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-rw-r--r--MAINTAINERS9
-rw-r--r--sound/soc/davinci/davinci-mcasp.c14
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c3
-rw-r--r--sound/soc/fsl/fsl_ssi.c5
-rw-r--r--sound/soc/intel/haswell/sst-haswell-ipc.c20
-rw-r--r--sound/soc/soc-utils.c9
6 files changed, 36 insertions, 24 deletions
diff --git a/MAINTAINERS b/MAINTAINERS
index 274f85405584..0581f47e959b 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -11239,7 +11239,6 @@ VOLTAGE AND CURRENT REGULATOR FRAMEWORK
M: Liam Girdwood <lgirdwood@gmail.com>
M: Mark Brown <broonie@kernel.org>
L: linux-kernel@vger.kernel.org
-W: http://opensource.wolfsonmicro.com/node/15
W: http://www.slimlogic.co.uk/?p=48
T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git
S: Supported
@@ -11368,17 +11367,15 @@ WM97XX TOUCHSCREEN DRIVERS
M: Mark Brown <broonie@kernel.org>
M: Liam Girdwood <lrg@slimlogic.co.uk>
L: linux-input@vger.kernel.org
-T: git git://opensource.wolfsonmicro.com/linux-2.6-touch
-W: http://opensource.wolfsonmicro.com/node/7
+W: https://github.com/CirrusLogic/linux-drivers/wiki
S: Supported
F: drivers/input/touchscreen/*wm97*
F: include/linux/wm97xx.h
WOLFSON MICROELECTRONICS DRIVERS
L: patches@opensource.wolfsonmicro.com
-T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
-T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
-W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
+T: git https://github.com/CirrusLogic/linux-drivers.git
+W: https://github.com/CirrusLogic/linux-drivers/wiki
S: Supported
F: Documentation/hwmon/wm83??
F: arch/arm/mach-s3c64xx/mach-crag6410*
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index add6bb99661d..7d45d98a861f 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
u8 rx_ser = 0;
u8 slots = mcasp->tdm_slots;
u8 max_active_serializers = (channels + slots - 1) / slots;
- int active_serializers, numevt, n;
+ int active_serializers, numevt;
u32 reg;
/* Default configuration */
if (mcasp->version < MCASP_VERSION_3)
@@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
* The number of words for numevt need to be in steps of active
* serializers.
*/
- n = numevt % active_serializers;
- if (n)
- numevt += (active_serializers - n);
+ numevt = (numevt / active_serializers) * active_serializers;
+
while (period_words % numevt && numevt > 0)
numevt -= active_serializers;
if (numevt <= 0)
@@ -1299,6 +1298,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
.ops = &davinci_mcasp_dai_ops,
.symmetric_samplebits = 1,
+ .symmetric_rates = 1,
},
{
.name = "davinci-mcasp.1",
@@ -1685,7 +1685,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
irq = platform_get_irq_byname(pdev, "common");
if (irq >= 0) {
- irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n",
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
dev_name(&pdev->dev));
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_common_irq_handler,
@@ -1702,7 +1702,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
irq = platform_get_irq_byname(pdev, "rx");
if (irq >= 0) {
- irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
dev_name(&pdev->dev));
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_rx_irq_handler,
@@ -1717,7 +1717,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
irq = platform_get_irq_byname(pdev, "tx");
if (irq >= 0) {
- irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
+ irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
dev_name(&pdev->dev));
ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
davinci_mcasp_tx_irq_handler,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5aeb6ed4827e..96f55ae75c71 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
- return -EINVAL;
+ ret = -EINVAL;
+ goto asrc_fail;
}
/* Common settings for corresponding Freescale CPU DAI driver */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 8ec6fb208ea0..37c5cd4d0e59 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
{
- return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
+ return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+ SND_SOC_DAIFMT_AC97;
}
static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
@@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
CCSR_SSI_SCR_TCH_EN);
}
- if (fmt & SND_SOC_DAIFMT_AC97)
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
fsl_ssi_setup_ac97(ssi_private);
return 0;
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index f6efa9d4acad..b27f25f70730 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -302,6 +302,10 @@ struct sst_hsw {
struct sst_hsw_ipc_dx_reply dx;
void *dx_context;
dma_addr_t dx_context_paddr;
+ enum sst_hsw_device_id dx_dev;
+ enum sst_hsw_device_mclk dx_mclk;
+ enum sst_hsw_device_mode dx_mode;
+ u32 dx_clock_divider;
/* boot */
wait_queue_head_t boot_wait;
@@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
trace_ipc_request("set device config", dev);
- config.ssp_interface = dev;
- config.clock_frequency = mclk;
- config.mode = mode;
- config.clock_divider = clock_divider;
+ hsw->dx_dev = config.ssp_interface = dev;
+ hsw->dx_mclk = config.clock_frequency = mclk;
+ hsw->dx_mode = config.mode = mode;
+ hsw->dx_clock_divider = config.clock_divider = clock_divider;
if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER)
config.channels = 4;
else
@@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw)
return -EIO;
}
- /* Set ADSP SSP port settings */
- ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0,
- SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
- SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ /* Set ADSP SSP port settings - sadly the FW does not store SSP port
+ settings as part of the PM context. */
+ ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk,
+ hsw->dx_mode, hsw->dx_clock_divider);
if (ret < 0)
dev_err(dev, "error: SSP re-initialization failed\n");
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 362c69ac1d6c..53dd085d3ee2 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec;
SNDRV_PCM_FMTBIT_S32_LE | \
SNDRV_PCM_FMTBIT_U32_LE | \
SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+/*
+ * The dummy CODEC is only meant to be used in situations where there is no
+ * actual hardware.
+ *
+ * If there is actual hardware even if it does not have a control bus
+ * the hardware will still have constraints like supported samplerates, etc.
+ * which should be modelled. And the data flow graph also should be modelled
+ * using DAPM.
+ */
static struct snd_soc_dai_driver dummy_dai = {
.name = "snd-soc-dummy-dai",
.playback = {
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-18 06:42:03 +0000