June 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Tegra and i.MX are mutually exclusive
by Arnd Bergmann 25 Jun '13

25 Jun '13

Both Tegra and i.MX provide ac97 operations, which can only exist once in the kernel. They can both be built as loadable modules (only one of them needs to be loaded anyway), but we have to disallow enabling them both as built-in. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 3843a18..5958948 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -99,6 +99,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC + depends on SND_SOC_TEGRA=n || (SND_SOC_TEGRA=m && m) help Say Y or M if you want to add support for codecs attached to the i.MX CPUs.

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[alsa-devel] [PATCH] snd-firewire-lib: remove unused header inclusion
by Takashi Sakamoto 25 Jun '13

25 Jun '13

spinlock is not used in amdtp.h. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index b680c5e..f6103d6 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -3,7 +3,6 @@ #include <linux/interrupt.h> #include <linux/mutex.h> -#include <linux/spinlock.h> #include "packets-buffer.h" /** -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - clean up code to reset hda link
by mengdong.lin＠intel.com 25 Jun '13

25 Jun '13

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a cleanup to the previous patch "reset hda link during system/ runtime suspend". In this patch - azx_enter_link_reset() and azx_exit_link_reset() are defined for entering and exiting the link reset respectively. azx_link_reset() is no longer used and replaced by azx_enter_link_reset(). - azx_reset() reuses the above two new functions for a link reset cycle Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f110c7..f39de90 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1121,7 +1121,7 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, #endif /* enter link reset */ -static void azx_reset_link(struct azx *chip) +static void azx_enter_link_reset(struct azx *chip) { unsigned long timeout; @@ -1134,11 +1134,22 @@ static void azx_reset_link(struct azx *chip) usleep_range(500, 1000); } -/* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +/* exit link reset */ +static void azx_exit_link_reset(struct azx *chip) { unsigned long timeout; + azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + +/* reset codec link */ +static int azx_reset(struct azx *chip, int full_reset) +{ if (!full_reset) goto __skip; @@ -1146,12 +1157,7 @@ static int azx_reset(struct azx *chip, int full_reset) azx_writeb(chip, STATESTS, STATESTS_INT_MASK); /* reset controller */ - azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_enter_link_reset(chip); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 @@ -1159,12 +1165,7 @@ static int azx_reset(struct azx *chip, int full_reset) usleep_range(500, 1000); /* Bring controller out of reset */ - azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - - timeout = jiffies + msecs_to_jiffies(100); - while (!azx_readb(chip, GCTL) && - time_before(jiffies, timeout)) - usleep_range(500, 1000); + azx_exit_link_reset(chip); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ usleep_range(1000, 1200); @@ -2908,7 +2909,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2961,7 +2962,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); - azx_reset_link(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); return 0; } -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Add In-driver connection info
by Wang Xingchao 25 Jun '13

25 Jun '13

Pin's connection list may change dynamically with hot-plug event on Intel Haswell chip. Users would see connections be "0" in codec#. when play audio on this pin, software driver choose converter from cache connections. So add "In-driver connection" info to avoid confuse when raw connections are different with cache connection. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_proc.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 0fee8fa..9760f00 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -504,6 +504,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, int conn_len) { int c, curr = -1; + const hda_nid_t *list; + int cache_len; if (conn_len > 1 && wid_type != AC_WID_AUD_MIX && @@ -521,6 +523,19 @@ static void print_conn_list(struct snd_info_buffer *buffer, } snd_iprintf(buffer, "\n"); } + + /* Get Cache connections info */ + cache_len = snd_hda_get_conn_list(codec, nid, &list); + if (cache_len != conn_len + || memcmp(list, conn, conn_len)) { + snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); + if (cache_len > 0) { + snd_iprintf(buffer, " "); + for (c = 0; c < cache_len; c++) + snd_iprintf(buffer, " 0x%02x", list[c]); + snd_iprintf(buffer, "\n"); + } + } } static void print_gpio(struct snd_info_buffer *buffer, -- 1.8.1.2

2 1

[alsa-devel] [PATCH v3 0/3] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 25 Jun '13

25 Jun '13

v3: - fixed documentation for adi,pll-clkdiv - droped implicit I2C_M_STOP - moved regcache_sync() to adau1761_reset() - added regcache_mark_dirty() - made gpio_pll_mode and array - dropped ADAU1701_CLKDIV_MCLK_LRCLK Thanks for the review, Lars! Daniel Mack (3): ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 21 ++ sound/soc/codecs/adau1701.c | 248 ++++++++++++++++----- 2 files changed, 214 insertions(+), 55 deletions(-) -- 1.8.1.4

5 11

Re: [alsa-devel] [PATCH - oss plugin update 1/2] oss: Add support of 32-bit and other formats
by Takashi Iwai 25 Jun '13

25 Jun '13

At Tue, 25 Jun 2013 03:38:11 +0100, Alex Wiggins wrote: > > Thank you for the prompt review Takashi. > > By #if/#endif in this case, do you mean #ifdef to see if the formats are > defined or something else? Not necessarily to be so fine grained. The point is that commenting out multiple lines of codes is a bad style. For disabling multiple lines of codes, use #if 0 or #ifndef XXX or whatever. Takashi > Good spot on the formats array - I guess I didn't have any soundcards with > more than 5 formats but you're right it needs to be increased now. > > I'll have a look at fixing these in the weekend and resubmitting. > > Best Regards, > > Alex > > > On 24 June 2013 08:54, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Sun, 23 Jun 2013 01:25:18 +0100, > > alex.d.wiggins(a)gmail.com wrote: > > > > > > From: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > > > > > Added support of 32-bit and other audio formats via 3 changes: > > > > > > - Additional audio format definitions > > > at the top of the file (if required); > > > > > > - Additional audio formats in oss_hw_params > > > (to allow as input to pass to OSS); > > > > > > - Additional audio formats in oss_hw_constraint > > > (to allow to report to ALSA). > > > > > > Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > > > > > diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c > > > index d43b44c..6d8dea4 100644 > > > --- a/oss/pcm_oss.c > > > +++ b/oss/pcm_oss.c > > > @@ -24,6 +24,32 @@ > > > #include <alsa/pcm_external.h> > > > #include <linux/soundcard.h> > > > > > > +/* definitions for additional OSS formats if required */ > > > +#ifndef AFMT_S32_LE > > > +#define AFMT_S32_LE 0x00001000 > > > +#endif > > > +#ifndef AFMT_S32_BE > > > +#define AFMT_S32_BE 0x00002000 > > > +#endif > > > +#ifndef AFMT_S24_LE > > > +#define AFMT_S24_LE 0x00008000 > > > +#endif > > > +#ifndef AFMT_S24_BE > > > +#define AFMT_S24_BE 0x00010000 > > > +#endif > > > +#ifndef AFMT_S24_PACKED > > > +#define AFMT_S24_PACKED 0x00040000 > > > +#endif > > > +#ifndef AFMT_FLOAT > > > +#define AFMT_FLOAT 0x00004000 > > > +#endif > > > +#ifndef AFMT_SPDIF_RAW > > > +#define AFMT_SPDIF_RAW 0x00020000 > > > +#endif > > > +#ifndef AFMT_VORBIS > > > +#define AFMT_VORBIS 0x00000800 > > > +#endif > > > + > > > typedef struct snd_pcm_oss { > > > snd_pcm_ioplug_t io; > > > char *device; > > > @@ -165,6 +191,143 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, > > > case SND_PCM_FORMAT_S16_BE: > > > oss->format = AFMT_S16_BE; > > > break; > > > + /* allow additional data formats (such as 32 bit) as input > > > + to pass to OSS (set in OSS driver) */ > > > + case SND_PCM_FORMAT_S32_LE: > > > + oss->format = AFMT_S32_LE; > > > + break; > > > + case SND_PCM_FORMAT_S32_BE: > > > + oss->format = AFMT_S32_BE; > > > + break; > > > + case SND_PCM_FORMAT_S8: > > > + oss->format = AFMT_S8; > > > + break; > > > + case SND_PCM_FORMAT_U16_LE: > > > + oss->format = AFMT_U16_LE; > > > + break; > > > + case SND_PCM_FORMAT_U16_BE: > > > + oss->format = AFMT_U16_BE; > > > + break; > > > + case SND_PCM_FORMAT_S24_LE: > > > + oss->format = AFMT_S24_LE; > > > + break; > > > + case SND_PCM_FORMAT_S24_BE: > > > + oss->format = AFMT_S24_BE; > > > + break; > > > + case SND_PCM_FORMAT_MU_LAW: > > > + oss->format = AFMT_MU_LAW; > > > + break; > > > + case SND_PCM_FORMAT_A_LAW: > > > + oss->format = AFMT_A_LAW; > > > + break; > > > + case SND_PCM_FORMAT_IMA_ADPCM: > > > + oss->format = AFMT_IMA_ADPCM; > > > + break; > > > + case SND_PCM_FORMAT_MPEG: > > > + oss->format = AFMT_MPEG; > > > + break; > > > + case SND_PCM_FORMAT_S24_3LE: > > > + oss->format = AFMT_S24_PACKED; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT: > > > + oss->format = AFMT_FLOAT; > > > + break; > > > + case SND_PCM_FORMAT_IEC958_SUBFRAME: > > > + oss->format = AFMT_SPDIF_RAW; > > > + break; > > > + /* following formats need defining checking native endian: > > > + case SND_PCM_FORMAT_S16: > > > + oss->format = AFMT_S16_NE; > > > + break; > > > + case SND_PCM_FORMAT_U16: > > > + oss->format = AFMT_U16_NE; > > > + break; > > > + case SND_PCM_FORMAT_S24: > > > + oss->format = AFMT_S24_NE; > > > + break; > > > + case SND_PCM_FORMAT_S32: > > > + oss->format = AFMT_S32_NE; > > > + */ > > > > Use #if/#endif instead of multi-line comments. > > > > > + /* following formats found in ALSA but not in OSS: > > > + case SND_PCM_FORMAT_U24_LE: > > > + oss->format = AFMT_U24_LE; > > > + break; > > > + case SND_PCM_FORMAT_U24_BE: > > > + oss->format = AFMT_U24_BE; > > > + break; > > > + case SND_PCM_FORMAT_U32_LE: > > > + oss->format = AFMT_U32_LE; > > > + break; > > > + case SND_PCM_FORMAT_U32_BE: > > > + oss->format = AFMT_U32_BE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT_LE: > > > + oss->format = AFMT_FLOAT_LE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT_BE: > > > + oss->format = AFMT_FLOAT_BE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT64_LE: > > > + oss->format = AFMT_FLOAT64_LE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT64_BE: > > > + oss->format = AFMT_FLOAT64_BE; > > > + break; > > > + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: > > > + oss->format = AFMT_SPDIF_RAW_LE; > > > + break; > > > + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: > > > + oss->format = AFMT_SPDIF_RAW_BE; > > > + break; > > > + case SND_PCM_FORMAT_GSM: > > > + oss->format = AFMT_GSM; > > > + break; > > > + case SND_PCM_FORMAT_SPECIAL: > > > + oss->format = AFMT_SPECIAL; > > > + break; > > > + case SND_PCM_FORMAT_S24_3BE: > > > + oss->format = AFMT_S24_PACKED_BE; > > > + break; > > > + case SND_PCM_FORMAT_U24_3LE: > > > + oss->format = AFMT_U24_PACKED; > > > + break; > > > + case SND_PCM_FORMAT_U24_3BE: > > > + oss->format = AFMT_U24_PACKED_BE; > > > + break; > > > + case SND_PCM_FORMAT_S20_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_S20_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U20_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U20_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_S18_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_S18_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U18_3LE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U18_3BE: > > > + oss->format = ; > > > + break; > > > + case SND_PCM_FORMAT_U24: > > > + oss->format = AFMT_U24_NE; > > > + break; > > > + case SND_PCM_FORMAT_U32: > > > + oss->format = AFMT_U32_NE; > > > + break; > > > + case SND_PCM_FORMAT_FLOAT64: > > > + oss->format = AFMT_FLOAT64; > > > + break; > > > + */ > > > > Ditto. > > > > > > > default: > > > fprintf(stderr, "*** OSS: unsupported format %s\n", > > snd_pcm_format_name(io->format)); > > > return -EINVAL; > > > @@ -273,6 +436,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > > > format[nformats++] = SND_PCM_FORMAT_S16_BE; > > > if (tmp & AFMT_MU_LAW) > > > format[nformats++] = SND_PCM_FORMAT_MU_LAW; > > > + /* allow additional data formats (such as 32 bit) > > > + to be reported from OSS to ALSA below */ > > > + if (tmp & AFMT_S32_LE) > > > + format[nformats++] = SND_PCM_FORMAT_S32_LE; > > > + if (tmp & AFMT_S32_BE) > > > + format[nformats++] = SND_PCM_FORMAT_S32_BE; > > > + if (tmp & AFMT_A_LAW) > > > + format[nformats++] = SND_PCM_FORMAT_A_LAW; > > > + if (tmp & AFMT_IMA_ADPCM) > > > + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; > > > + if (tmp & AFMT_S8) > > > + format[nformats++] = SND_PCM_FORMAT_S8; > > > + if (tmp & AFMT_U16_LE) > > > + format[nformats++] = SND_PCM_FORMAT_U16_LE; > > > + if (tmp & AFMT_U16_BE) > > > + format[nformats++] = SND_PCM_FORMAT_U16_BE ; > > > + if (tmp & AFMT_MPEG) > > > + format[nformats++] = SND_PCM_FORMAT_MPEG; > > > + if (tmp & AFMT_FLOAT) > > > + format[nformats++] = SND_PCM_FORMAT_FLOAT; > > > + if (tmp & AFMT_S24_LE) > > > + format[nformats++] = SND_PCM_FORMAT_S24_LE; > > > + if (tmp & AFMT_S24_BE) > > > + format[nformats++] = SND_PCM_FORMAT_S24_BE; > > > + if (tmp & AFMT_SPDIF_RAW) > > > + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; > > > + if (tmp & AFMT_S24_PACKED) > > > + format[nformats++] = SND_PCM_FORMAT_S24_3LE; > > > + /* following formats found in OSS but not in ALSA */ > > > + if (tmp & AFMT_AC3) > > > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > > > + if (tmp & AFMT_VORBIS) > > > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > > > + /* assume signed 16 bit (native CPU endian) if not found above */ > > > if (! nformats) > > > format[nformats++] = SND_PCM_FORMAT_S16; > > > if ((err = snd_pcm_ioplug_set_param_list(io, > > SND_PCM_IOPLUG_HW_FORMAT, > > > > You need to adjust format[] array size if you add more possible > > formats. > > > > > > thanks, > > > > Takashi > > > [2 <text/html; ISO-8859-1 (quoted-printable)>] >

