August 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] hda-emu: add python compiled files to gitignore
by David Henningsson 16 Aug '12

16 Aug '12

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- .gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/.gitignore b/.gitignore index 2281a23..5ab6918 100644 --- a/.gitignore +++ b/.gitignore @@ -1,6 +1,7 @@ *~ *.o *.a +*.pyc hda-emu hda-decode-verb hda-encode-verb -- 1.7.9.5

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Re: [alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Takashi Iwai 16 Aug '12

16 Aug '12

At Thu, 16 Aug 2012 09:37:18 +0900 (JST), etsuko suzuki wrote: > > Hi. > > I am Etsuko. > Kaz is my coworker. > > This was my question, so I had to post this mail myself. > However my mail was not posted from my E-mail address. Did you subscribe to the ML? Otherwise a post would need a manual approval and it misses sometimes timeframe that the ML server keeps a pending post. > Therefore he had posted this instead of me. > > I am challenging once again using the different mail address. > > ************************************************************* > > >It depends on what format is used in the original PCM stream. > > I recorded this file from 16bits wav file. OK, it's clear now. > >It'd make sense to have 00 in LSB if it was converted from 16 to 24bit. > > Does this mean the following? > > ex.) > > > 0000030 ffffc300 ffffc600 ffffca00 ffffd300 > ++====** > > ++ : padding > == : PCM data > ** : This is added when it was converted from 16 to 24bit. > Yes. When the original source is 16bit, ALSA-lib format conversion just shifts the bits. It's suboptimal, but fast. That's the reason of the left 00. > I'd like to ask one more. > I read the following, in order to know about wav file. > > http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx > > In this, I found the statement 'left-alinged'. > (In 'Details about wValidBitsPerSample') > > I understood 'left-alinged' was 'LSB padding'. > So I thought S24_LE was LSB padding. No. S24_LE has the MSB padding of 8 bits in 32 bit container. > Does the format in the ALSA driver differ from the format in wav file? You need to use S32_LE instead. HTH, Takashi

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[alsa-devel] [PATCH v2 0/4] ASoC machine driver for simple SoC with twl4030
by Peter Ujfalusi 15 Aug '12

15 Aug '12

Hello, Changes since v1: - Card name for cm-t35 is "cm-t3x" - Added Ack from Tony for the OMAP patch - Codec sysclock no longer needed by the machine driver The codec itself must know the mclk since it needs it for initial configuration and also in case only the vibra functionality is used. The codec driver ignores the machine driver provided clock, it always uses the mclk freq it got from the MFD core driver. Intro mail from v1: Create new unified ASoC machine driver (with DT support) for boards using simple configuration for audio and using the twl4030 codec. This driver will replace the following machine drivers (since they are basically the same): igep0020, omap3beagle, omap3evm and overo. This series will switch the boards I can identify to use the new unified ASoC machine driver. Changes under arch/arm/mach-omap2/ are trivial, I don't expect merge issues later. If it is OK for the maintainers I would like this to go via audio tree (with the McBSP and the twl4030-audio stack DT support). The series has been tested on BeagleBoard (with the McBSP and twl4030-audio DT support). Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-twl4030: Simple machine driver for TI SoC with twl4030 codec ARM: OMAP: twl-common: Add helper function to register the omap-twl4030 audio driver ARM: OMAP3: Switch to use the unified audio driver (omap-twl4030) for selected boards ASoC: Remove obsolete OMAP3 machine drivers .../devicetree/bindings/sound/omap-twl4030.txt | 17 ++ arch/arm/mach-omap2/board-cm-t35.c | 1 + arch/arm/mach-omap2/board-devkit8000.c | 1 + arch/arm/mach-omap2/board-igep0020.c | 1 + arch/arm/mach-omap2/board-omap3beagle.c | 1 + arch/arm/mach-omap2/board-omap3evm.c | 1 + arch/arm/mach-omap2/board-overo.c | 1 + arch/arm/mach-omap2/twl-common.c | 27 +++ arch/arm/mach-omap2/twl-common.h | 2 + include/linux/platform_data/omap-twl4030.h | 32 ++++ sound/soc/omap/Kconfig | 48 ++---- sound/soc/omap/Makefile | 10 +- sound/soc/omap/igep0020.c | 120 ------------- sound/soc/omap/omap-twl4030.c | 188 ++++++++++++++++++++ sound/soc/omap/omap3beagle.c | 150 ---------------- sound/soc/omap/omap3evm.c | 118 ------------ sound/soc/omap/overo.c | 122 ------------- 17 files changed, 287 insertions(+), 553 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 include/linux/platform_data/omap-twl4030.h delete mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/omap/omap-twl4030.c delete mode 100644 sound/soc/omap/omap3beagle.c delete mode 100644 sound/soc/omap/omap3evm.c delete mode 100644 sound/soc/omap/overo.c -- 1.7.8.6

