January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Set the card owner field
by Fabio Estevam 14 Jan '15

14 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> The following crash happens when trying to unload the snd_soc_imx_wm8962 module while playback is active: [ 208.666868] Unable to handle kernel paging request at virtc [ 208.674110] pgd = 80004000 [ 208.676867] [7f06541c] *pgd=4c334811, *pte=00000000, *ppte=00000000 [ 208.683211] Internal error: Oops: 80000007 [#1] SMP ARM [ 208.688445] Modules linked in: snd_soc_wm8962 snd_soc_fsl_ssi snd_soc_imx_audmux imx_pcm_fiq evbug] ... In order to avoid such problem, fill the card owner field as suggested by Lars-Peter Clausen: "But looking at the source it seems that this is a core feature of ALSA and at least for the card module itself it will do the ref-counting when a stream is started/stopped. And we even support setting the owner of a card in ASoC. It's just that pretty much no ASoC card driver bothers to set the owner field in the snd_soc_card struct. So this particular problem can be fixed by updating the imx-wm8962 driver to set the owner field." By doing as suggested, we no longer see the crash when attempting to unload the snd_soc_imx_wm8962 module while playback is active: $ modprobe -r snd_soc_imx_wm8962 modprobe: can't unload module snd_soc_imx_wm8962: Resource temporarily unavailable Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 4caacb0..cd146d4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) if (ret) goto clk_fail; data->card.num_links = 1; + data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_wm8962_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); -- 1.9.1

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[alsa-devel] [PATCH - alsa-utils 1/1] configure: allow contitional compilation of alsaucm
by Matthieu Crapet 14 Jan '15

14 Jan '15

If alsa-lib have been compiled with --disable-ucm, alsaucm can't be built. Detection is dynamic, no configure command line option is available for now. Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> diff --git a/Makefile.am b/Makefile.am index 248eb11..5bbe588 100644 --- a/Makefile.am +++ b/Makefile.am @@ -1,6 +1,6 @@ AM_CPPFLAGS=-I$(top_srcdir)/include -SUBDIRS = include alsactl alsaucm utils m4 po alsa-info +SUBDIRS = include alsactl utils m4 po alsa-info if ALSAMIXER SUBDIRS += alsamixer endif @@ -22,6 +22,9 @@ endif if HAVE_SEQ SUBDIRS += seq endif +if HAVE_UCM +SUBDIRS += alsaucm +endif EXTRA_DIST= TODO gitcompile AUTOMAKE_OPTIONS=foreign diff --git a/configure.ac b/configure.ac index 3c7032b..d953e5c 100644 --- a/configure.ac +++ b/configure.ac @@ -48,6 +48,8 @@ AC_CHECK_HEADERS([alsa/rawmidi.h], [have_rawmidi="yes"], [have_rawmidi="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/seq.h], [have_seq="yes"], [have_seq="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/use-case.h], [have_ucm="yes"], [have_ucm="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) @@ -55,6 +57,7 @@ AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") +AM_CONDITIONAL(HAVE_UCM, test "$have_ucm" = "yes") AM_CONDITIONAL(HAVE_SAMPLERATE, test "$have_samplerate" = "yes") dnl Check for librt -- 1.8.5.4

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[alsa-devel] [PATCH - alsalib 1/1] autotools: fix ucm partial build
by Matthieu Crapet 14 Jan '15

14 Jan '15

When --disable-ucm configure option is specified, don't install related include file. Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> diff --git a/include/Makefile.am b/include/Makefile.am index 0127d5c..4baa03a 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -5,7 +5,7 @@ alsaincludedir = ${includedir}/alsa alsainclude_HEADERS = asoundlib.h asoundef.h \ version.h global.h input.h output.h error.h \ - conf.h control.h iatomic.h use-case.h + conf.h control.h iatomic.h if BUILD_CTL_PLUGIN_EXT alsainclude_HEADERS += control_external.h @@ -17,7 +17,7 @@ if BUILD_PCM_PLUGIN alsainclude_HEADERS += pcm_plugin.h endif if BUILD_PCM_PLUGIN_RATE -alsainclude_HEADERS += pcm_rate.h +alsainclude_HEADERS += pcm_rate.h endif if BUILD_PCM_PLUGIN_EXTPLUG alsainclude_HEADERS += pcm_external.h pcm_extplug.h @@ -46,6 +46,10 @@ if BUILD_SEQ alsainclude_HEADERS += seq_event.h seq.h seqmid.h seq_midi_event.h endif +if BUILD_UCM +alsainclude_HEADERS += use-case.h +endif + if BUILD_ALISP alsainclude_HEADERS += alisp.h endif -- 1.8.5.4