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[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused variable
by Wang Xingchao 25 Jun '13

25 Jun '13

Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..0f7a380 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1891,7 +1891,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- 1.8.1.2

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[alsa-devel] [PATCH v4 0/4] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 24 Jun '13

24 Jun '13

v4: - added one more patch to move the firmware download, for more clarity. - postpone the firmware download to the first call of hw_params(), so we don't need to pass a default pll div in DT. - drop adi,pll-clkdiv entirely v3: - fixed documentation for adi,pll-clkdiv - droped implicit I2C_M_STOP - moved regcache_sync() to adau1761_reset() - added regcache_mark_dirty() - made gpio_pll_mode and array - dropped ADAU1701_CLKDIV_MCLK_LRCLK Daniel Mack (4): ASoC: codecs: adau1701: move firmware download to adau1701_reset() ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 12 + sound/soc/codecs/adau1701.c | 277 ++++++++++++++++----- 2 files changed, 232 insertions(+), 57 deletions(-) -- 1.8.1.4

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[alsa-devel] [PATCH 0/3] Support auto-mute via amp in HD-audio generic driver
by Takashi Iwai 24 Jun '13

24 Jun '13

Hi, this is a patch set to add the support of the headphone/line-out auto- mute via amp mute instead of pin-control values. The motivation of this is that we found that some machines don't honor the pin-control values but just ignore and keep speaker on. So, instead of muting via the pin control value, we need to mute via the amp value of the pin. The mechanism to add an extra mute check in software is relatively easy: just add the new mute flag and check it in the mixer put callback. The only tricky part is to update all relevant controls by jack plug/unplug or the auto-mute switch. For that, the new helper function, snd_ctl_vmaster_sync(), is added, so that the codec driver just updates the bit mask and call the sync at the last. There should be no behavior change unless the new flag is set. [PATCH 1/3] ALSA: vmaster: Add snd_ctl_sync_vmaster() helper function [PATCH 2/3] ALSA: hda - Add auto_mute_via_amp flag to generic parser [PATCH 3/3] ALSA: hda - Use auto_mute_via_amp=1 for VT1708 Takashi

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[alsa-devel] [GIT PULL] ASoC updates for v3.10
by Mark Brown 24 Jun '13

24 Jun '13

The following changes since commit 7d132055814ef17a6c7b69f342244c410a5e000f: Linux 3.10-rc6 (2013-06-15 11:51:07 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.10-rc7 for you to fetch changes up to b12a530682ec725ff7a95871d8003d1753e315b8: Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus (2013-06-19 19:49:23 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.10 A few small updates, one small core fix and several simple driver ones. The wm8962 changes are a little large but are pretty important. ---------------------------------------------------------------- Axel Lin (1): ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI Charles Keepax (1): ASoC: wm5110: Correct rate control for DSP4 Lars-Peter Clausen (1): ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table Mark Brown (4): Merge remote-tracking branch 'asoc/fix/adav80x' into asoc-linus Merge remote-tracking branch 'asoc/fix/adsp' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Nicolin Chen (2): ASoC: wm8962: Remove remaining direct register cache accesses ASoC: wm8962: Enable start-up and normal bias after reset in runtime resume sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8962.c | 35 ++++++++++++++++++++++++----------- sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/soc-dapm.c | 2 ++ 4 files changed, 28 insertions(+), 13 deletions(-)

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[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused variable
by Wang Xingchao 24 Jun '13

24 Jun '13

Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..0f7a380 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1891,7 +1891,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- 1.8.1.2

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[alsa-devel] [RFC PATCH] ALSA: hda - reset hda link during system/runtime suspend
by mengdong.lin＠intel.com 24 Jun '13

24 Jun '13

From: Mengdong Lin <mengdong.lin(a)intel.com> If all the codecs report ClkStopOK (OK to stop bus clock) after being put to D3, this patch will reset the HDA link before the controller is put to D3. So the link will be in reset during system or runtime suspend, the bus clock stops and the codecs are in D3(ClkStop) state. This may help to reduce power consumption by dozens of mW on some peripheral hda codecs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f089fa0..9f110c7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1120,6 +1120,20 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct snd_dma_buffer *dmab); #endif +/* enter link reset */ +static void azx_reset_link(struct azx *chip) +{ + unsigned long timeout; + + /* reset controller */ + azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); + + timeout = jiffies + msecs_to_jiffies(100); + while ((azx_readb(chip, GCTL) & ICH6_GCTL_RESET) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); +} + /* reset codec link */ static int azx_reset(struct azx *chip, int full_reset) { @@ -2894,6 +2908,7 @@ static int azx_suspend(struct device *dev) if (chip->initialized) snd_hda_suspend(chip->bus); azx_stop_chip(chip); + azx_reset_link(chip); if (chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; @@ -2946,6 +2961,7 @@ static int azx_runtime_suspend(struct device *dev) struct azx *chip = card->private_data; azx_stop_chip(chip); + azx_reset_link(chip); azx_clear_irq_pending(chip); return 0; } -- 1.7.10.4

3 2

Re: [alsa-devel] [PATCH - oss plugin update 2/2] oss: Add user variables to set period and buffer config
by Takashi Iwai 24 Jun '13