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[alsa-devel] [PATCH 03/22] ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements
by Lee Jones 15 Aug '12

15 Aug '12

If codec->control_data is not populated SoC Core assumes we want to use regmap, which fails catastrophically, as we don't have one: Unable to handle kernel NULL pointer dereference at virtual address 00000080 pgd = c0004000 [00000080] *pgd=00000000 Internal error: Oops: 17 [#1] PREEMPT SMP ARM Modules linked in: CPU: 1 Not tainted (3.5.0-rc6-00884-g0b2419e-dirty #130) PC is at regmap_read+0x10/0x5c LR is at hw_read+0x80/0x90 pc : [<c01a91b8>] lr : [<c0216804>] psr: 60000013 CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c79592..23b4018 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); -- 1.7.9.5

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[alsa-devel] [RFC PATCH 1/1] ALSA: hda - add a timer to find and suspend idle codecs
by mengdong.lin＠intel.com 15 Aug '12

15 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> (sorry, this patch is resent with title updated) This patch sets up a timer to monitor changes of the parameter 'power_save' and suspend any idle codecs if power save is re-enabled by the user. A codec won't be suspended if it becomes idle when power save is disabled. And it will remain unsuspended if power save is enabled later but no further user operations comes. So a timer is used here to find and suspend such idle codecs when power save is re-enabled. Now the timer has a fixed interval of 1 second. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 109 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 9 ++++ sound/pci/hda/hda_intel.c | 2 +- 3 files changed, 119 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 629131a..6bd8873 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -98,6 +98,8 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) +#define POWER_SAVE_MONITOR_INTERVAL 1000 +static void power_save_monitor_timeout(unsigned long data); #else static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 @@ -685,6 +687,9 @@ static int snd_hda_bus_free(struct hda_bus *bus) if (!bus) return 0; + + snd_hda_power_save_monitor_stop(bus); + if (bus->workq) flush_workqueue(bus->workq); if (bus->unsol) @@ -4558,6 +4563,106 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); + +#define BUS_POWER_SAVE(bus) \ + ((bus)->power_save && *(bus)->power_save ? 1 : 0) + +static void power_save_monitor_timeout(unsigned long data) +{ + int ret; + struct hda_bus *bus = (struct hda_bus *)data; + struct list_head *codec_list = &bus->codec_list; + struct hda_codec *codec, *next; + + if (BUS_POWER_SAVE(bus) == bus->power_save_enabled) + goto out; + + bus->power_save_enabled = BUS_POWER_SAVE(bus); + if (!bus->power_save_enabled) { + snd_printd("hda-bus: power save disabled\n"); + goto out; + } + + snd_printd("hda-bus: power save enabled\n"); + list_for_each_entry_safe(codec, next, codec_list, list) { + spin_lock(&codec->power_lock); + if (codec->power_on + && !codec->power_count + && !codec->power_transition) { + snd_printd("hda codec %d idle\n", codec->addr); + codec->power_transition = -1; /* avoid reentrance */ + queue_delayed_work(codec->bus->workq, + &codec->power_work, + msecs_to_jiffies(power_save(codec) * 