December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Building alsa-driver-1.0.23 for sis7012
by JD 04 Dec '10

04 Dec '10

Hi devels, I would like to build the driver for my sis audio chipset 7012, but I do not see it listed in the list of supported cards, which is printed out from ./configure --help So, if I configure --with-cards=sis7012 will it actually support my chipset? Best regards, JD

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[alsa-devel] [PATCH] ASoC: When disabling WM8994 FLL force a source selection
by Mark Brown 03 Dec '10

03 Dec '10

When we disable the WM8994 FLL code path sharing means that we end up writing out a configuration. Currently this is the currently active input and output frequency (which causes snd_soc_update_bits() to suppress actual writes both immediately and in the common case where we reenable the same configuration later) but we allow machine drivers to pass through a source of zero. Since the register values written are one less than the source constants this causes corruption of other bitfields in the register. Fix this by using the most recently configured FLL source when none is provided. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index af3a98a..b37e95c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1677,6 +1677,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, /* Allow no source specification when stopping */ if (freq_out) return -EINVAL; + src = wm8994->fll[id].src; break; case WM8994_FLL_SRC_MCLK1: case WM8994_FLL_SRC_MCLK2: -- 1.7.1

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[alsa-devel] [PATCH 0/3] ASoC: Laying the groundwork for compress_type overriding
by Dimitris Papastamos 03 Dec '10

03 Dec '10

This patch series is to prepare ASoC for allowing the compress_type of the codec to be overriden by the machine driver. Typically this will be done by providing an snd_soc_codec_conf struct which will contain the new compress_type in the machine driver. Dimitris Papastamos (3): ASoC: Add compress_type as a member to snd_soc_codec ASoC: Change the base value of compress_type ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf

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[alsa-devel] [PATCH] ASoC: Constify struct snd_soc_codec_driver
by Mark Brown 03 Dec '10

03 Dec '10

Allow the CODEC driver structure to be marked const by making all the APIs that use it do so. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 4 ++-- sound/soc/soc-cache.c | 13 +++++++------ sound/soc/soc-core.c | 5 +++-- 3 files changed, 12 insertions(+), 10 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7d53cc4..0eea08e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -265,7 +265,7 @@ int snd_soc_register_platform(struct device *dev, struct snd_soc_platform_driver *platform_drv); void snd_soc_unregister_platform(struct device *dev); int snd_soc_register_codec(struct device *dev, - struct snd_soc_codec_driver *codec_drv, + const struct snd_soc_codec_driver *codec_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_codec(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); @@ -453,7 +453,7 @@ struct snd_soc_codec { const char *name_prefix; int id; struct device *dev; - struct snd_soc_codec_driver *driver; + const struct snd_soc_codec_driver *driver; struct mutex mutex; struct snd_soc_card *card; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 6c6ced7..c0497e1 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -1078,7 +1078,7 @@ static int snd_soc_lzo_decompress_cache_block(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; @@ -1090,7 +1090,7 @@ static inline int snd_soc_lzo_get_blkindex(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, unsigned int reg) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; @@ -1101,7 +1101,7 @@ static inline int snd_soc_lzo_get_blkpos(struct snd_soc_codec *codec, static inline int snd_soc_lzo_get_blksize(struct snd_soc_codec *codec) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; @@ -1301,10 +1301,11 @@ static int snd_soc_lzo_cache_init(struct snd_soc_codec *codec) { struct snd_soc_lzo_ctx **lzo_blocks; size_t reg_size, bmp_size; - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; int ret, tofree, i, blksize, blkcount; const char *p, *end; unsigned long *sync_bmp; + const char *reg_default; ret = 0; codec_drv = codec->driver; @@ -1404,7 +1405,7 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) { int i; int ret; - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; unsigned int val; codec_drv = codec->driver; @@ -1504,7 +1505,7 @@ static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) { - struct snd_soc_codec_driver *codec_drv; + const struct snd_soc_codec_driver *codec_drv; size_t reg_size; codec_drv = codec->driver; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 41fe13d..35cdb8f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3487,8 +3487,9 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) * @codec: codec to register */ int snd_soc_register_codec(struct device *dev, - struct snd_soc_codec_driver *codec_drv, - struct snd_soc_dai_driver *dai_drv, int num_dai) + const struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) { size_t reg_size; struct snd_soc_codec *codec; -- 1.7.1