24 Jun '13

At Sun, 23 Jun 2013 01:25:19 +0100, alex.d.wiggins(a)gmail.com wrote: > > From: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > Added 4 user variables to control the OSS period and buffer config: > > - period_size_allowany (boolean): > if not set then queries OSS for period size > (by using SNDCTL_DSP_GETBLKSIZE), > if set to true then allows any power of 2 period size; > I'd suggest to put an underscore between "allow" and "any". Or, instead of adding a flag... > - period_size (integer): > sets the OSS period size (SND_PCM_IOPLUG_HW_PERIOD_BYTES); ... handle allowany when period_size < 0, for example. Takashi > - periods (integer): > sets the number of OSS periods (SND_PCM_IOPLUG_HW_PERIODS); > > - buffer_size (integer): > sets the OSS buffer size (SND_PCM_IOPLUG_HW_BUFFER_BYTES). > > Note that ALSA performs sanity checks to prevent conflicts. > > Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c > index 6d8dea4..7131e54 100644 > --- a/oss/pcm_oss.c > +++ b/oss/pcm_oss.c > @@ -60,6 +60,13 @@ typedef struct snd_pcm_oss { > unsigned int period_shift; > unsigned int periods; > unsigned int frame_bytes; > + /* user variables set using ALSA config files */ > + /* included here as this snd_pcm_oss_t struct > + is stored in io->private_data */ > + int user_period_size_allowany; > + long int user_period_size; > + long int user_periods; > + long int user_buffer_size; > } snd_pcm_oss_t; > > static snd_pcm_sframes_t oss_write(snd_pcm_ioplug_t *io, > @@ -406,6 +413,7 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > unsigned int nchannels; > unsigned int channel[6]; > /* period and buffer bytes must be power of two */ > + /* minimum: 2^8 = 256; maximum: 2^23 = 8388608 */ > static const unsigned int bytes_list[] = { > 1U<<8, 1U<<9, 1U<<10, 1U<<11, 1U<<12, 1U<<13, 1U<<14, 1U<<15, > 1U<<16, 1U<<17, 1U<<18, 1U<<19, 1U<<20, 1U<<21, 1U<<22, 1U<<23 > @@ -498,20 +506,96 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > if (err < 0) > return err; > > - /* period size (in power of two) */ > - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, > - ARRAY_SIZE(bytes_list), bytes_list); > - if (err < 0) > - return err; > - /* periods */ > - err = snd_pcm_ioplug_set_param_minmax(io, SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); > - if (err < 0) > - return err; > - /* buffer size (in power of two) */ > - err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, > - ARRAY_SIZE(bytes_list), bytes_list); > - if (err < 0) > - return err; > + /* period and buffer sizes: default behaviour below is to > + query period size from OSS */ > + > + /* consider user set paramaters > + (period_size_allowany; period_size; periods; buffer_size) */ > + /* note: ALSA performs sanity checks > + to prevent conflicting parameters */ > + > + /* period_size_allowany and period_size */ > + if (oss->user_period_size_allowany) { > + /* period_size_allowany flag set: > + allow any power of 2 period size */ > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, > + ARRAY_SIZE(bytes_list), bytes_list); > + if (err < 0) > + return err; > + } else { > + if (oss->user_period_size > 0) { > + /* period_size set to positive value: > + use user set period size */ > + const unsigned int user_period_size > + = oss->user_period_size; > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, 1, > + &user_period_size); > + } else { > + /* query OSS for period size (called fragment size) */ > + unsigned int fragsize; > + if (ioctl(oss->fd, SNDCTL_DSP_GETBLKSIZE, &fragsize) > + < 0) > + fprintf(stderr, > +"*** alsa-plugins-oss: error querying SNDCTL_DSP_GETBLKSIZE.\n"); > + > + /* note: second query of SNDCTL_DSP_GETBLKSIZE > + can give different result > + (i.e. 4096 instead of 2048) */ > + /* in oss_audio_core.c getblksize function, > + 4096 is the default return value: > + so second time probably not reliable */ > + > + /* period size (in power of two) */ > + unsigned int x = fragsize; > + /* check if fragsize is a power of two */ > + if ((x != 0) && ((x & (~x + 1)) == x)) > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, > + 1, &fragsize); > + else > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIOD_BYTES, > + ARRAY_SIZE(bytes_list), bytes_list); > + if (err < 0) > + return err; > + } > + } > + > + /* periods (number of periods per buffer) */ > + if (oss->user_periods > 0) { > + /* force number of periods to user set periods */ > + const unsigned int user_periods = oss->user_periods; > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_PERIODS, 1, &user_periods); > + if (err < 0) > + return err; > + } else { > + /* allow any number of periods between 2 and 1024 inclusive > + (ALSA can choose) */ > + err = snd_pcm_ioplug_set_param_minmax(io, > + SND_PCM_IOPLUG_HW_PERIODS, 2, 1024); > + if (err < 0) > + return err; > + } > + > + /* buffer_size */ > + if (oss->user_buffer_size > 0) { > + /* force buffer size to user set buffer_size */ > + const unsigned int user_buffer_size = oss->user_buffer_size; > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_BUFFER_BYTES, 1, &user_buffer_size); > + if (err < 0) > + return err; > + } else { > + /* buffer size (in power of two) */ > + err = snd_pcm_ioplug_set_param_list(io, > + SND_PCM_IOPLUG_HW_BUFFER_BYTES, > + ARRAY_SIZE(bytes_list), bytes_list); > + if (err < 0) > + return err; > + } > > return 0; > } > @@ -556,6 +640,11 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) > const char *device = "/dev/dsp"; > int err; > snd_pcm_oss_t *oss; > + /* default values for user variables*/ > + int user_period_size_allowany = 0; > + long int user_period_size = 0; > + long int user_periods = 0; > + long int user_buffer_size = 0; > > snd_config_for_each(i, next, conf) { > snd_config_t *n = snd_config_iterator_entry(i); > @@ -571,6 +660,39 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) > } > continue; > } > + > + /* user variables */ > + if (strcmp(id, "period_size_allowany") == 0) { > + if (snd_config_get_bool(n) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } else > + user_period_size_allowany = > + snd_config_get_bool(n); > + continue; > + } > + if (strcmp(id, "period_size") == 0) { > + if (snd_config_get_integer(n, &user_period_size) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } > + continue; > + } > + if (strcmp(id, "periods") == 0) { > + if (snd_config_get_integer(n, &user_periods) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } > + continue; > + } > + if (strcmp(id, "buffer_size") == 0) { > + if (snd_config_get_integer(n, &user_buffer_size) < 0) { > + SNDERR("Invalid type for %s", id); > + return -EINVAL; > + } > + continue; > + } > + > SNDERR("Unknown field %s", id); > return -EINVAL; > } > @@ -595,6 +717,12 @@ SND_PCM_PLUGIN_DEFINE_FUNC(oss) > goto error; > } > > + /* save user variables from ALSA config files */ > + oss->user_period_size_allowany = user_period_size_allowany; > + oss->user_period_size = user_period_size; > + oss->user_periods = user_periods; > + oss->user_buffer_size = user_buffer_size; > + > oss->io.version = SND_PCM_IOPLUG_VERSION; > oss->io.name = "ALSA <-> OSS PCM I/O Plugin"; > oss->io.poll_fd = oss->fd; > -- > 1.8.3.1 >

1 0

Re: [alsa-devel] [PATCH - oss plugin update 1/2] oss: Add support of 32-bit and other formats
by Takashi Iwai 24 Jun '13

24 Jun '13

At Sun, 23 Jun 2013 01:25:18 +0100, alex.d.wiggins(a)gmail.com wrote: > > From: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > Added support of 32-bit and other audio formats via 3 changes: > > - Additional audio format definitions > at the top of the file (if required); > > - Additional audio formats in oss_hw_params > (to allow as input to pass to OSS); > > - Additional audio formats in oss_hw_constraint > (to allow to report to ALSA). > > Signed-off-by: Alex Wiggins <alex.d.wiggins(a)gmail.com> > > diff --git a/oss/pcm_oss.c b/oss/pcm_oss.c > index d43b44c..6d8dea4 100644 > --- a/oss/pcm_oss.c > +++ b/oss/pcm_oss.c > @@ -24,6 +24,32 @@ > #include <alsa/pcm_external.h> > #include <linux/soundcard.h> > > +/* definitions for additional OSS formats if required */ > +#ifndef AFMT_S32_LE > +#define AFMT_S32_LE 0x00001000 > +#endif > +#ifndef AFMT_S32_BE > +#define AFMT_S32_BE 0x00002000 > +#endif > +#ifndef AFMT_S24_LE > +#define AFMT_S24_LE 0x00008000 > +#endif > +#ifndef AFMT_S24_BE > +#define AFMT_S24_BE 0x00010000 > +#endif > +#ifndef AFMT_S24_PACKED > +#define AFMT_S24_PACKED 0x00040000 > +#endif > +#ifndef AFMT_FLOAT > +#define AFMT_FLOAT 0x00004000 > +#endif > +#ifndef AFMT_SPDIF_RAW > +#define AFMT_SPDIF_RAW 0x00020000 > +#endif > +#ifndef AFMT_VORBIS > +#define AFMT_VORBIS 0x00000800 > +#endif > + > typedef struct snd_pcm_oss { > snd_pcm_ioplug_t io; > char *device; > @@ -165,6 +191,143 @@ static int oss_hw_params(snd_pcm_ioplug_t *io, > case SND_PCM_FORMAT_S16_BE: > oss->format = AFMT_S16_BE; > break; > + /* allow additional data formats (such as 32 bit) as input > + to pass to OSS (set in OSS driver) */ > + case SND_PCM_FORMAT_S32_LE: > + oss->format = AFMT_S32_LE; > + break; > + case SND_PCM_FORMAT_S32_BE: > + oss->format = AFMT_S32_BE; > + break; > + case SND_PCM_FORMAT_S8: > + oss->format = AFMT_S8; > + break; > + case SND_PCM_FORMAT_U16_LE: > + oss->format = AFMT_U16_LE; > + break; > + case SND_PCM_FORMAT_U16_BE: > + oss->format = AFMT_U16_BE; > + break; > + case SND_PCM_FORMAT_S24_LE: > + oss->format = AFMT_S24_LE; > + break; > + case SND_PCM_FORMAT_S24_BE: > + oss->format = AFMT_S24_BE; > + break; > + case SND_PCM_FORMAT_MU_LAW: > + oss->format = AFMT_MU_LAW; > + break; > + case SND_PCM_FORMAT_A_LAW: > + oss->format = AFMT_A_LAW; > + break; > + case SND_PCM_FORMAT_IMA_ADPCM: > + oss->format = AFMT_IMA_ADPCM; > + break; > + case SND_PCM_FORMAT_MPEG: > + oss->format = AFMT_MPEG; > + break; > + case SND_PCM_FORMAT_S24_3LE: > + oss->format = AFMT_S24_PACKED; > + break; > + case SND_PCM_FORMAT_FLOAT: > + oss->format = AFMT_FLOAT; > + break; > + case SND_PCM_FORMAT_IEC958_SUBFRAME: > + oss->format = AFMT_SPDIF_RAW; > + break; > + /* following formats need defining checking native endian: > + case SND_PCM_FORMAT_S16: > + oss->format = AFMT_S16_NE; > + break; > + case SND_PCM_FORMAT_U16: > + oss->format = AFMT_U16_NE; > + break; > + case SND_PCM_FORMAT_S24: > + oss->format = AFMT_S24_NE; > + break; > + case SND_PCM_FORMAT_S32: > + oss->format = AFMT_S32_NE; > + */ Use #if/#endif instead of multi-line comments. > + /* following formats found in ALSA but not in OSS: > + case SND_PCM_FORMAT_U24_LE: > + oss->format = AFMT_U24_LE; > + break; > + case SND_PCM_FORMAT_U24_BE: > + oss->format = AFMT_U24_BE; > + break; > + case SND_PCM_FORMAT_U32_LE: > + oss->format = AFMT_U32_LE; > + break; > + case SND_PCM_FORMAT_U32_BE: > + oss->format = AFMT_U32_BE; > + break; > + case SND_PCM_FORMAT_FLOAT_LE: > + oss->format = AFMT_FLOAT_LE; > + break; > + case SND_PCM_FORMAT_FLOAT_BE: > + oss->format = AFMT_FLOAT_BE; > + break; > + case SND_PCM_FORMAT_FLOAT64_LE: > + oss->format = AFMT_FLOAT64_LE; > + break; > + case SND_PCM_FORMAT_FLOAT64_BE: > + oss->format = AFMT_FLOAT64_BE; > + break; > + case SND_PCM_FORMAT_IEC958_SUBFRAME_LE: > + oss->format = AFMT_SPDIF_RAW_LE; > + break; > + case SND_PCM_FORMAT_IEC958_SUBFRAME_BE: > + oss->format = AFMT_SPDIF_RAW_BE; > + break; > + case SND_PCM_FORMAT_GSM: > + oss->format = AFMT_GSM; > + break; > + case SND_PCM_FORMAT_SPECIAL: > + oss->format = AFMT_SPECIAL; > + break; > + case SND_PCM_FORMAT_S24_3BE: > + oss->format = AFMT_S24_PACKED_BE; > + break; > + case SND_PCM_FORMAT_U24_3LE: > + oss->format = AFMT_U24_PACKED; > + break; > + case SND_PCM_FORMAT_U24_3BE: > + oss->format = AFMT_U24_PACKED_BE; > + break; > + case SND_PCM_FORMAT_S20_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_S20_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U20_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U20_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_S18_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_S18_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U18_3LE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U18_3BE: > + oss->format = ; > + break; > + case SND_PCM_FORMAT_U24: > + oss->format = AFMT_U24_NE; > + break; > + case SND_PCM_FORMAT_U32: > + oss->format = AFMT_U32_NE; > + break; > + case SND_PCM_FORMAT_FLOAT64: > + oss->format = AFMT_FLOAT64; > + break; > + */ Ditto. > default: > fprintf(stderr, "*** OSS: unsupported format %s\n", snd_pcm_format_name(io->format)); > return -EINVAL; > @@ -273,6 +436,40 @@ static int oss_hw_constraint(snd_pcm_oss_t *oss) > format[nformats++] = SND_PCM_FORMAT_S16_BE; > if (tmp & AFMT_MU_LAW) > format[nformats++] = SND_PCM_FORMAT_MU_LAW; > + /* allow additional data formats (such as 32 bit) > + to be reported from OSS to ALSA below */ > + if (tmp & AFMT_S32_LE) > + format[nformats++] = SND_PCM_FORMAT_S32_LE; > + if (tmp & AFMT_S32_BE) > + format[nformats++] = SND_PCM_FORMAT_S32_BE; > + if (tmp & AFMT_A_LAW) > + format[nformats++] = SND_PCM_FORMAT_A_LAW; > + if (tmp & AFMT_IMA_ADPCM) > + format[nformats++] = SND_PCM_FORMAT_IMA_ADPCM; > + if (tmp & AFMT_S8) > + format[nformats++] = SND_PCM_FORMAT_S8; > + if (tmp & AFMT_U16_LE) > + format[nformats++] = SND_PCM_FORMAT_U16_LE; > + if (tmp & AFMT_U16_BE) > + format[nformats++] = SND_PCM_FORMAT_U16_BE ; > + if (tmp & AFMT_MPEG) > + format[nformats++] = SND_PCM_FORMAT_MPEG; > + if (tmp & AFMT_FLOAT) > + format[nformats++] = SND_PCM_FORMAT_FLOAT; > + if (tmp & AFMT_S24_LE) > + format[nformats++] = SND_PCM_FORMAT_S24_LE; > + if (tmp & AFMT_S24_BE) > + format[nformats++] = SND_PCM_FORMAT_S24_BE; > + if (tmp & AFMT_SPDIF_RAW) > + format[nformats++] = SND_PCM_FORMAT_IEC958_SUBFRAME; > + if (tmp & AFMT_S24_PACKED) > + format[nformats++] = SND_PCM_FORMAT_S24_3LE; > + /* following formats found in OSS but not in ALSA */ > + if (tmp & AFMT_AC3) > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > + if (tmp & AFMT_VORBIS) > + format[nformats++] = SND_PCM_FORMAT_UNKNOWN; > + /* assume signed 16 bit (native CPU endian) if not found above */ > if (! nformats) > format[nformats++] = SND_PCM_FORMAT_S16; > if ((err = snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, You need to adjust format[] array size if you add more possible formats. thanks, Takashi

