July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [GIT PULL] ASoC updates for 3.1
by Liam Girdwood 05 Jul '11

05 Jul '11

The following changes since commit d21685ec258f803d3badae5eae821383a34815a9: Merge branch 'for-2.6.40' into for-2.6.41 (2011-05-30 10:54:18 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-3.1 Jarkko Nikula (1): ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes Ricardo Neri (4): ASoC: OMAP: Add CPU DAI driver for HDMI ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards ASoC: OMAP: Update Makefile and Kconfig for HDMI audio ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB. sound/soc/codecs/tlv320aic3x.c | 34 +++++--- sound/soc/codecs/twl6040.c | 4 +- sound/soc/omap/Kconfig | 11 +++ sound/soc/omap/Makefile | 4 + sound/soc/omap/omap-hdmi.c | 158 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi.h | 36 +++++++++ sound/soc/omap/omap4-hdmi-card.c | 129 +++++++++++++++++++++++++++++++ 7 files changed, 361 insertions(+), 15 deletions(-) create mode 100644 sound/soc/omap/omap-hdmi.c create mode 100644 sound/soc/omap/omap-hdmi.h create mode 100644 sound/soc/omap/omap4-hdmi-card.c

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[alsa-devel] [GIT PULL] ASoC fixes for 3.0
by Liam Girdwood 05 Jul '11

05 Jul '11

The following changes since commit 4a787a3ff3f419c23ab0a5cef677fa441356b818: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-05-20 11:25:32 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-3.0 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 2 files changed, 19 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] ASoC: core - Add platform read and write.
by Liam Girdwood 05 Jul '11

05 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Allow platform driver to perform IO. Intended for platform DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 30 ++++++++++++++++++++++++++++++ 2 files changed, 38 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6424b10..f30f3fe 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -312,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_default_writable_register(struct snd_soc_codec *codec, unsigned int reg); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg); +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -658,6 +662,10 @@ struct snd_soc_platform_driver { /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; + + /* platform IO - used for platform DAPM */ + unsigned int (*read)(struct snd_soc_platform *, unsigned int); + int (*write)(struct snd_soc_platform *, unsigned int, unsigned int); }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d49f0ed..d08abf4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1640,6 +1640,36 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg) +{ + unsigned int ret; + + if (!platform->driver->read) { + dev_err(platform->dev, "platform has no read back\n"); + return -1; + } + + ret = platform->driver->read(platform, reg); + dev_dbg(platform->dev, "read %x => %x\n", reg, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_platform_read); + +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val) +{ + if (!platform->driver->write) { + dev_err(platform->dev, "platform has no write back\n"); + return -1; + } + + dev_dbg(platform->dev, "write %x = %x\n", reg, val); + return platform->driver->write(platform, reg, val); +} +EXPORT_SYMBOL_GPL(snd_soc_platform_write); + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ASoc: mx27vis-aic32x4: Remove unneeded header
by Mark Brown 04 Jul '11

04 Jul '11

On Mon, Jul 04, 2011 at 04:01:24PM -0300, Fabio Estevam wrote: > There is no machine type check on this code, so remove the mach-types header. > > Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> This isn't really useful - if you're going to change something here you should change the code to check for the machine.

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[PATCH] ASoC: Manage WM8731 ACTIVE bit as a supply widget
by Mark Brown 04 Jul '11

04 Jul '11

Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 1 files changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b..59d9e59 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_insel_enum); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Active",WM8731_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], @@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, static const struct snd_soc_dapm_route wm8731_intercon[] = { {"DAC", NULL, "OSC", wm8731_check_osc}, {"ADC", NULL, "OSC", wm8731_check_osc}, + {"DAC", NULL, "ACTIVE"}, + {"ADC", NULL, "ACTIVE"}, /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, @@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* set active */ - snd_soc_write(codec, WM8731_ACTIVE, 0x0001); - - return 0; -} - -static void wm8731_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* deactivate */ - if (!codec->active) { - udelay(50); - snd_soc_write(codec, WM8731_ACTIVE, 0x0); - } -} - static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - snd_soc_write(codec, WM8731_ACTIVE, 0x0); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_ops wm8731_dai_ops = { - .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, -- 1.7.5.4