1000)); + } + spin_unlock(&codec->power_lock); + } + +out: + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} + +/** + * snd_hda_power_save_monitor_start - Setup a timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + * + * Setup a timer to monitor whether parameter power_save is changed by the user. + * A codec won't be suspended if it becomes idle when power save is disabled, + * and can remain unsuspended after power save is enabled later. + * This monitor will find and suspened such idle codecs when power save is + * re-enabled. + */ +void snd_hda_power_save_monitor_start(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: power save timer already inited\n"); + return; + } + + setup_timer(&bus->power_save_timer, + power_save_monitor_timeout, (unsigned long)bus); + bus->power_save_timer_inited = 1; + + snd_printd("hda-bus: start power save monitor\n"); + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_start); + +/** + * snd_hda_power_save_monitor_stop - Delete the timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + */ +void snd_hda_power_save_monitor_stop(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: stop power save monitor\n"); + ret = del_timer_sync(&bus->power_save_timer); + if (ret < 0) + snd_printd("hda-bus: error in del_timer_sync\n"); + bus->power_save_timer_inited = 0; + } + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_stop); #endif /* @@ -5045,6 +5150,8 @@ int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; + snd_hda_power_save_monitor_stop(bus); + list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); @@ -5069,6 +5176,8 @@ int snd_hda_resume(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { hda_call_codec_resume(codec); } + + snd_hda_power_save_monitor_start(bus); return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..8b88ec5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -653,6 +653,8 @@ struct hda_bus { struct hda_bus_unsolicited *unsol; char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ + /* timer to monitor changes of power_save parameter */ + struct timer_list power_save_timer; /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -667,6 +669,9 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ + /* power save flags */ + unsigned int power_save_enabled:1; /* enabled at last check */ + unsigned int power_save_timer_inited:1; /* timer is initialized */ }; /* @@ -1062,10 +1067,14 @@ void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); +extern void snd_hda_power_save_monitor_start(struct hda_bus *bus); +extern void snd_hda_power_save_monitor_stop(struct hda_bus *bus); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} +#define snd_hda_power_save_monitor_start(bus) {} +#define snd_hda_power_save_monitor_stop(bus) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..fe015d0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3236,7 +3236,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); - + snd_hda_power_save_monitor_start(chip->bus); return 0; out_free: -- 1.7.9.5

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[alsa-devel] [RFC PATCH 1/1] ALSA: hda - add a timer to find and suspend idle codecs
by mengdong.lin＠intel.com 14 Aug '12