1 0

[alsa-devel] [PATCH] Add M2Tech hiFace USB-SPDIF driver
by Antonio Ospite 24 Jun '13

24 Jun '13

Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- Hi, this is my first ALSA driver I still have some doubts and I marked them in the comments below with a XXX prefix, could someone please take a look at them? I plan on submitting a version 2 of the driver after I get some feedback, especially on those doubts. I also noticed that many drivers under sound/usb/ have text in the Kconfig entry stating "Say Y here to include support for ..." but I only see "[N/m/?]" as options when I run "make oldconfig" maybe because of some dependency, I am not sure, should such text be removed in order to avoid confusion? Thanks, Antonio sound/usb/Kconfig | 33 +++ sound/usb/Makefile | 2 +- sound/usb/hiface/Makefile | 2 + sound/usb/hiface/chip.c | 331 ++++++++++++++++++++++ sound/usb/hiface/chip.h | 34 +++ sound/usb/hiface/pcm.c | 666 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/hiface/pcm.h | 24 ++ 7 files changed, 1091 insertions(+), 1 deletion(-) create mode 100644 sound/usb/hiface/Makefile create mode 100644 sound/usb/hiface/chip.c create mode 100644 sound/usb/hiface/chip.h create mode 100644 sound/usb/hiface/pcm.c create mode 100644 sound/usb/hiface/pcm.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd7..2b206e8 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,38 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Say Y here to include support for M2Tech hiFace USB-SPDIF interface. + + M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output + Interface, see http://www.m2tech.biz/hiface.html + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc..abe668f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 0000000..463b136 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 0000000..dce74e9 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,331 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/initval.h> + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_AUTHOR("Antonio Ospite <ao2(a)amarulasolutions.com>"); +MODULE_DESCRIPTION("M2Tech HiFace USB audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{HiFace, Evo}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "HiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static struct hiface_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_dev_free(struct snd_device *device) +{ + struct hiface_chip *chip = device->device_data; + kfree(chip); + return 0; +} + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + static struct snd_device_ops ops = { + .dev_free = hiface_dev_free, + }; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot create alsa card.\n"); + return ret; + } + + strcpy(card->driver, DRIVER_NAME); + + if (quirk && quirk->device_name) + strcpy(card->shortname, quirk->device_name); + else + strcpy(card->shortname, "M2Tech generic audio"); + + sprintf(card->longname, "%s at %d:%d", card->shortname, + device->bus->busnum, device->devnum); + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_card_free(card); + return -ENOMEM; + } + + chip->dev = device; + chip->index = idx; + chip->card = card; + + ret = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (ret < 0) { + kfree(chip); + snd_card_free(card); + return ret; + } + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + pr_info("Probe " DRIVER_NAME " driver.\n"); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + snd_printk(KERN_ERR "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(&register_mutex); + for (i = 0; i < SNDRV_CARDS; i++) { + if (chips[i] && chips[i]->dev == device) { + if (chips[i]->shutdown) { + snd_printk(KERN_ERR CARD_NAME " device is in the shutdown state, cannot create a card instance\n"); + ret = -ENODEV; + goto err; + } + chip = chips[i]; + break; + } + } + if (!chip) { + /* it's a fresh one. + * now look for an empty slot and create a new card instance + */ + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i] && !chips[i]) { + ret = hiface_chip_create(device, i, quirk, + &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + break; + } + if (!chip) { + snd_printk(KERN_ERR "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + } + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + chips[chip->index] = chip; + chip->intf_count++; + + mutex_unlock(&register_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(&register_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + pr_debug("%s: called.\n", __func__); + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + chip->intf_count--; + if (chip->intf_count <= 0) { + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + mutex_lock(&register_mutex); + chips[chip->index] = NULL; + mutex_unlock(&register_mutex); + + chip->shutdown = true; + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); + } +} + +static struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 0000000..c2e9fda --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,34 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include <linux/usb.h> +#include <sound/core.h> + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + int intf_count; /* number of registered interfaces */ + int index; /* index in module parameter arrays */ + bool shutdown; + + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 0000000..34fe0ed --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,666 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#include <linux/slab.h> +#include <sound/pcm.h> + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_MAX_PACKET_SIZE 4096 +#define MAX_BUFSIZE (2 * PCM_N_URBS * PCM_MAX_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 rate; /* one of PCM_RATE_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates) - 2, /* by default rates up to 192000 are supported */ + .list = rates, + .mask = 0, +}; + +static const int rates_alsaid[] = { + SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000, + SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000, + SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000, + SNDRV_PCM_RATE_KNOT, SNDRV_PCM_RATE_KNOT }; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_BUFSIZE, + .period_bytes_min = PCM_MAX_PACKET_SIZE, + .period_bytes_max = MAX_BUFSIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + u8 rate_value[] = { 0x43, 0x4b, 0x42, 0x4a, 0x40, 0x48, 0x58, 0x68 }; + int ret; + struct usb_device *device = rt->chip->dev; + + /* We are already sure that the rate is supported here thanks to the + * contraints set in hiface_pcm_open(), but we have to retrieve the + * index to access rate_value. + */ + for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + if (rate == rates[rt->rate]) + break; + + if (unlikely(rt->rate == ARRAY_SIZE(rates))) { + pr_err("Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + 0xb0, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value[rt->rate], 0, NULL, 0, 100); + if (ret < 0) { + snd_printk(KERN_ERR "Error setting samplerate %d.\n", + rates[rt->rate]); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream( + struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + pr_debug("Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0, i; + + if (rt->stream_state == STREAM_DISABLED) { + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_MAX_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + pr_debug("%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + + +/* The hardware wants word-swapped 32-bit values */ +void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +static int hiface_pcm_playback(struct pcm_substream *sub, + struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + u8 *source; + unsigned int pcm_buffer_size; + + /* XXX Can an invalid format make it to here? + * Isn't ALSA checking pcm_hw.formats ? + */ + if (alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE) { + pr_err("Unsupported sample format\n"); + return -EINVAL; + } + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_MAX_PACKET_SIZE <= pcm_buffer_size) { + pr_debug("%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_MAX_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + pr_debug("%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_MAX_PACKET_SIZE - len); + } + sub->dma_off += PCM_MAX_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_MAX_PACKET_SIZE; + + return 0; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (usb_urb->status || rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) { + int ret; + + ret = hiface_pcm_playback(sub, out_urb); + if (ret < 0) { + spin_unlock_irqrestore(&sub->lock, flags); + goto out_fail; + } + if (sub->period_off >= sub->instance->runtime->period_size) { + sub->period_off %= sub->instance->runtime->period_size; + spin_unlock_irqrestore(&sub->lock, flags); + snd_pcm_period_elapsed(sub->instance); + } else { + spin_unlock_irqrestore(&sub->lock, flags); + } + } else { + memset(out_urb->buffer, 0, PCM_MAX_PACKET_SIZE); + spin_unlock_irqrestore(&sub->lock, flags); + } +out_fail: + usb_submit_urb(&out_urb->instance, GFP_ATOMIC); +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + /* XXX can we avoid setting hw.rates below? + * + * FWIU setting alsa_rt->hw.rates in the open callback can be + * useful in two cases: + * + * - different hardware revisions have different + * capabilities + * + * - different stream types must have the same capabilities + * + * but in our case it should not be needed as we only have one + * playback stream and we define the capabilities via + * constraints already to avoid invalid rates to be used. + * + * Moreover, following the code stolen from 6fire it seems + * impossible to get to setting alsa_rt->hw.rates here: + * + * hiface_pcm_init: + * rt->rate = ARRAY_SIZE(rates); + * + * hiface_pcm_open: + * if (rt->rate < ARRAY_SIZE(rates)) + * alsa_rt->hw.rates = rates_alsaid[rt->rate]; + * // So it does not get here on the first open + * + * hiface_pcm_prepare: + * hiface_pcm_set_rate: + * for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + * if (rate == rates[rt->rate]) + * break; + * + * hiface_pcm_close: + * rt->rate = ARRAY_SIZE(rates); + * // So it won't set alsa_rt->hw.rates at the next open either + * + * So the only way to set alsa_rt->hw.rates is to have two + * calls to the open callback, is that even possible? ALSA + * does not allow sharing PCM streams AFAIK and we use only + * one playback stream. + * + * Maybe the 6fire driver was setting alsa_rt->hw.rates in + * order to have the playback stream and the capture stream + * use the same rate? I.e. the first open,prepare,set_rate() + * sequence on one type of stream will decide the rate and the + * second open on the other type would restrict the supported + * rate to the one already in use, but I am not sure. + * + * If the rationale makes sense then we don't need to set + * alsa_rt->hw.rates and we can remove this and related code. + */ + if (rt->rate < ARRAY_SIZE(rates)) + alsa_rt->hw.rates = rates_alsaid[rt->rate]; + + sub = &rt->playback; + } + + if (!sub) { + mutex_unlock(&rt->stream_mutex); + pr_err("Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rate_max = 384000; + constraints_rates.count = ARRAY_SIZE(rates); + } + + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + pr_debug("%s: called.\n", __func__); + + mutex_lock(&rt->stream_mutex); + if (sub) { + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + /* all substreams closed? if so, stop streaming */ + if (!rt->playback.instance) { + hiface_pcm_stream_stop(rt); + rt->rate = ARRAY_SIZE(rates); + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_free_pages(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&sub->lock, flags); + sub->active = true; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&sub->lock, flags); + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_STATE_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_MAX_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + if (rt->playback.instance) { + snd_pcm_stop(rt->playback.instance, + SNDRV_PCM_STATE_XRUN); + } + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + pr_debug("%s: called.\n", __func__); + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + rt->rate = ARRAY_SIZE(rates); + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + pr_err("Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strcpy(pcm->name, "USB-SPDIF Audio"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MAX_BUFSIZE, MAX_BUFSIZE); + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 0000000..ddc9fe7 --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ -- 1.7.10.4

