September 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Remove return value checking for gpiochip_remove()
by Axel Lin 17 Sep '14

17 Sep '14

gpiochip_remove() will return void eventually. Thus this patch removes return value checking for gpiochip_remove(). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 5 +---- sound/soc/codecs/wm5100.c | 5 +---- sound/soc/codecs/wm8903.c | 6 +----- sound/soc/codecs/wm8962.c | 5 +---- sound/soc/codecs/wm8996.c | 6 +----- 5 files changed, 5 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9847473..dc978ad 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3274,11 +3274,8 @@ static void rt5677_init_gpio(struct i2c_client *i2c) static void rt5677_free_gpio(struct i2c_client *i2c) { struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c); - int ret; - ret = gpiochip_remove(&rt5677->gpio_chip); - if (ret != 0) - dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&rt5677->gpio_chip); } #else static void rt5677_init_gpio(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 7bb0d36..a01ad62 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2319,11 +2319,8 @@ static void wm5100_init_gpio(struct i2c_client *i2c) static void wm5100_free_gpio(struct i2c_client *i2c) { struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); - int ret; - ret = gpiochip_remove(&wm5100->gpio_chip); - if (ret != 0) - dev_err(&i2c->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm5100->gpio_chip); } #else static void wm5100_init_gpio(struct i2c_client *i2c) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index aa09848..c038b3e 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1877,11 +1877,7 @@ static void wm8903_init_gpio(struct wm8903_priv *wm8903) static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - int ret; - - ret = gpiochip_remove(&wm8903->gpio_chip); - if (ret != 0) - dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8903->gpio_chip); } #else static void wm8903_init_gpio(struct wm8903_priv *wm8903) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1098ae3..9077411 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3398,11 +3398,8 @@ static void wm8962_init_gpio(struct snd_soc_codec *codec) static void wm8962_free_gpio(struct snd_soc_codec *codec) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int ret; - ret = gpiochip_remove(&wm8962->gpio_chip); - if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8962->gpio_chip); } #else static void wm8962_init_gpio(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f16ff4f..b1dcc11 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2216,11 +2216,7 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) static void wm8996_free_gpio(struct wm8996_priv *wm8996) { - int ret; - - ret = gpiochip_remove(&wm8996->gpio_chip); - if (ret != 0) - dev_err(wm8996->dev, "Failed to remove GPIOs: %d\n", ret); + gpiochip_remove(&wm8996->gpio_chip); } #else static void wm8996_init_gpio(struct wm8996_priv *wm8996) -- 1.9.1

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[alsa-devel] [PATCH v6 00/10] Add mrfld DSP topology and widgets
by Subhransu S. Prusty 17 Sep '14

17 Sep '14

This patch series adds DPCM and DAPM widgets to handle DSP topology in the SST platform driver for merrifield. Changes in v6: Removed the init callback as per review comments from Takashi. Subhransu S. Prusty (2): ASoC: Export dapm_kcontrol_get_value ASoC: Intel: mrfld: Use snd_soc_dai_get_drvdata to derive drv data Vinod Koul (8): ASoC: Intel: mfld-pcm: don't call trigger ops to DSP for internal streams ASoC: Intel: mrfld: add bytes control for modules ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mfld-pcm: add control for powering up/down dsp ASoC: Intel: mrfld: add DSP core controls ASoC: Intel: mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: mrfld: add the DSP mixers include/sound/soc-dapm.h | 1 + sound/soc/intel/sst-atom-controls.c | 1259 ++++++++++++++++++++++++++++++- sound/soc/intel/sst-atom-controls.h | 558 ++++++++++++++ sound/soc/intel/sst-mfld-platform-pcm.c | 179 ++++- sound/soc/intel/sst-mfld-platform.h | 7 +- sound/soc/soc-dapm.c | 3 +- 6 files changed, 1970 insertions(+), 37 deletions(-) -- 1.9.0

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[alsa-devel] SGTL5000 on i.MX6 in I2S master mode
by Jean-Michel Hautbois 17 Sep '14

17 Sep '14

Hi, I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I am wondering if I forgot something in my DT or if I need to do something else : sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; }; &i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay"; codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; }; }; &iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>; imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDL_PAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT16__AUD5_TXC 0x130b0 MX6QDL_PAD_DISP0_DAT18__AUD5_TXFS 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; }; }; &ssi1 { status = "okay"; codec-handle = <&codec>; fsl,mode = "i2s-master"; fsl,ssi-asynchronous; }; FYI, AUD5_TXC is wirely connected to AUD5_RXC and AUD5_TXFS is connected to AUD5_RXFS. Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error The kernel is 3.17-rc5. Thanks for reading all this, and for any advice :). JM

