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[alsa-devel] 1ca3d59d5a3e762fb1c002bcd912914bc858b430 broke passthrough [Was: Re: ALSA 1.0.25 broke AC3 passthrough?]
by Jean-Yves Avenard 15 Feb '12

15 Feb '12

Hi again Allright.. Took me forever doing a bisect in alsa-kmirror while finding the proper revision in alsa-drivers to be able to compile it. So painful.. How do you guys ever manage to quickly do bisect ??? Anyhow... What broke passthrough for me (and others) is this commit: commit 1ca3d59d5a3e762fb1c002bcd912914bc858b430 Author: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Thu Aug 4 10:12:56 2011 -0500 ALSA: hda: option to enable arbitrary buffer/period sizes Add new parameter to disable rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and prevents users from specifying exact period/buffer sizes. For example for 44.1kHz, a period size set to 20ms will be rounded to 19.59ms. Tested and enabled on Intel HDA controllers. Option is disabled by default for other controllers. Tested-by: Wu Fengguang <fengguang.wu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> It's not an obvious breakage however and it only occurs with specific combinations of hardware buffer size (as seen for me in /proc/asound/card1/pcm3p/sub0/prealloc) and request of buffer time and period size. With the default ALSA hardware buffer size of 64kB, AC3/DTS works. With 4096, it doesn't and it will make the tested applications hang. both MythTV and mplayer (and I know of a few others), request by default a buffer time of 500ms with a period of 4 (requested respectively through snd_pcm_hw_params_set_buffer_time_near and snd_pcm_hw_params_set_periods_near) With those combinations: (4MB hardware buffer, 500ms and period of 4), trying to write audio with snd_pcm_writei will hang to eventually fail with the error "Input/output error" Disabling the functionality introduced in that changeset, fix it for v1.0.25 (using the following patch: diff --git a/pci/hda/hda_intel.c b/pci/hda/hda_intel.c index 9cbde2f..9227389 100644 --- a/pci/hda/hda_intel.c +++ b/pci/hda/hda_intel.c @@ -2773,8 +2773,8 @@ static int __devinit azx_create(struct snd_card *card, str } /* disable buffer size rounding to 128-byte multiples if supported */ - if (chip->driver_caps & AZX_DCAPS_BUFSIZE) - align_buffer_size = 0; +// if (chip->driver_caps & AZX_DCAPS_BUFSIZE) +// align_buffer_size = 0; /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) ) Hope this help Regards Jean-Yves Avenard

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Re: [alsa-devel] [PATCH 2/2] USB: Gadget: Add Audio Class 2.0 Driver
by Jassi Brar 15 Feb '12

15 Feb '12

On Mon, Aug 22, 2011 at 7:58 PM, Felipe Balbi <balbi(a)ti.com> wrote: > >> > audio.c doesn't do anything audio related. It's just adding the >> > configuration to composite layer. The real specification is implemented >> > in f_audio.c (should probably be called f_uac1.c, though). >> Agreed. Reusing audio.c is quite reasonable. >> So, I'll :- >> * Merge audio2.c into audio.c >> * Rename f_audio.c -> f_uac1.c >> * Add new f_audio2.c as f_uac2.c >> * Add Kconfig switch to compile UAC_1 (audio.c + f_uac1.c) or UAC_2 >> (audio.c + f_uac2.c) >> >> I'll wait for other comments and revise the patch after a few days. > > ok, sounds reasonable. > OK, after a 'few days' on some distant giant planet, here is the revised patchset accordingly. Cheers -jassi

3 5

Re: [alsa-devel] Bug report - patch_realtek.c - Laptop HP COMPAQ B1900 Series
by Joey Jiao 15 Feb '12

15 Feb '12

Btw, for the code modification, I modified model will in patch_realtek.c which works same as the hda-verb command. From: static const struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, {0x1a, AC_VERB_SET_PROC_COEF, 0x3040}, {} }; To: static const struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, {0x1a, AC_VERB_SET_PROC_COEF, 0x3040}, {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, {} }; 在 2012年2月15日上午9:18，Joey Jiao <joey.jiaojg(a)gmail.com> 写道： > Hi all, > I solved the speaker problem of B1900, and I have you can fix the > driver in next alsa release. > I use hda-verb tool to enable sound from speaker. > Commands as below: > sudo hda-verb /dev/snd/hwC0D0 0x01 0x716 0x01 > sudo hda-verb /dev/snd/hwC0D0 0x01 0x717 0x01 > > Perhaps you need to add a new model besides will. Meanwhile, auto > detect of headphone inserting is not function, so speaker and > headphone don't switch automatically. > > > 在 2012年2月15日上午8:50，Raymond Yau <superquad.vortex2(a)gmail.com> 写道： >> 2012/2/11, Jonathan Woithe <jwoithe(a)just42.net>: >>> Hi Joey >>> >>> For future reference, please email reports like this to the alsa-devel list. >>> The relevant developers are far more likely to respond to messages on the >>> list than they are to messages sent privately. >>> >>> I have copied this to the list to maximise its distribution. >>> >>> On Sat, Feb 11, 2012 at 09:17:05AM +0800, joey.jiaojg wrote: >>>> I'm reporting a bug related to ALSA and ALC260, which perhaps only >>>> you can solve. Appreciate if you can check. >>>> We got a lot of users using HP COMPAQ B1900 series which using ATI >>>> SB450 and ALC260 for linux systems like Debian, Ubuntu and etc. The >>>> problem is that there is no sound from speaker while enable >>>> model=will can hear from earphone. The problem and alsa-info.txt is >>>> reported here >>>> https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/186698, >>>> which thought be a bug of ALC260 driver bug. >>>> Hope you can check what we can do to solve it. >>> >>> From the above I am not entirely sure what the problem is. You seem to be >>> talking about two separate conditions. In the first (where I presume the >>> module is loaded without any parameters you say there's no sound from the >>> speaker. Do other aspects of the card work correctly in this case? >>> >>> The second situation is where "model=will" is specified as a module >>> parameter. Here you say you can get audio from the headphone, but again >>> what about other aspects of the soundcard - do they work normally? >>> >>> At a rough guess, it sounds like there are two underlying issues. The first >>> is that this system requires some model-specific handing in order to map >>> chip outputs correctly. The second is that if such model-specific details >>> are required then the driver preferrably would activate these automatically >>> without the need to specify "model=will". >>> >>> Another possibility is that there is something about this laptop system >>> which is confusing the auto-probing of the alc260 codec code. >>> >>> I'm not sure what is the best way of proceeding here. I suspect other more >>> active developers will have a streamlined approach to help rectify the >>> trouble. In the first instance I would tend to load the module with >>> "model=test" and experiment with the resulting alsamixer controls to >>> determine what chip outputs are connected to which pins. However, this is >>> quite "old school" now - while it's the approach I used when getting my >>> system working, my impression is that its use has been mostly deprecated by >>> the automatic parser which is now present. >>> >>> In the above-referenced launchpad report the question is asked about the >>> ALC260 datasheet. This is readily available on the net. I doubt it is all >>> that necessary now since I think the base infrastructure is pretty solid, >>> but if you do need it and have difficulty finding it please contact me >>> off-list. >>> >> >> The auto-parser require correct pin default >> >> Refer to alc260 General Description >> >> Realtek's proprietary impedance sensing and jack detect techniques >> allow device loads on inputs and outputs to be auto-detected. All >> analog IOs are input and output capable. Headphone amplifiers are also >> integrated at each analog output. All analog IOs can be re-tasked >> according to user's definitions, or automatically switched to suit the >> connecting device (Universal Audio Jack®). >> >> Reserve analog mixer architecture is backwards compatible with AC'97 > > > > -- > -Joey Jiao -- -Joey Jiao

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Use slow start for VMID
by Mark Brown 15 Feb '12

15 Feb '12

Improves performance on power up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e8f2396..b137ac9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -794,7 +794,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x3 << WM8994_VMID_RAMP_SHIFT)); + (0x2 << WM8994_VMID_RAMP_SHIFT)); wm_hubs_vmid_ena(codec); -- 1.7.9.rc1

1 0

Re: [alsa-devel] Bug report - patch_realtek.c - Laptop HP COMPAQ B1900 Series
by Jonathan Woithe 15 Feb '12

15 Feb '12

Hi Joey For future reference, please email reports like this to the alsa-devel list. The relevant developers are far more likely to respond to messages on the list than they are to messages sent privately. I have copied this to the list to maximise its distribution. On Sat, Feb 11, 2012 at 09:17:05AM +0800, joey.jiaojg wrote: > I'm reporting a bug related to ALSA and ALC260, which perhaps only > you can solve. Appreciate if you can check. > We got a lot of users using HP COMPAQ B1900 series which using ATI > SB450 and ALC260 for linux systems like Debian, Ubuntu and etc. The > problem is that there is no sound from speaker while enable > model=will can hear from earphone. The problem and alsa-info.txt is > reported here https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/186698, > which thought be a bug of ALC260 driver bug. > Hope you can check what we can do to solve it. _From the above I am not entirely sure what the problem is. You seem to be talking about two separate conditions. In the first (where I presume the module is loaded without any parameters you say there's no sound from the speaker. Do other aspects of the card work correctly in this case? The second situation is where "model=will" is specified as a module parameter. Here you say you can get audio from the headphone, but again what about other aspects of the soundcard - do they work normally? At a rough guess, it sounds like there are two underlying issues. The first is that this system requires some model-specific handing in order to map chip outputs correctly. The second is that if such model-specific details are required then the driver preferrably would activate these automatically without the need to specify "model=will". Another possibility is that there is something about this laptop system which is confusing the auto-probing of the alc260 codec code. I'm not sure what is the best way of proceeding here. I suspect other more active developers will have a streamlined approach to help rectify the trouble. In the first instance I would tend to load the module with "model=test" and experiment with the resulting alsamixer controls to determine what chip outputs are connected to which pins. However, this is quite "old school" now - while it's the approach I used when getting my system working, my impression is that its use has been mostly deprecated by the automatic parser which is now present. In the above-referenced launchpad report the question is asked about the ALC260 datasheet. This is readily available on the net. I doubt it is all that necessary now since I think the base infrastructure is pretty solid, but if you do need it and have difficulty finding it please contact me off-list. Regards jonathan

2 1

[alsa-devel] [PATCH] ASoC: core: Fix snd_soc_add_card_controls to pass correct data
by Stephen Warren 15 Feb '12

15 Feb '12

Commit 022658b "ASoC: core: Add support for DAI and machine kcontrols." created new functions to create controls for cards, codecs, platforms, and DAIs. The new snd_soc_add_card_controls() passed the wrong "data" to utility function snd_soc_add_controls(); it should be passing a codec, but instead ended up passing the soc_card. Fix this. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/soc-core.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8bd9995..e44b6f6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2095,9 +2095,13 @@ int snd_soc_add_card_controls(struct snd_soc_card *soc_card, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = soc_card->snd_card; + struct snd_soc_codec *codec = + list_first_entry(&soc_card->codec_dev_list, + struct snd_soc_codec, + card_list); return snd_soc_add_controls(card, soc_card->dev, controls, num_controls, - NULL, soc_card); + NULL, codec); } EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); -- 1.7.0.4

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[alsa-devel] [PATCH 0/3] ASoC: Relocate the OMAP HDMI codec to sound
by Ricardo Neri 14 Feb '12

14 Feb '12

Hello, Currently, the ASoC HDMI codec is embedded in the DSS HDMI driver. In order to give the OMAP HDMI code a more logical arrangement, the audio part is moved to /sound. Deep color support has been removed until it is supported by DSS so that no ugly register readings are done. The rest of the functionality is still supported. The relocation of the code is split into two parts: Part 1 is for DSS. It removes the HDMI codec from DSS and exposes DSS functionality to ASoC. Part 2 is the actual relocation of the codec under /sound. The purpose of this is split is to ease the integration of these patches first to DSS and then to ASoC. The DSS part has been submitted and can be consulted here: http://www.spinics.net/lists/linux-omap/msg64477.html Please note that these patches will not apply without the aforementioned DSS part. This work is also based on the patches previously sent for HDMI audio MCLK support: http://www.spinics.net/lists/linux-omap/msg64302.html and was implemented on top of: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git v3.3-rc3 Thanks, Ricardo Axel Castaneda Gonzalez (1): OMAPDSS/ASoC: Relocate ASoC HDMI codec. Part 2 Ricardo Neri (2): ASoC: OMAP: HDMI: Update codec name ASoC: OMAP: Update Kconfig and Makefile files for HDMI audio codec sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 5 + sound/soc/codecs/omap-hdmi.c | 422 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/Kconfig | 1 + sound/soc/omap/omap4-hdmi-card.c | 2 +- 5 files changed, 433 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/omap-hdmi.c

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[alsa-devel] ASoC update for 3.3
by Mark Brown 14 Feb '12

14 Feb '12

The following changes since commit a7c4183be2d6a7da8c97a9b671b5f3aed321127e: ASoC: wm8994: Disable line output discharge prior to ramping VMID (2012-02-08 19:52:00 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.3 for you to fetch changes up to 1987877d869027ab63dc9df515e11f19279a8091: ASoC: fsi: fixup fsi_pointer() calculation method (2012-02-09 10:50:53 +0000) ---------------------------------------------------------------- A simple fix from Morimoto-san for the pointer() operation in the FSI driver. ---------------------------------------------------------------- Kuninori Morimoto (1): ASoC: fsi: fixup fsi_pointer() calculation method sound/soc/sh/fsi.c | 6 +----- 1 files changed, 1 insertions(+), 5 deletions(-)

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[alsa-devel] [PATCH] ALSA: intel8x0: Fix default inaudible sound on Gateway M520
by Dan Chen 14 Feb '12

14 Feb '12

Hi Takashi, Attached please find a patch against current sound.git head that resolves a user's default inaudible sound on the Gateway M520. It's applicable to stable kernel trees, too. Thanks for considering the patch. Cheers. -Dan

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Re: [alsa-devel] TWL4030 low-latency
by Peter Ujfalusi 14 Feb '12

14 Feb '12

On 01/12/2012 02:09 AM, Jacinto Alvarez wrote: > Hi Peter, > > Thanks for your reply. > > Not sure what you mean under latency. For me the latency is the time > needed to play out the sample at the DMA pointer in main memory. > In case of McBSP2 (which has 1280 word long buffer) it is maximum of: > (1280/<number of channels>)/sampling frequency > In case of stereo sample it is 14.51ms with 44.1KHz, and 13.3ms with > 48KHz. > > > With latency I refer to the round-trip of the combined capture-playback > operation. The application I am using creates a couple of input/output > streams and bounds them together with snd_pcm_link(). The shortest time > I can get for this round-trip is 1024 frames (~23 ms) and I get that by > initializing the streams with 8 periods of 128 frames each. > > > In McBSP element mode the McBSP FIFO will be kept full all the time > (threshold is 0). This means that the McBSP will request one word from > the DMA if there's a single free slot in the FIFO. > > If you switch to threshold mode things will be different under the hood > (providing better power saving): > the McBSP threshold value will be calculated according to the period > size. In this mode the DMA will request for a chunk of data based on the > threshold value, so the FIFO will be filled with DMA bursts. You still > have the FIFO caused latency, but the system can rest between bursts. > I think if you use McBSP threshold mode with 3x 5ms periods you should > be fine (the ALSA buffer is going to be 15ms). > > > Is this threshold mode set by the application with the > snd_pcm_sw_params_set_start_threshold() command (e.g., passing a > non-zero parameter) or are you talking of some tweak INSIDE the driver's > code? It is McBSP driver specific. we do not have (yet) ALSA level control for it. The best fit would be some type of QoS, but I'm not sure of it. > Currently the application is initializing the streams with > snd_pcm_sw_params_set_start_threshold() and a value equal to the period > size (128 frames). Lets' try this (assuming you are using McBSP2): echo threshold > /sys/devices/platform/omap/omap-mcbsp.2/dma_op_mode Do you see difference in the round-trip? -- Péter

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Fix sidetone enumeration texts
by Mark Brown 14 Feb '12

14 Feb '12

The sidetone enumeration texts have left and right swapped. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 85f71a7..0fc7a56 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2001,7 +2001,7 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, return 0; } -static const char *st_text[] = { "None", "Right", "Left" }; +static const char *st_text[] = { "None", "Left", "Right" }; static const struct soc_enum str_enum = SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); -- 1.7.9.rc1

1 0

[alsa-devel] snd_dice: category ID
by Rolf Anderegg 13 Feb '12

13 Feb '12

Hi Clemens, I'd like to request the following patch to be applied on the firewire-kernel-streaming branch (current commit head 526795d69f5be6d058bdf89a65a303a1ee136aba). It involves handling Weiss DICE devices which use a dedicated category ID (other than TCAT's default Cat ID (0x04), which is not necessarily binding). Citing (one of many) TC's user guide: [...] The manufacturer is free to use the lower 5 bytes of the WWUID in any way. The DICE firmware, by default, uses the chip ID as follows: Upper 32 bits of WWUID +------------------------+--------+ | 24 bit OUI – 0x000166 | Cat | +------------------------+--------+ Category 8 bits: TCAT uses 0x04 for this field. Lower 32 bits of WWUID +----------------------------+----------------+ | 10 bit product identifier | 22 bit serial# | +----------------------------+----------------+ [...] Presumably, few of the other vendors have changed TCATs default Cat. ID. We have decided to change it to 0x00 since the beginning of our DICE product-line, alas this should be considered in snd-dice's dice_interface_check. Thanks in advance. Rolf Anderegg

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[alsa-devel] snd-hda-intel: Power management not enabled until audio is played
by Paul Menzel 13 Feb '12

13 Feb '12

Dear ALSA folks, playing with PowerTOP 1.97 I notice that it lists 100 % usage for the Audio device on several systems [1]. I made sure that power management is enabled. $ grep snd /etc/modprobe.d/* …: options snd_hda_intel power_save=1 $ more /sys/module/snd_hda_intel/parameters/power_save_controller Y $ more /sys/module/snd_hda_intel/parameters/power_save 0 Since I was on AC power pm-utils seems to reset that parameter to not save power. :/ $ sudo su # echo "1" > /sys/module/snd_hda_intel/parameters/power_save # exit But enabling that does not change anything right away. Only after using the device by playing some audio – `speaker-test` in this case, the power management seems to kick in. $ more /sys/module/snd_hda_intel/parameters/power_save* :::::::::::::: /sys/module/snd_hda_intel/parameters/power_save :::::::::::::: 1 :::::::::::::: /sys/module/snd_hda_intel/parameters/power_save_controller :::::::::::::: Y $ more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 405056 $ more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 409664 $ more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 413672 $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:08:46 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 464644 $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:08:52 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 470076 $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:09:04 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 482604 $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:10:41 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 9640 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 578904 $ date && speaker-test Sa 11. Feb 10:12:58 CET 2012 speaker-test 1.0.24.2 Wiedergabe-Gerät ist default Stream-Parameter sind 48000 Hz, S16_LE, 1 Kanäle Verwende 16 Oktaven rosa Rauschen Rate ist 48000 Hz (angefordert: 48000 Hz) Puffergröße von 2048 bis 8192 Periodengröße von 1024 bis 1024 Verwende maximale Puffergröße 8192 Perioden = 4 gesetzt: period_size = 1024 gesetzt: buffer_size = 8192 0 - Vorne links Zeit pro Periode = 2,828717 0 - Vorne links Zeit pro Periode = 2,985785 0 - Vorne links Zeit pro Periode = 2,985784 0 - Vorne links Zeit pro Periode = 2,985806 0 - Vorne links Zeit pro Periode = 2,985834 0 - Vorne links Zeit pro Periode = 2,985798 0 - Vorne links Zeit pro Periode = 2,985838 0 - Vorne links ^C $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:13:24 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 100732 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 651372 $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:13:29 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 105384 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 651372 $ date && more /sys/devices/pci0000\:00/0000\:00\:1b.0/sound/card0/hwC0D0/power_*_acct Sa 11. Feb 10:24:18 CET 2012 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_off_acct :::::::::::::: 753148 :::::::::::::: /sys/devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0/power_on_acct :::::::::::::: 652876 Is that an error in the driver? I have seen that with Realtek and Conexant devices. Please also find the output of `alsa-info.sh` [2] attached. There is no difference before and after playing some audio. $ diff -u alsa-info.txt.beforeplayingaudio alsa-info.txt.afterplayingaudio --- alsa-info.txt.beforeplayingaudio 2012-02-11 10:11:31.000000000 +0100 +++ alsa-info.txt.afterplayingaudio 2012-02-11 10:13:52.000000000 +0100 @@ -3,7 +3,7 @@ !!ALSA Information Script v 0.4.60 !!################################ -!!Script ran on: Sat Feb 11 09:11:29 UTC 2012 +!!Script ran on: Sat Feb 11 09:13:51 UTC 2012 !!Linux Distribution @@ -379,7 +379,7 @@ crw-rw----+ 1 root audio 116, 5 Feb 11 10:00 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Feb 11 10:00 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 3 Feb 11 10:00 /dev/snd/pcmC0D0c -crw-rw----+ 1 root audio 116, 2 Feb 11 10:00 /dev/snd/pcmC0D0p +crw-rw----+ 1 root audio 116, 2 Feb 11 10:12 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Feb 11 10:00 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Feb 11 10:00 /dev/snd/timer Thanks, Paul [1] http://lists.lesswatts.org/pipermail/discuss/2012-February/000687.html [2] http://alsa-project.org/main/index.php/Help_To_Debug

2 3

[alsa-devel] [PATCH 1/2] ASoC: tegra+alc5632: Added digital microphone DAPM widget.
by Leon Romanovsky 13 Feb '12

13 Feb '12

ALC5632 codec supports digital microphone. This patch adds DAPM widget. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/tegra/tegra_alc5632.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c000f51..2a27725 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -102,6 +102,7 @@ static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Digital Mic", NULL), }; static const struct snd_kcontrol_new tegra_alc5632_controls[] = { -- 1.7.3.4

2 1

Re: [alsa-devel] [PATCH v5 1/2] mxs-dma : move the mxs dma.h to a more common place
by Mark Brown 13 Feb '12

13 Feb '12

On Mon, Feb 13, 2012 at 01:25:55PM +0800, Huang Shijie wrote: > Move the header to a more common place. > The mxs dma engine is not only used in mx23/mx28, but also used > in mx50/mx6q. It will also be used in the future chips. Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com>

1 0

[alsa-devel] [PATCH] Trivial fixes in INSTALL file
by Vikram Narayanan 13 Feb '12

13 Feb '12

Changed from HG->git in the INSTALL file Signed-off-by: Vikram Narayanan <vikram186(a)gmail.com> --- diff --git a/INSTALL b/INSTALL index 202c491..544d5d2 100644 --- a/INSTALL +++ b/INSTALL @@ -24,11 +24,11 @@ package. The configure script from alsa-utils package probably cannot find header file asoundlib.h in $prefix/include/alsa directory (usually in /usr/include/alsa directory). -Compilation from HG sources +Compilation from git sources --------------------------- You need also GNU packages autoconf and automake installed in your system -to compile HG (Mercurial) sources of alsa-utils package. +to compile git sources of alsa-utils package. For compilation you can use these commands: @@ -40,7 +40,7 @@ For compilation you can use these commands: ./configure make -The included hgcompile script does this job for you. +The included gitcompile script does this job for you. Note: Some automake packages have missing aclocal program. Use newer version in the case. -- 1.7.4.1

2 2

Re: [alsa-devel] VT1705 : Mic not working with power_save
by Anisse Astier 13 Feb '12

13 Feb '12

(adding back list in Cc, it was lost) On Mon, 13 Feb 2012 10:02:27 +0800, <LydiaWang(a)viatech.com.cn> wrote : > > -----邮件原件----- > > 发件人: Anisse Astier [mailto:anisse@astier.eu] > > 发送时间: 2012年2月11日 1:56 > > 收件人: alsa-devel(a)alsa-project.org > > 抄送: Takashi Iwai; Lydia Wang > > 主题: VT1705 : Mic not working with power_save > > > > Hi, > > > > On the hardware described in attached alsa-info (VT1705), I have the > > following problem: > > - mic doesn't work > > - if, while recording, we switch "input source", mic works during the > > recording session. > > > > Passing power_save=0 as a parameter to snd-hda-intel solves the problem. > > > > It was reproduced on linus' master branch, tiwai's for-next branch, > > and 3.2 branch. > > > > Regards, > > > > Anisse > > Hi, > > The log showed that your machine had two microphones, one is an internal mic, > and the other is a rear mic. Which microphone did you use? The log showed > currently used mic is the internal mic. The rear mic was not plugged in a jack. > > Lydia On this hardware, there's only an internal mic, and no physical jack to plug another input.

3 5

[alsa-devel] Adding sound to BeagleBone
by Peter Bolch 13 Feb '12

13 Feb '12

Hi, what - beside making the hardware - has to be done in principle to add an external audio codec to BeagleBone board (OMAPAM3358 based, with I2S but without a codec on board)? To make things easy I will use a codec with available driver. Regards Peter

2 1

[alsa-devel] [PATCH] ASoC: wm9081: Use module_i2c_driver
by Mark Brown 12 Feb '12

12 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 23 +---------------------- 1 files changed, 1 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 13bfcbd..076c126 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1418,28 +1418,7 @@ static struct i2c_driver wm9081_i2c_driver = { }; #endif -static int __init wm9081_modinit(void) -{ - int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm9081_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm9081_modinit); - -static void __exit wm9081_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm9081_i2c_driver); -#endif -} -module_exit(wm9081_exit); - +module_i2c_driver(wm9081_i2c_driver); MODULE_DESCRIPTION("ASoC WM9081 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm9081: Move WM9081 IRQ platform data handling into I2C probe
by Mark Brown 12 Feb '12

12 Feb '12

Better style and better supports idle_bias_off which we're going to implement. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 20 +++++++++----------- 1 files changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 7b09b1f..5f464ab 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1258,7 +1258,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); int ret; - u16 reg; codec->control_data = wm9081->regmap; @@ -1268,16 +1267,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } - reg = 0; - if (wm9081->pdata.irq_high) - reg |= WM9081_IRQ_POL; - if (!wm9081->pdata.irq_cmos) - reg |= WM9081_IRQ_OP_CTRL; - snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, - WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); - - wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Enable zero cross by default */ snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, WM9081_LINEOUTZC, WM9081_LINEOUTZC); @@ -1395,6 +1384,15 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), sizeof(wm9081->pdata)); + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + regmap_update_bits(wm9081->regmap, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) -- 1.7.9.rc1

1 1

[alsa-devel] [PATCH 1/2] ASoC: wm8988: Convert to direct regmap API usage
by Mark Brown 12 Feb '12

12 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8988.c | 140 ++++++++++++++++++++++++++++++++++++++------- 1 files changed, 120 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 4ef9d4c..2470321 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -33,24 +33,89 @@ * We can't read the WM8988 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8988_reg[] = { - 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ - 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ - 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ - 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ - 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ - 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ - 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ - 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ - 0x0079, 0x0079, 0x0079, /* 40 */ +static const struct reg_default wm8988_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 5, 0x0008 }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 10, 0x00ff }, + { 11, 0x00ff }, + { 12, 0x000f }, + { 13, 0x000f }, + { 16, 0x0000 }, + { 17, 0x007b }, + { 18, 0x0000 }, + { 19, 0x0032 }, + { 20, 0x0000 }, + { 21, 0x00c3 }, + { 22, 0x00c3 }, + { 23, 0x00c0 }, + { 24, 0x0000 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0050 }, + { 35, 0x0050 }, + { 36, 0x0050 }, + { 37, 0x0050 }, + { 40, 0x0079 }, + { 41, 0x0079 }, + { 42, 0x0079 }, }; +static bool wm8988_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8988_LINVOL: + case WM8988_RINVOL: + case WM8988_LOUT1V: + case WM8988_ROUT1V: + case WM8988_ADCDAC: + case WM8988_IFACE: + case WM8988_SRATE: + case WM8988_LDAC: + case WM8988_RDAC: + case WM8988_BASS: + case WM8988_TREBLE: + case WM8988_RESET: + case WM8988_3D: + case WM8988_ALC1: + case WM8988_ALC2: + case WM8988_ALC3: + case WM8988_NGATE: + case WM8988_LADC: + case WM8988_RADC: + case WM8988_ADCTL1: + case WM8988_ADCTL2: + case WM8988_PWR1: + case WM8988_PWR2: + case WM8988_ADCTL3: + case WM8988_ADCIN: + case WM8988_LADCIN: + case WM8988_RADCIN: + case WM8988_LOUTM1: + case WM8988_LOUTM2: + case WM8988_ROUTM1: + case WM8988_ROUTM2: + case WM8988_LOUT2V: + case WM8988_ROUT2V: + case WM8988_LPPB: + return true; + default: + return false; + } +} + /* codec private data */ struct wm8988_priv { + struct regmap *regmap; unsigned int sysclk; - enum snd_soc_control_type control_type; struct snd_pcm_hw_constraint_list *sysclk_constraints; }; @@ -661,6 +726,7 @@ static int wm8988_mute(struct snd_soc_dai *dai, int mute) static int wm8988_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1; switch (level) { @@ -674,7 +740,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8988->regmap); /* VREF, VMID=2x5k */ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); @@ -730,7 +796,10 @@ static struct snd_soc_dai_driver wm8988_dai = { static int wm8988_suspend(struct snd_soc_codec *codec) { + struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); + wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_mark_dirty(wm8988->regmap); return 0; } @@ -745,7 +814,8 @@ static int wm8988_probe(struct snd_soc_codec *codec) struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); + codec->control_data = wm8988->regmap; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -781,9 +851,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .suspend = wm8988_suspend, .resume = wm8988_resume, .set_bias_level = wm8988_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8988_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8988_reg, .controls = wm8988_snd_controls, .num_controls = ARRAY_SIZE(wm8988_snd_controls), @@ -793,6 +860,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .num_dapm_routes = ARRAY_SIZE(wm8988_dapm_routes), }; +static struct regmap_config wm8988_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8988_LPPB, + .writeable_reg = wm8988_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8988_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8988_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8988_spi_probe(struct spi_device *spi) { @@ -804,18 +883,28 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) if (wm8988 == NULL) return -ENOMEM; - wm8988->control_type = SND_SOC_SPI; + wm8988->regmap = regmap_init_spi(spi, &wm8988_regmap); + if (IS_ERR(wm8988->regmap)) { + ret = PTR_ERR(wm8988->regmap); + dev_err(&spi->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8988); ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); + if (ret != 0) + regmap_exit(wm8988->regmap); return ret; } static int __devexit wm8988_spi_remove(struct spi_device *spi) { + struct wm8988_priv *wm8988 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); + regmap_exit(wm8988->regmap); return 0; } @@ -842,16 +931,27 @@ static __devinit int wm8988_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8988); - wm8988->control_type = SND_SOC_I2C; + + wm8988->regmap = regmap_init_i2c(i2c, &wm8988_regmap); + if (IS_ERR(wm8988->regmap)) { + ret = PTR_ERR(wm8988->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); + if (ret != 0) + regmap_exit(wm8988->regmap); + return ret; } static __devexit int wm8988_i2c_remove(struct i2c_client *client) { + struct wm8988_priv *wm8988 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8988->regmap); return 0; } -- 1.7.9.rc1