6 22

Re: [alsa-devel] [kbuild] sound/soc/soc-fw.c:945:9-16: ERROR: se is NULL but dereferenced.
by Fengguang Wu 23 Jun '13

23 Jun '13

On Sat, Jun 22, 2013 at 12:35:44PM -0300, Raphael S Carvalho wrote: > I'm in a Linux event now, and I can't clone the tree (probably the port for > that purpose isn't allowed) > Then I sending a simplified patch that shows which lines must be changed. Patch looks good to me. Thanks! Reviewed-by: Fengguang Wu <fengguang.wu(a)intel.com> > Signed-off: Raphael S. Carvalho <raphael.scarv(a)gmail.com> > > --- soc-fw_old.c 2013-06-22 12:13:48.000000000 -0300 > +++ soc-fw_new.c 2013-06-22 12:19:41.000000000 -0300 > @@ -893,7 +893,7 @@ > if (err < 0) { > dev_err(sfw->dev, "ASoC: could not create" > " values for %s\n", ec->hdr.name); > - goto err_se; > + goto err; > } > /* fall through to create texts */ > case SOC_CONTROL_TYPE_ENUM: > @@ -905,13 +905,13 @@ > if (err < 0) { > dev_err(sfw->dev, "ASoC: could not create" > " texts for %s\n", ec->hdr.name); > - goto err_se; > + goto err; > } > break; > default: > dev_err(sfw->dev, "ASoC: invalid enum control type %d for %s\n", > ec->hdr.index, ec->hdr.name); > - goto err_se; > + goto err; > } > > /* map standard io handlers and check for external handlers */ > @@ -928,21 +928,19 @@ > SOC_CONTROL_GET_ID_GET(ec->hdr.index), > SOC_CONTROL_GET_ID_PUT(ec->hdr.index), > SOC_CONTROL_GET_ID_INFO(ec->hdr.index)); > - goto err_se; > + goto err; > } > > err = soc_fw_init_kcontrol(sfw, kc); > if (err < 0) { > dev_err(sfw->dev, "ASoC: failed to init %s\n", > ec->hdr.name); > - goto err_se; > + goto err; > } > } > return kc; > > -err_se: > - kfree(kc); > - > +err: > /* free values and texts */ > if (se->dvalues) > kfree(se->dvalues); > @@ -950,7 +948,8 @@ > kfree(se->dtexts[i]); > > kfree(se); > - > +err_se: > + kfree(kc); > return NULL; > } > > > > On Sat, Jun 22, 2013 at 11:48 AM, Fengguang Wu <fengguang.wu(a)intel.com>wrote: > > > TO: Liam Girdwood <lrg(a)ti.com> > > CC: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > > > > > Hi Liam, > > > > FYI, there are coccinelle warnings in > > > > tree: git://gitorious.org/omap-audio/linux-audiopeter/topic/ti-audio-next-bnw-wip > > head: ac1981249f0c606cede98d6dab091ecb1ae480c9 > > commit: 6d98dbbcfb5da2176b3e60faaf86d2279790fe67 ASoC: firmware: Add > > support for dynamic components > > date: 5 weeks ago > > :::::: branch date: 5 days ago > > :::::: commit date: 5 weeks ago > > > > >> sound/soc/soc-fw.c:945:9-16: ERROR: se is NULL but dereferenced. > > -- > > >> sound/soc/soc-fw.c:1523:8-9: ERROR: reference preceded by free on line > > 1523 > > >> sound/soc/soc-fw.c:1443:8-9: ERROR: reference preceded by free on line > > 1443 > > >> sound/soc/soc-fw.c:1483:8-9: ERROR: reference preceded by free on line > > 1483 > > -- > > >> sound/soc/soc-fw.c:1528:6-8: ERROR: invalid reference to the index > > variable of the iterator on line 1513 > > >> sound/soc/soc-fw.c:1448:6-8: ERROR: invalid reference to the index > > variable of the iterator on line 1433 > > >> sound/soc/soc-fw.c:1488:6-8: ERROR: invalid reference to the index > > variable of the iterator on line 1473 > > > > git remote add omap-audio git://gitorious.org/omap-audio/linux-audio > > git remote update omap-audio > > git checkout 6d98dbbcfb5da2176b3e60faaf86d2279790fe67 > > vim +945 sound/soc/soc-fw.c > > > > 6d98dbbc Liam Girdwood 2012-12-05 929 goto > > err_se; > > 6d98dbbc Liam Girdwood 2012-12-05 930 } > > 6d98dbbc Liam Girdwood 2012-12-05 931 > > 6d98dbbc Liam Girdwood 2012-12-05 932 err = > > soc_fw_init_kcontrol(sfw, kc); > > 6d98dbbc Liam Girdwood 2012-12-05 933 if (err < 0) { > > 6d98dbbc Liam Girdwood 2012-12-05 934 > > dev_err(sfw->dev, "ASoC: failed to init %s\n", > > 6d98dbbc Liam Girdwood 2012-12-05 935 > > ec->hdr.name); > > 6d98dbbc Liam Girdwood 2012-12-05 936 goto > > err_se; > > 6d98dbbc Liam Girdwood 2012-12-05 937 } > > 6d98dbbc Liam Girdwood 2012-12-05 938 } > > 6d98dbbc Liam Girdwood 2012-12-05 939 return kc; > > 6d98dbbc Liam Girdwood 2012-12-05 940 > > 6d98dbbc Liam Girdwood 2012-12-05 941 err_se: > > 6d98dbbc Liam Girdwood 2012-12-05 942 kfree(kc); > > 6d98dbbc Liam Girdwood 2012-12-05 943 > > 6d98dbbc Liam Girdwood 2012-12-05 944 /* free values and texts */ > > 6d98dbbc Liam Girdwood 2012-12-05 @945 if (se->dvalues) > > 6d98dbbc Liam Girdwood 2012-12-05 946 kfree(se->dvalues); > > 6d98dbbc Liam Girdwood 2012-12-05 947 for (i = 0; i < ec->max; > > i++) > > 6d98dbbc Liam Girdwood 2012-12-05 948 > > kfree(se->dtexts[i]); > > 6d98dbbc Liam Girdwood 2012-12-05 949 > > 6d98dbbc Liam Girdwood 2012-12-05 950 kfree(se); > > 6d98dbbc Liam Girdwood 2012-12-05 951 > > 6d98dbbc Liam Girdwood 2012-12-05 952 return NULL; > > 6d98dbbc Liam Girdwood 2012-12-05 953 } > > > > --- > > 0-DAY kernel build testing backend Open Source Technology > > Center > > http://lists.01.org/mailman/listinfo/kbuild Intel > > Corporation > > _______________________________________________ > > kbuild mailing list > > kbuild(a)lists.01.org > > https://lists.01.org/mailman/listinfo/kbuild > >