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[alsa-devel] [PATCH 0/3] ASoC: Tegra: Fixes and SPDIF support
by Stephen Warren 04 Jul '11

04 Jul '11

A couple of small bug-fixes for the existing Tegra SPDIF support, and a new SPDIF driver. I've tested the SPDIF driver on Harmony. However, I'm not submitting the ASoC machine support patch for inclusion in mainline, since it relies on the board files registering some new platform devices, and we can't upstream any board file changes right now. Still, this may be useful for other board vendors to start working with internally. For people interested in testing/using this, I can push the branch I used for testing this on Harmony publicly sometime next week, hopefully. That branch contains the board changes to support this, and the machine driver change. Stephen Warren (3): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs ASoC: Tegra: I2S: s/clk_get_sys/clk_get/ ASoC: Tegra: Implement SPDIF CPU DAI sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_i2s.c | 10 +- sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 5 files changed, 862 insertions(+), 3 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Samuel, On Monday 04 July 2011 13:48:44 Samuel Ortiz wrote: > I don't see the value of those 3 inline functions. Removing them would make > the code actually more understandable (especially for the 2nd one). Is it OK for you if I fix these as a follow-up series, so I do not need to rewrite this series, and resend it? -- Péter

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Samuel, On Monday 04 July 2011 13:48:44 Samuel Ortiz wrote: > Hi Peter, > > On Tue, Jun 21, 2011 at 04:39:06PM +0300, Peter Ujfalusi wrote: > > +int twl6040_is_powered(struct twl6040 *twl6040) > > +{ > > + return twl6040->power_count; > > +} > > +EXPORT_SYMBOL(twl6040_is_powered); > > Do we really need to export this one ? No, we don't. At the end we do not need this function at all either. I'm going to remove it. > > +static inline int twl6040_get_rev(struct twl6040 *twl6040) > > +{ > > + return twl6040->rev; > > +} No need for this either, I'll remove it. > > + > > +static inline int twl6040_request_irq(struct twl6040 *twl6040, int irq, > > + irq_handler_t handler, > > + unsigned long irqflags, > > + const char *name, > > + void *data) > > +{ > > + if (!twl6040->irq_base) > > + return -EINVAL; > > + > > + return request_threaded_irq(twl6040->irq_base + irq, NULL, handler, > > + irqflags, name, data); > > +} > > + > > +static inline void twl6040_free_irq(struct twl6040 *twl6040, int irq, > > + void *data) > > +{ > > + if (!twl6040->irq_base) > > + return; > > + > > + free_irq(twl6040->irq_base + irq, data); > > +} > > I don't see the value of those 3 inline functions. Removing them would make > the code actually more understandable (especially for the 2nd one). The twl6040_request_irq is also used by the ASoC codec driver, and the vibra driver, but if you think it is better to replace those as well, I will do it right away. -- Péter

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Felipe, Samuel, On Monday 04 July 2011 14:53:30 Balbi, Felipe wrote: > > + ret = twl6040_request_irq(twl6040, TWL6040_IRQ_READY, > > + twl6040_naudint_handler, 0, > > + "twl6040_irq_ready", twl6040); > > why don't you use the normal request_threaded_irq() ?? This is a bit of > obfuscation IMO. I have left this 'macro' in since it is also used within the twl6040-vibra, and twl6040 ASoC codec driver to request the irq for them. But if you think it is better to replace the twl6040_request_irq with direct request_threaded_irq in those drivers as well, I'll do that. -- Péter

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[alsa-devel] [PATCH] ALSA: hda-intel - add position_fix quirk for Gateway NV53A
by Dmitry Torokhov 04 Jul '11

04 Jul '11

Without this option internal microphone does not seem to be working. $ lspci -vvnn | grep -A10 Audio 00:14.2 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (rev 40) Subsystem: Acer Incorporated [ALI] Device [1025:036e] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at d2300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel -- 01:05.1 Audio device [0403]: ATI Technologies Inc RS880 Audio Device [Radeon HD 4200] [1002:970f] Subsystem: Acer Incorporated [ALI] Device [1025:036e] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 19 Region 0: Memory at d2210000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a0] MSI: Enable- Count=1/1 Maskable- 64bit+ Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 486f6de..4fbcea6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2349,6 +2349,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1025, 0x036e, "Gateway NV53A", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),

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