1 1

[alsa-devel] [PATCH 1/4] ASoC: wm_hubs: Improve single ended line output enable performance
by Mark Brown 12 Feb '12

12 Feb '12

The enable of the single ended line outputs on wm_hubs devices performs better if the output is enabled prior to starting VMID. Since inactive outputs are held at VMID anyway there is little cost to doing this for unused outputs. Add callbacks into wm_hubs and keep track of which outputs are really active so we can disable them once we're active. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 4 ++ sound/soc/codecs/wm8994.c | 5 ++ sound/soc/codecs/wm_hubs.c | 113 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_hubs.h | 11 ++++ 4 files changed, 125 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index ab4685a..1e69f63 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1058,6 +1058,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; + wm_hubs_set_bias_level(codec, level); + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -1078,6 +1080,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(wm8993->regmap, false); regcache_sync(wm8993->regmap); + wm_hubs_vmid_ena(codec); + /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e6eebf7..21931a0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -787,6 +787,8 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | (0x3 << WM8994_VMID_RAMP_SHIFT)); + wm_hubs_vmid_ena(codec); + /* Remove discharge for line out */ snd_soc_update_bits(codec, WM8994_ANTIPOP_1, WM8994_LINEOUT1_DISCH | @@ -2074,6 +2076,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + wm_hubs_set_bias_level(codec, level); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -2168,6 +2172,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, wm8994->cur_fw = NULL; break; } + codec->dapm.bias_level = level; return 0; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a53daf6..f7650c5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -500,6 +500,36 @@ static int earpiece_event(struct snd_soc_dapm_widget *w, return 0; } +static int lineout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + bool *flag; + + switch (w->shift) { + case WM8993_LINEOUT1N_ENA_SHIFT: + flag = &hubs->lineout1n_ena; + break; + case WM8993_LINEOUT1P_ENA_SHIFT: + flag = &hubs->lineout1p_ena; + break; + case WM8993_LINEOUT2N_ENA_SHIFT: + flag = &hubs->lineout2n_ena; + break; + case WM8993_LINEOUT2P_ENA_SHIFT: + flag = &hubs->lineout2p_ena; + break; + default: + WARN(1, "Unknown line output"); + return -EINVAL; + } + + *flag = SND_SOC_DAPM_EVENT_ON(event); + + return 0; +} + static const struct snd_kcontrol_new in1l_pga[] = { SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), @@ -675,14 +705,18 @@ SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, line2p_mix, ARRAY_SIZE(line2p_mix)), -SND_SOC_DAPM_OUT_DRV("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUTPUT("SPKOUTLP"), SND_SOC_DAPM_OUTPUT("SPKOUTLN"), @@ -949,6 +983,11 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int jd_scthr, int jd_thr, int micbias1_lvl, int micbias2_lvl) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + hubs->lineout1_se = !lineout1_diff; + hubs->lineout2_se = !lineout2_diff; + if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, WM8993_LINEOUT1_MODE, @@ -978,6 +1017,64 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata); +void wm_hubs_vmid_ena(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int val = 0; + + if (hubs->lineout1_se) + val |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + val |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; + + /* Enable the line outputs while we power up */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, val, val); +} +EXPORT_SYMBOL_GPL(wm_hubs_vmid_ena); + +void wm_hubs_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int val; + + switch (level) { + case SND_SOC_BIAS_ON: + /* Turn off any unneded single ended outputs */ + val = 0; + + if (hubs->lineout1_se && hubs->lineout1n_ena) + val |= WM8993_LINEOUT1N_ENA; + + if (hubs->lineout1_se && hubs->lineout1p_ena) + val |= WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se && hubs->lineout2n_ena) + val |= WM8993_LINEOUT2N_ENA; + + if (hubs->lineout2_se && hubs->lineout2p_ena) + val |= WM8993_LINEOUT2P_ENA; + + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, + WM8993_LINEOUT1N_ENA | + WM8993_LINEOUT1P_ENA | + WM8993_LINEOUT2N_ENA | + WM8993_LINEOUT2P_ENA, + val); + + if (!hubs->lineout1n_ena && !hubs->lineout1p_ena && + !hubs->lineout2n_ena && !hubs->lineout2p_ena) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, 0); + break; + + default: + break; + } +} +EXPORT_SYMBOL_GPL(wm_hubs_set_bias_level); + MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index c674c7a..4140905 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -33,6 +33,14 @@ struct wm_hubs_data { bool class_w; u16 class_w_dcs; + bool lineout1_se; + bool lineout1n_ena; + bool lineout1p_ena; + + bool lineout2_se; + bool lineout2n_ena; + bool lineout2p_ena; + bool dcs_done_irq; struct completion dcs_done; }; @@ -46,5 +54,8 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); +extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec); +extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); #endif -- 1.7.9.rc1

1 3

[alsa-devel] [PATCH v2 00/12] ASoC: ALC5632: Capture path improvements
by Leon Romanovsky 12 Feb '12

12 Feb '12

This patches is bringing improved support of capture path to ALC5632 codec. --- Sorry for resending it, I rebased first patchset on the wrong branch. Andrey Danin (10): ASoC: alc5632: Coding style. Remove two extra empty lines. ASoC: alc5632: Fixed voice DAC volume step. ASoC: alc5632: Add voice DAC playback switch ASoC: alc5632: Rename capture switches to common scheme XXX2REC ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix ASoC: alc5632: Refactored DAPM routes to add voice support ASoC: alc5632: Add DMIC switches and controls ASoC: alc5632: Fix I2S digital interface power for recording ASoC: alc5632: Allow 8kHz stream support. ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix Paul Fertser (2): ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs sound/soc/codecs/alc5632.c | 183 +++++++++++++++++++++++++++++++------------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 130 insertions(+), 54 deletions(-) -- 1.7.3.4

2 18

[alsa-devel] [PATCH 00/12] ASoC: ALC5632: Capture path improvements
by Leon Romanovsky 11 Feb '12

11 Feb '12

This patches is bringing improved support of capture path to ALC5632 codec. Andrey Danin (10): ASoC: alc5632: Coding style. Remove two extra empty lines. ASoC: alc5632: Fixed voice DAC volume step. ASoC: alc5632: Add voice DAC playback switch ASoC: alc5632: Rename capture switches to common scheme XXX2REC ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix ASoC: alc5632: Refactored DAPM routes to add voice support ASoC: alc5632: Add DMIC switches and controls ASoC: alc5632: Fix I2S digital interface power for recording ASoC: alc5632: Allow 8kHz stream support. ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix Paul Fertser (2): ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs sound/soc/codecs/alc5632.c | 183 +++++++++++++++++++++++++++++++------------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 130 insertions(+), 54 deletions(-) -- 1.7.3.4

1 12

[alsa-devel] ALSA 1.0.25 broke AC3 passthrough?
by Jean-Yves Avenard 11 Feb '12

11 Feb '12

Hi Today, I installed 1.0.25 drivers and libraries... AC3 passthrough is completely broken. My hardware: NVidia GT430, hdmi output Device used: hdmi:CARD=NVidia,DEV=0,AES0=6,AES1=0x82,AES2=0x00,AES3=0x01 on Ubuntu 10.04 LTS (but using manually built alsa) It makes the application hangs or behaves incorrectly. All the time broken, no matter which app. Reverting back to 1.0.24 fixes the problem I haven't delved much into it yet, MythTV (I am the audio developer) after opening the audio device in passthrough mode, trying to write audio on the pcm just hangs and result in an I/O error after a little while snd_pcm_writei returns an error (not immediately) snd_pcm_state(pcm_handle) returns 3, snd_strerror(err) returns "Input/output error" mplayer just hangs... nothing obvious in the logs, audio device opens okay, i can't say I notice much difference in opening the device, allocating fragment and the lot. Anyone else has seen this? Jean-Yves

2 4

Re: [alsa-devel] [PATCH v3 1/2] mxs-dma : move the mxs dma.h to a more common place
by Shawn Guo 11 Feb '12

11 Feb '12

On Sat, Feb 11, 2012 at 07:36:50AM +0800, Huang Shijie wrote: > hi, > > On Sat, Feb 11, 2012 at 12:21 AM, Shawn Guo <shawn.guo(a)linaro.org> wrote: > > On Tue, Feb 07, 2012 at 06:54:34PM +0800, Huang Shijie wrote: > >> Move the header to a more common place. > >> The mxs dma engine is not only used in mx23/mx28, but also used > >> in mx50/mx6q. It will also be used in the future chips. > >> > >> Rename it to mxs-dma.h > >> > >> change mxs-dma driver, mxs-mmc driver, gpmi-nand driver, mxs-saif driver > >> to the new header file. > >> > >> Signed-off-by: Huang Shijie <b32955(a)freescale.com> > >> --- > >> drivers/dma/mxs-dma.c | 2 +- > >> drivers/mmc/host/mxs-mmc.c | 2 +- > >> drivers/mtd/nand/gpmi-nand/gpmi-nand.h | 2 +- > >> .../include/mach/dma.h => include/linux/mxs-dma.h | 0 > >> sound/soc/mxs/mxs-pcm.c | 2 +- > >> sound/soc/mxs/mxs-pcm.h | 2 +- > >> sound/soc/mxs/mxs-saif.c | 2 +- > >> 7 files changed, 6 insertions(+), 6 deletions(-) > >> rename arch/arm/mach-mxs/include/mach/dma.h => include/linux/mxs-dma.h (100%) > > > > I suggested you rename it include/linux/fsl/mxs-dma.h [1]. It seems > > you are ignoring the comment. Can you explain why it's not a > > reasonable suggestion? > > For gpmi-nand.h is placed in include/linux/mtd/ directory, so i think > it's better to place the mxs-dma.h > to similar include/linux/dma/ directory, but the directory does not exit. > Yes, from subsystem point of view, it makes sense too. > If i place it at include/linux/fsl/mxs-dma.h , i have to create a new > folder, I am not sure i have the right > to do so. But i can try it in the next version. > So either way works for me. But you need to create a folder anyway. So let's do it to test people. > thanks a lot. > > > > > > > And Wolfram and Mark told you that you need to Cc subsystem maintainers > > to collect their acks [2]. But you are still not Ccing Mark Brown and > > Chris Ball for sound/soc/ and drivers/mmc changes. So you are not > > listening. > > ok, i will CC them in next version. I　really did not know to whom I　should cc . > The file MAINTAINERS is created for this purpose. -- Regards, Shawn

1 0

[alsa-devel] [PATCH] ASoC: wm8993: Add register default for INPUTS_CLAMP
by Mark Brown 10 Feb '12

10 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index acc39ea..1e69f63 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -61,6 +61,7 @@ static struct reg_default wm8993_reg_defaults[] = { { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ { 19, 0x0010 }, /* R19 - GPIO1 */ { 20, 0x0000 }, /* R20 - IRQ_DEBOUNCE */ + { 21, 0x0000 }, /* R21 - Inputs Clamp */ { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ { 23, 0x0800 }, /* R23 - GPIO_POL */ { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] sound/soc/mxs/mxs-saif.c: use devm_ functions
by Julia Lawall 10 Feb '12

10 Feb '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> The various devm_ functions allocate memory that is released when a driver detaches. This patch uses these functions for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- patch against git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git from the branch for-3.4 sound/soc/mxs/mxs-saif.c | 51 ++++++++++------------------------------------- 1 file changed, 11 insertions(+), 40 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index f204dba..12be05b 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -630,7 +630,7 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - saif = kzalloc(sizeof(*saif), GFP_KERNEL); + saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -655,29 +655,16 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", ret); - goto failed_clk; + return ret; } iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - dev_err(&pdev->dev, "failed to get io resource: %d\n", - ret); - goto failed_get_resource; - } - if (!request_mem_region(iores->start, resource_size(iores), - "mxs-saif")) { - dev_err(&pdev->dev, "request_mem_region failed\n"); - ret = -EBUSY; - goto failed_get_resource; - } - - saif->base = ioremap(iores->start, resource_size(iores)); + saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; - goto failed_ioremap; + goto failed_get_resource; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -685,7 +672,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_ioremap; + goto failed_get_resource; } saif->dma_param.chan_num = dmares->start; @@ -694,14 +681,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_irq1; + goto failed_get_resource; } saif->dev = &pdev->dev; - ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif); + ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_irq1; + goto failed_get_resource; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -709,7 +697,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_irq2; + goto failed_get_resource; } platform_set_drvdata(pdev, saif); @@ -717,7 +705,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_register; + goto failed_get_resource; } saif->soc_platform_pdev = platform_device_alloc( @@ -740,36 +728,19 @@ failed_pdev_add: platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_register: -failed_get_irq2: - free_irq(saif->irq, saif); -failed_get_irq1: - iounmap(saif->base); -failed_ioremap: - release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); -failed_clk: - kfree(saif); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct mxs_saif *saif = platform_get_drvdata(pdev); platform_device_unregister(saif->soc_platform_pdev); - snd_soc_unregister_dai(&pdev->dev); - - iounmap(saif->base); - release_mem_region(res->start, resource_size(res)); - free_irq(saif->irq, saif); - clk_put(saif->clk); - kfree(saif); return 0; }

3 2

[alsa-devel] Regression 3.0 -> 3.1: no HDMI sound after suspend
by Mario 'BitKoenig' Holbe 10 Feb '12

10 Feb '12

Hello, since kernel 3.1 playing sound through HDMI via pulseaudio stopped working after suspend/resume on my Intel HDA (there is simply silence after resume). Same for 3.2. This worked up to 3.0 and it still works when I configure pulseaudio to play through Analog Stereo instead of HDMI, hence I guess the issue is more connected to ALSA than to pulseaudio. A work-around is to induce pulseaudio to re-open the underlying audio devices (`pasuspender true' does the trick). Here is the alsa-info.sh output for kernels 3.0 and 3.1 (no significant changes afaics): http://www.alsa-project.org/db/?f=67e9358c83963773f69905b12d7dba06c4c010ed http://www.alsa-project.org/db/?f=219dd70eee2160669f86c77c0585244b2decb5ef Please let me know if I can provide further information. Thanks for your work & best regards Mario -- We know that communication is a problem, but the company is not going to discuss it with the employees. -- Switching supervisor, AT&T Long Lines Division

1 0

[alsa-devel] [PATCH] ASoC: wm8993: Correct typos in regmap conversion
by Mark Brown 10 Feb '12

10 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f65339f..acc39ea 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -61,8 +61,8 @@ static struct reg_default wm8993_reg_defaults[] = { { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ { 19, 0x0010 }, /* R19 - GPIO1 */ { 20, 0x0000 }, /* R20 - IRQ_DEBOUNCE */ - { 21, 0x8000 }, /* R22 - GPIOCTRL 2 */ - { 22, 0x0800 }, /* R23 - GPIO_POL */ + { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 23, 0x0800 }, /* R23 - GPIO_POL */ { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ -- 1.7.9.rc1

1 0

[alsa-devel] VT1705 : Mic not working with power_save
by Anisse Astier 10 Feb '12

10 Feb '12

Hi, On the hardware described in attached alsa-info (VT1705), I have the following problem: - mic doesn't work - if, while recording, we switch "input source", mic works during the recording session. Passing power_save=0 as a parameter to snd-hda-intel solves the problem. It was reproduced on linus' master branch, tiwai's for-next branch, and 3.2 branch. Regards, Anisse

1 0

[alsa-devel] system announce port documentation
by Lieven Moors 10 Feb '12

10 Feb '12

Hi, I have been searching in vain for info or examples on how to use the system announce port of alsa sequencer. Is there somebody who could help me with that? A simple example, or a reference to a simple program that uses this port, would also be of great help. Thanks for your help, Greetings, lieven

2 2

Re: [alsa-devel] [PATCH v3 1/2] mxs-dma : move the mxs dma.h to a more common place
by Shawn Guo 10 Feb '12

10 Feb '12

On Tue, Feb 07, 2012 at 06:54:34PM +0800, Huang Shijie wrote: > Move the header to a more common place. > The mxs dma engine is not only used in mx23/mx28, but also used > in mx50/mx6q. It will also be used in the future chips. > > Rename it to mxs-dma.h > > change mxs-dma driver, mxs-mmc driver, gpmi-nand driver, mxs-saif driver > to the new header file. > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> > --- > drivers/dma/mxs-dma.c | 2 +- > drivers/mmc/host/mxs-mmc.c | 2 +- > drivers/mtd/nand/gpmi-nand/gpmi-nand.h | 2 +- > .../include/mach/dma.h => include/linux/mxs-dma.h | 0 > sound/soc/mxs/mxs-pcm.c | 2 +- > sound/soc/mxs/mxs-pcm.h | 2 +- > sound/soc/mxs/mxs-saif.c | 2 +- > 7 files changed, 6 insertions(+), 6 deletions(-) > rename arch/arm/mach-mxs/include/mach/dma.h => include/linux/mxs-dma.h (100%) I suggested you rename it include/linux/fsl/mxs-dma.h [1]. It seems you are ignoring the comment. Can you explain why it's not a reasonable suggestion? And Wolfram and Mark told you that you need to Cc subsystem maintainers to collect their acks [2]. But you are still not Ccing Mark Brown and Chris Ball for sound/soc/ and drivers/mmc changes. So you are not listening. Regards, Shawn [1] http://article.gmane.org/gmane.linux.kernel.mmc/12301 [2] http://article.gmane.org/gmane.linux.alsa.devel/93751

1 0

[alsa-devel] [PATCH] ASoC: Fix build error in sound/soc/omap/n810.c
by Axel Lin 10 Feb '12

10 Feb '12

Fix below build error which is introduced by commit 022658 "ASoC: core: Add support for DAI and machine kcontrols". CC [M] sound/soc/omap/n810.o sound/soc/omap/n810.c: In function 'n810_set_input': sound/soc/omap/n810.c:194: error: 'codec' undeclared (first use in this function) sound/soc/omap/n810.c:194: error: (Each undeclared identifier is reported only once sound/soc/omap/n810.c:194: error: for each function it appears in.) sound/soc/omap/n810.c:188: warning: unused variable 'card' make[3]: *** [sound/soc/omap/n810.o] Error 1 make[2]: *** [sound/soc/omap] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/omap/n810.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 1490227..c292bf0 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -191,7 +191,7 @@ static int n810_set_input(struct snd_kcontrol *kcontrol, return 0; n810_dmic_func = ucontrol->value.integer.value[0]; - n810_ext_control(codec); + n810_ext_control(&card->dapm); return 1; } -- 1.7.5.4

3 2

[alsa-devel] Multiple channels mapping
by Patrick Lai 10 Feb '12

10 Feb '12

Hi, The DSP I am working on supports channel assignments similar to ALSA route plug-in described in this link http://drona.csa.iisc.ernet.in /~uday/alsamch.shtml. I am currently looking a way to pass channel/speaker mapping information of PCM stream( i.c channel 0 -> Front Left speaker, channel 1 -> Front Right speaker) to ALSA driver. I looked at latest kernel tree and don't think there is already provision for my purpose. Can anyone confirm? If I were to improvise a way to pass channel/speaker mapping information, should I follow the same approach similar to how bits per sample and other hardware parameter is passed? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

4 6

Re: [alsa-devel] [PATCH RESEND] [trivial] ASoC: Fix typo in twl4030.c
by Mark Brown 10 Feb '12

10 Feb '12

On Thu, Feb 09, 2012 at 11:46:08PM +0900, Masanari Iida wrote: > Correct spelling "memroy" to "memory" in > sound/soc/codecs/twl4030.c Applied, thanks.

1 0

[alsa-devel] tea575x-tuner improvements & use in maxiradio
by Hans Verkuil 10 Feb '12

10 Feb '12

These patches improve the tea575x-tuner module to make it up to date with the latest V4L2 frameworks. The maxiradio driver has also been converted to use the tea575x-tuner and I've used that card to test it. Unfortunately, this card can't read the data pin, so the new hardware seek functionality has been tested only partially (yes, it seeks, but when it finds a channel I can't read back the frequency). Ondrej, are you able to test these patches for the sound cards that use this tea575x tuner? Note that these two patches rely on other work that I did and that hasn't been merged yet. So it is best to pull from my git tree: http://git.linuxtv.org/hverkuil/media_tree.git/shortlog/refs/heads/radio-pc… You can use the v4l-utils repository (http://git.linuxtv.org/v4l-utils.git) to test the drivers: the v4l2-compliance test should succeed and with v4l2-ctl you can test the hardware seek: To seek down: v4l2-ctl -d /dev/radio0 --freq-seek=dir=0 To seek up: v4l2-ctl -d /dev/radio0 --freq-seek=dir=1 To do the compliance test: v4l2-compliance -r /dev/radio0 Regards, Hans

2 9

Re: [alsa-devel] [RFC PATCH] ARM ASoC: add sound driver for imx27pdk using mc13783 codec
by Fabio Estevam 09 Feb '12

09 Feb '12

Hi Philippe, On Wed, Jan 25, 2012 at 1:45 PM, Philippe Rétornaz <philippe.retornaz(a)epfl.ch> wrote: > I have a working sound support for the mx31moboard boards available here: > http://github.com/retornaz/linux/commits/moboard-sound That's cool :-) > It's basically Sascha mc13783 codec driver (from his git tree) rebased on 3.2 > and some mx31moboad specific code. > > I can playback & record sound on our board (2 channels@16Khz) with the sdma. > > Sascha, do you have any plan to upstream the mc13783 codec driver ? In case Sascha doesn't have plans to upstream mc13783, is this something you could do? Thanks, Fabio Estevam

3 6

[alsa-devel] [PATCH] sound/soc/mxs/mxs-saif.c: use devm_ functions
by Julia Lawall 09 Feb '12

09 Feb '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> The various devm_ functions allocate memory that is released when a driver detaches. This patch uses these functions for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- patch against git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git sound/soc/mxs/mxs-saif.c | 51 ++++++++++------------------------------------- 1 file changed, 11 insertions(+), 40 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 76dc74d..63b0a81 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -632,7 +632,7 @@ static int mxs_saif_probe(struct platform_device *pdev) return ret; } - saif = kzalloc(sizeof(*saif), GFP_KERNEL); + saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -652,29 +652,16 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", ret); - goto failed_clk; + return ret; } iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - dev_err(&pdev->dev, "failed to get io resource: %d\n", - ret); - goto failed_get_resource; - } - if (!request_mem_region(iores->start, resource_size(iores), - "mxs-saif")) { - dev_err(&pdev->dev, "request_mem_region failed\n"); - ret = -EBUSY; - goto failed_get_resource; - } - - saif->base = ioremap(iores->start, resource_size(iores)); + saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; - goto failed_ioremap; + goto failed_get_resource; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -682,7 +669,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_ioremap; + goto failed_get_resource; } saif->dma_param.chan_num = dmares->start; @@ -691,14 +678,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_irq1; + goto failed_get_resource; } saif->dev = &pdev->dev; - ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif); + ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_irq1; + goto failed_get_resource; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -706,7 +694,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_irq2; + goto failed_get_resource; } platform_set_drvdata(pdev, saif); @@ -714,7 +702,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_register; + goto failed_get_resource; } saif->soc_platform_pdev = platform_device_alloc( @@ -737,36 +725,19 @@ failed_pdev_add: platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_register: -failed_get_irq2: - free_irq(saif->irq, saif); -failed_get_irq1: - iounmap(saif->base); -failed_ioremap: - release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); -failed_clk: - kfree(saif); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct mxs_saif *saif = platform_get_drvdata(pdev); platform_device_unregister(saif->soc_platform_pdev); - snd_soc_unregister_dai(&pdev->dev); - - iounmap(saif->base); - release_mem_region(res->start, resource_size(res)); - free_irq(saif->irq, saif); - clk_put(saif->clk); - kfree(saif); return 0; }

2 3

[alsa-devel] Submitting patches for alsa-utils
by Vikram Narayanan 09 Feb '12

09 Feb '12

Hello, Is there a separate mailing list for alsa-lib and alsa-utils? Or, Shall I submit patches to this list? ~Vikram

2 1

[alsa-devel] ASoC related suspend problems on OMAP3
by Grazvydas Ignotas 09 Feb '12

09 Feb '12

Hi, On OMAP3 pandora, system suspend stops working properly after using audio at least once: (just after boot): # echo mem > /sys/power/state [ 12.578186] PM: Entering mem sleep [ 12.678802] PM: suspend of devices complete after 92.834 msecs [ 12.688201] PM: late suspend of devices complete after 3.173 msecs [ 17.607971] Successfully put all powerdomains to target state # aplay /dev/zero ^C # echo mem > /sys/power/state [ 104.404663] PM: suspend of devices complete after 78.794 msecs [ 104.413909] PM: late suspend of devices complete after 3.021 msecs [ 106.601196] Powerdomain (per_pwrdm) didn't enter target state 1 [ 106.607421] Powerdomain (core_pwrdm) didn't enter target state 1 [ 106.613739] Could not enter target state in pm_suspend I'm seeing this on 3.2, unable to verify on current Linus HEAD as something else is preventing core/per low power states there. Any ideas what could be causing this? Perhaps some clock is left enabled? -- Gražvydas

4 6

[alsa-devel] [PATCH v3] ASoC: da7210: Add support for PLL and SRM
by Ashish Chavan 09 Feb '12

09 Feb '12

Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. This patch has been tested on DA7210 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v2: - Combined three tables of pll dividers in to one to minimize duplicate code - Revamped logic to fetch appropriate pll dividers - Replaced magic numbers with proper defines Changes since v1: - Removed unnecessary function to configure clock dividers - Changed data layout and code related to PLL dividers. Tried making it more readable and robust --- sound/soc/codecs/da7210.c | 230 ++++++++++++++++++++++++++++++++++++--------- 1 files changed, 187 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab38e93..f7dd409 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,6 +145,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -161,12 +162,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -205,6 +210,56 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* Index fields for PLL divider array */ +#define FREF_IDX (0 << 0) +#define DIV1_IDX (1 << 0) +#define DIV2_IDX (2 << 0) +#define DIV3_IDX (3 << 0) + +/* Number of PLL input frequencies supported */ +#define FREF_CNT (7 << 0) +/* Number of PLL dividers for each frequency */ +#define DIV_CNT (3 << 0) +/* Number of columns in PLL divider table */ +#define COL_CNT (DIV_CNT + 1) + +/* Index of first relevant row in PLL divider table */ +#define MASTER_2822400_DIV_OFFSET 0 +#define MASTER_3072000_DIV_OFFSET (MASTER_2822400_DIV_OFFSET + FREF_CNT) +#define SLAVE_SRM_DIV_OFFSET (MASTER_3072000_DIV_OFFSET + FREF_CNT) + +/* PLL dividers table */ +static const u32 da7210_pll_div[][COL_CNT] = { + /* for MASTER mode, fs = 44.1Khz */ + { 12000000, 0xE8, 0x6C, 0x2, }, /* MCLK=12Mhz */ + { 13000000, 0xDF, 0x28, 0xC, }, /* MCLK=13Mhz */ + { 13500000, 0xDB, 0x0A, 0xD, }, /* MCLK=13.5Mhz */ + { 14400000, 0xD4, 0x5A, 0x2, }, /* MCLK=14.4Mhz */ + { 19200000, 0xBB, 0x43, 0x9, }, /* MCLK=19.2Mhz */ + { 19680000, 0xB9, 0x6D, 0xA, }, /* MCLK=19.68Mhz */ + { 19800000, 0xB8, 0xFB, 0xB, }, /* MCLK=19.8Mhz */ + /* for MASTER mode, fs = 48Khz */ + { 12000000, 0xF3, 0x12, 0x7, }, /* MCLK=12Mhz */ + { 13000000, 0xE8, 0xFD, 0x5, }, /* MCLK=13Mhz */ + { 13500000, 0xE4, 0x82, 0x3, }, /* MCLK=13.5Mhz */ + { 14400000, 0xDD, 0x3A, 0x0, }, /* MCLK=14.4Mhz */ + { 19200000, 0xC1, 0xEB, 0x8, }, /* MCLK=19.2Mhz */ + { 19680000, 0xBF, 0xEC, 0x0, }, /* MCLK=19.68Mhz */ + { 19800000, 0xBF, 0x70, 0x0, }, /* MCLK=19.8Mhz */ + /* for SLAVE mode with SRM */ + { 12000000, 0xED, 0xBF, 0x5, }, /* MCLK=12Mhz */ + { 13000000, 0xE4, 0x13, 0x0, }, /* MCLK=13Mhz */ + { 13500000, 0xDF, 0xC6, 0x8, }, /* MCLK=13.5Mhz */ + { 14400000, 0xD8, 0xCA, 0x1, }, /* MCLK=14.4Mhz */ + { 19200000, 0xBE, 0x97, 0x9, }, /* MCLK=19.2Mhz */ + { 19680000, 0xBC, 0xAC, 0xD, }, /* MCLK=19.68Mhz */ + { 19800000, 0xBC, 0x35, 0xE, }, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -627,6 +682,8 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; + unsigned int mclk_rate; + int master; }; /* @@ -673,8 +730,9 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -707,43 +765,33 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; break; default: return -EINVAL; @@ -753,8 +801,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + } else { + /* PLL bypass mode, enable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -768,6 +823,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; @@ -776,9 +832,11 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -830,10 +888,120 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, row_idx, cnt; + + if (da7210->master) { + /* In PLL master mode, use master mode PLL dividers */ + switch (fout) { + case 2822400: + row_idx = MASTER_2822400_DIV_OFFSET; + break; + case 3072000: + row_idx = MASTER_3072000_DIV_OFFSET; + break; + default: + dev_err(codec_dai->dev, + "Unsupported PLL output frequency %d\n", fout); + return -EINVAL; + } + } else { + /* In PLL slave mode, use SRM mode PLL dividers */ + row_idx = SLAVE_SRM_DIV_OFFSET; + } + + for (cnt = 0; cnt < FREF_CNT; cnt++) { + if (fref == da7210_pll_div[row_idx + cnt][FREF_IDX]) { + pll_div1 = da7210_pll_div[row_idx + cnt][DIV1_IDX]; + pll_div2 = da7210_pll_div[row_idx + cnt][DIV2_IDX]; + pll_div3 = da7210_pll_div[row_idx + cnt][DIV3_IDX]; + break; + } + } + if (cnt >= FREF_CNT) + goto err; + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + if (da7210->master) { + /* In master mode, no need to enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, + DA7210_PLL_EN); + } else { + /* In slave mode, enable SRM and PLL */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -872,24 +1040,13 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ + + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); - /* - * make sure that DA7210 use bypass mode before start up - */ snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* * ADC settings @@ -964,24 +1121,11 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ); /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* As suggested by Dialog */ snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); -- 1.7.1