1 0

[alsa-devel] [GIT PULL] ASoC updates for v3.11
by Mark Brown 21 Jun '13

21 Jun '13

The following changes since commit 7d132055814ef17a6c7b69f342244c410a5e000f: Linux 3.10-rc6 (2013-06-15 11:51:07 -1000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.11 for you to fetch changes up to 84bbc4aa2de97143db0f6ef0c3fa0bd84f73b207: Merge remote-tracking branch 'asoc/topic/x86' into asoc-next (2013-06-17 17:20:33 +0100) ---------------------------------------------------------------- ASoC: Updates for v3.11 Not a big release subsystem wise, the main changes have been some nice improvements on the driver side: - Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500. - Generalisation of the Bluetooth and HDMI stub drivers. - New CODEC drivers for SSM2518 and RT5640. - New machine driver for Tegra CPUs with RT5640. ---------------------------------------------------------------- Andrew Bresticker (1): ASoC: max98090: add digital mic mux to record path Axel Lin (2): ASoC: ssm2518: Fix trivial typo in checking tx_mask and rx_mask values ASoC: adav80x: Select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI Bard Liao (1): ASoC: add RT5640 CODEC driver Barry Song (2): ASoC: dfbmcs320: make the driver common for other BT modules ASoC: omap-hdmi-codec: make the driver common for other SoCs Boris BREZILLON (1): ASoC: atmel-ssc: prepare clk before calling enable Charles Keepax (10): ASoC: wm5102: Correct OSR control name for EPOUT mfd: wm5102: Manually apply register patch mfd: arizona: Change fast_start pdata name to better reflect functionality ASoC: bells: Hookup DMICs for Bells ASoC: wm0010: Set IRQ as a wake source ASoC: wm0010: Clear IRQ as wake source and include missing header ASoC: arizona: Hookup SYSCLK to inputs and noise generators mfd: wm5102: Expose DRE control registers ASoC: wm5102: Expose controls for DRE ASoC: wm5110: Correct rate control for DSP4 Dan Carpenter (2): ASoC: wm_adsp: memory leak in wm_adsp_create_control() ASoC: wm8994: missing break in wm8994_get_fll_config() Daniel Mack (3): ASoC: adau1701: refactor firmware loading function ASoC: codecs: adau1701: add DT bindings ASoC: codecs: adau1701: factor out firmware reset Dimitris Papastamos (2): ASoC: wm_adsp: Expose coefficient blocks as ALSA binary controls ASoC: wm_adsp: Ensure set controls are synced on each boot Emil Goode (1): ASoC: Fix double assignment of .owner in struct snd_soc_card Fabio Baltieri (15): ASoC: ux500: register controls to card instead of codec ASoC: ux500: drop clock gating widgets from machine driver ASoC: ab8500-codec: Add missing ad_to_slot definitions ASoC: ab8500-codec: Set tx dai slots from tx_mask ASoC: ab8500-codec: Set rx dai slots from rx_mask ASoC: ux500: Drop pinctrl sleep support ASoC: ab8500-codec: Move codec ops on a separate structure ASoC: ux500: Drop dangling struct i2s_controller ASoC: ux500: Drop unused code from msp headers ASoC: ux500: Add missing mop500_ab8500.h include ASoC: ux500: Drop redundant msp id enumerations ASoC: ux500: Ensure consistent configuration between DAIs ASoC: ux500: Move DMA parameters into ux500_msp ASoC: ux500: Set DMA address during device init ASoC: ux500: Add DMA slave config prepare routine Fabio Estevam (15): ASoC: imx-sgtl5000: Do not enter the error path on success ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable ASoC: mxs-sgtl5000: Remove unneeded fields from snd_soc_dai_link ASoC: fsl: imx-audmux: Let device core handle pinctrl ASoC: mxs: mxs-saif: Let device core handle pinctrl ASoC: sgtl5000: Fix comment about register addresses ASoC: sgtl5000: Convert to use regmap directly ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable ASoC: mxs-sgtl5000: Remove unneeded fields from snd_soc_dai_link ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe() ASoC: sgtl5000: Fix driver probe after reset ASoC: sgtl5000: Let the codec acquire its clock ASoC: sgtl5000: Use i2c_get_clientdata() ASoC: sgtl5000: Add 'clocks' entry as a required propery ASoC: imx-sgtl5000: Use devm_clk_get() Geert Uytterhoeven (1): blackfin: bf533-stamp: Remove bogus "||" Kuninori Morimoto (1): ASoC: fsi: fixup sparse errors Lars-Peter Clausen (36): ASoC: spear: Setup dma data in DAI probe ASoC: spear: Use generic dmaengine PCM ASoC: jz4740-i2s: Use clk_prepare_enable/clk_disable_unprepare ASoC: ep93xx: Setup dma data in DAI probe ASoC: ep93xx: Use generic dmaengine PCM ASoC: core: Move snd_soc_set_runtime_hwparams() to soc-pcm.c ASoC: core: Add helper function to initialize the runtime pcm ASoC: core: Use kasprintf instead of opencoding it ASoC: ep93xx-i2s: Staticize non exported struct ASoC: kirkwood-dma: Staticize non exported struct ASoC: spear: spdif_in: Staticize non exported struct ASoC: spear: spdif_out: Staticize unexported function ASoC: mmp-pcm: Staticize non exported structs and functions ASoC: mmp-sspa: Staticize non exported struct ASoC: bf5xx-tdm-pcm: Staticize non exported struct ASoC: mop500_ab8500: Staticize non exported functions ASoC: mop500: Staticize non exported struct ASoC: sn95031: Staticize non exported struct ASoC: davinci-sffsdr: Staticize non exported struct ASoC: Fix early event callback list iteration ASoC: Add ssm2518 support ASoC: blackfin: Remove unused bf5xx-{i2s, tdm, ac97}-pcm.h ASoC: blackfin: bf5xx-i2s: Use dev_{err, dbg} instead of pr_{error, debug} ASoC: blackfin: bf5xx-sport: Allow setting rx and tx mask independently ASoC: blackfin: bf5xx-i2s: Allocate buffer only as large as requested ASoC: blackfin: bf5xx-i2s-pcm: Use snd_pcm_lib_preallocate_pages_for_all() ASoC: blackfin: bf5xx-i2s: Add support for TDM mode ASoC: blackfin: Switch bf5xx-ad193x from bf5xx-tdm to bf5xx-i2s ASoC: blackfin: Switch bf5xx-ad1836 from bf5xx-tdm to bf5xx-i2s ASoC: blackfin: Remove bf5xx-tdm driver blackfin: Remove references to the bf5x_tdm driver ASoC: dapm: Setup private_free callback for dapm kcontrols ASoC: dapm: Remove unnecessary loop ASoC: dapm: Add a helper function to free a DAPM path ASoC: dapm: Remove unused long_name field from snd_soc_dapm_path struct ASoC: dapm: Add snd_soc_dapm_switch to the power up/down sequence table Marek Belisko (3): ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. ASoC: spdif_transmitter: Add DT support. ASoC: spdif_receiver: Add DT support. Marek Vasut (1): ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order Mark Brown (68): ASoC: sam9g20ek: Let device core handle pinctrl ASoC: dapm: Provide early event callbacks for power up and down ASoC: wm8994: Support EFS mode for FLL mfd: arizona: Read the device identification information after boot mfd: arizona: Add missing cleanup on remove mfd: arizona: Disable interrupts during suspend mfd: Add device tree bindings for Arizona class devices mfd: arizona: Support use of external DCVDD ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A ASoC: arizona: Provide simple DAI ops for autoconfiguring interfaces ASoC: wm5102: Stub hookup for Slimbus interface ASoC: wm5110: Stub hookup for Slimbus interface Merge tag 'v3.10-rc2' into asoc-ux500 mfd: wm5110: Make DSPn_STATUS_3 readable Merge tag 'mfd-arizona-v3.10-rc2' of git://git.kernel.org/.../broonie/misc into asoc-arizona ASoC: wm8994: Remove restore of DAC enable state ASoC: wm8994: Ensure lambda is zeroed for WM8994 ASoC: wm8994: Defer declaration of open circuit microphones ASoC: wm8994: Add digital loopback paths ASoC: wm8994: Remove unused variable ASoC: wm8994: Allow debounce before MICDET identification ASoC: sgtl5000: Make device cache only when powered off ASoC: max98090: Guard runtime PM callbacks Merge branch 'topic/fsl' of git://git.kernel.org/.../broonie/sound into asoc-sgtl5000 ASoC: wm8962: Restore device state after reset in runtime resume Merge branch 'fix/wm8962' of git://git.kernel.org/.../broonie/sound into asoc-wm8962 ASoC: SPEAr: Add dependency on dmaengine helpers ASoC: SPEAr: Hide component drivers in Kconfig ASoC: spear: Remove nonexistant EVM options ASoC: spear: Normalise module names Merge remote-tracking branch 'asoc/fix/adav80x' into asoc-linus Merge remote-tracking branch 'asoc/fix/adsp' into asoc-linus Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8962' into asoc-linus Merge remote-tracking branch 'asoc/topic/88pm860x' into asoc-next Merge remote-tracking branch 'asoc/topic/adau1701' into asoc-next Merge remote-tracking branch 'asoc/topic/adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/atmel' into asoc-next Merge remote-tracking branch 'asoc/topic/blackfin' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next Merge remote-tracking branch 'asoc/topic/dfmbcs320' into asoc-next Merge remote-tracking branch 'asoc/topic/dwc' into asoc-next Merge remote-tracking branch 'asoc/topic/ep93xx' into asoc-next Merge remote-tracking branch 'asoc/topic/fsi' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next Merge remote-tracking branch 'asoc/topic/hdmi' into asoc-next Merge remote-tracking branch 'asoc/topic/jz4740' into asoc-next Merge remote-tracking branch 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branch 'asoc/topic/max98090' into asoc-next Merge remote-tracking branch 'asoc/topic/mxs' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5640' into asoc-next Merge remote-tracking branch 'asoc/topic/samsung' into asoc-next Merge remote-tracking branch 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branch 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branch 