9 files changed, 66 insertions, 17 deletions

diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index 0b8dcb5cd729..90a0264f7538 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*/
 };
 /*
@@ -625,11 +625,10 @@ static int tlv320aic23_resume(struct platform_device *pdev)
 {
         struct snd_soc_device *socdev = platform_get_drvdata(pdev);
         struct snd_soc_codec *codec = socdev->card->codec;
-        int i;
         u16 reg;
 
         /* Sync reg_cache with the hardware */
-        for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+        for (reg = 0; reg < TLV320AIC23_RESET; reg++) {
                 u16 val = tlv320aic23_read_reg_cache(codec, reg);
                 tlv320aic23_write(codec, reg, val);
         }
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 2dee9839be86..653a362425df 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 5735945788bf..6a829eef2a4f 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 9114c263077b..13aa380de162 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 ad219aaf7cb8..0cd06f5dd356 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 5cbbdc80fde3..1f35c6fcf5fd 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 3c06c401d0fb..105a77eeded0 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 7ff04ad2a97e..0a1b2f64bbee 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 8de6f9dec4a2..66d4c165f99b 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -973,9 +973,19 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
                         if (!w->power_check)
                                 continue;
 
-                        power = w->power_check(w);
-                        if (power)
-                                sys_power = 1;
+                        /* If we're suspending then pull down all the 
+                         * power. */
+                        switch (event) {
+                        case SND_SOC_DAPM_STREAM_SUSPEND:
+                                power = 0;
+                                break;
+
+                        default:
+                                power = w->power_check(w);
+                                if (power)
+                                        sys_power = 1;
+                                break;
+                        }
 
                         if (w->power == power)
                                 continue;
@@ -999,8 +1009,12 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
                 case SND_SOC_DAPM_STREAM_RESUME:
                         sys_power = 1;
                         break;
+                case SND_SOC_DAPM_STREAM_SUSPEND:
+                        sys_power = 0;
+                        break;
                 case SND_SOC_DAPM_STREAM_NOP:
                         sys_power = codec->bias_level != SND_SOC_BIAS_STANDBY;
+                        break;
                 default:
                         break;
                 }
@@ -2072,9 +2086,9 @@ 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;
 }