2 5

[alsa-devel] [PATCH v2] ALSA: PCM - Add PCM creation API for internal PCMs.
by Liam Girdwood 09 Feb '12

09 Feb '12

The new ASoC dynamic PCM core needs to create PCMs and substreams that are for use by internal ASoC drivers only and not visible to userspace for direct IO. These new PCMs are similar to regular PCMs expect they have no device nodes or procfs entries. The ASoC component drivers use them in exactly the same way as regular PCMs for PCM and DAI operations. The intention is that a dynamic PCM based driver will register both regular PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace however the internal PCMs will be used by the driver to route digital audio through numerous back end DAI links (with potentially a DSP providing different hw_params, DAI formats based on the regular front end PCM params) to devices like CODECs, MODEMs, Bluetooth, FM, DMICs, etc This patch adds a new snd_pcm_new_internal() API call to create the internal PCM without device nodes or procfs. It also adds adds a new internal flag to snd_pcm. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- Changes since V1 - Renamed to Internal PCM instead of Dynamic PCM wrt ALSA. - Removed similar PCM creation code and use flag for procfs/device creation. include/sound/pcm.h | 4 ++ sound/core/pcm.c | 100 +++++++++++++++++++++++++++++++++++--------------- 2 files changed, 74 insertions(+), 30 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0cf91b2..b33993f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -454,6 +454,7 @@ struct snd_pcm { void *private_data; void (*private_free) (struct snd_pcm *pcm); struct device *dev; /* actual hw device this belongs to */ + bool internal; /* pcm is for internal use only */ #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) struct snd_pcm_oss oss; #endif @@ -475,6 +476,9 @@ extern const struct file_operations snd_pcm_f_ops[2]; int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm **rpcm); +int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm ** rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca87..138122e 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -650,7 +650,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) pstr->stream = stream; pstr->pcm = pcm; pstr->substream_count = substream_count; - if (substream_count > 0) { + if (substream_count > 0 && !pcm->internal) { err = snd_pcm_stream_proc_init(pstr); if (err < 0) { snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n"); @@ -674,15 +674,18 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) pstr->substream = substream; else prev->next = substream; - err = snd_pcm_substream_proc_init(substream); - if (err < 0) { - snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n"); - if (prev == NULL) - pstr->substream = NULL; - else - prev->next = NULL; - kfree(substream); - return err; + + if (!pcm->internal) { + err = snd_pcm_substream_proc_init(substream); + if (err < 0) { + snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n"); + if (prev == NULL) + pstr->substream = NULL; + else + prev->next = NULL; + kfree(substream); + return err; + } } substream->group = &substream->self_group; spin_lock_init(&substream->self_group.lock); @@ -696,25 +699,9 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) EXPORT_SYMBOL(snd_pcm_new_stream); -/** - * snd_pcm_new - create a new PCM instance - * @card: the card instance - * @id: the id string - * @device: the device index (zero based) - * @playback_count: the number of substreams for playback - * @capture_count: the number of substreams for capture - * @rpcm: the pointer to store the new pcm instance - * - * Creates a new PCM instance. - * - * The pcm operators have to be set afterwards to the new instance - * via snd_pcm_set_ops(). - * - * Returns zero if successful, or a negative error code on failure. - */ -int snd_pcm_new(struct snd_card *card, const char *id, int device, - int playback_count, int capture_count, - struct snd_pcm ** rpcm) +static int _snd_pcm_new(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, bool internal, + struct snd_pcm ** rpcm) { struct snd_pcm *pcm; int err; @@ -735,6 +722,7 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, } pcm->card = card; pcm->device = device; + pcm->internal = internal; if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -756,8 +744,60 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, return 0; } +/** + * snd_pcm_new - create a new PCM instance + * @card: the card instance + * @id: the id string + * @device: the device index (zero based) + * @playback_count: the number of substreams for playback + * @capture_count: the number of substreams for capture + * @rpcm: the pointer to store the new pcm instance + * + * Creates a new PCM instance. + * + * The pcm operators have to be set afterwards to the new instance + * via snd_pcm_set_ops(). + * + * Returns zero if successful, or a negative error code on failure. + */ +int snd_pcm_new(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, struct snd_pcm ** rpcm) +{ + return _snd_pcm_new(card, id, device, playback_count, capture_count, + false, rpcm); +} EXPORT_SYMBOL(snd_pcm_new); +/** + * snd_pcm_new_internal - create a new internal PCM instance + * @card: the card instance + * @id: the id string + * @device: the device index (zero based - shared with normal PCMs) + * @playback_count: the number of substreams for playback + * @capture_count: the number of substreams for capture + * @rpcm: the pointer to store the new pcm instance + * + * Creates a new internal PCM instance with no userspace device or procfs + * entries. This is used by ASoC Back End PCMs in order to create a PCM that + * will only be used internally by kernel drivers. i.e. it cannot be opened + * by userspace. It provides existing ASoC components drivers with a substream + * and access to any private data. + * + * The pcm operators have to be set afterwards to the new instance + * via snd_pcm_set_ops(). + * + * Returns zero if successful, or a negative error code on failure. + */ +int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm ** rpcm) +{ + return _snd_pcm_new(card, id, device, playback_count, capture_count, + true, rpcm); +} + +EXPORT_SYMBOL(snd_pcm_new_internal); + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -994,7 +1034,7 @@ static int snd_pcm_dev_register(struct snd_device *device) } for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; - if (pcm->streams[cidx].substream == NULL) + if (pcm->streams[cidx].substream == NULL || pcm->internal) continue; switch (cidx) { case SNDRV_PCM_STREAM_PLAYBACK: -- 1.7.5.4

3 5

[alsa-devel] [PATCH 1/3] ASoC: core: Allow CODECs to set ignore_pmdown_time in the driver struct
by Mark Brown 09 Feb '12

09 Feb '12

This is usually not a use case dependant flag anyway. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + 2 files changed, 3 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ecd5d72..d1e7a6a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -657,6 +657,8 @@ struct snd_soc_codec_driver { /* codec stream completion event */ int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a3b487..0b9974f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3380,6 +3380,7 @@ int snd_soc_register_codec(struct device *dev, codec->volatile_register = codec_drv->volatile_register; codec->readable_register = codec_drv->readable_register; codec->writable_register = codec_drv->writable_register; + codec->ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; -- 1.7.9.rc1

3 4

[alsa-devel] [PATCH] ASoC: pcm: If pmdown_time is zero then shut down DAPM immediately
by Mark Brown 09 Feb '12

09 Feb '12

Since we've already got logic to special case immediate teardown of the stream we may as well use it if the pmdown_time has been set to zero by the application layer instead of scheduling a work item with zero delay. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef28dd5..d7bb268 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -369,7 +369,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (codec->ignore_pmdown_time || + if (!rtd->pmdown_time || codec->ignore_pmdown_time || rtd->dai_link->ignore_pmdown_time) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, -- 1.7.9.rc1

2 1

[alsa-devel] [PATCH v2 0/7] MFD: twl6040: Conversion to i2c driver
by Peter Ujfalusi 09 Feb '12

09 Feb '12

Hello, Changes since v1 (RFC series): - make use of regmap in the converted twl6040 driver [1] - use module_i2c_driver to reduce line of code - Use regulator_bulk functions for now in the twl6040 mfd driver. [2] [1] Initially I'm not using the register cache offered by regmap, it will come as a followup series [2] When the driver supports more power levels, the bulk_enable/disable will be removed, and the VIO/V2V1 supplies are going to be handled separately. As of now it is better to enable them with the bulk call. Still in the to-do list: Implement proper regulator support on SDP4430, and PandaBoard for the needed supplies by the twl6040. The supply source is the same on both boards: TWL6040 VDD2V1 <- TWL6030 V2V1 SMPS TWL6040 VDDVIO <- TWL6030 V1V8 SMPS There seams to be no control implemented in the regulator driver for these SMPS regulators, so this need to be done first. The reason that I'm sending this series (even when I know that the proper regulator support is missing for SDP4430/PandaBoard) is to get feedback for the rest of the series, so I can include the fixes in the v3 series. ------ Intro mail from the original series: This series will convert the twl6040 MFD driver to an i2c driver. Compared to older twl4030/5030/TPS the twl6040 is a standalone audio IC. It is better if the twl6040-core (and all of it's child devices) does not depend on the twl-core since it has nothing to with it. With this conversion the dependency on twl can be dropped from the twl6040 driver stack (core, vibra, audio). Between the first and second patch the audio will not probe on OMAP4, but I felt it is better this way at least for the first RFC series to not have too big change within one patch. The two patch can be squashed together later if no objections. Regards, Peter --- Peter Ujfalusi (7): MFD: twl-core: Detach twl6040 from the pmic mfd driver MFD: twl6040: Convert to i2c driver, and separate it from twl core ASoC: twl6040: Remove dependency on twl4030 from Kconfig OMAP: 4430sdp: Correct fixed regulator device ID OMAP: sdp4430: Add fixed regulator for twl6040 needs OMAP: omap4panda: Add fixed regulator for twl6040 needs MFD: TWL6040: Add regulator support for VIO, V2V1 supplies arch/arm/mach-omap2/board-4430sdp.c | 77 ++++++++++++++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 75 ++++++++++++++-- arch/arm/mach-omap2/twl-common.c | 37 +++++++- arch/arm/mach-omap2/twl-common.h | 10 +-- drivers/input/misc/Kconfig | 1 - drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 2 +- drivers/mfd/twl-core.c | 58 +++++++----- drivers/mfd/twl6040-core.c | 155 +++++++++++++++++++++---------- include/linux/i2c/twl.h | 12 --- include/linux/mfd/twl6040.h | 29 ++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/twl6040.c | 3 +- 14 files changed, 345 insertions(+), 122 deletions(-) -- 1.7.8.4

2 15

Re: [alsa-devel] [PATCH] [trivial] twl4030: Fix typo in twl4030.c
by Peter Ujfalusi 09 Feb '12

09 Feb '12

Hi, On 02/08/2012 01:53 PM, Masanari Iida wrote: > Correct spelling "memroy" to "memory" in > sound/soc/codecs/twl4030.c > > Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> > --- > sound/soc/codecs/twl4030.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c > index 3039ba2..d7eee0d 100644 > --- a/sound/soc/codecs/twl4030.c > +++ b/sound/soc/codecs/twl4030.c > @@ -2221,7 +2221,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) > > twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); > if (twl4030 == NULL) { > - printk("Can not allocate memroy\n"); > + printk("Can not allocate memory\n"); > return -ENOMEM; > } > snd_soc_codec_set_drvdata(codec, twl4030); Could you please resend the patch with the ASoC maintainers on CC (Liam, Mark), and the alsa-devel list. Thank you. Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com>

1 0

[alsa-devel] ASoC updates for 3.3
by Mark Brown 08 Feb '12

08 Feb '12

The following changes since commit 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e: ASoC: wm_hubs: Correct line input to line output 2 paths (2012-02-01 23:55:56 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.3 for you to fetch changes up to a7c4183be2d6a7da8c97a9b671b5f3aed321127e: ASoC: wm8994: Disable line output discharge prior to ramping VMID (2012-02-08 19:52:00 +0000) ---------------------------------------------------------------- A few small WM8994 updates to go on top of the previous lot of things that were sent. They collide with some -next work so I'd really like to get them into 3.3-rc3 if possible to merge back up into the -next code. All driver specific and unexciting in the grand scheme of things. ---------------------------------------------------------------- Mark Brown (3): ASoC: wm8994: Enabling VMID should take a runtime PM reference ASoC: wm8994: Fix typo in VMID ramp setting ASoC: wm8994: Disable line output discharge prior to ramping VMID sound/soc/codecs/wm8994.c | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: core: Convert single bit wide bitfields into bools
by Mark Brown 08 Feb '12

08 Feb '12

It's what they are, we've just been using these for historical reasons. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 12 ++++++------ include/sound/soc-dapm.h | 26 +++++++++++++------------- include/sound/soc.h | 32 ++++++++++++++++---------------- 3 files changed, 35 insertions(+), 35 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1a2b08c..53f9a1f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -208,7 +208,7 @@ struct snd_soc_dai_driver { /* DAI capabilities */ struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; - unsigned int symmetric_rates:1; + bool symmetric_rates; /* probe ordering - for components with runtime dependencies */ int probe_order; @@ -230,13 +230,13 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active:1; /* stream is in use */ - unsigned int playback_active:1; /* stream is in use */ - unsigned int symmetric_rates:1; + bool capture_active; /* stream is in use */ + bool playback_active; /* stream is in use */ + bool symmetric_rates; struct snd_pcm_runtime *runtime; unsigned int active; - unsigned char pop_wait:1; - unsigned char probed:1; + bool pop_wait; + bool probed; /* DAI DMA data */ void *playback_dma_data; diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index db8435a..99661c0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -456,9 +456,9 @@ struct snd_soc_dapm_path { struct snd_kcontrol *kcontrol; /* status */ - u32 connect:1; /* source and sink widgets are connected */ - u32 walked:1; /* path has been walked */ - u32 weak:1; /* path ignored for power management */ + bool connect; /* source and sink widgets are connected */ + bool walked; /* path has been walked */ + bool weak; /* path ignored for power management */ int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -488,16 +488,16 @@ struct snd_soc_dapm_widget { unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ - unsigned char power:1; /* block power status */ - unsigned char invert:1; /* invert the power bit */ - unsigned char active:1; /* active stream on DAC, ADC's */ - unsigned char connected:1; /* connected codec pin */ - unsigned char new:1; /* cnew complete */ - unsigned char ext:1; /* has external widgets */ - unsigned char force:1; /* force state */ - unsigned char ignore_suspend:1; /* kept enabled over suspend */ - unsigned char new_power:1; /* power from this run */ - unsigned char power_checked:1; /* power checked this run */ + bool power; /* block power status */ + bool invert; /* invert the power bit */ + bool active; /* active stream on DAC, ADC's */ + bool connected; /* connected codec pin */ + bool new; /* cnew complete */ + bool ext; /* has external widgets */ + bool force; /* force state */ + bool ignore_suspend; /* kept enabled over suspend */ + bool new_power; /* power from this run */ + bool power_checked; /* power checked this run */ int subseq; /* sort within widget type */ int (*power_check)(struct snd_soc_dapm_widget *w); diff --git a/include/sound/soc.h b/include/sound/soc.h index d1e7a6a..466b1f3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -571,14 +571,14 @@ struct snd_soc_codec { /* runtime */ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; - unsigned int cache_bypass:1; /* Suppress access to the cache */ - unsigned int suspended:1; /* Codec is in suspend PM state */ - unsigned int probed:1; /* Codec has been probed */ - unsigned int ac97_registered:1; /* Codec has been AC97 registered */ - unsigned int ac97_created:1; /* Codec has been created by SoC */ - unsigned int sysfs_registered:1; /* codec has been sysfs registered */ - unsigned int cache_init:1; /* codec cache has been initialized */ - unsigned int using_regmap:1; /* using regmap access */ + bool cache_bypass; /* Suppress access to the cache */ + bool suspended; /* Codec is in suspend PM state */ + bool probed; /* Codec has been probed */ + bool ac97_registered; /* Codec has been AC97 registered */ + bool ac97_created; /* Codec has been created by SoC */ + bool sysfs_registered; /* codec has been sysfs registered */ + bool cache_init; /* codec cache has been initialized */ + bool using_regmap; /* using regmap access */ u32 cache_only; /* Suppress writes to hardware */ u32 cache_sync; /* Cache needs to be synced to hardware */ @@ -598,7 +598,7 @@ struct snd_soc_codec { /* dapm */ struct snd_soc_dapm_context dapm; - unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ + bool ignore_pmdown_time; /* pmdown_time is ignored at stop */ #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_codec_root; @@ -712,8 +712,8 @@ struct snd_soc_platform { struct device *dev; struct snd_soc_platform_driver *driver; - unsigned int suspended:1; /* platform is suspended */ - unsigned int probed:1; + bool suspended; /* platform is suspended */ + bool probed; struct snd_soc_card *card; struct list_head list; @@ -737,13 +737,13 @@ struct snd_soc_dai_link { unsigned int dai_fmt; /* format to set on init */ /* Keep DAI active over suspend */ - unsigned int ignore_suspend:1; + bool ignore_suspend; /* Symmetry requirements */ - unsigned int symmetric_rates:1; + bool symmetric_rates; /* pmdown_time is ignored at stop */ - unsigned int ignore_pmdown_time:1; + bool ignore_pmdown_time; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); @@ -876,8 +876,8 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; - unsigned int complete:1; - unsigned int dev_registered:1; + bool complete; + bool dev_registered; long pmdown_time; -- 1.7.9.rc1

2 4

[alsa-devel] [PATCH] ASoC: wm8962: Support mono playback and record
by Mark Brown 08 Feb '12

08 Feb '12

Make sure we generate enough BCLKs for I2S style modes by always having a minimum of two channels worth of clocks for the BCLK. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0f6bd06..85f71a7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2536,6 +2536,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, int adctl3 = 0; wm8962->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + wm8962->bclk *= 2; + wm8962->lrclk = params_rate(params); for (i = 0; i < ARRAY_SIZE(sr_vals); i++) { @@ -2913,14 +2916,14 @@ static struct snd_soc_dai_driver wm8962_dai = { .name = "wm8962", .playback = { .stream_name = "Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8962_RATES, .formats = WM8962_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8962_RATES, .formats = WM8962_FORMATS, -- 1.7.9.rc1

1 0

[alsa-devel] [RFC/PATCH] ALSA: PCM - Add PCM creation API for ASoC dynamic PCMs.
by Liam Girdwood 08 Feb '12

08 Feb '12

The new ASoC dynamic PCM core needs to create PCMs and substreams that are for use by internal ASoC drivers only and not visible to userspace for direct IO. These new PCMs are similar to regular PCMs expect they have no device nodes or procfs entries. The ASoC component drivers use them in exactly the same way as regular PCMs for PCM and DAI operations. The intention is that a dynamic PCM based driver will register both regular PCMs and dynamic PCMs. The regular PCMs will be used for all IO with userspace however the dynamic PCMs will be used by the driver to route digital audio through numerous back end DAI links (with potentially a DSP providing different hw_params, DAI formats based on the regular front end PCM params) to devices like CODECs, MODEMs, Bluetooth, FM, DMICs, etc This patch adds a new snd_pcm_new_dynamic() API call to create the dynamic PCM without device nodes or procfs. It also adds snd_pcm_new_stream_dynamic() and snd_pcm_dev_register_dynamic() which are used to create the dynamic PCM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/pcm.h | 3 + sound/core/pcm.c | 145 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 148 insertions(+), 0 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0cf91b2..1597f89 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -475,6 +475,9 @@ extern const struct file_operations snd_pcm_f_ops[2]; int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm **rpcm); +int snd_pcm_new_dynamic(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm ** rpcm); int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count); int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree); diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca87..8e2a429 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -41,6 +41,7 @@ static DEFINE_MUTEX(register_mutex); static int snd_pcm_free(struct snd_pcm *pcm); static int snd_pcm_dev_free(struct snd_device *device); static int snd_pcm_dev_register(struct snd_device *device); +static int snd_pcm_dev_register_dynamic(struct snd_device *device); static int snd_pcm_dev_disconnect(struct snd_device *device); static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) @@ -697,6 +698,60 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) EXPORT_SYMBOL(snd_pcm_new_stream); /** + * snd_pcm_new_stream_dynamic - create a new dynamic PCM stream + * @pcm: the pcm instance + * @stream: the stream direction, SNDRV_PCM_STREAM_XXX + * @substream_count: the number of substreams + * + * Creates a new dynamic stream for the pcm. Dynamic PCM streams have no procfs + * entries or device entries and are hence not directly controllable by + * userspace. + * The corresponding stream on the pcm must have been empty before + * calling this, i.e. zero must be given to the argument of + * snd_pcm_new(). + * + * Returns zero if successful, or a negative error code on failure. + */ +static int snd_pcm_new_stream_dynamic(struct snd_pcm *pcm, int stream, + int substream_count) +{ + int idx; + struct snd_pcm_str *pstr = &pcm->streams[stream]; + struct snd_pcm_substream *substream, *prev; + + pstr->stream = stream; + pstr->pcm = pcm; + pstr->substream_count = substream_count; + + prev = NULL; + for (idx = 0, prev = NULL; idx < substream_count; idx++) { + substream = kzalloc(sizeof(*substream), GFP_KERNEL); + if (substream == NULL) { + snd_printk(KERN_ERR "Cannot allocate BE PCM substream\n"); + return -ENOMEM; + } + substream->pcm = pcm; + substream->pstr = pstr; + substream->number = idx; + substream->stream = stream; + sprintf(substream->name, "subdevice #%i", idx); + substream->buffer_bytes_max = UINT_MAX; + if (prev == NULL) + pstr->substream = substream; + else + prev->next = substream; + + substream->group = &substream->self_group; + spin_lock_init(&substream->self_group.lock); + INIT_LIST_HEAD(&substream->self_group.substreams); + list_add_tail(&substream->link_list, &substream->self_group.substreams); + atomic_set(&substream->mmap_count, 0); + prev = substream; + } + return 0; +} + +/** * snd_pcm_new - create a new PCM instance * @card: the card instance * @id: the id string @@ -735,6 +790,7 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, } pcm->card = card; pcm->device = device; + if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -758,6 +814,72 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, EXPORT_SYMBOL(snd_pcm_new); +/** + * snd_pcm_new_dynamic - create a new dynamic PCM instance for ASoC BE DAI link + * @card: the card instance + * @id: the id string + * @device: the device index (zero based - shared with normal PCMs) + * @playback_count: the number of substreams for playback + * @capture_count: the number of substreams for capture + * @rpcm: the pointer to store the new pcm instance + * + * Creates a new dynamic PCM instance with no userspace device or procfs + * entries. This is used by ASoC Back End PCMs in order to create a PCM that + * will only be used internally by kernel drivers. i.e. it cannot be opened + * by userspace. It provides existing ASoC components drivers with a substream + * and access to any private data. + * + * The pcm operators have to be set afterwards to the new instance + * via snd_pcm_set_ops(). + * + * Returns zero if successful, or a negative error code on failure. + */ +int snd_pcm_new_dynamic(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm ** rpcm) +{ + struct snd_pcm *pcm; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_pcm_dev_free, + .dev_register = snd_pcm_dev_register_dynamic, + .dev_disconnect = snd_pcm_dev_disconnect, + }; + + if (snd_BUG_ON(!card)) + return -ENXIO; + if (rpcm) + *rpcm = NULL; + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (pcm == NULL) { + snd_printk(KERN_ERR "Cannot allocate dynamic PCM\n"); + return -ENOMEM; + } + pcm->card = card; + pcm->device = device; + + if (id) + strlcpy(pcm->id, id, sizeof(pcm->id)); + if ((err = snd_pcm_new_stream_dynamic(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { + snd_pcm_free(pcm); + return err; + } + if ((err = snd_pcm_new_stream_dynamic(pcm, SNDRV_PCM_STREAM_CAPTURE, capture_count)) < 0) { + snd_pcm_free(pcm); + return err; + } + + if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { + snd_pcm_free(pcm); + return err; + } + if (rpcm) + *rpcm = pcm; + return 0; +} + +EXPORT_SYMBOL(snd_pcm_new_dynamic); + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -1035,6 +1157,29 @@ static int snd_pcm_dev_register(struct snd_device *device) return 0; } +static int snd_pcm_dev_register_dynamic(struct snd_device *device) +{ + int err; + struct snd_pcm_notify *notify; + struct snd_pcm *pcm; + + if (snd_BUG_ON(!device || !device->device_data)) + return -ENXIO; + pcm = device->device_data; + mutex_lock(&register_mutex); + err = snd_pcm_add(pcm); + if (err) { + mutex_unlock(&register_mutex); + return err; + } + + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_register(pcm); + + mutex_unlock(&register_mutex); + return 0; +} + static int snd_pcm_dev_disconnect(struct snd_device *device) { struct snd_pcm *pcm = device->device_data; -- 1.7.5.4

2 2

Re: [alsa-devel] FW: [PATCH v3] ASoC: Add support for TI LM49453 Audio codec
by M R Swami Reddy 08 Feb '12

08 Feb '12

> -----Original Message----- > From: Mark Brown [mailto:broonie@opensource.wolfsonmicro.com] > Sent: Tuesday, February 07, 2012 10:34 PM > To: Reddy, MR Swami > Cc: Girdwood, Liam; alsa-devel(a)alsa-project.org > Subject: Re: [PATCH v3] ASoC: Add support for TI LM49453 Audio codec > > On Mon, Feb 06, 2012 at 06:20:26AM -0800, Reddy, MR Swami wrote: > >> Changes made in v3: >> o Updated the lm49453_set_dai_pll() as per review comments in v2 patch. >> o Removed pll disable code in _set_dai_pll(). > > This doesn't really seem to address the issue at all - you still have the problems with the set_pll() function not doing anything it's supposed to do with the input and output frequencies, and now there's no way to disable the PLL. OK. So its ideal to remove the _set_dai_pll(), as its not doing anything here. Thanks Swami

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[alsa-devel] [RESEND PATCH] davinci: CQ93VC: remove machine specific header file inclusion from codec driver
by Manjunath Hadli 08 Feb '12

08 Feb '12

remove unnecessary inclusion of machine specific header file mach/dm365.h from cq93vc.c voice codec driver which comes in the way of platform code consolidation. Signed-off-by: Manjunath Hadli <manjunath.hadli(a)ti.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> Cc: <alsa-devel(a)alsa-project.org> Cc: <linux-kernel(a)vger.kernel.org> --- sound/soc/codecs/cq93vc.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 46dbfd0..ff4eb7a 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -38,8 +38,6 @@ #include <sound/soc.h> #include <sound/initval.h> -#include <mach/dm365.h> - static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, unsigned int reg) { -- 1.6.2.4

3 3

[alsa-devel] [PATCH 1/15] sound/soc/mxs/mxs-saif.c: add missing iounmap
by Julia Lawall 08 Feb '12

08 Feb '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Add missing iounmap in error handling code, in a case where the function already preforms iounmap on some other execution path. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression e; statement S,S1; int ret; @@ e = $ioremap\|ioremap_nocache$(...) ... when != iounmap(e) if (<+...e...+>) S ... when any when != iounmap(e) *if (...) { ... when != iounmap(e) return ...; } ... when any iounmap(e); // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/mxs/mxs-saif.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 049e543..5ee0adb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -680,7 +680,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_ioremap; + goto failed_get_resource; } saif->dma_param.chan_num = dmares->start; @@ -739,6 +739,7 @@ failed_register: failed_get_irq2: free_irq(saif->irq, saif); failed_get_irq1: +failed_get_resource: iounmap(saif->base); failed_ioremap: release_mem_region(iores->start, resource_size(iores));