'asoc/topic/spdif' into asoc-next Merge remote-tracking branch 'asoc/topic/spear' into asoc-next Merge remote-tracking branch 'asoc/topic/ssm2518' into asoc-next Merge remote-tracking branch 'asoc/topic/tegra' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Merge remote-tracking branch 'asoc/topic/wm0010' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8994' into asoc-next Merge remote-tracking branch 'asoc/topic/x86' into asoc-next Michał Mirosław (1): ASoC: fix kernel message grepability Nicolin Chen (5): ASoC: WM8962: Create default platform data structure ASoC: wm8962: Add device tree binding ASoC: fsl: add imx-wm8962 machine driver ASoC: wm8962: Remove remaining direct register cache accesses ASoC: wm8962: Enable start-up and normal bias after reset in runtime resume Paul Bolle (2): ASoC: remove saarb and tavorevb3 machine drivers ASoC: OMAP: Remove obsolete Makefile line Rajeev Kumar (3): ASoC: dwc: debug message correction. ASoC: dwc: Folder path correction in file header. ASoC: Add Kconfig and Makefile to support SPEAr audio driver Sachin Kamat (7): ASoC: 88pm860x: Remove redundant platform_set_drvdata() ASoC: ep93xx: Remove redundant platform_set_drvdata() ASoC: jz4740: Remove redundant platform_set_drvdata() ASoC: omap-mcbsp: Remove redundant platform_set_drvdata() ASoC: Samsung: Remove redundant platform_set_drvdata() ASoC: mid-x86: Remove redundant platform_set_drvdata() ASoC: davinci: Remove redundant use of of_match_ptr macro Sekhar Nori (1): ASoC: davinci: remove sffsdr machine support Shawn Guo (6): ASoC: mxs: move to use generic DMA helper ASoC: fsl: remove use of imx-pcm-audio from fsl_ssi ASoC: fsl: remove use of imx-pcm-audio from imx-ssi ASoC: fsl: create function imx_pcm_fiq_exit() ASoC: fsl: remove use of imx-fiq-pcm-audio from imx-ssi ASoC: fsl: remove imx-pcm driver Stephen Warren (7): ASoC: tegra: always use clk_get() in utility code ASoC: tegra: implement suspend/resume for Tegra30 AHUB ASoC: tegra: implement suspend/resume for Tegra30 I2S ASoC: tegra: add runtime PM to resume functions ASoC: rt5640: add device tree support ASoC: tegra: add tegra+RT5640 machine driver ASoC: rt5640: fix sparse warnings Tushar Behera (1): ASoC: spear: Convert to use devm_ioremap_resource Uwe Kleine-König (3): ASoC: SPEAr spdif_{in,out}: use devm for clk and a few more cleanups ASoC: simplify registration of snd-soc-dummy device ASoC: SPEAr spdif_{in,out}: fix fallout of previous cleanup Documentation/devicetree/bindings/mfd/arizona.txt | 62 + .../devicetree/bindings/sound/adi,adau1701.txt | 23 + .../devicetree/bindings/sound/imx-audio-wm8962.txt | 46 + .../devicetree/bindings/sound/mxs-saif.txt | 17 +- .../bindings/sound/nvidia,tegra-audio-rt5640.txt | 71 + Documentation/devicetree/bindings/sound/rt5640.txt | 30 + .../devicetree/bindings/sound/sgtl5000.txt | 3 + .../devicetree/bindings/sound/spdif-receiver.txt | 10 + .../bindings/sound/spdif-transmitter.txt | 10 + .../devicetree/bindings/sound/ssm2518.txt | 20 + Documentation/devicetree/bindings/sound/wm8962.txt | 23 + arch/blackfin/mach-bf527/boards/ad7160eval.c | 12 - arch/blackfin/mach-bf527/boards/ezkit.c | 32 +- arch/blackfin/mach-bf533/boards/ezkit.c | 12 - arch/blackfin/mach-bf533/boards/stamp.c | 35 +- arch/blackfin/mach-bf537/boards/stamp.c | 30 +- arch/blackfin/mach-bf548/boards/ezkit.c | 28 +- arch/blackfin/mach-bf561/boards/ezkit.c | 14 +- arch/blackfin/mach-bf609/boards/ezkit.c | 2 +- drivers/mfd/arizona-core.c | 234 ++- drivers/mfd/arizona-i2c.c | 10 +- drivers/mfd/arizona-spi.c | 10 +- drivers/mfd/arizona.h | 12 + drivers/mfd/wm5102-tables.c | 9 +- drivers/mfd/wm5110-tables.c | 8 + drivers/misc/atmel-ssc.c | 8 +- include/linux/mfd/arizona/core.h | 2 + include/linux/mfd/arizona/pdata.h | 2 +- include/linux/mfd/arizona/registers.h | 47 + include/linux/mfd/wm8994/pdata.h | 5 + include/linux/mfd/wm8994/registers.h | 8 + include/linux/platform_data/ssm2518.h | 22 + include/sound/rt5640.h | 22 + include/sound/soc-dapm.h | 3 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/sam9g20_wm8731.c | 9 - sound/soc/blackfin/Kconfig | 47 +- sound/soc/blackfin/Makefile | 4 - sound/soc/blackfin/bf5xx-ac97-pcm.c | 1 - sound/soc/blackfin/bf5xx-ac97-pcm.h | 26 - sound/soc/blackfin/bf5xx-ac97.c | 8 +- sound/soc/blackfin/bf5xx-ad1836.c | 19 +- sound/soc/blackfin/bf5xx-ad193x.c | 40 +- sound/soc/blackfin/bf5xx-ad1980.c | 1 - sound/soc/blackfin/bf5xx-ad73311.c | 1 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 183 +- sound/soc/blackfin/bf5xx-i2s-pcm.h | 21 +- sound/soc/blackfin/bf5xx-i2s.c | 129 +- sound/soc/blackfin/bf5xx-sport.c | 10 +- sound/soc/blackfin/bf5xx-sport.h | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 1 - sound/soc/blackfin/bf5xx-tdm-pcm.c | 345 ---- sound/soc/blackfin/bf5xx-tdm-pcm.h | 18 - sound/soc/blackfin/bf5xx-tdm.c | 328 --- sound/soc/blackfin/bf5xx-tdm.h | 23 - sound/soc/cirrus/Kconfig | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 16 +- sound/soc/cirrus/ep93xx-i2s.c | 16 +- sound/soc/cirrus/ep93xx-pcm.c | 138 +- sound/soc/codecs/88pm860x-codec.c | 9 +- sound/soc/codecs/Kconfig | 18 +- sound/soc/codecs/Makefile | 14 +- sound/soc/codecs/ab8500-codec.c | 85 +- sound/soc/codecs/ab8500-codec.h | 42 +- sound/soc/codecs/adau1701.c | 69 +- sound/soc/codecs/arizona.c | 7 + sound/soc/codecs/arizona.h | 3 +- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +- sound/soc/codecs/jz4740.c | 2 - sound/soc/codecs/max98090.c | 24 +- sound/soc/codecs/rt5640.c | 2128 ++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2092 +++++++++++++++++++ sound/soc/codecs/sgtl5000.c | 267 ++- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/spdif_receiver.c | 10 + .../{spdif_transciever.c => spdif_transmitter.c} | 10 + sound/soc/codecs/ssm2518.c | 856 ++++++++ sound/soc/codecs/ssm2518.h | 20 + sound/soc/codecs/wm0010.c | 10 + sound/soc/codecs/wm5102.c | 205 +- sound/soc/codecs/wm5110.c | 192 ++ sound/soc/codecs/wm8962.c | 143 +- sound/soc/codecs/wm8994.c | 188 +- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_adsp.c | 461 ++++- sound/soc/codecs/wm_adsp.h | 3 + sound/soc/davinci/Kconfig | 10 - sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-mcasp.c | 4 +- sound/soc/davinci/davinci-sffsdr.c | 181 -- sound/soc/dwc/designware_i2s.c | 6 +- sound/soc/fsl/Kconfig | 17 +- sound/soc/fsl/Makefile | 13 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 +- sound/soc/fsl/imx-audmux.c | 8 - sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 + sound/soc/fsl/imx-pcm.c | 145 -- sound/soc/fsl/imx-pcm.h | 10 +- sound/soc/fsl/imx-sgtl5000.c | 39 +- sound/soc/fsl/imx-ssi.c | 44 +- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/imx-wm8962.c | 323 +++ sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/jz4740/jz4740-i2s.c | 17 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mid-x86/mfld_machine.c | 1 - sound/soc/mxs/mxs-pcm.c | 18 - sound/soc/mxs/mxs-pcm.h | 7 - sound/soc/mxs/mxs-saif.c | 37 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 10 +- sound/soc/omap/Makefile | 1 - sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 - sound/soc/pxa/Kconfig | 20 - sound/soc/pxa/Makefile | 4 - sound/soc/pxa/mmp-pcm.c | 6 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/saarb.c | 190 -- sound/soc/pxa/tavorevb3.c | 189 -- sound/soc/pxa/zylonite.c | 1 - sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/bells.c | 14 + sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/smdk_wm8580pcm.c | 1 - sound/soc/samsung/smdk_wm8994pcm.c | 1 - sound/soc/sh/fsi.c | 10 +- sound/soc/soc-core.c | 106 +- sound/soc/soc-dapm.c | 110 +- sound/soc/soc-pcm.c | 91 +- sound/soc/soc-utils.c | 13 +- sound/soc/spear/Kconfig | 9 + sound/soc/spear/Makefile | 8 + sound/soc/spear/spdif_in.c | 31 +- sound/soc/spear/spdif_out.c | 43 +- sound/soc/spear/spear_pcm.c | 152 +- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra30_ahub.c | 25 + sound/soc/tegra/tegra30_i2s.c | 26 + sound/soc/tegra/tegra_asoc_utils.c | 23 +- sound/soc/tegra/tegra_rt5640.c | 257 +++ sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/mop500_ab8500.c | 62 +- sound/soc/ux500/ux500_msp_dai.c | 11 +- sound/soc/ux500/ux500_msp_dai.h | 4 - sound/soc/ux500/ux500_msp_i2s.c | 88 +- sound/soc/ux500/ux500_msp_i2s.h | 74 +- sound/soc/ux500/ux500_pcm.c | 30 + 157 files changed, 8786 insertions(+), 3053 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/arizona.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1701.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5640.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt create mode 100644 Documentation/devicetree/bindings/sound/ssm2518.txt create mode 100644 include/linux/platform_data/ssm2518.h create mode 100644 include/sound/rt5640.h delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-tdm.c delete mode 100644 sound/soc/blackfin/bf5xx-tdm.h rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) create mode 100644 sound/soc/codecs/rt5640.c create mode 100644 sound/soc/codecs/rt5640.h rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) create mode 100644 sound/soc/codecs/ssm2518.c create mode 100644 sound/soc/codecs/ssm2518.h delete mode 100644 sound/soc/davinci/davinci-sffsdr.c delete mode 100644 sound/soc/fsl/imx-pcm.c create mode 100644 sound/soc/fsl/imx-wm8962.c delete mode 100644 sound/soc/pxa/saarb.c delete mode 100644 sound/soc/pxa/tavorevb3.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/tegra/tegra_rt5640.c

