
+ a simple rule like this one should do the trick (it does for me):
+ ACTION=="add", KERNEL=="controlC0", \
+ RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
config SND_OMAP_SOC_OSK5912
tristate "SoC Audio support for omap osk5912"
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 5735945788b..6a829eef2a4 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -195,8 +195,12 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
else
omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
- omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
- omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+ if (!(cpu_class_is_omap1())) {
+ omap_set_dma_src_burst_mode(prtd->dma_ch,
+ OMAP_DMA_DATA_BURST_16);
+ omap_set_dma_dest_burst_mode(prtd->dma_ch,
+ OMAP_DMA_DATA_BURST_16);
+ }
return 0;
}
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 9114c263077..13aa380de16 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -144,4 +144,4 @@ module_exit(omap3evm_soc_exit);
MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM");
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index ad219aaf7cb..0cd06f5dd35 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -134,7 +134,7 @@ static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
* |P| <--- TWL4030 <--------- Line In and MICs
*/
static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
- SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("PCM DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
0, 0, NULL, 0, omap3pandora_hp_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -181,6 +181,7 @@ static int omap3pandora_out_init(struct snd_soc_codec *codec)
snd_soc_dapm_nc_pin(codec, "CARKITR");
snd_soc_dapm_nc_pin(codec, "HFL");
snd_soc_dapm_nc_pin(codec, "HFR");
+ snd_soc_dapm_nc_pin(codec, "VIBRA");
ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
ARRAY_SIZE(omap3pandora_out_dapm_widgets));
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 5cbbdc80fde..1f35c6fcf5f 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -75,11 +75,19 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
{
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
dma_addr_t pos = prtd->dma_pos;
+ unsigned int limit;
int ret;
pr_debug("Entered %s\n", __func__);
- while (prtd->dma_loaded < prtd->dma_limit) {
+ if (s3c_dma_has_circular()) {
+ limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period;
+ } else
+ limit = prtd->dma_limit;
+
+ pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit);
+
+ while (prtd->dma_loaded < limit) {
unsigned long len = prtd->dma_period;
pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
@@ -123,7 +131,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
snd_pcm_period_elapsed(substream);
spin_lock(&prtd->lock);
- if (prtd->state & ST_RUNNING) {
+ if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) {
prtd->dma_loaded--;
s3c24xx_pcm_enqueue(substream);
}
@@ -164,6 +172,11 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
printk(KERN_ERR "failed to get dma channel\n");
return ret;
}
+
+ /* use the circular buffering if we have it available. */
+ if (s3c_dma_has_circular())
+ s3c2410_dma_setflags(prtd->params->channel,
+ S3C2410_DMAF_CIRCULAR);
}
s3c2410_dma_set_buffdone_fn(prtd->params->channel,
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
index 3c06c401d0f..105a77eeded 100644
--- a/sound/soc/s3c24xx/s3c64xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -220,6 +220,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev)
goto err;
}
+ clk_enable(i2s->iis_cclk);
+
ret = s3c_i2sv2_probe(pdev, dai, i2s, 0);
if (ret)
goto err_clk;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7ff04ad2a97..0a1b2f64bbe 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -834,6 +834,9 @@ EXPORT_SYMBOL_GPL(snd_soc_resume_device);
#define soc_resume NULL
#endif
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
static void snd_soc_instantiate_card(struct snd_soc_card *card)
{
struct platform_device *pdev = container_of(card->dev,
@@ -877,6 +880,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
ac97 = 1;
}
+ for (i = 0; i < card->num_links; i++) {
+ if (!card->dai_link[i].codec_dai->ops)
+ card->dai_link[i].codec_dai->ops = &null_dai_ops;
+ }
+
/* If we have AC97 in the system then don't wait for the
* codec. This will need revisiting if we have to handle
* systems with mixed AC97 and non-AC97 parts. Only check for
@@ -2329,9 +2337,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
return 0;
}
-static struct snd_soc_dai_ops null_dai_ops = {
-};
-
/**
* snd_soc_register_dai - Register a DAI with the ASoC core
*
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f79711b9fa5..d89f6dc0090 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
/* connected jack or spk ? */
if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
- widget->id == snd_soc_dapm_line)
+ (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources)))
return 1;
}
@@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
return 1;
/* connected jack ? */
- if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
+ if (widget->id == snd_soc_dapm_mic ||
+ (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks)))
return 1;
}
@@ -2071,9 +2072,9 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
}
}
}
- mutex_unlock(&codec->mutex);
dapm_power_widgets(codec, event);
+ mutex_unlock(&codec->mutex);
dump_dapm(codec, __func__);
return 0;
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-29 01:48:08 +0000