6 29

[alsa-devel] concept of substream...
by Vinod Koul 08 Feb '12

08 Feb '12

Hi, I was trying to understand the concept of substream. So can we have a device which can have support say 5 substreams and these are controlled (start/stop) independent of each other. At any instance we can have any number of substreams active and runtime we should be able to activate/deactivate a particular substream. Is that doable with this concept or should i look at something else? -- ~Vinod

3 5

Re: [alsa-devel] [PATCH 1/6] hda_intel: Add Lynx Point HD Audio Controller DeviceIDs
by Takashi Iwai 08 Feb '12

08 Feb '12

At Wed, 8 Feb 2012 00:35:33 +0000, Heasley, Seth wrote: > > >> I'm also in favor of not rocking the boat, but I'm curious what > >changing this would entail, and what would be the advantage or rationale > >for making the change? > > > >Reading the LPIB register, and reading the DMA position buffer, are two > >different methods of reading the current playback (or recording) > >position, i e which sample is currently being played back. > > > >The problem is that some chipsets prefer one method over the other, i e, > >only one of the methods work reliably. And if the other method only > >breaks occasionally, this can be quite difficult to detect and track > >down - we will have users complaining about their audio sometimes either > >sounding distorted, maybe not working at all, or just once in a while > >glitches. All of these symptoms can have many causes, so deducing that > >to a broken playback position is time consuming. That's why I think it's > >worth the extra question, to get it right from the start. > > > >So...thanks in advance for looking it up for us? :-) > > I'm inquiring about it, but I can say that it seems to work just fine using the DMA position buffer. OK, I merged the patch now as is. Please let me know if you find a problem. thanks, Takashi

1 0

[alsa-devel] [PATCH] Add support for Echo3G devices. Thanks to unimatrix.
by Luke Yelavich 08 Feb '12

08 Feb '12

From: Luke Yelavich <luke.yelavich(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/433573 Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- src/conf/cards/Echo_Echo3G.conf | 318 +++++++++++++++++++++++++++++++++++++++ src/conf/cards/Makefile.am | 1 + 2 files changed, 319 insertions(+), 0 deletions(-) create mode 100644 src/conf/cards/Echo_Echo3G.conf diff --git a/src/conf/cards/Echo_Echo3G.conf b/src/conf/cards/Echo_Echo3G.conf new file mode 100644 index 0000000..766f13f --- /dev/null +++ b/src/conf/cards/Echo_Echo3G.conf @@ -0,0 +1,318 @@ +# +# Configuration for the Echo3G driver +# + +<confdir:pcm/front.conf> +Echo_Echo3G.pcm.front.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 0 +} + +<confdir:pcm/rear.conf> +Echo_Echo3G.pcm.rear.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 4 +} + +<confdir:pcm/center_lfe.conf> +Echo_Echo3G.pcm.center_lfe.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 2 +} + +<confdir:pcm/side.conf> +Echo_Echo3G.pcm.side.0 { + @args [ CARD ] + @args.CARD { + type string + } + type hw + card $CARD + device 0 + subdevice 6 +} + +<confdir:pcm/surround40.conf> +Echo_Echo3G.pcm.surround40.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + ] +} + +<confdir:pcm/surround41.conf> +Echo_Echo3G.pcm.surround41.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 0 } + ] +} + +<confdir:pcm/surround50.conf> +Echo_Echo3G.pcm.surround50.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 1 } + ] +} + +<confdir:pcm/surround51.conf> +Echo_Echo3G.pcm.surround51.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 0 } + { slave 2 channel 1 } + ] +} + +<confdir:pcm/surround71.conf> +Echo_Echo3G.pcm.surround71.0 { + @args [ CARD ] + @args.CARD { + type string + } + type multi + slaves [ + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.front.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.rear.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.center_lfe.0:CARD=" $CARD + ] + } + channels 2 + } + { + pcm { + @func concat + strings [ + "cards.Echo_Echo3G.pcm.side.0:CARD=" $CARD + ] + } + channels 2 + } + ] + bindings [ + { slave 0 channel 0 } + { slave 0 channel 1 } + { slave 1 channel 0 } + { slave 1 channel 1 } + { slave 2 channel 0 } + { slave 2 channel 1 } + { slave 3 channel 0 } + { slave 3 channel 1 } + ] +} + +<confdir:pcm/iec958.conf> +Echo_Echo3G.pcm.iec958.0 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { + type string + } + @args.AES0 { + type integer + } + @args.AES1 { + type integer + } + @args.AES2 { + type integer + } + @args.AES3 { + type integer + } + type hooks + slave.pcm { + type hw + card $CARD + device 1 + } + + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + preserve true + optional true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + ] + } +} diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index b7190e7..ee7991b 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -19,6 +19,7 @@ cfg_files = aliases.conf \ CMI8738-MC8.conf \ CMI8788.conf \ CS46xx.conf \ + Echo_Echo3G.conf \ EMU10K1.conf \ EMU10K1X.conf \ ENS1370.conf \ -- 1.7.9

2 1

[alsa-devel] Request for help CM109 on arm board
by Jonathan Andrews 08 Feb '12

08 Feb '12

Hi all, I've written an application on the PC that works with alsa, it uses 8 mono 22050Hz 16 bit audio channels via alsa. I'm trying to port the application to an Arm board with only a CM109 via USB. I have a working alsa library, I can play raw audio via the aplay utility and record via arecord ok. My problem is I don't understand the alsa configuration in enough detail to get it right. At the moment I have an alsa.conf cribbed from google, seems to work but only as a passthough. # cat /usr/share/alsa/alsa.conf_old pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } ARM / # lsmod snd_usb_audio 65715 0 - Live 0xbf082000 snd_hwdep 4756 1 snd_usb_audio, Live 0xbf07b000 snd_pcm 54351 1 snd_usb_audio, Live 0xbf062000 snd_timer 14870 1 snd_pcm, Live 0xbf058000 snd_page_alloc 3336 1 snd_pcm, Live 0xbf052000 snd_usbmidi_lib 14853 1 snd_usb_audio, Live 0xbf048000 snd_rawmidi 14657 1 snd_usbmidi_lib, Live 0xbf03d000 snd_seq_device 4675 1 snd_rawmidi, Live 0xbf036000 cm109 6943 0 - Live 0xbf011000 dm9601 5583 0 - Live 0xbf00a000 usbnet 11888 1 dm9601, Live 0xbf000000 ARM / # cat /proc/asound/cards 0 [default ]: USB-Audio - Generic USB Audio Device Generic USB Audio Device at usb-ep93xx-3, full speed ARM / # cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer ARM / # I do not have enough of a machine or distribution to use the alsaconf utility. Could someone help me with an alsa.conf that will do what roughly what I am after : IE to allow an application to open N channels each 22050Hz 16 bit mono to be mixed and resampled for an output via a CM109 that only provides 44.1 or 48Khz 16 bit interleaved ... phew ... Thanks, Jon

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[alsa-devel] [PATCH 1/2] ASoC: Convert kirkwood-t5325 to use snd_soc_register_card()
by Axel Lin 07 Feb '12

07 Feb '12

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I don't have this hardware, I'd appreciate if someone can test this patch. Axel arch/arm/mach-kirkwood/t5325-setup.c | 6 ++++ sound/soc/kirkwood/kirkwood-t5325.c | 47 +++++++++++++++++---------------- 2 files changed, 30 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-kirkwood/t5325-setup.c b/arch/arm/mach-kirkwood/t5325-setup.c index 966b2b3..f9d2a11 100644 --- a/arch/arm/mach-kirkwood/t5325-setup.c +++ b/arch/arm/mach-kirkwood/t5325-setup.c @@ -106,6 +106,11 @@ static struct platform_device hp_t5325_button_device = { } }; +static struct platform_device hp_t5325_audio_device = { + .name = "t5325-audio", + .id = -1, +}; + static unsigned int hp_t5325_mpp_config[] __initdata = { MPP0_NF_IO2, MPP1_SPI_MOSI, @@ -179,6 +184,7 @@ static void __init hp_t5325_init(void) kirkwood_sata_init(&hp_t5325_sata_data); kirkwood_ehci_init(); platform_device_register(&hp_t5325_button_device); + platform_device_register(&hp_t5325_audio_device); i2c_register_board_info(0, i2c_board_info, ARRAY_SIZE(i2c_board_info)); kirkwood_audio_init(); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 16b83e5..ce4f408 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -80,7 +80,6 @@ static struct snd_soc_dai_link t5325_dai[] = { }, }; - static struct snd_soc_card t5325 = { .name = "t5325", .owner = THIS_MODULE, @@ -93,38 +92,40 @@ static struct snd_soc_card t5325 = { .num_dapm_routes = ARRAY_SIZE(t5325_route), }; -static struct platform_device *t5325_snd_device; - -static int __init t5325_init(void) +static int __devinit t5325_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &t5325; int ret; - if (!machine_is_t5325()) - return 0; - - t5325_snd_device = platform_device_alloc("soc-audio", -1); - if (!t5325_snd_device) - return -ENOMEM; - - platform_set_drvdata(t5325_snd_device, - &t5325); - - ret = platform_device_add(t5325_snd_device); - if (ret) { - printk(KERN_ERR "%s: platform_device_add failed\n", __func__); - platform_device_put(t5325_snd_device); - } + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -module_init(t5325_init); -static void __exit t5325_exit(void) +static int __devexit t5325_remove(struct platform_device *pdev) { - platform_device_unregister(t5325_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_exit(t5325_exit); + +static struct platform_driver t5325_driver = { + .driver = { + .name = "t5325-audio", + .owner = THIS_MODULE, + }, + .probe = t5325_probe, + .remove = __devexit_p(t5325_remove), +}; + +module_platform_driver(t5325_driver); MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:t5325-audio"); -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: wm8994: Fix typo in VMID ramp setting
by Mark Brown 07 Feb '12

07 Feb '12

The VMID ramp rate is supposed to be 3b, not 0x11. Fix that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 81795eb..b75a652 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -785,7 +785,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_RAMP_MASK, WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x11 << WM8994_VMID_RAMP_SHIFT)); + (0x3 << WM8994_VMID_RAMP_SHIFT)); /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, -- 1.7.9.rc1

2 2

[alsa-devel] [RESUBMIT][PATCH v2 0/4] Amstrad Delta: access MODEM_RESET GPIO pin over a regulator
by Janusz Krzysztofik 07 Feb '12

07 Feb '12

The Amstrad Delta on-board latch2 bit named MODEM_NRESET, now available as a GPIO pin AMS_DELTA_GPIO_PIN_NMODEM_RESET, is used to power up/down (bring into/out of a reset state) two distinct on-board devices simultaneously: the modem, and the voice codec. As a consequence, that bit is, or can be, manipulated concurrently by two drivers, or their platform provided hooks. Instead of updating those drivers to use the gpiolib API as a new method of controlling the MODEM_NRESET pin state, like it was done to other drivers accessing latch2 pins, and still being vulnerable to potential concurrency conflicts, or trying to solve that sharing issue with a custom piece of code, set up a fixed regulator device on top of that GPIO pin and update both drivers to manipulate that regulator, not the GPIO pin directly. I'm resubmitting v2 of this series, originally posted on 30 Dec 2011, with Mark's Ack already collected, now rebased on top of current linux-omap/omap1 tip, commit 967809bd7faf71ddc29c8081e0f21db8b201a0f4. Since patch 2/4, "ASoC: cx20442: add bias control over a platform provided regulator", has already been applied by Liam (thanks!) and is already present in mainline, I've omitted it from this submission. v2 changes against initial version: * in both the codec and the modem callbacks, don't track the regulator enable/disable state, compare new target bias level (the codec case) or power state (the modem case) with the old value instead; thanks to Mark Brown who suggested this solution, * a few other minor changes, mostly stylistic. Janusz Krzysztofik (3): ARM: OMAP1: ams-delta: set up regulator over modem reset GPIO pin ARM: OMAP1: ams-delta: update the modem to use regulator API ASoC: OMAP: ams-delta: drop .set_bias_level callback arch/arm/mach-omap1/Kconfig | 2 + arch/arm/mach-omap1/board-ams-delta.c | 94 +++++++++++++++++++-- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/omap/ams-delta.c | 32 ------- 4 files changed, 87 insertions(+), 42 deletions(-) -- 1.7.3.4

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Re: [alsa-devel] linux-next: Tree for Feb 7 (sound/soc)
by Randy Dunlap 07 Feb '12

07 Feb '12

On 02/06/2012 10:09 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120206: > > > The sound-asoc tree lost its build failure. on x86_64: ERROR: "regmap_update_bits_check" [sound/soc/snd-soc-core.ko] undefined! Full randconfig file is attached. -- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

1 0

Re: [alsa-devel] [PATCH v3] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 07 Feb '12

07 Feb '12

On Mon, Feb 06, 2012 at 06:20:26AM -0800, Reddy, MR Swami wrote: > Changes made in v3: > o Updated the lm49453_set_dai_pll() as per review comments in v2 patch. > o Removed pll disable code in _set_dai_pll(). This doesn't really seem to address the issue at all - you still have the problems with the set_pll() function not doing anything it's supposed to do with the input and output frequencies, and now there's no way to disable the PLL.

1 0

[alsa-devel] [PATCH] ASoC: wm_hubs: Convert most output drivers to OUT_DRV widgets
by Mark Brown 07 Feb '12

07 Feb '12

No practical impact but now we have the control type we may as well use it for the slightly nicer sequencing. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 24 ++++++++++++------------ 1 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e469483..a53daf6 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -656,10 +656,10 @@ SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), SND_SOC_DAPM_SUPPLY("TSHUT", WM8993_POWER_MANAGEMENT_2, 14, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, - NULL, 0), -SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, + NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, line1_mix, ARRAY_SIZE(line1_mix)), @@ -675,14 +675,14 @@ SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, line2p_mix, ARRAY_SIZE(line2p_mix)), -SND_SOC_DAPM_PGA("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUTLP"), SND_SOC_DAPM_OUTPUT("SPKOUTLN"), -- 1.7.9.rc1

1 0

[alsa-devel] [patch] ALSA: hda ca0132 - remove unused error handling code
by Dan Carpenter 07 Feb '12

07 Feb '12

We never save the return value from chipio_write() so "err" is always zero here, not less than zero. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- The other option would be to save the return value. I don't have the hardware and I worried that there was a reason we don't save it. The current code is misleading because it looks like it's checked until you notice that the "err = " bit is missing. diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 35abe3c..92f1f0c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -733,8 +733,6 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol, /* *valp 0 is mute, 1 is unmute */ data = (data & 0x7f) | (*valp ? 0 : 0x80); chipio_write(codec, REG_CODEC_MUTE, data); - if (err < 0) - return err; spec->curr_hp_switch = *valp; @@ -775,8 +773,6 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol, /* *valp 0 is mute, 1 is unmute */ data = (data & 0xef) | (*valp ? 0 : 0x10); chipio_write(codec, REG_CODEC_MUTE, data); - if (err < 0) - return err; spec->curr_speaker_switch = *valp; @@ -824,14 +820,10 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol, val = 31 - left_vol; data = (data & 0xe0) | val; chipio_write(codec, REG_CODEC_HP_VOL_L, data); - if (err < 0) - return err; val = 31 - right_vol; data = (data & 0xe0) | val; chipio_write(codec, REG_CODEC_HP_VOL_R, data); - if (err < 0) - return err; spec->curr_hp_volume[0] = left_vol; spec->curr_hp_volume[1] = right_vol;

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Remove even more unused code
by Mark Brown 07 Feb '12

07 Feb '12

Now that we're not implement GPIO or microphone support there's no users for the read() function any more. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic3x.c | 24 ------------------------ 1 files changed, 0 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 07efbed..8d20f6e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -121,30 +121,6 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x02, /* 100 */ }; -/* - * read from the aic3x register space. Only use for this function is if - * wanting to read volatile bits from those registers that has both read-only - * and read/write bits. All other cases should use snd_soc_read. - */ -static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, - u8 *value) -{ - u8 *cache = codec->reg_cache; - - if (codec->cache_only) - return -EINVAL; - if (reg >= AIC3X_CACHEREGNUM) - return -1; - - codec->cache_bypass = 1; - *value = snd_soc_read(codec, reg); - codec->cache_bypass = 0; - - cache[reg] = *value; - - return 0; -} - #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ -- 1.7.9.rc1

3 3

[alsa-devel] [PATCH v2 0/4] Add device tree support of audio codec (ALC5632) on PAZ00 board
by Leon Romanovsky 07 Feb '12

07 Feb '12

The following patchset is going to: * add device tree support of audio codec (ALC5632) on PAZ00 board. * enable headset autodetection via device tree. * enable headset microphone via device tree. Leon Romanovsky (4): ARM: dt: tegra: Enable device tree audio codec on PAZ00 board. ARM: dt: tegra: Enable headset autodetection on PAZ00 board. ASoC: Tegra+ALC5632: Enable headset autodetection on PAZ00 board. ASoC: tegra: Remove unused DAPM route structure. arch/arm/boot/dts/tegra-paz00.dts | 30 ++++++++++++++++---- sound/soc/tegra/tegra_alc5632.c | 55 +++++++++++++++++++++++------------- 2 files changed, 59 insertions(+), 26 deletions(-) --- Chages since v1: * Headset autodetection commit was split tp be two independant commits. * Main commit(enable device tree audio codec) was forgotten in previous patchset. -- 1.7.3.4

5 14

Re: [alsa-devel] [PATCH 1/6] hda_intel: Add Lynx Point HD Audio Controller DeviceIDs
by David Henningsson 07 Feb '12

07 Feb '12

On 02/07/2012 01:13 AM, Heasley, Seth wrote: >>>> This patch adds the HD Audio DeviceIDs for the Intel Lynx Point PCH. >>> >>> Thanks. As we asked for oak trail: can you confirm you prefer DMA >>> Position buffers for detecting current playback/recording position, >> over >>> using the LPIB register? >> >> Yes, thanks for the notice. >> >> I guess this won't need it since it's the successor of PPT. >> But it's always good to check it for the new hardware, indeed. > > I'm also in favor of not rocking the boat, but I'm curious what changing this would entail, and what would be the advantage or rationale for making the change? Reading the LPIB register, and reading the DMA position buffer, are two different methods of reading the current playback (or recording) position, i e which sample is currently being played back. The problem is that some chipsets prefer one method over the other, i e, only one of the methods work reliably. And if the other method only breaks occasionally, this can be quite difficult to detect and track down - we will have users complaining about their audio sometimes either sounding distorted, maybe not working at all, or just once in a while glitches. All of these symptoms can have many causes, so deducing that to a broken playback position is time consuming. That's why I think it's worth the extra question, to get it right from the start. So...thanks in advance for looking it up for us? :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Bring WM8994 accessory detection up to date
by Mark Brown 06 Feb '12

06 Feb '12

Make the mechanism used for WM8994 more like that for WM1811 and WM8958: provide the logic to distinguish between headphone and headset and hard code the reporting of sensible SND_JACK values. Should integration with other detection mechanisms be required we can add appropriate callbacks (though some integrations should be able to use the subsystem ones). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 82 +++++++++++++++++++++++++++++++++----------- sound/soc/codecs/wm8994.h | 5 +-- 2 files changed, 63 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7df3159..298c5ae 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2927,8 +2927,6 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) * @codec: WM8994 codec * @jack: jack to report detection events on * @micbias: microphone bias to detect on - * @det: value to report for presence detection - * @shrt: value to report for short detection * * Enable microphone detection via IRQ on the WM8994. If GPIOs are * being used to bring out signals to the processor then only platform @@ -2939,43 +2937,63 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) * and micbias2_lvl platform data members. */ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, - int micbias, int det, int shrt) + int micbias) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_micdet *micdet; struct wm8994 *control = wm8994->wm8994; - int reg; + int reg, ret; - if (control->type != WM8994) + if (control->type != WM8994) { + dev_warn(codec->dev, "Not a WM8994\n"); return -EINVAL; + } switch (micbias) { case 1: micdet = &wm8994->micdet[0]; + if (jack) + ret = snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS1"); + else + ret = snd_soc_dapm_disable_pin(&codec->dapm, + "MICBIAS1"); break; case 2: micdet = &wm8994->micdet[1]; + if (jack) + ret = snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS1"); + else + ret = snd_soc_dapm_disable_pin(&codec->dapm, + "MICBIAS1"); break; default: + dev_warn(codec->dev, "Invalid MICBIAS %d\n", micbias); return -EINVAL; - } + } - dev_dbg(codec->dev, "Configuring microphone detection on %d: %x %x\n", - micbias, det, shrt); + if (ret != 0) + dev_warn(codec->dev, "Failed to configure MICBIAS%d: %d\n", + micbias, ret); + + dev_dbg(codec->dev, "Configuring microphone detection on %d %p\n", + micbias, jack); /* Store the configuration */ micdet->jack = jack; - micdet->det = det; - micdet->shrt = shrt; + micdet->detecting = true; /* If either of the jacks is set up then enable detection */ if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) reg = WM8994_MICD_ENA; - else + else reg = 0; snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg); + snd_soc_dapm_sync(&codec->dapm); + return 0; } EXPORT_SYMBOL_GPL(wm8994_mic_detect); @@ -3001,20 +3019,42 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) dev_dbg(codec->dev, "Microphone status: %x\n", reg); report = 0; - if (reg & WM8994_MIC1_DET_STS) - report |= priv->micdet[0].det; - if (reg & WM8994_MIC1_SHRT_STS) - report |= priv->micdet[0].shrt; + if (reg & WM8994_MIC1_DET_STS) { + if (priv->micdet[0].detecting) + report = SND_JACK_HEADSET; + } + if (reg & WM8994_MIC1_SHRT_STS) { + if (priv->micdet[0].detecting) + report = SND_JACK_HEADPHONE; + else + report |= SND_JACK_BTN_0; + } + if (report) + priv->micdet[0].detecting = false; + else + priv->micdet[0].detecting = true; + snd_soc_jack_report(priv->micdet[0].jack, report, - priv->micdet[0].det | priv->micdet[0].shrt); + SND_JACK_HEADSET | SND_JACK_BTN_0); report = 0; - if (reg & WM8994_MIC2_DET_STS) - report |= priv->micdet[1].det; - if (reg & WM8994_MIC2_SHRT_STS) - report |= priv->micdet[1].shrt; + if (reg & WM8994_MIC2_DET_STS) { + if (priv->micdet[1].detecting) + report = SND_JACK_HEADSET; + } + if (reg & WM8994_MIC2_SHRT_STS) { + if (priv->micdet[1].detecting) + report = SND_JACK_HEADPHONE; + else + report |= SND_JACK_BTN_0; + } + if (report) + priv->micdet[1].detecting = false; + else + priv->micdet[1].detecting = true; + snd_soc_jack_report(priv->micdet[1].jack, report, - priv->micdet[1].det | priv->micdet[1].shrt); + SND_JACK_HEADSET | SND_JACK_BTN_0); return IRQ_HANDLED; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c3a4247..f996d14 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -35,7 +35,7 @@ typedef void (*wm8958_micdet_cb)(u16 status, void *data); int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, - int micbias, int det, int shrt); + int micbias); int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micdet_cb cb, void *cb_data); @@ -46,8 +46,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec); struct wm8994_micdet { struct snd_soc_jack *jack; - int det; - int shrt; + bool detecting; }; /* codec private data */ -- 1.7.9.rc1

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[alsa-devel] [PATCH] ASoC: wm8994: We don't need to runtime resume by default
by Mark Brown 06 Feb '12

06 Feb '12

This is the default state that the runtime PM infrastructure expects so instead just kick the runtime PM core to suspend us if we're not doing anything (as is default). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 553b367..1653000 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3402,7 +3402,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) WM8994_IRQ_MIC1_DET; pm_runtime_enable(codec->dev); - pm_runtime_resume(codec->dev); + pm_runtime_idle(codec->dev); /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Unsuspend the device while reading GPIO statuses
by Mark Brown 06 Feb '12

06 Feb '12

Otherwise we might get an error if the GPIO is configured as an input since that makes the register volatile and a suspended device can't be read from. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1653000..a0eaf93 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3554,6 +3554,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->fll_locked_irq = false; } + /* Make sure we can read from the GPIOs if they're inputs */ + pm_runtime_get_sync(codec->dev); + /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically * at runtime we can deal with that then. @@ -3582,6 +3585,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->lrclk_shared[1] = 0; } + pm_runtime_put(codec->dev); + /* Latch volume updates (right only; we always do left then right). */ snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); -- 1.7.9.rc1

1 0

[alsa-devel] HDA irq understanding
by Raffaele Recalcati 06 Feb '12

06 Feb '12

Hi, I know my question is quite easy for this ml, but I hope to get a little help. I'm an embedded developer and I'm not so good with x86. I'm trying to load the system and hear mp3 decoding getting worst, but no way on my "Intel(R) Core(TM)2 Duo CPU P8400 @ 2.26GHz" very basic system. I'm trying to understand why I can't. Using trace (the complete one is here www.opensurf.it/trace12-02-04-19-14-48.txt_orig.lzma ) I get: grep HDA trace12-02-04-19-14-48.txt .. cpu-100.sh-26486 [000] 9490.976257: irq_handler_entry: irq=21 handler=HDA Intel cpu-100.sh-26474 [000] 9490.984236: irq_handler_entry: irq=21 handler=HDA Intel cpu-100.sh-26467 [000] 9490.992220: irq_handler_entry: irq=21 handler=HDA Intel cpu-100.sh-26502 [000] 9491.088042: irq_handler_entry: irq=21 handler=HDA Intel .. almost every 10msec pulseaudio reads from /dev/snd/pcmC0D0p, mplayer reads from pulseaudio. How can I create context switch problem in this situation and trace is well ? Thanks, Raffaele --------------------- Loading the system --------------------- where I have lauched cpu-100.sh 20 times $ cat cpu-100.sh #!/bin/bash while : ; do true done I have also launched root $ ping -l 100000 -q -s 10 -f localhost & once --- www.opensurf.it

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[alsa-devel] [PATCH] ASoC: core - Add support for DAI and machine kcontrols.
by Liam Girdwood 06 Feb '12