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[alsa-devel] [PATCH] ASoC: rt5677: Add the TDM function
by Oder Chiou 17 Sep '14

17 Sep '14

The patch adds the TDM function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 54 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 02bc8bd..1d4719f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -3107,6 +3107,59 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, return 0; } +static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 12); + + switch (slots) { + case 4: + val |= (1 << 10); + break; + case 6: + val |= (2 << 10); + break; + case 8: + val |= (3 << 10); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 8); + break; + case 24: + val |= (2 << 8); + break; + case 32: + val |= (3 << 8); + break; + case 16: + default: + break; + } + + switch (dai->id) { + case RT5677_AIF1: + snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + break; + case RT5677_AIF2: + snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + break; + default: + break; + } + + return 0; +} + static int rt5677_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -3357,6 +3410,7 @@ static struct snd_soc_dai_ops rt5677_aif_dai_ops = { .set_fmt = rt5677_set_dai_fmt, .set_sysclk = rt5677_set_dai_sysclk, .set_pll = rt5677_set_dai_pll, + .set_tdm_slot = rt5677_set_tdm_slot, }; static struct snd_soc_dai_driver rt5677_dai[] = { -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH/alsa-lib trivial] hwdep: add missing declaration for snd_hwdep_poll_descriptors_count()
by Takashi Sakamoto 17 Sep '14

17 Sep '14

The alsa-lib has 'snd_hwdep_poll_descriptors_count()' but hwdep.h has no declaration for the function. No applications can call the function just by including the header. This commit adds a line to the file for this function. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/hwdep.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/hwdep.h b/include/hwdep.h index 6496fa2..ad8bb49 100644 --- a/include/hwdep.h +++ b/include/hwdep.h @@ -102,6 +102,7 @@ typedef struct _snd_hwdep snd_hwdep_t; int snd_hwdep_open(snd_hwdep_t **hwdep, const char *name, int mode); int snd_hwdep_close(snd_hwdep_t *hwdep); int snd_hwdep_poll_descriptors(snd_hwdep_t *hwdep, struct pollfd *pfds, unsigned int space); +int snd_hwdep_poll_descriptors_count(snd_hwdep_t *hwdep); int snd_hwdep_poll_descriptors_revents(snd_hwdep_t *hwdep, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); int snd_hwdep_nonblock(snd_hwdep_t *hwdep, int nonblock); int snd_hwdep_info(snd_hwdep_t *hwdep, snd_hwdep_info_t * info); -- 1.9.1

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[alsa-devel] [PATCH 08/22] ALSA: hda - Replace strnicmp with strncasecmp
by Rasmus Villemoes 17 Sep '14

17 Sep '14

The kernel used to contain two functions for length-delimited, case-insensitive string comparison, strnicmp with correct semantics and a slightly buggy strncasecmp. The latter is the POSIX name, so strnicmp was renamed to strncasecmp, and strnicmp made into a wrapper for the new strncasecmp to avoid breaking existing users. To allow the compat wrapper strnicmp to be removed at some point in the future, and to avoid the extra indirection cost, do s/strnicmp/strncasecmp/g. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rasmus Villemoes <linux(a)rasmusvillemoes.dk> --- sound/pci/hda/hda_sysfs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index e207909..9b49f15 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -514,7 +514,7 @@ enum { static inline int strmatch(const char *a, const char *b) { - return strnicmp(a, b, strlen(b)) == 0; + return strncasecmp(a, b, strlen(b)) == 0; } /* parse the contents after the line "[codec]" -- 2.0.4

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Re: [alsa-devel] coverity fix in alsa-libs
by Takashi Iwai 17 Sep '14

17 Sep '14

At Wed, 17 Sep 2014 03:17:08 +0000 (GMT), Renu Tyagi wrote: > > Hi Takashi, > File in which changes are being made : rawmidi.c > Bug type - Handle h to be closed in case of error before returning > PFA patch. By some reason, your post doesn't appear on alsa-devel ML. Did you subscribe the ML? If not, please subscribe and repost. Also, what about other patches you made? Last but not least: please prepare a patch to be applicable via git-am, including subject and from tags and the changelog content, and don't forget your sign-off to the patch, just like a patch to Linux kernel. thanks, Takashi > > Community Code: > if (err >= 0) > err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); > if (err < 0) > return err; > if (inputp) { > (*inputp)->dl_handle = h; h = NULL; > > Recommended Code : > if (err >= 0) > err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); > if (err < 0){ > if (h) > snd_dlclose(h); > return err; > } > if (inputp) { > (*inputp)->dl_handle = h; h = NULL; > > > Thanks & Regards, > Renu Tyagi > [2 22578.patch <application/octet-stream (base64)>] >