14 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch sets up a timer to monitor changes of the parameter 'power_save' and suspend any idle codecs if power save is re-enabled by the user. A codec won't be suspended if it becomes idle when power save is disabled. And it will remain unsuspended if power save is enabled later but no further user operations comes. So a timer is used here to find and suspend such idle codecs when power save is re-enabled. Now the timer has a fixed interval of 1 second. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 109 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 9 ++++ sound/pci/hda/hda_intel.c | 2 +- 3 files changed, 119 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 629131a..6bd8873 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -98,6 +98,8 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) +#define POWER_SAVE_MONITOR_INTERVAL 1000 +static void power_save_monitor_timeout(unsigned long data); #else static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 @@ -685,6 +687,9 @@ static int snd_hda_bus_free(struct hda_bus *bus) if (!bus) return 0; + + snd_hda_power_save_monitor_stop(bus); + if (bus->workq) flush_workqueue(bus->workq); if (bus->unsol) @@ -4558,6 +4563,106 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); + +#define BUS_POWER_SAVE(bus) \ + ((bus)->power_save && *(bus)->power_save ? 1 : 0) + +static void power_save_monitor_timeout(unsigned long data) +{ + int ret; + struct hda_bus *bus = (struct hda_bus *)data; + struct list_head *codec_list = &bus->codec_list; + struct hda_codec *codec, *next; + + if (BUS_POWER_SAVE(bus) == bus->power_save_enabled) + goto out; + + bus->power_save_enabled = BUS_POWER_SAVE(bus); + if (!bus->power_save_enabled) { + snd_printd("hda-bus: power save disabled\n"); + goto out; + } + + snd_printd("hda-bus: power save enabled\n"); + list_for_each_entry_safe(codec, next, codec_list, list) { + spin_lock(&codec->power_lock); + if (codec->power_on + && !codec->power_count + && !codec->power_transition) { + snd_printd("hda codec %d idle\n", codec->addr); + codec->power_transition = -1; /* avoid reentrance */ + queue_delayed_work(codec->bus->workq, + &codec->power_work, + msecs_to_jiffies(power_save(codec) * 1000)); + } + spin_unlock(&codec->power_lock); + } + +out: + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} + +/** + * snd_hda_power_save_monitor_start - Setup a timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + * + * Setup a timer to monitor whether parameter power_save is changed by the user. + * A codec won't be suspended if it becomes idle when power save is disabled, + * and can remain unsuspended after power save is enabled later. + * This monitor will find and suspened such idle codecs when power save is + * re-enabled. + */ +void snd_hda_power_save_monitor_start(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: power save timer already inited\n"); + return; + } + + setup_timer(&bus->power_save_timer, + power_save_monitor_timeout, (unsigned long)bus); + bus->power_save_timer_inited = 1; + + snd_printd("hda-bus: start power save monitor\n"); + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_start); + +/** + * snd_hda_power_save_monitor_stop - Delete the timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + */ +void snd_hda_power_save_monitor_stop(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: stop power save monitor\n"); + ret = del_timer_sync(&bus->power_save_timer); + if (ret < 0) + snd_printd("hda-bus: error in del_timer_sync\n"); + bus->power_save_timer_inited = 0; + } + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_stop); #endif /* @@ -5045,6 +5150,8 @@ int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; + snd_hda_power_save_monitor_stop(bus); + list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); @@ -5069,6 +5176,8 @@ int snd_hda_resume(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { hda_call_codec_resume(codec); } + + snd_hda_power_save_monitor_start(bus); return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..8b88ec5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -653,6 +653,8 @@ struct hda_bus { struct hda_bus_unsolicited *unsol; char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ + /* timer to monitor changes of power_save parameter */ + struct timer_list power_save_timer; /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -667,6 +669,9 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ + /* power save flags */ + unsigned int power_save_enabled:1; /* enabled at last check */ + unsigned int power_save_timer_inited:1; /* timer is initialized */ }; /* @@ -1062,10 +1067,14 @@ void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); +extern void snd_hda_power_save_monitor_start(struct hda_bus *bus); +extern void snd_hda_power_save_monitor_stop(struct hda_bus *bus); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} +#define snd_hda_power_save_monitor_start(bus) {} +#define snd_hda_power_save_monitor_stop(bus) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..fe015d0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3236,7 +3236,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); - + snd_hda_power_save_monitor_start(chip->bus); return 0; out_free: -- 1.7.9.5

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[alsa-devel] [PATCH v2 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 14 Aug '12

14 Aug '12

Hello, Changes since v1: - Get the MCLK frequencey from twl-core driver (via new API) - hs_extmute_disable_level parameter has been removed - empty of_find_node_by_name() in of.h for !CONFIG_OF builds Mark: the extmute GPIO handling (when it is used) remained in the codec driver for now. I can think more on how to add support for this type of mute control. If you can point me to other codec needing this it would help on designing something which is common enough for other users. As for now I have a separate series to add GPIO controlled output amps (hp/spk). I will send that to review soon. Introl mail from v1: The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform). If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (14): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl-core: Add API to query the HFCLK rate MFD: twl4030-audio: Get audio MCLK via twl-core API instead of pdata dt: Add empty of_find_node_by_name() function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 46 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +-- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 20 ++++- drivers/mfd/twl-core.c | 32 ++++++ drivers/mfd/twl4030-audio.c | 107 +++++++++++++------- include/linux/i2c/twl.h | 3 +- include/linux/of.h | 6 + sound/soc/codecs/twl4030.c | 103 +++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 10 files changed, 261 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.8.6