06 Feb '12

Currently ASoC can only add kcontrols using codec and platform component device handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for SoC card machine drivers too. This allows the kcontrol to have a direct handle to the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily get it's private data. This change makes snd_soc_add_controls() static and wraps it in the folowing calls (card and dai are new) :- snd_soc_add_card_controls() snd_soc_add_codec_controls() snd_soc_add_dai_controls() snd_soc_add_platform_controls() This patch also does a lot of small mechanical changes in individual codec drivers to replace snd_soc_add_controls() with snd_soc_add_codec_controls(). It also updates the McBSP DAI driver to use snd_soc_add_dai_controls(). Finally, it updates the existing machine drivers that register controls to either :- 1) Use snd_soc_add_card_controls() where no direct codec control is required. 2) Use snd_soc_add_codec_controls() where there is direct codec control. In the case of 1) above we also update the machine drivers to get the correct component data pointers from the kcontrol (rather than getting the machine pointer via the codec pointer). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 6 ++- sound/soc/codecs/ad1836.c | 6 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 8 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/max98088.c | 4 +- sound/soc/codecs/max98095.c | 6 +- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/uda134x.c | 6 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8904.c | 16 +++--- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 14 ++-- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 6 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 4 +- sound/soc/codecs/wm8994.c | 12 ++-- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 4 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 6 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/mid-x86/mfld_machine.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 15 ++--- sound/soc/omap/omap-mcbsp.c | 8 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 25 ++++---- sound/soc/pxa/corgi.c | 14 ++--- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/poodle.c | 14 ++--- sound/soc/pxa/tosa.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 4 +- sound/soc/samsung/s3c24xx_simtec.c | 6 +- sound/soc/soc-core.c | 108 +++++++++++++++++++++++------------- 60 files changed, 203 insertions(+), 173 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0992dff..9e96b4b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -368,10 +368,14 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, void *data, char *long_name, const char *prefix); -int snd_soc_add_controls(struct snd_soc_codec *codec, +int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_add_platform_controls(struct snd_soc_platform *platform, const struct snd_kcontrol_new *controls, int num_controls); +int snd_soc_add_card_controls(struct snd_soc_card *soc_card, + const struct snd_kcontrol_new *controls, int num_controls); +int snd_soc_add_dai_controls(struct snd_soc_dai *dai, + const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 982d201..12e3b41 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -277,7 +277,7 @@ static int ad1836_probe(struct snd_soc_codec *codec) if (ad1836->type == AD1836) { /* left/right diff:PGA/MUX */ snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); - ret = snd_soc_add_controls(codec, ad1836_controls, + ret = snd_soc_add_codec_controls(codec, ad1836_controls, ARRAY_SIZE(ad1836_controls)); if (ret) return ret; @@ -285,11 +285,11 @@ static int ad1836_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00); } - ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2); + ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2); if (ret) return ret; - ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs); + ret = snd_soc_add_codec_controls(codec, ad183x_adc_controls, num_adcs); if (ret) return ret; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 9bba7f8..8c39ddd 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -228,7 +228,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - snd_soc_add_controls(codec, ad1980_snd_ac97_controls, + snd_soc_add_codec_controls(codec, ad1980_snd_ac97_controls, ARRAY_SIZE(ad1980_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 971ba45..51e0d05 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1259,7 +1259,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) pdata->drc_setting[i]); } - snd_soc_add_controls(codec, adau1373_drc_controls, + snd_soc_add_codec_controls(codec, adau1373_drc_controls, pdata->num_drc); val = 0; @@ -1284,7 +1284,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) } if (!lineout_differential) { - snd_soc_add_controls(codec, adau1373_lineout2_controls, + snd_soc_add_codec_controls(codec, adau1373_lineout2_controls, ARRAY_SIZE(adau1373_lineout2_controls)); } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 9e809e05..dd15516 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -382,7 +382,7 @@ static int ak4535_probe(struct snd_soc_codec *codec) /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, ak4535_snd_controls, + snd_soc_add_codec_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); return 0; } diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5ef70b5..16bd1e7 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -476,7 +476,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_controls(codec, ak4642_snd_controls, + snd_soc_add_codec_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index a53b152..5fb7c2a 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,7 +628,7 @@ static int ak4671_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_controls(codec, ak4671_snd_controls, + snd_soc_add_codec_controls(codec, ak4671_snd_controls, ARRAY_SIZE(ak4671_snd_controls)); ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 3feee56..94089b4 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -925,22 +925,22 @@ static int alc5623_probe(struct snd_soc_codec *codec) switch (alc5623->id) { case 0x21: - snd_soc_add_controls(codec, alc5621_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls, ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: - snd_soc_add_controls(codec, alc5622_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls, ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: - snd_soc_add_controls(codec, alc5623_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls, ARRAY_SIZE(alc5623_vol_snd_controls)); break; default: return -EINVAL; } - snd_soc_add_controls(codec, alc5623_snd_controls, + snd_soc_add_codec_controls(codec, alc5623_snd_controls, ARRAY_SIZE(alc5623_snd_controls)); snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 390e437..9b0ab2f 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -994,7 +994,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) switch (alc5632->id) { case 0x5c: - snd_soc_add_controls(codec, alc5632_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls, ARRAY_SIZE(alc5632_vol_snd_controls)); break; default: diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 06d2ea1..064cd6a 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -157,7 +157,7 @@ static int cq93vc_probe(struct snd_soc_codec *codec) codec->control_data = davinci_vc; /* Set controls */ - snd_soc_add_controls(codec, cq93vc_snd_controls, + snd_soc_add_codec_controls(codec, cq93vc_snd_controls, ARRAY_SIZE(cq93vc_snd_controls)); /* Off, with power on */ diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0555366..6baccd2 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -521,7 +521,7 @@ static int cs4270_probe(struct snd_soc_codec *codec) } /* Add the non-DAPM controls */ - ret = snd_soc_add_controls(codec, cs4270_snd_controls, + ret = snd_soc_add_codec_controls(codec, cs4270_snd_controls, ARRAY_SIZE(cs4270_snd_controls)); if (ret < 0) { dev_err(codec->dev, "failed to add controls\n"); diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f6fe846..bf71412 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -513,7 +513,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) /* Power-up sequence requires 85 uS */ udelay(85); - return snd_soc_add_controls(codec, cs4271_snd_controls, + return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); } diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 3190392..ba4fafb 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -179,7 +179,7 @@ static int lm4857_probe(struct snd_soc_codec *codec) codec->control_data = lm4857->i2c; - ret = snd_soc_add_controls(codec, lm4857_controls, + ret = snd_soc_add_codec_controls(codec, lm4857_controls, ARRAY_SIZE(lm4857_controls)); if (ret) return ret; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 006efcf..af7324b 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1908,7 +1908,7 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec) max98088->eq_enum.texts = max98088->eq_texts; max98088->eq_enum.max = max98088->eq_textcnt; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); } @@ -2030,7 +2030,7 @@ static int max98088_probe(struct snd_soc_codec *codec) max98088_handle_pdata(codec); - snd_soc_add_controls(codec, max98088_snd_controls, + snd_soc_add_codec_controls(codec, max98088_snd_controls, ARRAY_SIZE(max98088_snd_controls)); err_access: diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index fcfa749..0bb511a 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1284,7 +1284,7 @@ static const struct snd_soc_dapm_route max98095_audio_map[] = { static int max98095_add_widgets(struct snd_soc_codec *codec) { - snd_soc_add_controls(codec, max98095_snd_controls, + snd_soc_add_codec_controls(codec, max98095_snd_controls, ARRAY_SIZE(max98095_snd_controls)); return 0; @@ -1984,7 +1984,7 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) max98095->eq_enum.texts = max98095->eq_texts; max98095->eq_enum.max = max98095->eq_textcnt; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); } @@ -2139,7 +2139,7 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) max98095->bq_enum.texts = max98095->bq_texts; max98095->bq_enum.max = max98095->bq_textcnt; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add Biquad control: %d\n", ret); } diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index dcf6f2a..3a2ba3d 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -253,7 +253,7 @@ static const struct snd_kcontrol_new max9877_controls[] = { /* This function is called from ASoC machine driver */ int max9877_add_controls(struct snd_soc_codec *codec) { - return snd_soc_add_controls(codec, max9877_controls, + return snd_soc_add_codec_controls(codec, max9877_controls, ARRAY_SIZE(max9877_controls)); } EXPORT_SYMBOL_GPL(max9877_add_controls); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index f99baa0..3a09faa 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -871,7 +871,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); - snd_soc_add_controls(codec, sn95031_snd_controls, + snd_soc_add_codec_controls(codec, sn95031_snd_controls, ARRAY_SIZE(sn95031_snd_controls)); return 0; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 333dd98..de2b205 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -548,7 +548,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, SSM2602_ROUT1V, ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); - ret = snd_soc_add_controls(codec, ssm2602_snd_controls, + ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); if (ret) return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index cc0566c..982e437 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -355,7 +355,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, stac9766_snd_ac97_controls, + snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index dfa41a9..16d55f9 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -593,7 +593,7 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); - snd_soc_add_controls(codec, tlv320aic23_snd_controls, + snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); return 0; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index a038dae..802064b 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -389,7 +389,7 @@ static int aic26_probe(struct snd_soc_codec *codec) /* register controls */ dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_controls(codec, aic26_snd_controls, + err = snd_soc_add_codec_controls(codec, aic26_snd_controls, ARRAY_SIZE(aic26_snd_controls)); WARN_ON(err < 0); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index eb401ef..37bb248 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -679,7 +679,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, aic32x4_snd_controls, + snd_soc_add_codec_controls(codec, aic32x4_snd_controls, ARRAY_SIZE(aic32x4_snd_controls)); aic32x4_add_widgets(codec); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 492f22f..400d1ec 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1426,10 +1426,10 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - snd_soc_add_controls(codec, aic3x_snd_controls, + snd_soc_add_codec_controls(codec, aic3x_snd_controls, ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) - snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f0aad26..4e679e8 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1440,7 +1440,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) /* Only add the FIFO controls, if we have valid IRQ number */ if (dac33->irq >= 0) - snd_soc_add_controls(codec, dac33_mode_snd_controls, + snd_soc_add_codec_controls(codec, dac33_mode_snd_controls, ARRAY_SIZE(dac33_mode_snd_controls)); err_power: diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 363b99d..6fe4aa3 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -351,10 +351,10 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) data = i2c_get_clientdata(tpa6130a2_client); if (data->id == TPA6140A2) - return snd_soc_add_controls(codec, tpa6140a2_controls, + return snd_soc_add_codec_controls(codec, tpa6140a2_controls, ARRAY_SIZE(tpa6140a2_controls)); else - return snd_soc_add_controls(codec, tpa6130a2_controls, + return snd_soc_add_codec_controls(codec, tpa6130a2_controls, ARRAY_SIZE(tpa6130a2_controls)); } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 8f4f469..797b0dd 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -531,15 +531,15 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: - ret = snd_soc_add_controls(codec, uda1340_snd_controls, + ret = snd_soc_add_codec_controls(codec, uda1340_snd_controls, ARRAY_SIZE(uda1340_snd_controls)); break; case UDA134X_UDA1341: - ret = snd_soc_add_controls(codec, uda1341_snd_controls, + ret = snd_soc_add_codec_controls(codec, uda1341_snd_controls, ARRAY_SIZE(uda1341_snd_controls)); break; case UDA134X_UDA1345: - ret = snd_soc_add_controls(codec, uda1345_snd_controls, + ret = snd_soc_add_codec_controls(codec, uda1345_snd_controls, ARRAY_SIZE(uda1345_snd_controls)); break; default: diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 44aacf9..3d868dc 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -464,7 +464,7 @@ static int wl1273_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, wl1273); - r = snd_soc_add_controls(codec, wl1273_controls, + r = snd_soc_add_codec_controls(codec, wl1273_controls, ARRAY_SIZE(wl1273_controls)); if (r) kfree(wl1273); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index ff95e62c..4fe9d19 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -599,7 +599,7 @@ static int wm8737_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - snd_soc_add_controls(codec, wm8737_snd_controls, + snd_soc_add_codec_controls(codec, wm8737_snd_controls, ARRAY_SIZE(wm8737_snd_controls)); wm8737_add_widgets(codec); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19374a9..fb09d1c 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -643,7 +643,7 @@ static int wm8770_probe(struct snd_soc_codec *codec) /* mute all DACs */ snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10, 0x10); - snd_soc_add_controls(codec, wm8770_snd_controls, + snd_soc_add_codec_controls(codec, wm8770_snd_controls, ARRAY_SIZE(wm8770_snd_controls)); snd_soc_dapm_new_controls(&codec->dapm, wm8770_dapm_widgets, ARRAY_SIZE(wm8770_dapm_widgets)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 14afc11..9f5f186 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1176,11 +1176,11 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) switch (wm8904->devtype) { case WM8904: - snd_soc_add_controls(codec, wm8904_adc_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_adc_snd_controls, ARRAY_SIZE(wm8904_adc_snd_controls)); - snd_soc_add_controls(codec, wm8904_dac_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_dac_snd_controls, ARRAY_SIZE(wm8904_dac_snd_controls)); - snd_soc_add_controls(codec, wm8904_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_snd_controls, ARRAY_SIZE(wm8904_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8904_adc_dapm_widgets, @@ -1201,7 +1201,7 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) break; case WM8912: - snd_soc_add_controls(codec, wm8904_dac_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_dac_snd_controls, ARRAY_SIZE(wm8904_dac_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets, @@ -2020,7 +2020,7 @@ static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts; wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts; - ret = snd_soc_add_controls(codec, &control, 1); + ret = snd_soc_add_codec_controls(codec, &control, 1); if (ret != 0) dev_err(codec->dev, "Failed to add ReTune Mobile control: %d\n", ret); @@ -2033,7 +2033,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) int ret, i; if (!pdata) { - snd_soc_add_controls(codec, wm8904_eq_controls, + snd_soc_add_codec_controls(codec, wm8904_eq_controls, ARRAY_SIZE(wm8904_eq_controls)); return; } @@ -2061,7 +2061,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) wm8904->drc_enum.max = pdata->num_drc_cfgs; wm8904->drc_enum.texts = wm8904->drc_texts; - ret = snd_soc_add_controls(codec, &control, 1); + ret = snd_soc_add_codec_controls(codec, &control, 1); if (ret != 0) dev_err(codec->dev, "Failed to add DRC mode control: %d\n", ret); @@ -2075,7 +2075,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) if (pdata->num_retune_mobile_cfgs) wm8904_handle_retune_mobile_pdata(codec); else - snd_soc_add_controls(codec, wm8904_eq_controls, + snd_soc_add_codec_controls(codec, wm8904_eq_controls, ARRAY_SIZE(wm8904_eq_controls)); } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index ae1933e..d2883af 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -717,7 +717,7 @@ static int wm8940_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_controls(codec, wm8940_snd_controls, + ret = snd_soc_add_codec_controls(codec, wm8940_snd_controls, ARRAY_SIZE(wm8940_snd_controls)); if (ret) return ret; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 8d4ea43..0502c9d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -920,11 +920,11 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->dsp_active = -1; - snd_soc_add_controls(codec, wm8958_mbc_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); - snd_soc_add_controls(codec, wm8958_vss_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_vss_snd_controls, ARRAY_SIZE(wm8958_vss_snd_controls)); - snd_soc_add_controls(codec, wm8958_enh_eq_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_enh_eq_snd_controls, ARRAY_SIZE(wm8958_enh_eq_snd_controls)); @@ -958,7 +958,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->mbc_enum.max = pdata->num_mbc_cfgs; wm8994->mbc_enum.texts = wm8994->mbc_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add MBC mode controls: %d\n", ret); @@ -986,7 +986,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_enum.max = pdata->num_vss_cfgs; wm8994->vss_enum.texts = wm8994->vss_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add VSS mode controls: %d\n", ret); @@ -1015,7 +1015,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add VSS HPFmode controls: %d\n", @@ -1045,7 +1045,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add enhanced EQ controls: %d\n", diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e5caae3..840d720 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -940,7 +940,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); - snd_soc_add_controls(codec, wm8960_snd_controls, + snd_soc_add_codec_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 4f20c72..05ea7c2 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1022,7 +1022,7 @@ static int wm8961_probe(struct snd_soc_codec *codec) wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8961_snd_controls, + snd_soc_add_codec_controls(codec, wm8961_snd_controls, ARRAY_SIZE(wm8961_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets, ARRAY_SIZE(wm8961_dapm_widgets)); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a20e2b7..3831764 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2403,13 +2403,13 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_add_controls(codec, wm8962_snd_controls, + snd_soc_add_codec_controls(codec, wm8962_snd_controls, ARRAY_SIZE(wm8962_snd_controls)); if (pdata && pdata->spk_mono) - snd_soc_add_controls(codec, wm8962_spk_mono_controls, + snd_soc_add_codec_controls(codec, wm8962_spk_mono_controls, ARRAY_SIZE(wm8962_spk_mono_controls)); else - snd_soc_add_controls(codec, wm8962_spk_stereo_controls, + snd_soc_add_codec_controls(codec, wm8962_spk_stereo_controls, ARRAY_SIZE(wm8962_spk_stereo_controls)); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index e538eda..9d24235 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1356,7 +1356,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_controls(codec, wm8990_snd_controls, + snd_soc_add_codec_controls(codec, wm8990_snd_controls, ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 7ee40da..9ac31ba 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1297,7 +1297,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_controls(codec, wm8991_snd_controls, + snd_soc_add_codec_controls(codec, wm8991_snd_controls, ARRAY_SIZE(wm8991_snd_controls)); snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 592f18e..045fe38 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1549,13 +1549,13 @@ static int wm8993_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - snd_soc_add_controls(codec, wm8993_snd_controls, + snd_soc_add_codec_controls(codec, wm8993_snd_controls, ARRAY_SIZE(wm8993_snd_controls)); if (wm8993->pdata.num_retune_configs != 0) { dev_dbg(codec->dev, "Using ReTune Mobile\n"); } else { dev_dbg(codec->dev, "No ReTune Mobile, using normal EQ\n"); - snd_soc_add_controls(codec, wm8993_eq_controls, + snd_soc_add_codec_controls(codec, wm8993_eq_controls, ARRAY_SIZE(wm8993_eq_controls)); } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 93d27b6..b0f3c8f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2861,7 +2861,7 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; - ret = snd_soc_add_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(wm8994->codec->dev, @@ -2914,7 +2914,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) wm8994->drc_enum.max = pdata->num_drc_cfgs; wm8994->drc_enum.texts = wm8994->drc_texts; - ret = snd_soc_add_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(wm8994->codec->dev, @@ -2930,7 +2930,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) if (pdata->num_retune_mobile_cfgs) wm8994_handle_retune_mobile_pdata(wm8994); else - snd_soc_add_controls(wm8994->codec, wm8994_eq_controls, + snd_soc_add_codec_controls(wm8994->codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { @@ -3638,7 +3638,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_handle_pdata(wm8994); wm_hubs_add_analogue_controls(codec); - snd_soc_add_controls(codec, wm8994_snd_controls, + snd_soc_add_codec_controls(codec, wm8994_snd_controls, ARRAY_SIZE(wm8994_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets, ARRAY_SIZE(wm8994_dapm_widgets)); @@ -3664,7 +3664,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - snd_soc_add_controls(codec, wm8958_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); @@ -3686,7 +3686,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM1811: - snd_soc_add_controls(codec, wm8958_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index c8aada5..8e397cc 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2137,7 +2137,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995_update_class_w(codec); - snd_soc_add_controls(codec, wm8995_snd_controls, + snd_soc_add_codec_controls(codec, wm8995_snd_controls, ARRAY_SIZE(wm8995_snd_controls)); snd_soc_dapm_new_controls(&codec->dapm, wm8995_dapm_widgets, ARRAY_SIZE(wm8995_dapm_widgets)); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index d8da10f..2814e26 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2763,7 +2763,7 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec) wm8996->retune_mobile_enum.max = wm8996->num_retune_mobile_texts; wm8996->retune_mobile_enum.texts = wm8996->retune_mobile_texts; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add ReTune Mobile controls: %d\n", ret); @@ -2961,7 +2961,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else - snd_soc_add_controls(codec, wm8996_eq_controls, + snd_soc_add_codec_controls(codec, wm8996_eq_controls, ARRAY_SIZE(wm8996_eq_controls)); /* If the TX LRCLK pins are not in LRCLK mode configure the diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a6bab39..7b09b1f 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1287,7 +1287,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); - snd_soc_add_controls(codec, wm9081_eq_controls, + snd_soc_add_codec_controls(codec, wm9081_eq_controls, ARRAY_SIZE(wm9081_eq_controls)); } diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a2b9208..e8280ee 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -433,7 +433,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_add_controls(codec, wm9090_controls, + snd_soc_add_codec_controls(codec, wm9090_controls, ARRAY_SIZE(wm9090_controls)); if (wm9090->pdata.lin1_diff) { @@ -442,7 +442,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) } else { snd_soc_dapm_add_routes(dapm, audio_map_in1_se, ARRAY_SIZE(audio_map_in1_se)); - snd_soc_add_controls(codec, wm9090_in1_se_controls, + snd_soc_add_codec_controls(codec, wm9090_in1_se_controls, ARRAY_SIZE(wm9090_in1_se_controls)); } @@ -452,7 +452,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) } else { snd_soc_dapm_add_routes(dapm, audio_map_in2_se, ARRAY_SIZE(audio_map_in2_se)); - snd_soc_add_controls(codec, wm9090_in2_se_controls, + snd_soc_add_codec_controls(codec, wm9090_in2_se_controls, ARRAY_SIZE(wm9090_in2_se_controls)); } diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 40c92ea..cacc6a8 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -351,7 +351,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) if (ret) goto reset_err; - snd_soc_add_controls(codec, wm9705_snd_ac97_controls, + snd_soc_add_codec_controls(codec, wm9705_snd_ac97_controls, ARRAY_SIZE(wm9705_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b7b31f8..7faa2d4 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -637,7 +637,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm9712_snd_ac97_controls, + snd_soc_add_codec_controls(codec, wm9712_snd_ac97_controls, ARRAY_SIZE(wm9712_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2b8479b..2d22cc7 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1216,7 +1216,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - snd_soc_add_controls(codec, wm9713_snd_ac97_controls, + snd_soc_add_codec_controls(codec, wm9713_snd_ac97_controls, ARRAY_SIZE(wm9713_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2a61094..32a5f82 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -895,7 +895,7 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); - snd_soc_add_controls(codec, analogue_snd_controls, + snd_soc_add_codec_controls(codec, analogue_snd_controls, ARRAY_SIZE(analogue_snd_controls)); snd_soc_dapm_new_controls(dapm, analogue_dapm_widgets, diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 6f77eef..2937e54 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -235,7 +235,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Headphones"); snd_soc_dapm_enable_pin(dapm, "Mic"); - ret_val = snd_soc_add_controls(codec, mfld_snd_controls, + ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); if (ret_val) { pr_err("soc_add_controls failed %d", ret_val); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a04a433..f406285 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -572,7 +572,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "AGCOUT"); /* Add virtual switch */ - ret = snd_soc_add_controls(codec, ams_delta_audio_controls, + ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls, ARRAY_SIZE(ams_delta_audio_controls)); if (ret) dev_warn(card->dev, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fc6209b..630b419 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -55,9 +55,8 @@ static int n810_spk_func; static int n810_jack_func; static int n810_dmic_func; -static void n810_ext_control(struct snd_soc_codec *codec) +static void n810_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int hp = 0, line1l = 0; switch (n810_jack_func) { @@ -102,7 +101,7 @@ static int n810_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - n810_ext_control(codec); + n810_ext_control(&codec->dapm); return clk_enable(sys_clkout2); } @@ -142,13 +141,13 @@ static int n810_get_spk(struct snd_kcontrol *kcontrol, static int n810_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (n810_spk_func == ucontrol->value.integer.value[0]) return 0; n810_spk_func = ucontrol->value.integer.value[0]; - n810_ext_control(codec); + n810_ext_control(&card->dapm); return 1; } @@ -164,13 +163,13 @@ static int n810_get_jack(struct snd_kcontrol *kcontrol, static int n810_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (n810_jack_func == ucontrol->value.integer.value[0]) return 0; n810_jack_func = ucontrol->value.integer.value[0]; - n810_ext_control(codec); + n810_ext_control(&card->dapm); return 1; } @@ -186,7 +185,7 @@ static int n810_get_input(struct snd_kcontrol *kcontrol, static int n810_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (n810_dmic_func == ucontrol->value.integer.value[0]) return 0; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 0173719..1287b87 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -744,17 +744,17 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { omap_mcbsp3_set_st_ch1_volume), }; -int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id) +int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai) { if (!cpu_is_omap34xx()) return -ENODEV; - switch (mcbsp_id) { + switch (dai->id) { case 1: /* McBSP 2 */ - return snd_soc_add_controls(codec, omap_mcbsp2_st_controls, + return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls, ARRAY_SIZE(omap_mcbsp2_st_controls)); case 2: /* McBSP 3 */ - return snd_soc_add_controls(codec, omap_mcbsp3_st_controls, + return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls, ARRAY_SIZE(omap_mcbsp3_st_controls)); default: break; diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 65cde9d3..476fe2a 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -59,6 +59,6 @@ enum omap_mcbsp_div { #define NUM_LINKS 5 #endif -int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id); +int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index ad16db5..04e4d25 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -59,9 +59,8 @@ static int rx51_spk_func; static int rx51_dmic_func; static int rx51_jack_func; -static void rx51_ext_control(struct snd_soc_codec *codec) +static void rx51_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int hp = 0, hs = 0, tvout = 0; switch (rx51_jack_func) { @@ -102,11 +101,11 @@ static int rx51_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 0; } @@ -138,13 +137,13 @@ static int rx51_get_spk(struct snd_kcontrol *kcontrol, static int rx51_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (rx51_spk_func == ucontrol->value.integer.value[0]) return 0; rx51_spk_func = ucontrol->value.integer.value[0]; - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 1; } @@ -184,13 +183,13 @@ static int rx51_get_input(struct snd_kcontrol *kcontrol, static int rx51_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (rx51_dmic_func == ucontrol->value.integer.value[0]) return 0; rx51_dmic_func = ucontrol->value.integer.value[0]; - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 1; } @@ -206,13 +205,13 @@ static int rx51_get_jack(struct snd_kcontrol *kcontrol, static int rx51_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (rx51_jack_func == ucontrol->value.integer.value[0]) return 0; rx51_jack_func = ucontrol->value.integer.value[0]; - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 1; } @@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINE1R"); /* Add RX-51 specific controls */ - err = snd_soc_add_controls(codec, aic34_rx51_controls, + err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls, ARRAY_SIZE(aic34_rx51_controls)); if (err < 0) return err; @@ -314,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); - err = omap_mcbsp_st_add_controls(codec, 1); + err = omap_mcbsp_st_add_controls(rtd->cpu_dai); if (err < 0) return err; @@ -335,7 +334,7 @@ static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm) { int err; - err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb, + err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb, ARRAY_SIZE(aic34_rx51_controlsb)); if (err < 0) return err; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index bc21944..863367a 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -45,10 +45,8 @@ static int corgi_jack_func; static int corgi_spk_func; -static void corgi_ext_control(struct snd_soc_codec *codec) +static void corgi_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: @@ -104,7 +102,7 @@ static int corgi_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - corgi_ext_control(codec); + corgi_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -173,13 +171,13 @@ static int corgi_get_jack(struct snd_kcontrol *kcontrol, static int corgi_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (corgi_jack_func == ucontrol->value.integer.value[0]) return 0; corgi_jack_func = ucontrol->value.integer.value[0]; - corgi_ext_control(codec); + corgi_ext_control(&card->dapm); return 1; } @@ -193,13 +191,13 @@ static int corgi_get_spk(struct snd_kcontrol *kcontrol, static int corgi_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (corgi_spk_func == ucontrol->value.integer.value[0]) return 0; corgi_spk_func = ucontrol->value.integer.value[0]; - corgi_ext_control(codec); + corgi_ext_control(&card->dapm); return 1; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 3f7a8ec..aace19e 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -411,7 +411,7 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "VINR"); /* Add magician specific controls */ - err = snd_soc_add_controls(codec, uda1380_magician_controls, + err = snd_soc_add_codec_controls(codec, uda1380_magician_controls, ARRAY_SIZE(uda1380_magician_controls)); if (err < 0) return err; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index fd0ed10..d2cc8173 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -43,10 +43,8 @@ static int poodle_jack_func; static int poodle_spk_func; -static void poodle_ext_control(struct snd_soc_codec *codec) +static void poodle_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -81,7 +79,7 @@ static int poodle_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - poodle_ext_control(codec); + poodle_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -152,13 +150,13 @@ static int poodle_get_jack(struct snd_kcontrol *kcontrol, static int poodle_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (poodle_jack_func == ucontrol->value.integer.value[0]) return 0; poodle_jack_func = ucontrol->value.integer.value[0]; - poodle_ext_control(codec); + poodle_ext_control(&card->dapm); return 1; } @@ -172,13 +170,13 @@ static int poodle_get_spk(struct snd_kcontrol *kcontrol, static int poodle_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (poodle_spk_func == ucontrol->value.integer.value[0]) return 0; poodle_spk_func = ucontrol->value.integer.value[0]; - poodle_ext_control(codec); + poodle_ext_control(&card->dapm); return 1; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 564ef08..2aec63f 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -197,7 +197,7 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "MONOOUT"); /* add tosa specific controls */ - err = snd_soc_add_controls(codec, tosa_controls, + err = snd_soc_add_codec_controls(codec, tosa_controls, ARRAY_SIZE(tosa_controls)); if (err < 0) return err; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7ac0ba2..e34f4e8 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -298,7 +298,7 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) if (ret) return ret; - ret = snd_soc_add_controls(codec, neo1973_gta02_wm8753_controls, + ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls, ARRAY_SIZE(neo1973_gta02_wm8753_controls)); if (ret) return ret; @@ -338,7 +338,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return ret; /* add neo1973 specific controls */ - ret = snd_soc_add_controls(codec, neo1973_wm8753_controls, + ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls, ARRAY_SIZE(neo1973_wm8753_controls)); if (ret) return ret; diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index a253bcc..656d5af 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -134,18 +134,18 @@ static const struct snd_kcontrol_new amp_unmute_controls[] = { void simtec_audio_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; if (pdata->amp_gpio > 0) { pr_debug("%s: adding amp routes\n", __func__); - snd_soc_add_controls(codec, amp_unmute_controls, + snd_soc_add_card_controls(card, amp_unmute_controls, ARRAY_SIZE(amp_unmute_controls)); } if (pdata->amp_gain[0] > 0) { pr_debug("%s: adding amp controls\n", __func__); - snd_soc_add_controls(codec, amp_gain_controls, + snd_soc_add_card_controls(card, amp_gain_controls, ARRAY_SIZE(amp_gain_controls)); } } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b48cf4f..cbaf595 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -992,7 +992,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } if (driver->controls) - snd_soc_add_controls(codec, driver->controls, + snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); if (driver->dapm_routes) snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, @@ -1453,13 +1453,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } - /* We should have a non-codec control add function but we don't */ if (card->controls) - snd_soc_add_controls(list_first_entry(&card->codec_dev_list, - struct snd_soc_codec, - card_list), - card->controls, - card->num_controls); + snd_soc_add_card_controls(card, card->controls, card->num_controls); if (card->dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, @@ -2007,9 +2002,28 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, } EXPORT_SYMBOL_GPL(snd_soc_cnew); +static int snd_soc_add_controls(struct snd_card *card, struct device *dev, + const struct snd_kcontrol_new *controls, int num_controls, + const char *prefix, void *data) +{ + int err, i; + + for (i = 0; i < num_controls; i++) { + const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, data, + control->name, prefix)); + if (err < 0) { + dev_err(dev, "Failed to add %s: %d\n", control->name, err); + return err; + } + } + + return 0; +} + /** - * snd_soc_add_controls - add an array of controls to a codec. - * Convienience function to add a list of controls. Many codecs were + * snd_soc_add_codec_controls - add an array of controls to a codec. + * Convenience function to add a list of controls. Many codecs were * duplicating this code. * * @codec: codec to add controls to @@ -2018,31 +2032,19 @@ EXPORT_SYMBOL_GPL(snd_soc_cnew); * * Return 0 for success, else error. */ -int snd_soc_add_controls(struct snd_soc_codec *codec, +int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; - int err, i; - for (i = 0; i < num_controls; i++) { - const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, codec, - control->name, - codec->name_prefix)); - if (err < 0) { - dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, control->name, err); - return err; - } - } - - return 0; + return snd_soc_add_controls(card, codec->dev, controls, num_controls, + codec->name_prefix, codec); } -EXPORT_SYMBOL_GPL(snd_soc_add_controls); +EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); /** * snd_soc_add_platform_controls - add an array of controls to a platform. - * Convienience function to add a list of controls. + * Convenience function to add a list of controls. * * @platform: platform to add controls to * @controls: array of controls to add @@ -2054,23 +2056,53 @@ int snd_soc_add_platform_controls(struct snd_soc_platform *platform, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = platform->card->snd_card; - int err, i; - for (i = 0; i < num_controls; i++) { - const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, platform, - control->name, NULL)); - if (err < 0) { - dev_err(platform->dev, "Failed to add %s %d\n",control->name, err); - return err; - } - } - - return 0; + return snd_soc_add_controls(card, platform->dev, controls, num_controls, + NULL, platform); } EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls); /** + * snd_soc_add_card_controls - add an array of controls to a SoC card. + * Convenience function to add a list of controls. + * + * @soc_card: SoC card to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_card_controls(struct snd_soc_card *soc_card, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = soc_card->snd_card; + + return snd_soc_add_controls(card, soc_card->dev, controls, num_controls, + NULL, soc_card); +} +EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); + +/** + * snd_soc_add_dai_controls - add an array of controls to a DAI. + * Convienience function to add a list of controls. + * + * @dai: DAI to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_dai_controls(struct snd_soc_dai *dai, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = dai->card->snd_card; + + return snd_soc_add_controls(card, dai->dev, controls, num_controls, + NULL, dai); +} +EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls); + +/** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control * @uinfo: control element information -- 1.7.5.4