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[alsa-devel] [RFC 0/2] ASoC: core: Option to reorder widget power sequence
by Peter Ujfalusi 03 Dec '10

03 Dec '10

Hello, Quite some time ago I have faced similar issue with the sequence of the DAPM register writes [1] For that issue, I used a workaround within the twl4030 codec driver. Now I have again issue with the register write sequence. The dac33 codec emmits a pop noise, when user enable the analog bypass path. The route cause in this codec are the DAPM update power and register write sequence: 1. User enable the bypass 2. DAPM power up sequence takes place 2.1 codec DAPM widgets got powered on, codec is enabled 2.2 External speaker enabled 3. The bit enabling the bypass got changed. Step 3 cause a pop, which propagate to the speaker, since it is already enabled. The first patch implements a new flag for the core to reorder the sequence to: 1. User enable the bypass 2. The bit enabling the bypass got changed. 3. DAPM power up sequence takes place 3.1 codec DAPM widgets got powered on, codec is enabled 3.2 External speaker enabled The default sequence is kept, codec driver must ask to have the reordered sequence to avoid any side effect on other codecs. [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030024.html Peter --- Peter Ujfalusi (2): ASoC: core: Reordered DAPM update power on widgets ASoC: tlv320dac33: Request for reordered update power sequence include/sound/soc-dapm.h | 1 + sound/soc/codecs/tlv320dac33.c | 1 + sound/soc/soc-dapm.c | 27 +++++++++++++++++++-------- 3 files changed, 21 insertions(+), 8 deletions(-) -- 1.7.3.2

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[alsa-devel] netbook: pcspeaker doesn't work since 2.6.35 see attached alsa-info-files
by Halim Sahin 03 Dec '10

03 Dec '10

Hi, The subject sais it :-(. In alsamixer I can't umute or set any pc speaker related settings since 2.6.35. The control for pc speaker is gone :-(. Now I started the last working kernel 2.6.34, run alsa-info. Then repeated same with 2.6.36. can you please help to solve this? For accessibility software a working pc-speaker is really useful :-). Regards Halim

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[alsa-devel] [PATCH] ASoC: Don't oops in soc_probe_aux_dev in case of missing codec
by Jarkko Nikula 03 Dec '10

03 Dec '10

Blind copy of codec finding algorithm from soc_bind_dai_link does not work in soc_probe_aux_dev if matching codec name is not found. In that case the code falls through and tries to start the probing procedure with invalid codec pointer. Fix this and add an error print showing the codec name that cannot be found. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/soc-core.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eb950f7..1fd5e72 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1609,7 +1609,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_codec *codec; const char *temp; - int ret = 0; + int ret = -ENODEV; /* find CODEC from registered CODECs*/ list_for_each_entry(codec, &codec_list, list) { @@ -1620,10 +1620,14 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) ret = -EBUSY; goto out; } - break; + goto found; } } + /* codec not found */ + dev_err(card->dev, "asoc: codec %s not found", aux_dev->codec_name); + goto out; +found: if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; -- 1.7.2.3

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[alsa-devel] [PATCH] ASoC: Fix inconsistent meaning of default case while checking alc5623->id
by Axel Lin 03 Dec '10