2 1

[alsa-devel] [PATCH 0/6] Convert JZ4740 to dmaengine
by Lars-Peter Clausen 21 Jun '13

21 Jun '13

Hi, This series replaces the custom JZ4740 DMA API with a dmaengine driver. This is done in 3 steps: 1) Add a dmaengine driver which wraps the custom JZ4740 DMA API 2) Update all users of the JZ4740 DMA API to use dmaengine instead 3) Remove the custom API and move all direct hardware access to the dmaengine driver. The first two patches in the series also make sure that the clock of the DMA core is enabled. Since the patches in this series depend on each other I'd prefer if they could all go through the DMA tree. There are a few minor changes to the patch 3 in v2 of this series, all other patches are identical to v1. - Lars Lars-Peter Clausen (4): dma: Add a jz4740 dmaengine driver MIPS: jz4740: Register jz4740 DMA device ASoC: jz4740: Use the generic dmaengine PCM driver MIPS: jz4740: Remove custom DMA API Maarten ter Huurne (2): MIPS: jz4740: Correct clock gate bit for DMA controller MIPS: jz4740: Acquire and enable DMA controller clock arch/mips/include/asm/mach-jz4740/dma.h | 56 --- arch/mips/include/asm/mach-jz4740/platform.h | 1 + arch/mips/jz4740/Makefile | 2 +- arch/mips/jz4740/board-qi_lb60.c | 1 + arch/mips/jz4740/clock.c | 2 +- arch/mips/jz4740/dma.c | 287 ------------- arch/mips/jz4740/platform.c | 21 + drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/dma-jz4740.c | 617 +++++++++++++++++++++++++++ sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 48 +-- sound/soc/jz4740/jz4740-pcm.c | 310 +------------- sound/soc/jz4740/jz4740-pcm.h | 20 - 14 files changed, 676 insertions(+), 697 deletions(-) delete mode 100644 arch/mips/jz4740/dma.c create mode 100644 drivers/dma/dma-jz4740.c delete mode 100644 sound/soc/jz4740/jz4740-pcm.h -- 1.8.2.1

3 17

[alsa-devel] [PATCH 0/4] ALSA: usb: misc consistency fixes
by Antonio Ospite 21 Jun '13

21 Jun '13

Hi, when working on the hiFace driver (which I plan on re-sending later today BTW) I've been reading the code of other drivers and I've stumbled on a couple of minor inconsistencies which were also spotted during the review rounds for the hiFace driver. I am sending some patches to address these issues, so that new drivers authors won't do the same mistakes I did when copying code over. Please take a closer look at the ones about vmalloc buffers, do we need an actual Tested-by for these? Patches are against 3.10-rc6. Thanks, Antonio Antonio Ospite (4): ALSA: snd-usb-caiaq: remove the unused snd_card_used variable ALSA: snd-usb-caiaq: use vmalloc buffers ALSA: snd-usb-6fire: use vmalloc buffers ALSA: usb: uniform style used in MODULE_SUPPORTED_DEVICE() sound/usb/6fire/chip.c | 2 +- sound/usb/6fire/pcm.c | 12 +++++------- sound/usb/caiaq/audio.c | 12 +++++------- sound/usb/caiaq/device.c | 31 +++++++++++++++---------------- sound/usb/usx2y/usbusx2y.c | 2 +- 5 files changed, 27 insertions(+), 32 deletions(-) -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

2 5

[alsa-devel] [patch] ALSA: vx_core: off by one in vx_read_status()
by Dan Carpenter 21 Jun '13

21 Jun '13

This code is older than git, and I haven't tested it, but if size == SIZE_MAX_STATUS then we would write one space past the end of the rmh->Stat[] array. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index c39961c..8359689 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -205,7 +205,7 @@ static int vx_read_status(struct vx_core *chip, struct vx_rmh *rmh) if (size < 1) return 0; - if (snd_BUG_ON(size > SIZE_MAX_STATUS)) + if (snd_BUG_ON(size >= SIZE_MAX_STATUS)) return -EINVAL; for (i = 1; i <= size; i++) {

2 1

[alsa-devel] WM8741 DAC with AT91SAM9G20
by Michael Lathion 21 Jun '13

21 Jun '13

Hi all, I'm trying to use a WM8742 dac instead of WM8731 on a custom board based on AT91SAM9G20EK (developpment board from Atmel for AT91SAM9G20 soc) with Linux Kernel 2.6.38. I have the driver working in alsa (driver compiled, mixer up in alsa, ...) but I can't have the ssc interface working. When trying to play sound with aplay I get only the masterclock and bitclock but no frame clock (LRCLK), see output below. What I've done is modifying the /sound/soc/atmel/sam9g20_wm8731.c file but it seems something is not configured as it should (probably related to WM8742 being slave and the WM8731 used on original dev board was master for clock distribution). My modified sam9g20_wm8741.c is enclosed to this mail, i suspect something is wrongly setup in CPU DAI configuration but can't find what. I'm trying to set this up from some weeks now but can't find the right config so if someone have any idea on what can has to be done it could be great! Thanks for help Regards Michael Linux kernel 2.6.38 AT91SAM9G20 processor WM8741 DAC Kernel boot output : ASoC: Mclk set to 22579200 wm8741-codec 0-001b: Successful registration FOX_Board_wm8741 : PB1003A: wm8741__init() called wm8741-codec 0-001b: wm8741_set_dai_sysclk info: freq=22579200 asoc: wm8741 <-> atmel-ssc-dai.0 mapping ok ALSA device list: #0: DAISY16 When trying play sound kernel driver print: wm8741-codec 0-001b: wm8741_set_dai_fmt: Format=0, Clock Inv=0 wm8741-codec 0-001b: wm8741_hw_params: bit size param = 2

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Remove unused variable
by Wang Xingchao 21 Jun '13

21 Jun '13

Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7803ddd..0f7a380 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1891,7 +1891,6 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec, /* override pins connection list */ snd_printdd("hdmi: haswell: override pin connection 0x%x\n", nid); - nconns = max(spec->num_cvts, 4); snd_hda_override_conn_list(codec, nid, spec->num_cvts, spec->cvt_nids); } -- 1.8.1.2

1 1

Re: [alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Mark Brown 21 Jun '13

21 Jun '13

On Wed, Jun 19, 2013 at 08:26:12PM +0200, Tomasz Figa wrote: > On Wednesday 19 of June 2013 18:40:47 Mark Brown wrote: > > - ret = pd->get_signal(plchan->cd); > > + ret = (pd->get_signal)(plchan->cd); > Hmm, that's strange. The former is a completely valid piece of code... I know, hence... > > to get it to build which makes me suspect the compiler a bit as well... ...my comment about suspecting the compiler. > > I was applying this to -next, are there any other dependencies I need or > > anything? > Hmm, I've been testing this on top of my common clock framework and device > tree patches, but I don't think this had any effect. Did you add necessary > clkdev lookups to the clock driver? No, I didn't - that's most likely it, I didn't really investigate. I didn't test the watchdog stuff as the clocks didn't get sent to me. > In Samsung CCF alias notation it looks like this: > + ALIAS(HCLK_DMA1, "dma-pl080s.1", "apb_pclk"), > + ALIAS(HCLK_DMA0, "dma-pl080s.0", "apb_pclk"), > Not sure how hard it will be to add such lookups to the old clock driver, > though. It's pretty much the same providing you know which clock needs to be used. > I will test this applied directly on top of current linux-next when I find > some time, but for now you might check out my v3.11-devel branch on my > github: > https://github.com/tom3q/linux.git Will try to get round to it.

2 4

[alsa-devel] [PATCH v2 0/3] ASoC: codecs: some more improvements for adau1701
by Daniel Mack 21 Jun '13

21 Jun '13

Sorry Lars, it's been a while since the last round. I hope I addressed all the details you mentioned. Thanks for having another look, Daniel Daniel Mack (3): ASoC: codecs: adau1701: allow configuration of PLL mode pins ASoC: codecs: adau1701: switch to direct regmap API usage ASoC: codecs: adau1701: add support for pin muxing .../devicetree/bindings/sound/adi,adau1701.txt | 18 ++ sound/soc/codecs/adau1701.c | 250 ++++++++++++++++----- sound/soc/codecs/adau1701.h | 4 + 3 files changed, 216 insertions(+), 56 deletions(-) -- 1.8.1.4

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Re: [alsa-devel] [PATCH V6 6/6] clk: exynos5250: Add enum entries for divider clock of i2s1 and i2s2
by Padma Venkat 21 Jun '13

21 Jun '13

Hi Mike, On Wed, Jun 12, 2013 at 10:15 PM, Mike Turquette <mturquette(a)linaro.org> wrote: > Quoting Padmavathi Venna (2013-06-12 01:07:43) >> This patch adds enum entries for div_i2s1 and div_i2s2 which are >> required for i2s1 and i2s2 controllers. >> >> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> > > Looks good. Did you want me to take the clk patches or just gathering > Acks? Please take the patches :). Thanks Padma > > Regards, > Mike > >> --- >> drivers/clk/samsung/clk-exynos5250.c | 5 +++-- >> 1 files changed, 3 insertions(+), 2 deletions(-) >> >> diff --git a/drivers/clk/samsung/clk-exynos5250.c b/drivers/clk/samsung/clk-exynos5250.c >> index 5c97e75..7c68850 100644 >> --- a/drivers/clk/samsung/clk-exynos5250.c >> +++ b/drivers/clk/samsung/clk-exynos5250.c >> @@ -87,6 +87,7 @@ enum exynos5250_clks { >> sclk_mmc0, sclk_mmc1, sclk_mmc2, sclk_mmc3, sclk_sata, sclk_usb3, >> sclk_jpeg, sclk_uart0, sclk_uart1, sclk_uart2, sclk_uart3, sclk_pwm, >> sclk_audio1, sclk_audio2, sclk_spdif, sclk_spi0, sclk_spi1, sclk_spi2, >> + div_i2s1, div_i2s2, >> >> /* gate clocks */ >> gscl0 = 256, gscl1, gscl2, gscl3, gscl_wa, gscl_wb, smmu_gscl0, >> @@ -291,8 +292,8 @@ struct samsung_div_clock exynos5250_div_clks[] __initdata = { >> DIV(none, "div_pcm1", "sclk_audio1", DIV_PERIC4, 4, 8), >> DIV(none, "div_audio2", "mout_audio2", DIV_PERIC4, 16, 4), >> DIV(none, "div_pcm2", "sclk_audio2", DIV_PERIC4, 20, 8), >> - DIV(none, "div_i2s1", "sclk_audio1", DIV_PERIC5, 0, 6), >> - DIV(none, "div_i2s2", "sclk_audio2", DIV_PERIC5, 8, 6), >> + DIV(div_i2s1, "div_i2s1", "sclk_audio1", DIV_PERIC5, 0, 6), >> + DIV(div_i2s2, "div_i2s2", "sclk_audio2", DIV_PERIC5, 8, 6), >> DIV(sclk_pixel, "div_hdmi_pixel", "sclk_vpll", DIV_DISP1_0, 28, 4), >> DIV_A(none, "armclk", "div_arm", DIV_CPU0, 28, 3, "armclk"), >> DIV_F(none, "div_mipi1_pre", "div_mipi1", >> -- >> 1.7.4.4

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