3 12

[alsa-devel] [PATCH] mfd: wm8994: Mark MICBIAS register as readable
by Mark Brown 06 Feb '12

06 Feb '12

Omitted in the regmap conversion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm8994-regmap.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm8994-regmap.c b/drivers/mfd/wm8994-regmap.c index c598ae6..47743fa 100644 --- a/drivers/mfd/wm8994-regmap.c +++ b/drivers/mfd/wm8994-regmap.c @@ -968,6 +968,7 @@ static bool wm8994_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case WM8994_DC_SERVO_READBACK: + case WM8994_MICBIAS: case WM8994_WRITE_SEQUENCER_CTRL_1: case WM8994_WRITE_SEQUENCER_CTRL_2: case WM8994_AIF1_ADC2_LEFT_VOLUME: -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 0/2] sound: soc: trivial stuff for tlv320aic3x
by Felipe Contreras 06 Feb '12

06 Feb '12

Not important, but nice to have :) Felipe Contreras (2): sound: soc: add alias for tlv320aic3x sound: soc: tlv320aic3x: remove unused code sound/soc/codecs/tlv320aic3x.c | 37 +------------------------------------ sound/soc/codecs/tlv320aic3x.h | 9 --------- 2 files changed, 1 insertions(+), 45 deletions(-) -- 1.7.9

3 6

[alsa-devel] [PATCH] ASoC: DAPM - Clean up header information.
by Liam Girdwood 06 Feb '12

06 Feb '12

Fix some spelling mistakes in the header and remove the todo items. Most todo items are now available as kcontrol options now anyway. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 10 ++-------- 1 files changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e2a9797..87732d5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -14,19 +14,13 @@ * dynamic configuration of codec internal audio paths and active * DACs/ADCs. * o Platform power domain - can support external components i.e. amps and - * mic/meadphone insertion events. + * mic/headphone insertion events. * o Automatic Mic Bias support * o Jack insertion power event initiation - e.g. hp insertion will enable * sinks, dacs, etc - * o Delayed powerdown of audio susbsystem to reduce pops between a quick + * o Delayed power down of audio subsystem to reduce pops between a quick * device reopen. * - * Todo: - * o DAPM power change sequencing - allow for configurable per - * codec sequences. - * o Support for analogue bias optimisation. - * o Support for reduced codec oversampling rates. - * o Support for reduced codec bias currents. */ #include <linux/module.h> -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: DAPM - Export mixer|mux_update_power() to public API.
by Liam Girdwood 06 Feb '12

06 Feb '12

Allow for the operation of custom mixer and mux DAPM widgets that can call snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly after updating their status. This is useful with complex DAPM Mixer operations where we need to do additional work in addition to setting a few mixer register bits. Changes since V1:- o Remove the unused "change" flag parameter from snd_soc_dapm_mux_update_power(). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++++++ sound/soc/soc-dapm.c | 21 +++++++++------------ 2 files changed, 15 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d26a9b7..7794737 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -365,6 +365,12 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, const char *stream, int event); void snd_soc_dapm_shutdown(struct snd_soc_card *card); +/* external DAPM widget events */ +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int connect); +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e); + /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3ad1f59..e2a9797 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1659,9 +1659,8 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int change, - int mux, struct soc_enum *e) +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -1671,9 +1670,6 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_value_mux) return -ENODEV; - if (!change) - return 0; - /* find dapm widget path assoc with kcontrol */ list_for_each_entry(path, &widget->dapm->card->paths, list) { if (path->kcontrol != kcontrol) @@ -1702,9 +1698,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; @@ -1733,6 +1730,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); /* show dapm widget status in sys fs */ static ssize_t dapm_widget_show(struct device *dev, @@ -2315,7 +2313,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mixer_update_power(widget, kcontrol, connect); + snd_soc_dapm_mixer_update_power(widget, kcontrol, connect); widget->dapm->update = NULL; } @@ -2406,7 +2404,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } @@ -2467,8 +2465,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, widget->value = ucontrol->value.enumerated.item[0]; - dapm_mux_update_power(widget, kcontrol, change, - widget->value, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); } } @@ -2571,7 +2568,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: DAPM - Export mixer|mux_update_power() to public API.
by Liam Girdwood 06 Feb '12

06 Feb '12

Allow for the operation of custom mixer and mux DAPM widgets that can call snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly after updating their status. This is useful with complex DAPM Mixer operations where we need to do additional work in addition to setting a few mixer register bits. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 7 +++++++ sound/soc/soc-dapm.c | 16 +++++++++------- 2 files changed, 16 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d26a9b7..4f12faa 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -365,6 +365,13 @@ int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, const char *stream, int event); void snd_soc_dapm_shutdown(struct snd_soc_card *card); +/* external DAPM widget events */ +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int connect); +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int change, + int mux, struct soc_enum *e); + /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3ad1f59..49b944a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1659,7 +1659,7 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int change, int mux, struct soc_enum *e) { @@ -1702,9 +1702,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; @@ -1733,6 +1734,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); /* show dapm widget status in sys fs */ static ssize_t dapm_widget_show(struct device *dev, @@ -2315,7 +2317,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mixer_update_power(widget, kcontrol, connect); + snd_soc_dapm_mixer_update_power(widget, kcontrol, connect); widget->dapm->update = NULL; } @@ -2406,7 +2408,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, change, mux, e); widget->dapm->update = NULL; } @@ -2467,8 +2469,8 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, widget->value = ucontrol->value.enumerated.item[0]; - dapm_mux_update_power(widget, kcontrol, change, - widget->value, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, change, + widget->value, e); } } @@ -2571,7 +2573,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, change, mux, e); widget->dapm->update = NULL; } -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: samsung: Enable accessory detect for WM8994 on Littlemill
by Mark Brown 06 Feb '12

06 Feb '12

The WM8994 has a different accessory detect architecture, call its setup function too. We ignore the errors and the driver will check for chip type so it's safe to call the setup functions for both architectures. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/littlemill.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 9dd818b..e741685 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -189,6 +189,9 @@ static int littlemill_late_probe(struct snd_soc_card *card) /* This will check device compatibility itself */ wm8958_mic_detect(codec, &littlemill_headset, NULL, NULL); + /* As will this */ + wm8994_mic_detect(codec, &littlemill_headset, 1); + return 0; } -- 1.7.9.rc1

1 0

Re: [alsa-devel] [Pkg-alsa-devel] Bug#658762: arecord: manpage has typos in examples
by Elimar Riesebieter 06 Feb '12

06 Feb '12

forwarded 658762 alsa-devel(a)alsa-project.org tags 658762 patch thanks * Yann Dirson [120205 22:42 +0100]: > Package: alsa-utils > Version: 1.0.24.2-4 > Severity: normal > > The examples in manpage talk about -max-file_time, while the correct > name is --max-file-time (2 typos packed in one option, at least twice > in the manpage). Patch attached Elimar -- We all know Linux is great... it does infinite loops in 5 seconds. -Linus Torvalds

1 0

Re: [alsa-devel] [Pkg-alsa-devel] Bug#658760: alsactl -h advertises non-existent command "names"
by Elimar Riesebieter 06 Feb '12

06 Feb '12

forwarded 658760 alsa-devel(a)alsa-project.org tags 658760 patch thanks * Yann Dirson [120205 22:18 +0100]: > Package: alsa-utils > Version: 1.0.24.2-4 > Severity: normal > > # alsactl -h > Usage: alsactl <options> command > > [...] > > Available commands: > store <card #> save current driver setup for one or each soundcards > to configuration file > restore <card #> load current driver setup for one or each soundcards > from configuration file > init <card #> initialize driver to a default state > names <card #> dump information about all the known present (sub-)devices > into configuration file (DEPRECATED) > > > Not only that command is deprecated, but it appears it does not exist > any more. Patch attached Elimar -- Numeric stability is probably not all that important when you're guessing;-)

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Enabling VMID should take a runtime PM reference
by Mark Brown 06 Feb '12

06 Feb '12

We can enable VMID independently of the bias in some use cases so we need to ensure that the core device is powered up. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 365c16f..48fe205 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -770,6 +770,8 @@ static void vmid_reference(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + pm_runtime_get_sync(codec->dev); + wm8994->vmid_refcount++; dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n", @@ -837,6 +839,8 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); } + + pm_runtime_put(codec->dev); } static int vmid_event(struct snd_soc_dapm_widget *w, -- 1.7.9.rc1

1 0

[alsa-devel] Sending "ASoC: Ensure we generate a driver name" to stable
by Paul Fertser 06 Feb '12

06 Feb '12

Hi, Can you please consider sending f0e8ed858edb327802ee65fd695cc1538286226f (ASoC: Ensure we generate a driver name) for inclusion in the Greg's stable tree as it's essential to get nice alsa-lib autoconfiguration for the supported cards. -- Be free, use free (http://www.gnu.org/philosophy/free-sw.html) software! mailto:fercerpav@gmail.com

2 1

[alsa-devel] ASoC updates for 3.3
by Mark Brown 06 Feb '12

06 Feb '12

The following changes since commit 1ae5cbc52e7c6619a3f44b87809fd25370df31bb: ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine (2012-01-30 13:22:13 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 for you to fetch changes up to 43b6cec27e1e50a1de3eff47e66e502f3fe7e66e: ASoC: wm_hubs: Correct line input to line output 2 paths (2012-02-01 23:55:56 +0000) ---------------------------------------------------------------- The only particularly remarkable change here is the one for handling of the Android suspend ignore code for idle_bias_off CODECs. That one is actually a regression fix as some of the new power savings that have been introduced confused the suspend ignore code, making devices that are active for non-audio reasons look like they are idle causing them to be suspended instead of being kept active. ---------------------------------------------------------------- Axel Lin (1): ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode Mark Brown (4): ASoC: wm_hubs: Fix routing of input PGAs to line output mixer ASoC: wm8994: Remove ASoC level register cache sync ASoC: core: Better support for idle_bias_off suspend ignores ASoC: wm_hubs: Correct line input to line output 2 paths Susan Gao (1): ASoC: wm8962: Fix word length configuration sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/wm8962.c | 6 +++--- sound/soc/codecs/wm8994.c | 5 ----- sound/soc/codecs/wm_hubs.c | 8 ++++---- sound/soc/soc-core.c | 11 +++++++++++ 5 files changed, 19 insertions(+), 13 deletions(-)

2 4

[alsa-devel] Jack detection on HDA vs SoC
by Dylan Reid 06 Feb '12

06 Feb '12

under hda I see: new, control-based jack notification, and auto-switching of the output/input done in the driver. under soc it is: /dev/input used for jack notification, with the switching _not_ done in the driver. Is there a plan for the soc code to use the control-based method? Is there a consensus on whether new drivers should auto-switch or not? Thanks, Dylan

4 6

Re: [alsa-devel] [PATCH 1/7 v2] dmaengine: add a simple dma library
by Guennadi Liakhovetski 06 Feb '12

06 Feb '12

Hi Shimoda-san On Fri, 27 Jan 2012, Shimoda, Yoshihiro wrote: > Hello Guennadi-san, > > 2012/01/26 23:56, Guennadi Liakhovetski wrote: > > This patch adds a library of functions, helping to implement dmaengine > > drivers for hardware, unable to handle scatter-gather lists natively. > > The first version of this driver only supports memcpy and slave DMA > > operation. > > > > Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> > > --- > > > > v2: > > > > 1. switch from using a tasklet to threaded IRQ, which allowed to > > I tested this driver on 2 environment, but it could not work correctly. > - renesas_usbhs driver with shdma driver on the mackerel board > - renesas_usbhs driver with experimental SUDMAC driver > > In this case, should we modify the renesas_usbhs driver? Yes, you do. Please, see the respective version of the shdma patch. It doesn't request channel IRQs any more directly, instead, it uses a dma-simple wrapper function. Also operations change a bit. Thanks Guennadi > For reference, if I changed the threaded IRQ to tasklet, > the 2 environment worked correctly: > ============== > diff --git a/drivers/dma/dma-simple.c b/drivers/dma/dma-simple.c > index 49d8f7d..65a5170 100644 > --- a/drivers/dma/dma-simple.c > +++ b/drivers/dma/dma-simple.c > @@ -715,16 +715,18 @@ static irqreturn_t chan_irq(int irq, void *dev) > > spin_lock(&schan->chan_lock); > > - ret = ops->chan_irq(schan, irq) ? IRQ_WAKE_THREAD : IRQ_NONE; > + ret = ops->chan_irq(schan, irq) ? IRQ_HANDLED : IRQ_NONE; > + if (ret == IRQ_HANDLED) > + tasklet_schedule(&schan->tasklet); > > spin_unlock(&schan->chan_lock); > > return ret; > } > > -static irqreturn_t chan_irqt(int irq, void *dev) > +static void chan_irqt(unsigned long dev) > { > - struct dma_simple_chan *schan = dev; > + struct dma_simple_chan *schan = (struct dma_simple_chan *)dev; > const struct dma_simple_ops *ops = > to_simple_dev(schan->dma_chan.device)->ops; > struct dma_simple_desc *sdesc; > @@ -744,15 +746,13 @@ static irqreturn_t chan_irqt(int irq, void *dev) > spin_unlock_irq(&schan->chan_lock); > > simple_chan_ld_cleanup(schan, false); > - > - return IRQ_HANDLED; > } > > int dma_simple_request_irq(struct dma_simple_chan *schan, int irq, > unsigned long flags, const char *name) > { > - int ret = request_threaded_irq(irq, chan_irq, chan_irqt, > - flags, name, schan); > + int ret = request_irq(irq, chan_irq, flags, name, schan); > + tasklet_init(&schan->tasklet, chan_irqt, (unsigned long)schan); > > schan->irq = ret < 0 ? ret : irq; > > diff --git a/include/linux/dma-simple.h b/include/linux/dma-simple.h > index 5336674..beb1489 100644 > --- a/include/linux/dma-simple.h > +++ b/include/linux/dma-simple.h > @@ -68,6 +68,7 @@ struct dma_simple_chan { > int id; /* Raw id of this channel */ > int irq; /* Channel IRQ */ > enum dma_simple_pm_state pm_state; > + struct tasklet_struct tasklet; > }; > > /** > ============== > > Best regards, > Yoshihiro Shimoda > --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

3 11

Re: [alsa-devel] [PATCH v2] ASoC: Add support for TI LM49453 Audio codec
by M R Swami Reddy 06 Feb '12

06 Feb '12

Mark Brown Wrote: > > --- > > Changes made in V2 > > o Removed fll and vco reference frequency settings from _set_dai_pll() > > o Reworked chip enable and disable in _STANDBY and _OFF modes in > > m49453_set_bias_level, as per the review comments. >You say this is version 2 but there's been rather more than two versions > posted... Thats right. I started numbering the patch version with previous one, so mentioned it as v2. The version number will be continued. > > +static int lm49453_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, > > + int source, unsigned int freq_in, > > + unsigned int freq_out) { >This is now a bit odd. I will update it as per the other drivers formate. > > + state->in = freq_in; > > + state->out = freq_out; >It stores but otherwise ignores the configuration which was passed in (which > seems more than a little odd). Will be removed in the next patch. > > + /* Always disable the PLL - it is not safe to leave it running > > + * while reprogramming it. > > + */ > > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, > > + LM49453_PMC_SETUP_PLL_EN, 0); > > + > > + if (!freq_in || !freq_out) > > + return 0; > > + > > + > > + /* All done, turn it on */ > > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, pwr_mask, > > + pwr_mask); >Then if the PLL was already enabled it bounces the power briefly. >Really it seems like this should just be merged in with set_sysclk() - you're >not actually configuring the PLL at all, just turning it on and off, at which >point it's just another SYSCLK source. Ok. The PLL disable code will be removed and the PLL enable code will be moved to _sysclk(). >Otherwise this seems good, just this clock configuration stuff to sort out. OK. Thank you. Do I need to resend the patch. Thanks Swami

1 0

[alsa-devel] [PATCH 0/7 v2] extract a simple dmaengine library from shdma.c
by Guennadi Liakhovetski 06 Feb '12

06 Feb '12

This is v2 of the simple DMA dmaengine library. It addresses Paul's comments to v1 - thanks, adds some comments. Guennadi Liakhovetski (7): dmaengine: add a simple dma library dma: shdma: prepare for simple DMA conversion mmc: sh_mmcif: remove unneeded struct sh_mmcif_dma, prepare for simple DMA mmc: sh_mobile_sdhi: prepare for conversion to simple DMA serial: sh-sci: prepare for conversion to simple DMA ASoC: SIU: prepare for conversion to simple DMA dma: shdma: convert to the simple DMA library drivers/dma/Kconfig | 4 + drivers/dma/Makefile | 1 + drivers/dma/dma-simple.c | 873 ++++++++++++++++++++++++++++ drivers/dma/shdma.c | 1138 +++++++++---------------------------- drivers/dma/shdma.h | 45 +- drivers/mmc/host/sh_mmcif.c | 24 +- drivers/mmc/host/sh_mobile_sdhi.c | 8 +- drivers/tty/serial/sh-sci.c | 8 +- include/linux/dma-simple.h | 124 ++++ include/linux/mmc/sh_mmcif.h | 8 +- include/linux/sh_dma.h | 39 +- sound/soc/sh/siu_pcm.c | 8 +- 12 files changed, 1334 insertions(+), 946 deletions(-) create mode 100644 drivers/dma/dma-simple.c create mode 100644 include/linux/dma-simple.h -- 1.7.2.5 Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

3 19

[alsa-devel] [PATCH] Sound, emu8000: Remove duplicate linux/moduleparam.h include from emu8000_patch.c
by Jesper Juhl 06 Feb '12

06 Feb '12

The header 'linux/moduleparam.h' is included twice in 'sound/isa/sb/emu8000_patch.c'. Once is enough. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/isa/sb/emu8000_patch.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index e09f144..c99c607 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -22,7 +22,6 @@ #include "emu8000_local.h" #include <asm/uaccess.h> #include <linux/moduleparam.h> -#include <linux/moduleparam.h> static int emu8000_reset_addr; module_param(emu8000_reset_addr, int, 0444); -- 1.7.9 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

2 1

[alsa-devel] [patch] ALSA: hda/realtek - cleanup a condition
by Dan Carpenter 06 Feb '12

06 Feb '12

Sparse complains that "spec->multiout.dac_nids" is a pointer. sound/pci/hda/patch_realtek.c:2321:37: error: incompatible types for operation (>) sound/pci/hda/patch_realtek.c:2321:37: left side has type unsigned short const [usertype] *dac_nids sound/pci/hda/patch_realtek.c:2321:37: right side has type int I think the intent here was just to check if it was non-NULL, but please double check. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- I don't have this hardware. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8e82be..7da4fd7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2318,7 +2318,7 @@ static int alc_build_pcms(struct hda_codec *codec) "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - if (spec->multiout.dac_nids > 0) { + if (spec->multiout.dac_nids) { p = spec->stream_analog_playback; if (!p) p = &alc_pcm_analog_playback;

3 2

Re: [alsa-devel] [PATCH v2] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 04 Feb '12

04 Feb '12

On Fri, Feb 03, 2012 at 10:09:24PM +0530, M R Swami Reddy wrote: > --- > Changes made in V2 > o Removed fll and vco reference frequency settings from _set_dai_pll() > o Reworked chip enable and disable in _STANDBY and _OFF modes in > m49453_set_bias_level, as per the review comments. You say this is version 2 but there's been rather more than two versions posted... > +static int lm49453_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, > + int source, unsigned int freq_in, > + unsigned int freq_out) > +{ This is now a bit odd. > + state->in = freq_in; > + state->out = freq_out; It stores but otherwise ignores the configuration which was passed in (which seems more than a little odd). > + /* Always disable the PLL - it is not safe to leave it running > + * while reprogramming it. > + */ > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, > + LM49453_PMC_SETUP_PLL_EN, 0); > + > + if (!freq_in || !freq_out) > + return 0; > + > + > + /* All done, turn it on */ > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, pwr_mask, > + pwr_mask); Then if the PLL was already enabled it bounces the power briefly. Really it seems like this should just be merged in with set_sysclk() - you're not actually configuring the PLL at all, just turning it on and off, at which point it's just another SYSCLK source. Otherwise this seems good, just this clock configuration stuff to sort out.

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Don't bring up to STANDBY by default
by Mark Brown 03 Feb '12

03 Feb '12

In cases where we should enter STANDBY DAPM will power us up, otherwise there is no need to power up and we can remain in OFF. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7723f3c..b28a09c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2736,8 +2736,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { if (!wm8994->fll_suspend[i].out) continue; -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH ] ASoC: Add support for TI LM49453 Audio codec
by Reddy, MR Swami 03 Feb '12