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[alsa-devel] [PATCH 22/22] Documentation: Add the AB8500 CODEC device to the MFD AB8500 doc
by Lee Jones 14 Aug '12

14 Aug '12

Here we document the AB8500 Audio CODEC in the existing MFD AB8500 document. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- Documentation/devicetree/bindings/mfd/ab8500.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/Documentation/devicetree/bindings/mfd/ab8500.txt b/Documentation/devicetree/bindings/mfd/ab8500.txt index 69e757a..ce83c8d 100644 --- a/Documentation/devicetree/bindings/mfd/ab8500.txt +++ b/Documentation/devicetree/bindings/mfd/ab8500.txt @@ -23,6 +23,7 @@ Device IRQ Names Supply Names Description ab8500-bm : : : Battery Manager ab8500-btemp : : : Battery Temperature ab8500-charger : : : Battery Charger +ab8500-codec : : : Audio Codec ab8500-fg : : : Fuel Gauge ab8500-gpadc : HW_CONV_END : vddadc : Analogue to Digital Converter SW_CONV_END : : @@ -52,6 +53,14 @@ Optional child device properties: supplied in the interrupts property - <supply_name>-supply : contains a phandle to the regulator supply node in Device Tree +Non-standard child device properties: + - Audio CODEC: + - stericsson,amic[1|2]-type-single-ended : Single-ended Analoge Mic (default: differential) + - stericsson,amic1a-bias-vamic2 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,amic1b-bias-vamic2 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,amic2-bias-vamic1 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,earpeice-cmv : Earpeice voltage (only: 950 | 1100 | 1270 | 1580) + ab8500@5 { compatible = "stericsson,ab8500"; reg = <5>; /* mailbox 5 is i2c */ @@ -110,6 +119,12 @@ ab8500@5 { compatible = "stericsson,ab8500-pwm"; }; + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; + ab8500-regulators { compatible = "stericsson,ab8500-regulator"; -- 1.7.9.5

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[alsa-devel] Automated test suite for HDA codecs, now up and running for daily builds
by David Henningsson 14 Aug '12

14 Aug '12

As some of you know, I've been working on extending hda-emu to become an automated test suite. A little status report follows: * Daniel Manrique provided me with alsa-info from ~110 machines that Canonical have certified. These alsa-infos are now upstreamed (except for a few which were marked confidential or never reached the market), and distributed with hda-emu [1]. * The test suite (which also is a part of hda-emu) loops over all these codecs, and sends a few commands to hda-emu for each one of them. E g, it tries setting volume controls and verifies that the value read back is the same it tried to set. It also looks for errors from hda-emu after each command sent. * I already run daily imports of Takashi's sound tarballs to provide automatic DKMS-based builds. What's new is that this import script [3] now also runs the test suite on that code. It uses the summary of this test run as commit message, which is then imported into Launchpad [2], and sent out by email (to anyone subscribed to that branch). I've just set this up, so maybe there will be some bumps before the import script runs correctly, but hopefully everything will run smoothly soon, and that we'll catch some regressions before they affect anyone. :-) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=commit;h=fb3b… [2] https://code.launchpad.net/~ubuntu-audio-dev/alsa-driver/tiwai-trunk2 [3] https://code.launchpad.net/~ubuntu-audio-dev/alsa-driver/filter-sound-script

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[alsa-devel] [PATCH 1/2] hda-emu: configure.ac: Allow relative paths in hdadir
by David Henningsson 14 Aug '12

14 Aug '12

Previously, adding a path that was relative could fail because the symlinks created were broken. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 2 ++ 1 file changed, 2 insertions(+) diff --git a/configure.ac b/configure.ac index 315e759..18118f3 100644 --- a/configure.ac +++ b/configure.ac @@ -51,6 +51,8 @@ AC_ARG_WITH(hdadir, [path where HD-audio kernel files are stored]), hdadir="$withval", hdadir="") +hdadir="$( cd "$hdadir" && pwd )" + if test -n "$hdadir"; then test -d "$hdadir" || \ -- 1.7.9.5

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