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-rw-r--r--sound/soc/omap/Kconfig13
-rw-r--r--sound/soc/omap/omap-pcm.c8
-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.c2
6 files changed, 44 insertions, 9 deletions
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/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 8de6f9dec4a..d89f6dc0090 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -2072,9 +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;
}