03 Feb '12

ASoC: Add support for TI LM49453 Audio codec Signed-off-by: M R Swami Reddy <MR.Swami.Reddy(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lm49453.c | 1629 ++++++++ sound/soc/codecs/lm49453.h | 381 ++ 4 files changed, 2016 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6943e24..ed5f66c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C + select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C @@ -184,6 +185,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_LM49453 + tristate + config SND_SOC_MAX98088 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 379bc55..7bbfdb2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o +snd-soc-lm49453-objs := lm49453.o snd-soc-max98088-objs := max98088.o snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o @@ -106,6 +107,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c new file mode 100644 index 0000000..a650edb --- /dev/null +++ b/sound/soc/codecs/lm49453.c @@ -0,0 +1,1629 @@ +/* + * lm49453.c - LM49453 ALSA Soc Audio driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * 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; version 2 of the License. + * + * Initially based on sound/soc/codecs/wm8350.c + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "lm49453.h" + +static struct reg_default lm49453_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x00 }, + { 22, 0x00 }, + { 23, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 46, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 51, 0x00 }, + { 56, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x00 }, + { 70, 0x00 }, + { 71, 0x00 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 85, 0x00 }, + { 85, 0x00 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x00 }, + { 90, 0x00 }, + { 91, 0x00 }, + { 92, 0x00 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x01 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, + { 100, 0x00 }, + { 101, 0x00 }, + { 102, 0x00 }, + { 103, 0x01 }, + { 105, 0x01 }, + { 106, 0x00 }, + { 107, 0x01 }, + { 107, 0x00 }, + { 108, 0x00 }, + { 109, 0x00 }, + { 110, 0x00 }, + { 111, 0x02 }, + { 112, 0x02 }, + { 113, 0x00 }, + { 121, 0x80 }, + { 122, 0xBB }, + { 123, 0x80 }, + { 124, 0xBB }, + { 128, 0x00 }, + { 130, 0x00 }, + { 131, 0x00 }, + { 132, 0x00 }, + { 133, 0x0A }, + { 134, 0x0A }, + { 135, 0x0A }, + { 136, 0x0F }, + { 137, 0x00 }, + { 138, 0x73 }, + { 139, 0x33 }, + { 140, 0x73 }, + { 141, 0x33 }, + { 142, 0x73 }, + { 143, 0x33 }, + { 144, 0x73 }, + { 145, 0x33 }, + { 146, 0x73 }, + { 147, 0x33 }, + { 148, 0x73 }, + { 149, 0x33 }, + { 150, 0x73 }, + { 151, 0x33 }, + { 152, 0x00 }, + { 153, 0x00 }, + { 154, 0x00 }, + { 155, 0x00 }, + { 176, 0x00 }, + { 177, 0x00 }, + { 178, 0x00 }, + { 179, 0x00 }, + { 180, 0x00 }, + { 181, 0x00 }, + { 182, 0x00 }, + { 183, 0x00 }, + { 184, 0x00 }, + { 185, 0x00 }, + { 186, 0x00 }, + { 189, 0x00 }, + { 188, 0x00 }, + { 194, 0x00 }, + { 195, 0x00 }, + { 196, 0x00 }, + { 197, 0x00 }, + { 200, 0x00 }, + { 201, 0x00 }, + { 202, 0x00 }, + { 203, 0x00 }, + { 204, 0x00 }, + { 205, 0x00 }, + { 208, 0x00 }, + { 209, 0x00 }, + { 210, 0x00 }, + { 211, 0x00 }, + { 213, 0x00 }, + { 214, 0x00 }, + { 215, 0x00 }, + { 216, 0x00 }, + { 217, 0x00 }, + { 218, 0x00 }, + { 219, 0x00 }, + { 221, 0x00 }, + { 222, 0x00 }, + { 224, 0x00 }, + { 225, 0x00 }, + { 226, 0x00 }, + { 227, 0x00 }, + { 228, 0x00 }, + { 229, 0x00 }, + { 230, 0x13 }, + { 231, 0x00 }, + { 232, 0x80 }, + { 233, 0x0C }, + { 234, 0xDD }, + { 235, 0x00 }, + { 236, 0x04 }, + { 237, 0x00 }, + { 238, 0x00 }, + { 239, 0x00 }, + { 240, 0x00 }, + { 241, 0x00 }, + { 242, 0x00 }, + { 243, 0x00 }, + { 244, 0x00 }, + { 245, 0x00 }, + { 248, 0x00 }, + { 249, 0x00 }, + { 254, 0x00 }, + { 255, 0x00 }, +}; + +struct pll_state { + unsigned int in; + unsigned int out; +}; + +/* codec private data */ +struct lm49453_priv { + struct regmap *regmap; + struct pll_state pll1, pll2; + int fs_rate; + int sysclk; +}; + +/* capture path controls */ + +static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); + +static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +/* MUX Controls */ +static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; + +static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; + +static const struct soc_enum lm49453_adcl_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + ARRAY_SIZE(lm49453_adcl_mux_text), + lm49453_adcl_mux_text); + +static const struct soc_enum lm49453_adcr_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + ARRAY_SIZE(lm49453_adcr_mux_text), + lm49453_adcr_mux_text); + +static const struct snd_kcontrol_new lm49453_adcl_mux_control = + SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); + +static const struct snd_kcontrol_new lm49453_adcr_mux_control = + SOC_DAPM_ENUM("ADC Right Mux", lm49453_adcr_enum); + +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), +}; + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { +/* Sidetone supports mono only */ +SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, + 0, 0x3F, 0, digital_tlv), +}; + +static const struct snd_kcontrol_new lm49453_snd_controls[] = { + /* mic1 and mic2 supports mono only */ + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, + 0, digital_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, + 0, digital_tlv), + + SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, + LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, + LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), + SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), + SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum), + + /* Capture path filter enable */ + SOC_SINGLE("DMIC1 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 0, 1, 0), + SOC_SINGLE("DMIC2 HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 1, 1, 0), + SOC_SINGLE("ADC HPFilter Switch", LM49453_P0_ADC_FX_ENABLES_REG, + 2, 1, 0), + + SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, + LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, + LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, + LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, + LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, + 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 6, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 6, 3, 0, port_tlv), + + SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 2, 3, 0, port_tlv), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = { + + /* All end points HP,EP, LS, Lineout and Haptic */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("LSOUTL"), + SND_SOC_DAPM_OUTPUT("LSOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTR"), + + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0), + + /* playback path driver enables */ + SND_SOC_DAPM_OUT_DRV("Headset Switch", + LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earpiece Switch", + LM49453_P0_EP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Left Switch", + LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Right Switch", + LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0), + + + SND_SOC_DAPM_PGA("AUXL Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUXR Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("DMIC1 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC1 Right", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("DMIC2 Right", "Capture", SND_SOC_NOPM, 1, 0), + + SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcl_mux_control), + SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcr_mux_control), + + SND_SOC_DAPM_MUX("Mic1 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control), + + SND_SOC_DAPM_MUX("Mic2 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 2, 0), + SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 3, 0), + SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 7, 0), + + /* Port1 TX controls */ + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Port2 TX controls */ + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Port1 Playback", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Port2 Playback", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 1, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Port1 Capture", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Port2 Capture", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 2, 0, NULL, 0), + + /* Sidetone Mixer */ + SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0, + &lm49453_sidetone_mixer_controls, + ARRAY_SIZE(lm49453_sidetone_mixer_controls)), + + /* DAC MIXERS */ + SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_left_mixer, + ARRAY_SIZE(lm49453_headset_left_mixer)), + SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_right_mixer, + ARRAY_SIZE(lm49453_headset_right_mixer)), + SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_left_mixer, + ARRAY_SIZE(lm49453_lineout_left_mixer)), + SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_right_mixer, + ARRAY_SIZE(lm49453_lineout_right_mixer)), + SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_left_mixer, + ARRAY_SIZE(lm49453_speaker_left_mixer)), + SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_right_mixer, + ARRAY_SIZE(lm49453_speaker_right_mixer)), + SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_left_mixer, + ARRAY_SIZE(lm49453_haptic_left_mixer)), + SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_right_mixer, + ARRAY_SIZE(lm49453_haptic_right_mixer)), + + /* Capture Mixer */ + SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx1_mixer, + ARRAY_SIZE(lm49453_port1_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx2_mixer, + ARRAY_SIZE(lm49453_port1_tx2_mixer)), + SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx3_mixer, + ARRAY_SIZE(lm49453_port1_tx3_mixer)), + SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx4_mixer, + ARRAY_SIZE(lm49453_port1_tx4_mixer)), + SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx5_mixer, + ARRAY_SIZE(lm49453_port1_tx5_mixer)), + SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx6_mixer, + ARRAY_SIZE(lm49453_port1_tx6_mixer)), + SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx7_mixer, + ARRAY_SIZE(lm49453_port1_tx7_mixer)), + SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx8_mixer, + ARRAY_SIZE(lm49453_port1_tx8_mixer)), + + SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx1_mixer, + ARRAY_SIZE(lm49453_port2_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx2_mixer, + ARRAY_SIZE(lm49453_port2_tx2_mixer)), +}; + +static const struct snd_soc_dapm_route lm49453_audio_map[] = { + /* Port SDI mapping */ + { "PORT1_1_RX", NULL, "PORT1_SDI" }, + { "PORT1_2_RX", NULL, "PORT1_SDI" }, + { "PORT1_3_RX", NULL, "PORT1_SDI" }, + { "PORT1_4_RX", NULL, "PORT1_SDI" }, + { "PORT1_5_RX", NULL, "PORT1_SDI" }, + { "PORT1_6_RX", NULL, "PORT1_SDI" }, + { "PORT1_7_RX", NULL, "PORT1_SDI" }, + { "PORT1_8_RX", NULL, "PORT1_SDI" }, + + { "PORT2_1_RX", NULL, "PORT2_SDI" }, + { "PORT2_2_RX", NULL, "PORT2_SDI" }, + + /* HP mapping */ + { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPL Mixer", "ADCL Switch", "ADC Left" }, + { "HPL Mixer", "ADCR Switch", "ADC Right" }, + { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HPL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPL DAC", NULL, "HPL Mixer" }, + + { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPR Mixer", "ADCL Switch", "ADC Left" }, + { "HPR Mixer", "ADCR Switch", "ADC Right" }, + { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" }, + { "HPR Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPR DAC", NULL, "HPR Mixer" }, + + { "HPOUTL", "Headset Switch", "HPL DAC"}, + { "HPOUTR", "Headset Switch", "HPR DAC"}, + + /* EP map */ + { "EPOUT", "Earpiece Switch", "HPL DAC" }, + + /* Speaker map */ + { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSL Mixer", "ADCL Switch", "ADC Left" }, + { "LSL Mixer", "ADCR Switch", "ADC Right" }, + { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSL DAC", NULL, "LSL Mixer" }, + + { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSR Mixer", "ADCL Switch", "ADC Left" }, + { "LSR Mixer", "ADCR Switch", "ADC Right" }, + { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSR DAC", NULL, "LSR Mixer" }, + + { "LSOUTL", "Speaker Left Switch", "LSL DAC"}, + { "LSOUTR", "Speaker Left Switch", "LSR DAC"}, + + /* Haptic map */ + { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAL Mixer", "ADCL Switch", "ADC Left" }, + { "HAL Mixer", "ADCR Switch", "ADC Right" }, + { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HAL DAC", NULL, "HAL Mixer" }, + + { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAR Mixer", "ADCL Switch", "ADC Left" }, + { "HAR Mixer", "ADCR Switch", "ADC Right" }, + { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAR Mixer", "Sideton Switch", "Sidetone" }, + + { "HAR DAC", NULL, "HAR Mixer" }, + + { "HAOUTL", "Haptic Left Switch", "HAL DAC" }, + { "HAOUTR", "Haptic Right Switch", "HAR DAC" }, + + /* Lineout map */ + { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOL Mixer", "ADCL Switch", "ADC Left" }, + { "LOL Mixer", "ADCR Switch", "ADC Right" }, + { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOL DAC", NULL, "LOL Mixer" }, + + { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOR Mixer", "ADCL Switch", "ADC Left" }, + { "LOR Mixer", "ADCR Switch", "ADC Right" }, + { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOR DAC", NULL, "LOR Mixer" }, + + { "LOOUTL", NULL, "LOL DAC" }, + { "LOOUTR", NULL, "LOR DAC" }, + + /* TX map */ + /* Port1 mappings */ + { "Port1_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_3 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_3 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_4 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_4 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_5 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_5 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_6 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_6 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_7 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_7 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_8 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_8 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "P1_1_TX", NULL, "Port1_1 Mixer" }, + { "P1_2_TX", NULL, "Port1_2 Mixer" }, + { "P1_3_TX", NULL, "Port1_3 Mixer" }, + { "P1_4_TX", NULL, "Port1_4 Mixer" }, + { "P1_5_TX", NULL, "Port1_5 Mixer" }, + { "P1_6_TX", NULL, "Port1_6 Mixer" }, + { "P1_7_TX", NULL, "Port1_7 Mixer" }, + { "P1_8_TX", NULL, "Port1_8 Mixer" }, + + { "P2_1_TX", NULL, "Port2_1 Mixer" }, + { "P2_2_TX", NULL, "Port2_2 Mixer" }, + + { "PORT1_SDO", NULL, "P1_1_TX"}, + { "PORT1_SDO", NULL, "P1_2_TX"}, + { "PORT1_SDO", NULL, "P1_3_TX"}, + { "PORT1_SDO", NULL, "P1_4_TX"}, + { "PORT1_SDO", NULL, "P1_5_TX"}, + { "PORT1_SDO", NULL, "P1_6_TX"}, + { "PORT1_SDO", NULL, "P1_7_TX"}, + { "PORT1_SDO", NULL, "P1_8_TX"}, + + { "PORT2_SDO", NULL, "P2_1_TX"}, + { "PORT2_SDO", NULL, "P2_2_TX"}, + + { "Mic1 Input", NULL, "AMIC1" }, + { "Mic2 Input", NULL, "AMIC2" }, + + { "AUXL Input", NULL, "AUXL" }, + { "AUXR Input", NULL, "AUXR" }, + + /* AUX connections */ + { "ADCL Mux", "Aux_L", "AUXL Input" }, + { "ADCL Mux", "MIC1", "Mic1 Input" }, + + { "ADCR Mux", "Aux_R", "AUXR Input" }, + { "ADCR Mux", "MIC2", "Mic2 Input" }, + + /* ADC connection */ + { "ADC Left", NULL, "ADCL Mux"}, + { "ADC Right", NULL, "ADCR Mux"}, + + { "DMIC1 Left", NULL, "DMIC1DAT"}, + { "DMIC1 Right", NULL, "DMIC1DAT"}, + { "DMIC2 Left", NULL, "DMIC2DAT"}, + { "DMIC2 Right", NULL, "DMIC2DAT"}, + + /* TX Paths */ + { "Port1 Capture", NULL, "P1_1_TX" }, + { "Port1 Capture", NULL, "P1_2_TX" }, + { "Port1 Capture", NULL, "P1_3_TX" }, + { "Port1 Capture", NULL, "P1_4_TX" }, + { "Port1 Capture", NULL, "P1_5_TX" }, + { "Port1 Capture", NULL, "P1_6_TX" }, + { "Port1 Capture", NULL, "P1_7_TX" }, + { "Port1 Capture", NULL, "P1_8_TX" }, + + { "Port2 Capture", NULL, "P2_1_TX" }, + { "Port2 Capture", NULL, "P2_2_TX" }, + + /* RX Paths */ + { "Port1 Playback", NULL, "P1_1_RX" }, + { "Port1 Playback", NULL, "P1_2_RX" }, + { "Port1 Playback", NULL, "P1_3_RX" }, + { "Port1 Playback", NULL, "P1_4_RX" }, + { "Port1 Playback", NULL, "P1_5_RX" }, + { "Port1 Playback", NULL, "P1_6_RX" }, + { "Port1 Playback", NULL, "P1_7_RX" }, + { "Port1 Playback", NULL, "P1_8_RX" }, + + { "Port2 Playback", NULL, "P2_1_RX" }, + { "Port2 Playback", NULL, "P2_2_RX" }, + + /* Sidetone map */ + { "Sidetone Mixer", NULL, "ADC Left" }, + { "Sidetone Mixer", NULL, "ADC Right" }, + { "Sidetone Mixer", NULL, "DMIC1 Left" }, + { "Sidetone Mixer", NULL, "DMIC1 Right" }, + { "Sidetone Mixer", NULL, "DMIC2 Left" }, + { "Sidetone Mixer", NULL, "DMIC2 Right" }, + + { "Sidetone", "Sidetone Switch", "Sidetone Mixer" }, +}; + +static int lm49453_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + struct pll_state *state; + unsigned int pwr_mask; + + switch (pll_id) { + case LM49453_PLL_1: + state = &lm49453->pll1; + pwr_mask = LM49453_PMC_SETUP_PLL_EN; + break; + case LM49453_PLL_2: + state = &lm49453->pll2; + pwr_mask = LM49453_PMC_SETUP_PLL_P2_EN; + break; + default: + return -EINVAL; + } + + state->in = freq_in; + state->out = freq_out; + + /* Always disable the PLL - it is not safe to leave it running + * while reprogramming it. + */ + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_PLL_EN, 0); + + if (!freq_in || !freq_out) + return 0; + + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQL_REG, (state->in & 0xff)); + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQH_REG, + ((state->in >> 8) & 0xff)); + + snd_soc_write(codec, LM49453_P0_VCO_TARGETLL_REG, + state->out & 0xff); + snd_soc_write(codec, LM49453_P0_VCO_TARGETLH_REG, + (state->out >> 8) & 0xff); + snd_soc_write(codec, LM49453_P0_VCO_TARGETHL_REG, + (state->out >> 16) & 0xff); + snd_soc_write(codec, LM49453_P0_VCO_TARGETHH_REG, + (state->out >> 24) & 0xff); + + /* All done, turn it on */ + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, pwr_mask, + pwr_mask); + + return 0; +} + +static int lm49453_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 clk_div = 0; + u16 pll_clk = 0; + + lm49453->fs_rate = params_rate(params); + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (lm49453->fs_rate) { + case 8000: + case 16000: + case 32000: + case 24000: + case 48000: + clk_div = 256; + break; + case 11025: + case 22050: + case 44100: + clk_div = 216; + break; + case 96000: + clk_div = 127; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div); + + switch (lm49453->sysclk) { + case 12288000: + case 26000000: + case 19000000: + /* PLL CLK slection */ + pll_clk = ~BIT(4); + break; + case 48000: + case 32576: + /* FLL CLK slection */ + pll_clk = BIT(4); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk); + + return 0; +} + +static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + int aif_val; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | + ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, + LM49453_AUDIO_PORT1_BASIC_FMT_MASK, aif_val); + + return 0; +} + +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 12288000: + case 26000000: + case 19200000: + case 48000: + case 32576: + lm49453->sysclk = freq; + return 0; + default: + return -EINVAL; + } +} + +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), + (mute ? (BIT(1)|BIT(0)) : 0)); + return 0; +} + +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(3)|BIT(2), + (mute ? (BIT(3)|BIT(2)) : 0)); + return 0; +} + +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(5)|BIT(4), + (mute ? (BIT(5)|BIT(4)) : 0)); + return 0; +} + +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(4), + (mute ? BIT(4) : 0)); + return 0; +} + +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(7)|BIT(6), + (mute ? (BIT(7)|BIT(6)) : 0)); + return 0; +} + +static int lm49453_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(lm49453->regmap); + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, 0); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +/* Formates supported by LM49453 driver. */ +#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops lm49453_headset_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_pll = lm49453_set_dai_pll, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_hp_mute, +}; + +static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { + .digital_mute = lm49453_ls_mute, + .hw_params = lm49453_hw_params, +}; + +static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { + .digital_mute = lm49453_ha_mute, + .hw_params = lm49453_hw_params, +}; + +static struct snd_soc_dai_ops lm49453_ep_dai_ops = { + .digital_mute = lm49453_ep_mute, + .hw_params = lm49453_hw_params, +}; + +static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { + .digital_mute = lm49453_lo_mute, + .hw_params = lm49453_hw_params, +}; + +/* LM49453 dai structure. */ +struct snd_soc_dai_driver lm49453_dai[] = { + { + .name = "LM49453 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_headset_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "LM49453 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_speaker_dai_ops, + }, + { + .name = "LM49453 Haptic", + .playback = { + .stream_name = "Haptic", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_haptic_dai_ops, + }, + { + .name = "LM49453 Earpiece", + .playback = { + .stream_name = "Earpiece", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_ep_dai_ops, + }, + { + .name = "LM49453 line out", + .playback = { + .stream_name = "Lineout", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_lineout_dai_ops, + }, +}; + +static int lm49453_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int lm49453_resume(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + regcache_sync(lm49453->regmap); + + lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + lm49453_set_bias_level(codec, codec->dapm.suspend_bias_level); + + return 0; +} + +static int lm49453_probe(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + codec->control_data = lm49453->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int lm49453_remove(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { + .probe = lm49453_probe, + .remove = lm49453_remove, + .suspend = lm49453_suspend, + .resume = lm49453_resume, + .set_bias_level = lm49453_set_bias_level, + .controls = lm49453_snd_controls, + .num_controls = ARRAY_SIZE(lm49453_snd_controls), + .dapm_widgets = lm49453_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets), + .dapm_routes = lm49453_audio_map, + .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), + .idle_bias_off = true, +}; + +static const struct regmap_config lm49453_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = LM49453_MAX_REGISTER, + .reg_defaults = lm49453_reg_defs, + .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int lm49453_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm49453_priv *lm49453; + int ret = 0; + + lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv), + GFP_KERNEL); + + if (lm49453 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm49453); + + lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config); + if (IS_ERR(lm49453->regmap)) { + ret = PTR_ERR(lm49453->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_lm49453, + lm49453_dai, ARRAY_SIZE(lm49453_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(lm49453->regmap); + return ret; + } + + return ret; +} + +static int __devexit lm49453_i2c_remove(struct i2c_client *client) +{ + struct lm49453_priv *lm49453 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(lm49453->regmap); + return 0; +} + +static const struct i2c_device_id lm49453_i2c_id[] = { + { "lm49453", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); + +static struct i2c_driver lm49453_i2c_driver = { + .driver = { + .name = "lm49453", + .owner = THIS_MODULE, + }, + .probe = lm49453_i2c_probe, + .remove = __devexit_p(lm49453_i2c_remove), + .id_table = lm49453_i2c_id, +}; + +module_i2c_driver(lm49453_i2c_driver); + +MODULE_DESCRIPTION("ASoC LM49453 driver"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy(a)ti.com"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h new file mode 100644 index 0000000..c62fd38 --- /dev/null +++ b/sound/soc/codecs/lm49453.h @@ -0,0 +1,381 @@ +/* + * lm49453.h - LM49453 ALSA Soc Audio drive + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * 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; version 2 of the License. + * + */ + +#ifndef _LM49453_H +#define _LM49453_H + +#include <linux/bitops.h> + +/* LM49453_P0 register space for page0 */ +#define LM49453_P0_PMC_SETUP_REG 0x00 +#define LM49453_P0_PLL_CLK_SEL1_REG 0x01 +#define LM49453_P0_PLL_CLK_SEL2_REG 0x02 +#define LM49453_P0_PMC_CLK_DIV_REG 0x03 +#define LM49453_P0_HSDET_CLK_DIV_REG 0x04 +#define LM49453_P0_DMIC_CLK_DIV_REG 0x05 +#define LM49453_P0_ADC_CLK_DIV_REG 0x06 +#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07 +#define LM49453_P0_PLL_HF_M_REG 0x08 +#define LM49453_P0_PLL_LF_M_REG 0x09 +#define LM49453_P0_PLL_NL_REG 0x0A +#define LM49453_P0_PLL_N_MODL_REG 0x0B +#define LM49453_P0_PLL_N_MODH_REG 0x0C +#define LM49453_P0_PLL_P1_REG 0x0D +#define LM49453_P0_PLL_P2_REG 0x0E +#define LM49453_P0_FLL_REF_FREQL_REG 0x0F +#define LM49453_P0_FLL_REF_FREQH_REG 0x10 +#define LM49453_P0_VCO_TARGETLL_REG 0x11 +#define LM49453_P0_VCO_TARGETLH_REG 0x12 +#define LM49453_P0_VCO_TARGETHL_REG 0x13 +#define LM49453_P0_VCO_TARGETHH_REG 0x14 +#define LM49453_P0_PLL_CONFIG_REG 0x15 +#define LM49453_P0_DAC_CLK_SEL_REG 0x16 +#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17 + +/* Analog Mixer Input Stages */ +#define LM49453_P0_MICL_REG 0x20 +#define LM49453_P0_MICR_REG 0x21 +#define LM49453_P0_EP_REG 0x24 +#define LM49453_P0_DIS_PKVL_FB_REG 0x25 + +/* Analog Mixer Output Stages */ +#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E + +/*ADC or DAC */ +#define LM49453_P0_ADC_DSP_REG 0x30 +#define LM49453_P0_DAC_DSP_REG 0x31 + +/* EFFECTS ENABLES */ +#define LM49453_P0_ADC_FX_ENABLES_REG 0x33 + +/* GPIO */ +#define LM49453_P0_GPIO1_REG 0x38 +#define LM49453_P0_GPIO2_REG 0x39 +#define LM49453_P0_GPIO3_REG 0x3A +#define LM49453_P0_HAP_CTL_REG 0x3B +#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C +#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D +#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E +#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F + +/* DIGITAL MIXER */ +#define LM49453_P0_DMIX_CLK_SEL_REG 0x40 +#define LM49453_P0_PORT1_RX_LVL1_REG 0x41 +#define LM49453_P0_PORT1_RX_LVL2_REG 0x42 +#define LM49453_P0_PORT2_RX_LVL_REG 0x43 +#define LM49453_P0_PORT1_TX1_REG 0x44 +#define LM49453_P0_PORT1_TX2_REG 0x45 +#define LM49453_P0_PORT1_TX3_REG 0x46 +#define LM49453_P0_PORT1_TX4_REG 0x47 +#define LM49453_P0_PORT1_TX5_REG 0x48 +#define LM49453_P0_PORT1_TX6_REG 0x49 +#define LM49453_P0_PORT1_TX7_REG 0x4A +#define LM49453_P0_PORT1_TX8_REG 0x4B +#define LM49453_P0_PORT2_TX1_REG 0x4C +#define LM49453_P0_PORT2_TX2_REG 0x4D +#define LM49453_P0_STN_SEL_REG 0x4F +#define LM49453_P0_DACHPL1_REG 0x50 +#define LM49453_P0_DACHPL2_REG 0x51 +#define LM49453_P0_DACHPR1_REG 0x52 +#define LM49453_P0_DACHPR2_REG 0x53 +#define LM49453_P0_DACLOL1_REG 0x54 +#define LM49453_P0_DACLOL2_REG 0x55 +#define LM49453_P0_DACLOR1_REG 0x56 +#define LM49453_P0_DACLOR2_REG 0x57 +#define LM49453_P0_DACLSL1_REG 0x58 +#define LM49453_P0_DACLSL2_REG 0x59 +#define LM49453_P0_DACLSR1_REG 0x5A +#define LM49453_P0_DACLSR2_REG 0x5B +#define LM49453_P0_DACHAL1_REG 0x5C +#define LM49453_P0_DACHAL2_REG 0x5D +#define LM49453_P0_DACHAR1_REG 0x5E +#define LM49453_P0_DACHAR2_REG 0x5F + +/* AUDIO PORT 1 (TDM) */ +#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63 +#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64 +#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65 +#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66 +#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67 +#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68 +#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69 + +/* AUDIO PORT 2 */ +#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A +#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B +#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C +#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D +#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E +#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F +#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70 + +/* SAMPLE RATE */ +#define LM49453_P0_PORT1_SR_LSB_REG 0x79 +#define LM49453_P0_PORT1_SR_MSB_REG 0x7A +#define LM49453_P0_PORT2_SR_LSB_REG 0x7B +#define LM49453_P0_PORT2_SR_MSB_REG 0x7C + +/* EFFECTS - HPFs */ +#define LM49453_P0_HPF_REG 0x80 + +/* EFFECTS ADC ALC */ +#define LM49453_P0_ADC_ALC1_REG 0x82 +#define LM49453_P0_ADC_ALC2_REG 0x83 +#define LM49453_P0_ADC_ALC3_REG 0x84 +#define LM49453_P0_ADC_ALC4_REG 0x85 +#define LM49453_P0_ADC_ALC5_REG 0x86 +#define LM49453_P0_ADC_ALC6_REG 0x87 +#define LM49453_P0_ADC_ALC7_REG 0x88 +#define LM49453_P0_ADC_ALC8_REG 0x89 +#define LM49453_P0_DMIC1_LEVELL_REG 0x8A +#define LM49453_P0_DMIC1_LEVELR_REG 0x8B +#define LM49453_P0_DMIC2_LEVELL_REG 0x8C +#define LM49453_P0_DMIC2_LEVELR_REG 0x8D +#define LM49453_P0_ADC_LEVELL_REG 0x8E +#define LM49453_P0_ADC_LEVELR_REG 0x8F +#define LM49453_P0_DAC_HP_LEVELL_REG 0x90 +#define LM49453_P0_DAC_HP_LEVELR_REG 0x91 +#define LM49453_P0_DAC_LO_LEVELL_REG 0x92 +#define LM49453_P0_DAC_LO_LEVELR_REG 0x93 +#define LM49453_P0_DAC_LS_LEVELL_REG 0x94 +#define LM49453_P0_DAC_LS_LEVELR_REG 0x95 +#define LM49453_P0_DAC_HA_LEVELL_REG 0x96 +#define LM49453_P0_DAC_HA_LEVELR_REG 0x97 +#define LM49453_P0_SOFT_MUTE_REG 0x98 +#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99 +#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A +#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B + +/*DIGITAL MIC1 */ +#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2 + +/*DIGITAL MIC2 */ +#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5 + +/* ADC DECIMATOR */ +#define LM49453_P0_ADC_DECIMATOR_REG 0xB6 + +/* DAC CONFIGURE */ +#define LM49453_P0_DAC_CONFIG_REG 0xB7 + +/* SIDETONE */ +#define LM49453_P0_STN_VOL_ADCL_REG 0xB8 +#define LM49453_P0_STN_VOL_ADCR_REG 0xB9 +#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA +#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB +#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC +#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD + +/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */ +#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2 +#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3 +#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4 +#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5 + +/* ADC ALC EFFECT MONITORS (Read Only) */ +#define LM49453_P0_ADC_LVLMONL_REG 0xC8 +#define LM49453_P0_ADC_LVLMONR_REG 0xC9 +#define LM49453_P0_ADC_ALCMONL_REG 0xCA +#define LM49453_P0_ADC_ALCMONR_REG 0xCB +#define LM49453_P0_ADC_MUTED_REG 0xCC +#define LM49453_P0_DAC_MUTED_REG 0xCD + +/* HEADSET DETECT */ +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0 +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3 +#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4 +#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5 +#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6 +#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7 +#define LM49453_P0_HSD_IRQ1_REG 0xD8 +#define LM49453_P0_HSD_IRQ2_REG 0xD9 +#define LM49453_P0_HSD_IRQ3_REG 0xDA +#define LM49453_P0_HSD_IRQ4_REG 0xDB +#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC +#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD +#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE +#define LM49453_P0_HSD_R_HPLL_REG 0xE0 +#define LM49453_P0_HSD_R_HPLH_REG 0xE1 +#define LM49453_P0_HSD_R_HPLU_REG 0xE2 +#define LM49453_P0_HSD_R_HPRL_REG 0xE3 +#define LM49453_P0_HSD_R_HPRH_REG 0xE4 +#define LM49453_P0_HSD_R_HPRU_REG 0xE5 +#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6 +#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7 +#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8 +#define LM49453_P0_HSD_RO_FINALL_REG 0xE9 +#define LM49453_P0_HSD_RO_FINALH_REG 0xEA +#define LM49453_P0_HSD_RO_FINALU_REG 0xEB +#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC +#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED +#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE +#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5 + +/* I/O PULLDOWN CONFIG */ +#define LM49453_P0_PULL_CONFIG1_REG 0xF8 +#define LM49453_P0_PULL_CONFIG2_REG 0xF9 +#define LM49453_P0_PULL_CONFIG3_REG 0xFA + +/* RESET */ +#define LM49453_P0_RESET_REG 0xFE + +/* PAGE */ +#define LM49453_PAGE_REG 0xFF + +#define LM49453_MAX_REGISTER (0xFF+1) + +/* LM49453_P0_PMC_SETUP_REG (0x00h) */ +#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0)) +#define LM49453_PMC_SETUP_PLL_EN BIT(2) +#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3) +#define LM49453_PMC_SETUP_PLL_FLL BIT(4) +#define LM49453_PMC_SETUP_MCLK_OVER BIT(5) +#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6) +#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7) + +/* Chip Enable bits */ +#define LM49453_CHIP_EN_SHUTDOWN 0x00 +#define LM49453_CHIP_EN_NO_HSD_DETECT 0x01 +#define LM49453_CHIP_EN_HSD_DETECT 0x02 +#define LM49453_CHIP_EN_INVALID_HSD 0x03 + +/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */ +#define LM49453_CLK_SEL1_MCLK_SEL 0x11 +#define LM49453_CLK_SEL1_RTC_SEL 0x11 +#define LM49453_CLK_SEL1_PORT1_SEL 0x10 +#define LM49453_CLK_SEL1_PORT2_SEL 0x11 + +/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */ +#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38 + +/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */ +#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001 + +/* LM49453_P0_VCO_TARGETLL_REG (0x11) */ +#define LM49453_VCO_TARGET_VAL 0x8ca0001 + +/* LM49453_P0_ADC_DSP_REG (0x30h) */ +#define LM49453_ADC_DSP_ADC_MUTEL BIT(0) +#define LM49453_ADC_DSP_ADC_MUTER BIT(1) +#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2) +#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3) +#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4) +#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5) +#define LM49453_ADC_DSP_MUTE_ALL 0x3F + +/* LM49453_P0_DAC_DSP_REG (0x31h) */ +#define LM49453_DAC_DSP_MUTE_ALL 0xFF + +/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */ +#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3)) +#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3) +#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4) + +/* LM49453_P0_RESET_REG (0xFEh) */ +#define LM49453_RESET_REG_RST BIT(0) + +/* Page select register bits (0xFF) */ +#define LM49453_PAGE0_SELECT 0x0 +#define LM49453_PAGE1_SELECT 0x1 + +/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */ +#define LM49453_JACK_DISABLE 0x00 +#define LM49453_JACK_CONFIG1 0x01 +#define LM49453_JACK_CONFIG2 0x02 +#define LM49453_JACK_CONFIG3 0x03 +#define LM49453_JACK_CONFIG4 0x04 +#define LM49453_JACK_CONFIG5 0x05 + +/* Page 1 REGISTERS */ + +/* SIDETONE */ +#define LM49453_P1_SIDETONE_SA0L_REG 0x80 +#define LM49453_P1_SIDETONE_SA0H_REG 0x81 +#define LM49453_P1_SIDETONE_SAB0U_REG 0x82 +#define LM49453_P1_SIDETONE_SB0L_REG 0x83 +#define LM49453_P1_SIDETONE_SB0H_REG 0x84 +#define LM49453_P1_SIDETONE_SH0L_REG 0x85 +#define LM49453_P1_SIDETONE_SH0H_REG 0x86 +#define LM49453_P1_SIDETONE_SH0U_REG 0x87 +#define LM49453_P1_SIDETONE_SA1L_REG 0x88 +#define LM49453_P1_SIDETONE_SA1H_REG 0x89 +#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A +#define LM49453_P1_SIDETONE_SB1L_REG 0x8B +#define LM49453_P1_SIDETONE_SB1H_REG 0x8C +#define LM49453_P1_SIDETONE_SH1L_REG 0x8D +#define LM49453_P1_SIDETONE_SH1H_REG 0x8E +#define LM49453_P1_SIDETONE_SH1U_REG 0x8F +#define LM49453_P1_SIDETONE_SA2L_REG 0x90 +#define LM49453_P1_SIDETONE_SA2H_REG 0x91 +#define LM49453_P1_SIDETONE_SAB2U_REG 0x92 +#define LM49453_P1_SIDETONE_SB2L_REG 0x93 +#define LM49453_P1_SIDETONE_SB2H_REG 0x94 +#define LM49453_P1_SIDETONE_SH2L_REG 0x95 +#define LM49453_P1_SIDETONE_SH2H_REG 0x96 +#define LM49453_P1_SIDETONE_SH2U_REG 0x97 +#define LM49453_P1_SIDETONE_SA3L_REG 0x98 +#define LM49453_P1_SIDETONE_SA3H_REG 0x99 +#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A +#define LM49453_P1_SIDETONE_SB3L_REG 0x9B +#define LM49453_P1_SIDETONE_SB3H_REG 0x9C +#define LM49453_P1_SIDETONE_SH3L_REG 0x9D +#define LM49453_P1_SIDETONE_SH3H_REG 0x9E +#define LM49453_P1_SIDETONE_SH3U_REG 0x9F +#define LM49453_P1_SIDETONE_SA4L_REG 0xA0 +#define LM49453_P1_SIDETONE_SA4H_REG 0xA1 +#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2 +#define LM49453_P1_SIDETONE_SB4L_REG 0xA3 +#define LM49453_P1_SIDETONE_SB4H_REG 0xA4 +#define LM49453_P1_SIDETONE_SH4L_REG 0xA5 +#define LM49453_P1_SIDETONE_SH4H_REG 0xA6 +#define LM49453_P1_SIDETONE_SH4U_REG 0xA7 +#define LM49453_P1_SIDETONE_SA5L_REG 0xA8 +#define LM49453_P1_SIDETONE_SA5H_REG 0xA9 +#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA +#define LM49453_P1_SIDETONE_SB5L_REG 0xAB +#define LM49453_P1_SIDETONE_SB5H_REG 0xAC +#define LM49453_P1_SIDETONE_SH5L_REG 0xAD +#define LM49453_P1_SIDETONE_SH5H_REG 0xAE +#define LM49453_P1_SIDETONE_SH5U_REG 0xAF + +/* CHARGE PUMP CONFIG */ +#define LM49453_P1_CP_CONFIG1_REG 0xB0 +#define LM49453_P1_CP_CONFIG2_REG 0xB1 +#define LM49453_P1_CP_CONFIG3_REG 0xB2 +#define LM49453_P1_CP_CONFIG4_REG 0xB3 +#define LM49453_P1_CP_LA_VTH1L_REG 0xB4 +#define LM49453_P1_CP_LA_VTH1M_REG 0xB5 +#define LM49453_P1_CP_LA_VTH2L_REG 0xB6 +#define LM49453_P1_CP_LA_VTH2M_REG 0xB7 +#define LM49453_P1_CP_LA_VTH3L_REG 0xB8 +#define LM49453_P1_CP_LA_VTH3H_REG 0xB9 +#define LM49453_P1_CP_CLK_DIV_REG 0xBA + +/* DAC */ +#define LM49453_P1_DAC_CHOP_REG 0xC0 + +#define LM49453_PLL_1 1 +#define LM49453_PLL_2 2 +#endif