03 Dec '10

In alc5623_i2c_probe(), the default case for checking alc5623->id behaves the same as case 0x23. However, In alc5623_probe() the default case for checking alc5623->id becomes to be the same as case 0x21. This makes the meaning of default case inconsistent. Since we have checked codec id in alc5623_i2c_probe() by comparing vid2 with id->driver_data, it is not possible to run into the default case now. In case we may add more supported devices to alc5623_i2c_table in the future, this patch changes the default case return -EINVAL to let people know that they should not run into this case. They should also add a new case accordingly for the new id. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Or alternatively, we can simply remove the default cases. Axel sound/soc/codecs/alc5623.c | 10 +++++++--- 1 files changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9783e7e..4f377c9 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -925,7 +925,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) } switch (alc5623->id) { - default: case 0x21: snd_soc_add_controls(codec, rt5621_vol_snd_controls, ARRAY_SIZE(rt5621_vol_snd_controls)); @@ -938,6 +937,8 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, alc5623_vol_snd_controls, ARRAY_SIZE(alc5623_vol_snd_controls)); break; + default: + return -EINVAL; } snd_soc_add_controls(codec, alc5623_snd_controls, @@ -950,7 +951,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); switch (alc5623->id) { - default: case 0x21: case 0x22: snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets, @@ -962,6 +962,8 @@ static int alc5623_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, intercon_spk, ARRAY_SIZE(intercon_spk)); break; + default: + return -EINVAL; } return ret; @@ -1039,10 +1041,12 @@ static int alc5623_i2c_probe(struct i2c_client *client, case 0x22: alc5623_dai.name = "alc5622-hifi"; break; - default: case 0x23: alc5623_dai.name = "alc5623-hifi"; break; + default: + kfree(alc5623); + return -EINVAL; } i2c_set_clientdata(client, alc5623); -- 1.7.2

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[alsa-devel] [PATCH] ASoC: sh: fsi: clean up SPDIF defines
by Kuninori Morimoto 03 Dec '10

03 Dec '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/fsi.c | 17 ++++++++++++++--- 1 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 507e709..76ef3f3 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -50,13 +50,20 @@ /* DO_FMT */ /* DI_FMT */ +#define CR_BWS_24 (0x0 << 20) /* FSI2 */ +#define CR_BWS_16 (0x1 << 20) /* FSI2 */ +#define CR_BWS_20 (0x2 << 20) /* FSI2 */ + +#define CR_DTMD_PCM (0x0 << 8) /* FSI2 */ +#define CR_DTMD_SPDIF_PCM (0x1 << 8) /* FSI2 */ +#define CR_DTMD_SPDIF_STREAM (0x2 << 8) /* FSI2 */ + #define CR_MONO (0x0 << 4) #define CR_MONO_D (0x1 << 4) #define CR_PCM (0x2 << 4) #define CR_I2S (0x3 << 4) #define CR_TDM (0x4 << 4) #define CR_TDM_D (0x5 << 4) -#define CR_SPDIF 0x00100120 /* DOFF_CTL */ /* DIFF_CTL */ @@ -93,6 +100,10 @@ #define IR (1 << 4) /* Interrupt Reset */ #define FSISR (1 << 0) /* Software Reset */ +/* OUT_SEL (FSI2) */ +#define DMMD (1 << 4) /* SPDIF output timing 0: Biphase only */ + /* 1: Biphase and serial */ + /* FIFO_SZ */ #define FIFO_SZ_MASK 0x7 @@ -828,10 +839,10 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "This FSI can not use SPDIF\n"); return -EINVAL; } - data = CR_SPDIF; + data = CR_BWS_16 | CR_DTMD_SPDIF_PCM | CR_PCM; io->chan_num = 2; fsi_spdif_clk_ctrl(fsi, 1); - fsi_reg_mask_set(fsi, OUT_SEL, 0x0010, 0x0010); + fsi_reg_mask_set(fsi, OUT_SEL, DMMD, DMMD); break; default: dev_err(dai->dev, "unknown format.\n"); -- 1.7.1

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[alsa-devel] [PATCH 0/4] ASoC: sh: fsi: remove runtime reg check
by Kuninori Morimoto 03 Dec '10

03 Dec '10

Dear Mark, Liam These are "remove runtime reg check" patches from fsi_xx function. Kuninori Morimoto (4): ASoC: sh: fsi: remove fsi_master_write ASoC: sh: fsi: change fsi->mst_ctrl to master->a/b_mclk ASoC: sh: fsi: remove runtime register check from fsi_reg_xxx ASoC: sh: fsi: remove runtime register check from fsi_master_xxx current runtime check is just useless Best regards -- Kuninori Morimoto

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