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration
by Peter Ujfalusi 14 Jan '15

14 Jan '15

In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha <jsarha(a)ti.com> (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- 2.1.0

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[alsa-devel] [PATCH v9 0/4] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 14 Jan '15

14 Jan '15

Based on 3.19.0-rc3. v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (4): drm/i2c: tda998x: Add DT support for audio drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter drm/i2c: tda998x: set cts_n according to the sample width .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 213 +++++++++++++++++++-- include/sound/tda998x.h | 23 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 175 +++++++++++++++++ 7 files changed, 424 insertions(+), 12 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

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[alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 14 Jan '15

14 Jan '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. Changes since previous version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (16): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 362 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 159 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: fix incorrect indentation on if statement
by Colin King 14 Jan '15

14 Jan '15

From: Colin Ian King <colin.king(a)canonical.com> Minor style issue, statement needs an extra tab indentation on the if statement. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 61d5dc2..dd40ca9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -166,7 +166,7 @@ static int usX2Y_urb_play_prepare(struct snd_usX2Y_substream *subs, /* set the buffer pointer */ urb->transfer_buffer = runtime->dma_area + subs->hwptr * usX2Y->stride; if ((subs->hwptr += count) >= runtime->buffer_size) - subs->hwptr -= runtime->buffer_size; + subs->hwptr -= runtime->buffer_size; } else urb->transfer_buffer = subs->tmpbuf; -- 2.1.4

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[alsa-devel] [PATCH - UCM 1/2] control: enable octal and hexadecimal parse
by han.lu＠intel.com 14 Jan '15

14 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..acaf734 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,10 +59,10 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; - strtol(p, &p, 10); + strtol(p, &p, 0); } if (*p == '%') { val = (long)convert_prange1(strtod(s, NULL), min, max); @@ -87,10 +87,10 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; - strtol(p, &p, 10); + strtol(p, &p, 0); } if (*p == '%') { val = (long long)convert_prange1(strtod(s, NULL), min, max); -- 2.1.0

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[alsa-devel] [PATCH - UCM 1/2] control: enable octal and hexadecimal parse
by han.lu＠intel.com 14 Jan '15

14 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Use zero-base for strtol(), so get_integer() and get_integer64() can parse decimal, octal and hexadecimal data from input string. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..8d6c385 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,7 +59,7 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; strtol(p, &p, 10); @@ -87,7 +87,7 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; strtol(p, &p, 10); -- 1.9.1

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[alsa-devel] Regression with .set_bias_level in pcm512x?
by Peter Rosin 14 Jan '15

14 Jan '15

Hi! I'm working on adding support for DAIFMT_CBM_CFM and DAIFMT_CBM_CFS to pcm512x, and it works nicely in the 3.10 kernel + Atmel-patches were I have developed the code. Now I'm trying to forward-port it and test it on 3.18 + other-Atmel-patches in preparation for submission. However, there seems to be a regression in that the (pre-existing) pcm512x_set_bias_level callback is never called in the 3.18 version. I see that there has been some core work on set_bias_level, and suspect it has something to do with dapm and components, but I think someone else will find the problem much quicker than I am able to. I have feebly tried to revert 0bd2ac3dae74ee25c5ea171cb572731c7a89c248 and 68f831c2724ab72c0088471b2ed1dc99e81948ef, but that didn't help. That also seems a bit backwards. So, what needs to be done in the pcm512x driver to have set_bias_level called? Please help. Cheers, Peter

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