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[alsa-devel] [PATCH] ASoC: fsl_spdif: don't change the root clock rate of spdif in driver
by Shengjiu Wang 17 Sep '14

17 Sep '14

The spdif root clock may be used by other module or defined with CLK_SET_RATE_GATE, so we can't change the clock rate in driver. In this patch remove the clk_set_rate and clk_round_rate to protect the clock. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 24 ++---------------------- 1 file changed, 2 insertions(+), 22 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 70acfe4..f2e4595 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -377,7 +377,6 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, unsigned long csfs = 0; u32 stc, mask, rate; u8 clk, txclk_df, sysclk_df; - int ret; switch (sample_rate) { case 32000: @@ -419,21 +418,6 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, sysclk_df = spdif_priv->sysclk_df[rate]; - /* Don't mess up the clocks from other modules */ - if (clk != STC_TXCLK_SPDIF_ROOT) - goto clk_set_bypass; - - /* - * The S/PDIF block needs a clock of 64 * fs * txclk_df. - * So request 64 * fs * (txclk_df + 1) to get rounded. - */ - ret = clk_set_rate(spdif_priv->txclk[rate], 64 * sample_rate * (txclk_df + 1)); - if (ret) { - dev_err(&pdev->dev, "failed to set tx clock rate\n"); - return ret; - } - -clk_set_bypass: dev_dbg(&pdev->dev, "expected clock rate = %d\n", (64 * sample_rate * txclk_df * sysclk_df)); dev_dbg(&pdev->dev, "actual clock rate = %ld\n", @@ -1056,7 +1040,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, { const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; bool is_sysclk = clk == spdif_priv->sysclk; - u64 rate_ideal, rate_actual, sub; + u64 rate_actual, sub; u32 sysclk_dfmin, sysclk_dfmax; u32 txclk_df, sysclk_df, arate; @@ -1066,11 +1050,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, for (sysclk_df = sysclk_dfmin; sysclk_df <= sysclk_dfmax; sysclk_df++) { for (txclk_df = 1; txclk_df <= 128; txclk_df++) { - rate_ideal = rate[index] * (txclk_df + 1) * 64; - if (round) - rate_actual = clk_round_rate(clk, rate_ideal); - else - rate_actual = clk_get_rate(clk); + rate_actual = clk_get_rate(clk); arate = rate_actual / 64; arate /= txclk_df * sysclk_df; -- 1.7.9.5

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[alsa-devel] Using codec pins to configure part of audio routing subsystem
by Anatol Pomozov 17 Sep '14

17 Sep '14

Hi Here is reiteration from my previous question [1] I have following task. I have realtek rt5677 codec. Codec has 6 pins. Some of the pins are configured as output and used to manipulate part of audio routing subsystem. In my case one pin enables headphones amp, another pin gates DMIC clock. I can easily do it by hard-coding this logic to rt5677 driver. Now I an trying to pull this board-specific logic out of the codec driver code and move it to device tree. I was told (without much explanation) to do it either 2 ways: 1) export codec pins as gpio and then create fixed regulator and tie it to codec via SND_SOC_DAPM_REGULATOR_SUPPLY. I sent a question [1] to clarify if it is the right way. 2) create ALSA controls for pins. I believe it means I need to create controls with names "GPIO1", "GPIO2"..."GPIO6" and then use as part of audio_routing list. What is the right way to achieve my goal. Are there examples of configuration similar to mine? I am sure it is an easy task that was already implemented by someone, but I can't google any useful result. [1] http://comments.gmane.org/gmane.linux.alsa.devel/127508

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[alsa-devel] [PATCH 1/1 linux-next] ASoC: cs35l32: remove second linux/slab.h inclusion
by Fabian Frederick 17 Sep '14

17 Sep '14

linux/slab.h was included twice. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/soc/codecs/cs35l32.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 76f628b..c125925 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -25,7 +25,6 @@ #include <linux/regulator/consumer.h> #include <linux/gpio/consumer.h> #include <linux/of_device.h> -#include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 2.1.0

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