3 7

Re: [alsa-devel] soc-dsp programming model for loopbacks
by Liam Girdwood 02 Feb '12

02 Feb '12

On Wed, 2012-02-01 at 11:50 +0000, Girdwood, Liam wrote: > On 1 February 2012 09:07, Vinod Koul <vinod.koul(a)linux.intel.com> wrote: > > On Wed, 2012-01-25 at 17:07 +0000, Liam Girdwood wrote: > >> We could eventually remove steps 2 and 4 for the FE DAI link, and look > >> at hard coding the hw_params() in the mach driver for the loopback > >> link. > >> That way the only user space driven actions would be to configure the > >> mixers in the CODEC and DSP for the correct route. DAPM would then > >> detect the path and Dynamic PCM would use the hard coded configuration > >> or bespoke mach driver logic to configure the loopback DAI link based > >> on > >> use case. This would have to be done after the basic Dynamic PCM > >> infrastructure was upstream though (unless you have a patch atm). > > Hi, > > Sorry to join the party late :-) > > > No worries ;-) > > > Liam, if we configure the hw_params in the machine driver statically, > > represent the DSP using a map along with a codec kind of modeling. > > In OMAP we reconfigure (or fixup) some of the hw_params() in the machine driver for a BE DAI link that cant operate at the requested FE params. The ABE takes care of any conversion though. > > Would the BEs be triggered on from soc_dsp_runtime_update() when the > > loopback is established thru the DSP. > > This way we avoid all the "virtual" FEs. Use loopback to turn on codec > > and DSP (thru BEs) > > > > Would this make sense, or I need more coffee :) In the code base atm we have to use a FE pcm to trigger the BE loopback DAIs. however, the intention is to use Mark's CODEC <-> CODEC DAI link work to support the loopback with hard coded params in the machine driver. This would save the need for any FE pcm operations. Regards Liam > > > > -- > > ~Vinod > > >

3 5

[alsa-devel] [RFC 0/7] MFD: twl6040: Conversion to i2c driver
by Peter Ujfalusi 02 Feb '12

02 Feb '12

Hello, This series will convert the twl6040 MFD driver to an i2c driver. Compared to older twl4030/5030/TPS the twl6040 is a standalone audio IC. It is better if the twl6040-core (and all of it's child devices) does not depend on the twl-core since it has nothing to with it. With this conversion the dependency on twl can be dropped from the twl6040 driver stack (core, vibra, audio). Between the first and second patch the audio will not probe on OMAP4, but I felt it is better this way at least for the first RFC series to not have too big change within one patch. The two patch can be squashed together later if no objections. Regards, Peter --- Peter Ujfalusi (7): MFD: twl-core: Detach twl6040 from the pmic mfd driver MFD: twl6040: Convert to i2c driver, and separate it from twl core ASoC: twl6040: Remove dependency on twl4030 from Kconfig OMAP: 4430sdp: Correct fixed regulator device ID OMAP: sdp4430: Add fixed regulator for twl6040 needs OMAP: omap4panda: Add fixed regulator for twl6040 needs MFD: TWL6040: Add regulator support for VIO, V2V1 supplies arch/arm/mach-omap2/board-4430sdp.c | 77 +++++++++++++-- arch/arm/mach-omap2/board-generic.c | 2 +- arch/arm/mach-omap2/board-omap4panda.c | 75 +++++++++++++-- arch/arm/mach-omap2/twl-common.c | 37 ++++++- arch/arm/mach-omap2/twl-common.h | 10 +- drivers/input/misc/Kconfig | 1 - drivers/input/misc/twl6040-vibra.c | 4 +- drivers/mfd/Kconfig | 2 +- drivers/mfd/twl-core.c | 58 +++++++----- drivers/mfd/twl6040-core.c | 165 ++++++++++++++++++++++++++------ include/linux/i2c/twl.h | 12 --- include/linux/mfd/twl6040.h | 27 +++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/twl6040.c | 3 +- 14 files changed, 375 insertions(+), 100 deletions(-) -- 1.7.8.4

2 20

[alsa-devel] [PATCH] regulator: wm8350: Don't specify consumer supplies with struct device
by Mark Brown 02 Feb '12

02 Feb '12

Very, very deprecated. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/wm8350-regulator.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/regulator/wm8350-regulator.c b/drivers/regulator/wm8350-regulator.c index 6894009..4f46067 100644 --- a/drivers/regulator/wm8350-regulator.c +++ b/drivers/regulator/wm8350-regulator.c @@ -1544,7 +1544,7 @@ int wm8350_register_led(struct wm8350 *wm8350, int lednum, int dcdc, int isink, return -ENOMEM; } - led->isink_consumer.dev = &pdev->dev; + led->isink_consumer.dev_name = dev_name(&pdev->dev); led->isink_consumer.supply = "led_isink"; led->isink_init.num_consumer_supplies = 1; led->isink_init.consumer_supplies = &led->isink_consumer; @@ -1559,7 +1559,7 @@ int wm8350_register_led(struct wm8350 *wm8350, int lednum, int dcdc, int isink, return ret; } - led->dcdc_consumer.dev = &pdev->dev; + led->dcdc_consumer.dev_name = dev_name(&pdev->dev); led->dcdc_consumer.supply = "led_vcc"; led->dcdc_init.num_consumer_supplies = 1; led->dcdc_init.consumer_supplies = &led->dcdc_consumer; -- 1.7.9.rc1

2 1

[alsa-devel] [PATCH] ASoC: pcm: Improve error logging
by Mark Brown 02 Feb '12

02 Feb '12

Use the standard logging macros and use dev_ variants where we can, also reporting error codes whenever we report an error. These changes (the error codes in particular) make it noticeably easier to figure out what went wrong just from the basic dmesg output. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 39 ++++++++++++++++++++++----------------- 1 files changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fabd33e..ef28dd5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -123,8 +123,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); + dev_err(cpu_dai->dev, "can't open interface %s: %d\n", + cpu_dai->name, ret); goto out; } } @@ -132,7 +132,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->open) { ret = platform->driver->ops->open(substream); if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + dev_err(platform->dev, "can't open platform %s: %d\n", + platform->name, ret); goto platform_err; } } @@ -140,8 +141,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); + dev_err(codec_dai->dev, "can't open codec %s: %d\n", + codec_dai->name, ret); goto codec_dai_err; } } @@ -149,7 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + pr_err("asoc: %s startup failed: %d\n", + rtd->dai_link->name, ret); goto machine_err; } } @@ -413,7 +415,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); + pr_err("asoc: machine prepare error: %d\n", ret); goto out; } } @@ -421,7 +423,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->prepare) { ret = platform->driver->ops->prepare(substream); if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); + dev_err(platform->dev, "platform prepare error: %d\n", + ret); goto out; } } @@ -429,7 +432,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); + dev_err(codec_dai->dev, "DAI prepare error: %d\n", + ret); goto out; } } @@ -437,7 +441,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + dev_err(cpu_dai->dev, "DAI prepare error: %d\n", + ret); goto out; } } @@ -484,7 +489,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); + pr_err("asoc: machine hw_params failed: %d\n", ret); goto out; } } @@ -492,8 +497,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (codec_dai->driver->ops->hw_params) { ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); + dev_err(codec_dai->dev, "can't set %s hw params: %d\n", + codec_dai->name, ret); goto codec_err; } } @@ -501,8 +506,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (cpu_dai->driver->ops->hw_params) { ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); + dev_err(cpu_dai->dev, "%s hw params failed: %d\n", + cpu_dai->name, ret); goto interface_err; } } @@ -510,8 +515,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (platform->driver->ops && platform->driver->ops->hw_params) { ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); + dev_err(platform->dev, "%s hw params failed: %d\n", + platform->name, ret); goto platform_err; } } -- 1.7.9.rc1

2 1

[alsa-devel] Some questions related to ALSA based duplex audio processing
by public-hk 02 Feb '12

02 Feb '12

Hi, I have just started to develop a duplex audio processing application based on ALSA. My development goals are - of course - maximum stability as well as lowest possible delay (latency). So, here is what I did on my Ubuntu laptop: 1) Setup of device/soundcard "hw:0,0" for processing at a samplerate of 44.1 kHz, buffersize 128 samples. 2) I created one input PCM device and one output PCM device handle 3) I use the function "snd_async_add_pcm_handler" to install a callback function (ASYNC mode), one for input, one for output. 4) I use "snd_pcm_link" to synchronize both pcm handles. 5) I use the mmap'ed area access My motivation to work the way I do is to maximize the control of processing behavior and to minimize the latency. E.g., I have understood from the docs that the snd_pcm_read/snd_pcm_write functions do nothing else than addressing the memory mapped areas. As a conclusion, I realize this access myself to have more control about it. And by using the async callbacks, I do not have to deal with blocking or non-blocking read functions or polling related issues. So far, I have realized the playback part and noticed some aspects which are not clear to me from the ALSA documentation: a) What is the most efficient way to realize duplex audio processing, the async way with mmap'ed read/write that I follow? b) At first, I created the pcm handles as follows: "snd_pcm_open (.., SND_PCM_ASYNC);" When starting to process audio (call of "snd_pcm_start"), I realized that the registered callback functions do not get called. I had to change this to "snd_pcm_open (.., 0);". Is that really intended, it seems contradictory? c) The function "snd_pcm_info_get_sync" is supposed to return a description of the synchronization behavior of a soundcard. In my case, I called this function for two soundcards (one USB and the laptop integrated soundcard). In both cases, the returned content is all zeros. Should not this be different for both devices? d) By default, it seems that the callbacks for audio frames come within a thread with normal priority. On Windows OS, I am used to increase the thread priority whenever starting my threads, but it is commonly not a good idea to increase the process priority. In the ALSA latency-example, the PROCESS priority is increased (which can be done only with superuser priv.). What is the recommended way in Linux to achieve lower latencies? In the future I will have to deal with synchronization of input and output since the samples arrive/leave my application in two independent callback functions. So what will I do: Once input samples are available I will store these in one buffer, and these samples will be output the next time when there is space in the output ring buffer. In order to minimize the latency, I would need to know more about the intended exact behavior of the ALSA lib which I did not find in the documentation: I) If linking input and output with "snd_pcm_link", I have understood that the changes of state for input and output PCM handle will occur synchronously. That is, when doing the operation such as "snd_pcm_prepare" on one of the handles, both handles will be affected. However, what does this means for audio processing on the frame level: If I use two callbacks for signal processing for input and output respectively installed based on the function "snd_async_add_pcm_handler", will these callbacks occur simultaneously? On Windows OS (speaking of ASIO sound) there is one callback in which input and output is handled simultaneously. Can I somehow setup ALSA to have a similar behavor? Thank you for any assistance and best regards HK

3 7

Re: [alsa-devel] [PATCH v4 1/5] OMAP4: hwmod: Add names for DMIC memory address space
by Peter Ujfalusi 02 Feb '12

02 Feb '12

Hi Paul, On 01/25/2012 11:23 PM, Paul Walmsley wrote: > Okay, thanks, queued for 3.3-rc. It seams this patch did not made it to 3.3-rc2. We still have broken audio out of box on SDP4430/Blaze. Can you make sure that this will be queued for rc3? Thank you, Péter

2 2

[alsa-devel] [PATCH] ASoC: wm_hubs: Correct line input to line output 2 paths
by Mark Brown 02 Feb '12

02 Feb '12

The second line output mixer has the controls for the line input bypasses in the opposite order. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 57514dd2..b08d04c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 1/2] ASoC: Use dai_fmt in afeb9260 machine driver
by Axel Lin 01 Feb '12

01 Feb '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/snd-soc-afeb9260.c | 23 ++--------------------- 1 files changed, 2 insertions(+), 21 deletions(-) diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 4ca667d..cb0130c 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -46,29 +46,8 @@ static int afeb9260_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S| - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return err; - } - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return err; - } - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -129,6 +108,8 @@ static struct snd_soc_dai_link afeb9260_dai = { .platform_name = "atmel_pcm-audio", .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &afeb9260_ops, }; -- 1.7.5.4

2 2

[alsa-devel] [PATCH 2/3] ARM: ux500: Add DMA-channels for MSP
by Ola Lilja 01 Feb '12

01 Feb '12

Add DMA-channels for MSP0, MSP1, MSP2 and MSP3. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/devices-db8500.c | 6 ++++++ arch/arm/mach-ux500/include/mach/hardware.h | 2 ++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ux500/devices-db8500.c b/arch/arm/mach-ux500/devices-db8500.c index a7c6cdc..6e66d37 100644 --- a/arch/arm/mach-ux500/devices-db8500.c +++ b/arch/arm/mach-ux500/devices-db8500.c @@ -101,6 +101,9 @@ static const dma_addr_t dma40_tx_map[DB8500_DMA_NR_DEV] = { [DB8500_DMA_DEV41_SD_MM3_TX] = -1, [DB8500_DMA_DEV42_SD_MM4_TX] = -1, [DB8500_DMA_DEV43_SD_MM5_TX] = -1, + [DB8500_DMA_DEV14_MSP2_TX] = U8500_MSP2_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV30_MSP1_TX] = U8500_MSP1_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV31_MSP0_TX_SLIM0_CH0_TX] = U8500_MSP0_BASE + MSP_TX_RX_REG_OFFSET, }; /* Mapping between source event lines and physical device address */ @@ -133,6 +136,9 @@ static const dma_addr_t dma40_rx_map[DB8500_DMA_NR_DEV] = { [DB8500_DMA_DEV41_SD_MM3_RX] = -1, [DB8500_DMA_DEV42_SD_MM4_RX] = -1, [DB8500_DMA_DEV43_SD_MM5_RX] = -1, + [DB8500_DMA_DEV14_MSP2_RX] = U8500_MSP2_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV30_MSP3_RX] = U8500_MSP3_BASE + MSP_TX_RX_REG_OFFSET, + [DB8500_DMA_DEV31_MSP0_RX_SLIM0_CH0_RX] = U8500_MSP0_BASE + MSP_TX_RX_REG_OFFSET, }; /* Reserved event lines for memcpy only */ diff --git a/arch/arm/mach-ux500/include/mach/hardware.h b/arch/arm/mach-ux500/include/mach/hardware.h index b6ba26a..d93d6db 100644 --- a/arch/arm/mach-ux500/include/mach/hardware.h +++ b/arch/arm/mach-ux500/include/mach/hardware.h @@ -30,6 +30,8 @@ #include <mach/db8500-regs.h> #include <mach/db5500-regs.h> +#define MSP_TX_RX_REG_OFFSET 0 + #ifndef __ASSEMBLY__ #include <mach/id.h> -- 1.7.8.3

2 1

[alsa-devel] [PATCH 1/3] ARM: ux500: Add placeholder for clk_set_parent
by Ola Lilja 01 Feb '12

01 Feb '12

Calling clk_set_parent (e.g. from Ux500 ASoC-driver) generates build-errors. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- arch/arm/mach-ux500/clock.c | 7 +++++++ arch/arm/mach-ux500/clock.h | 1 + 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-ux500/clock.c b/arch/arm/mach-ux500/clock.c index 7379075..ec35f0a 100644 --- a/arch/arm/mach-ux500/clock.c +++ b/arch/arm/mach-ux500/clock.c @@ -223,6 +223,13 @@ int clk_set_rate(struct clk *clk, unsigned long rate) } EXPORT_SYMBOL(clk_set_rate); +int clk_set_parent(struct clk *clk, struct clk *parent) +{ + /*TODO*/ + return -ENOSYS; +} +EXPORT_SYMBOL(clk_set_parent); + static void clk_prcmu_enable(struct clk *clk) { void __iomem *cg_set_reg = __io_address(U8500_PRCMU_BASE) diff --git a/arch/arm/mach-ux500/clock.h b/arch/arm/mach-ux500/clock.h index 0744907..d776ada 100644 --- a/arch/arm/mach-ux500/clock.h +++ b/arch/arm/mach-ux500/clock.h @@ -21,6 +21,7 @@ struct clkops { void (*enable) (struct clk *); void (*disable) (struct clk *); unsigned long (*get_rate) (struct clk *); + int (*set_parent)(struct clk *, struct clk *); }; /** -- 1.7.8.3

2 1

[alsa-devel] [PATCH] ALSA: E-mu 1616m PCIe support
by Sergey Korolev 01 Feb '12

01 Feb '12

diff -urdN linux-2.6.36.orig/sound/pci/emu10k1/emu10k1_main.c linux-2.6.36/sound/pci/emu10k1/emu10k1_main.c --- linux-2.6.36.orig/sound/pci/emu10k1/emu10k1_main.c 2010-10-20 22:30:22.000000000 +0200 +++ linux-2.6.36/sound/pci/emu10k1/emu10k1_main.c 2011-01-21 17:48:11.440000537 +0100 @@ -1415,6 +1415,14 @@ .ca0108_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 new revision */ + /* Tested 21th Jan 2011. */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40071102, + .driver = "Audigy2", .name = "E-mu 1010 PCIe", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU1010B}, /* EMU 1010 PCIe */ /* Tested by James(a)superbug.co.uk 8th July 2005. */ /* This is MAEM8810, 0202 is MAEM8820 */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40011102,

1 0

[alsa-devel] [PATCH] ASoC: blackfin: Use dai_fmt
by Lars-Peter Clausen 01 Feb '12

01 Feb '12

Use the dai_link's dai_fmt attribute to setup the DAI format instead of doing this manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ad1836.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad193x.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad73311.c | 29 ++++------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 20 ++++---------------- sound/soc/blackfin/bfin-eval-adau1373.c | 13 ++----------- sound/soc/blackfin/bfin-eval-adau1701.c | 16 +++++----------- sound/soc/blackfin/bfin-eval-adav80x.c | 13 ++----------- 7 files changed, 27 insertions(+), 98 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 60962ce..d542d40 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -40,20 +40,8 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; /* set cpu DAI channel mapping */ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), @@ -68,6 +56,9 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { .hw_params = bf5xx_ad1836_hw_params, }; +#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { { .name = "ad1836", @@ -77,6 +68,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, { .name = "ad1836", @@ -86,6 +78,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 2d8d82d..0e55e9f 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,18 +60,6 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, break; } - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, SND_SOC_CLOCK_IN); @@ -92,6 +80,9 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, return 0; } +#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_ops bf5xx_ad193x_ops = { .hw_params = bf5xx_ad193x_hw_params, }; @@ -105,6 +96,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, { .name = "ad193x", @@ -114,6 +106,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 8e49508..61cc91d 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -145,29 +145,8 @@ static int bf5xx_probe(struct snd_soc_card *card) return 0; } -static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - - -static struct snd_soc_ops bf5xx_ad73311_ops = { - .hw_params = bf5xx_ad73311_hw_params, -}; +#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { { @@ -177,7 +156,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, { .name = "ad73311", @@ -186,7 +165,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 0303032..df3ac73 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -49,7 +49,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int clk = 0; int ret = 0; @@ -75,21 +74,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, break; } - /* - * CODEC is master for BCLK and LRC in this configuration. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -102,6 +86,10 @@ static struct snd_soc_ops bf5xx_ssm2602_ops = { .hw_params = bf5xx_ssm2602_hw_params, }; +/* CODEC is master for BCLK and LRC in this configuration. */ +#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { { .name = "ssm2602", diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 26b271c..f3adbdb 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -67,21 +67,10 @@ static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; int pll_rate; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - switch (params_rate(params)) { case 48000: case 8000: @@ -143,6 +132,8 @@ static struct snd_soc_dai_link bfin_eval_adau1373_dai = { .codec_name = "adau1373.0-001a", .ops = &bfin_eval_adau1373_ops, .init = bfin_eval_adau1373_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card bfin_eval_adau1373 = { diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index c0064fa..b0531fc 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -37,20 +37,9 @@ static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, SND_SOC_CLOCK_IN); @@ -61,6 +50,9 @@ static struct snd_soc_ops bfin_eval_adau1701_ops = { .hw_params = bfin_eval_adau1701_hw_params, }; +#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { { .name = "adau1701", @@ -70,6 +62,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, { .name = "adau1701", @@ -79,6 +72,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 4ef079f..84b0998 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -34,20 +34,9 @@ static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, 27000000, params_rate(params) * 256); if (ret) @@ -88,6 +77,8 @@ static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { .platform_name = "bfin-i2s-pcm-audio", .init = bfin_eval_adav80x_codec_init, .ops = &bfin_eval_adav80x_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }, }; -- 1.7.8.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm9712: Remove driver specific version
by Mark Brown 01 Feb '12

01 Feb '12

I don't think it's been updated since the driver was merged. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b7b31f8..247f19a 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -22,8 +22,6 @@ #include <sound/soc.h> #include "wm9712.h" -#define WM9712_VERSION "0.4" - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, @@ -619,8 +617,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); -- 1.7.9.rc1

1 1

[alsa-devel] [PATCH] ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
by David Henningsson 01 Feb '12

01 Feb '12

Since I'm a little new in HDMI land and the patch is non-trivial, I wouldn't mind some comments on this one. Also thanks to Jaroslav who cleared a few things out about the logical and physical devices which helped me figure out what I should do about it :-) Anyway, here's the patch. Some newer chips have more than one HDMI output, but usually not all of them are exposed as physical jacks. Removing the unused PCM devices (as indicated by BIOS in the pin config default) will reduce user confusion as they currently have to choose between several HDMI devices, some of them not working anyway. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 14

[alsa-devel] [PATCH] ALSA: HDA: Fix duplicated output to more than one codec
by David Henningsson 01 Feb '12

01 Feb '12

This typo caused the wrong codec's nid to be checked for wcaps type. As a result, sometimes speakers would duplicate the output sent to HDMI output. Cc: stable(a)kernel.org BugLink: https://bugs.launchpad.net/bugs/924320 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4df72c0..c2c65f6 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1447,7 +1447,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < c->cvt_setups.used; i++) { p = snd_array_elem(&c->cvt_setups, i); if (!p->active && p->stream_tag == stream_tag && - get_wcaps_type(get_wcaps(codec, p->nid)) == type) + get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; } } -- 1.7.8.3

2 1

[alsa-devel] [RFC][PATCH] ASoC: cs42l73: Fix Output [X|A|V]SP_SCLK Sourcing Mode setting for master mode
by Axel Lin 01 Feb '12

01 Feb '12

For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Brian, I think current code of setting priv->config[id].spc for master mode is wrong: priv->config[id].spc &= MCK_SCLK_64FS; will set priv->config[id].spc to be 0. I don't have this hardware, I'd appreciate you can review and test this patch. Regards, Axel sound/soc/codecs/cs42l73.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 9d38db8..78979b3 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc &= MCK_SCLK_64FS; + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; -- 1.7.5.4

3 2

[alsa-devel] [PATCH] ASoC: wm8962: Fix word length configuration
by Mark Brown 01 Feb '12

01 Feb '12

From: Susan Gao <sgao(a)opensource.wolfsonmicro.com> Signed-off-by: Susan Gao <sgao(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index de74af7..c713d55 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2594,13 +2594,13 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: break; case SNDRV_PCM_FORMAT_S20_3LE: - aif0 |= 0x40; + aif0 |= 0x4; break; case SNDRV_PCM_FORMAT_S24_LE: - aif0 |= 0x80; + aif0 |= 0x8; break; case SNDRV_PCM_FORMAT_S32_LE: - aif0 |= 0xc0; + aif0 |= 0xc; break; default: return -EINVAL; -- 1.7.9.rc1

1 0

Re: [alsa-devel] soc-dsp programming model for loopbacks
by Liam Girdwood 01 Feb '12

01 Feb '12

On Tue, 2012-01-24 at 13:36 -0800, Pierre-Louis Bossart wrote: > Hi, > I've been doing some work on audio loopback (FM radio, BT, Modem -> audio > codec), and I am somewhat confused by the soc-dsp programming model. Let me > take the FM radio example: > > If the FM radio is directly connected to the audio codec, my understanding > is that changing the routing with codec controls will trigger the DAPM > logic, which will turn on everything that needs to be on. Very simple, no > matter if the FM-codec link is analog or digital. > Yep, this is correct. soc-dsp (now know as Dynamic PCM) takes care of any digital links that require the host CPU to perform any PCM or DAI operations (based again on DAPM graph status). i.e. on OMAP we can start the McBSP to FM Digital DAI based on the DAPM graph status and then configure the link format, speed etc to optimal values for the ABE and FM. It's not required for the analog path as DAPM will switch on the correct components for FM analog to be played on the CODEC Speaker output. > Now if the FM radio routing is handled with a digital loopback on the > application processor audio dsp (omap-abe, Intel SST, etc), then soc-dsp > will need to be used. And for a simple FM playback, I need to > 1. configure the audio codec routing for output selection > 2. open a virtual front-end for FM capture > 3. configure the DSP routing to link capture front-end to I2S1 backend (FM > interface) > 4. open a virtual front-end for FM playback > 5. configure the DSP routing to link playback front-end to I2S2 backend > (codec interface) > Using Dynamic PCM here really depends on your hardware capabilities. On OMAP, the host CPU configures and manages all DAIs so Dynamic PCM manages the DAI and PCM operations. We also have the need to re-write the hw_params and DAI formats between the FEs and BEs for optimum performance. > That seems complicated. To some extent, having the ability to have two > back-ends connected together would make more sense, and would simplify the > programming model a great deal. User-space code would be similar for > loopbacks internal to the codecs or handled on the application processor. > This would apply to Bluetooth and modem connections as well. Without this > capability, we will end-up with multiple 'virtual' front-ends (6 in my > case), making user-space code quite complex. It's not really that much extra code, it's really just open(), hw_params() and pcm_start() on the FE. Steps 1,3 & 5 are compulsory anyway for any audio device, but we need to configure format, rate, DAI clocking for the loopback. Having the FE PCM gives us this capability, although we dont transfer any data to or from the host with the FE so it's use is only for configuration. We could eventually remove steps 2 and 4 for the FE DAI link, and look at hard coding the hw_params() in the mach driver for the loopback link. That way the only user space driven actions would be to configure the mixers in the CODEC and DSP for the correct route. DAPM would then detect the path and Dynamic PCM would use the hard coded configuration or bespoke mach driver logic to configure the loopback DAI link based on use case. This would have to be done after the basic Dynamic PCM infrastructure was upstream though (unless you have a patch atm). > Looking at the current soc-dsp code, I saw that each back-end is supposed to > have at least one front-end client, and the impact of my proposal seems > fairly important. Before I start looking further into code changes, I wanted > to see if my understanding is correct and if there are other ideas to > simplify loopbacks. Atm with the current code base we can support routing many FE PCMs to many BE PCMs. We can support the BE loopback, but this does require the FE to be opened atm in order to configure the DAI link. Regards Liam

4 5

[alsa-devel] [PATCH] ALSA: hda - Fix calling cs_automic twice for Cirrus codecs.
by Dylan Reid 01 Feb '12

01 Feb '12

If cs_automic is called twice (like it is during init) while the mic is present, it will over-write the last_input with the new one, causing it to switch back to the automic input when the mic is unplugged. This leaves the driver in a state (cur_input, last_input, and automix_idx the same) where the internal mic can not be selected until it is rebooted without the mic attached. Check that the mic hasn't already been switched to before setting last_input. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0e99357..bc5a993 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec) change_cur_input(codec, !spec->automic_idx, 0); } else { if (present) { - spec->last_input = spec->cur_input; - spec->cur_input = spec->automic_idx; + if (spec->cur_input != spec->automic_idx) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } } else { spec->cur_input = spec->last_input; } -- 1.7.9.rc0

2 1

[alsa-devel] [PATCH] ALSA: HDA: Fix jack creation for codecs with front and rear Line In
by David Henningsson 01 Feb '12

01 Feb '12

If a codec has both a front and a rear Line In, two controls both named "Line Jack" will be created, which causes parsing to fail. While a long term solution might be to name the jacks differently, this extra check is consistent with what is already being done in many auto-parsers, and will also protect against other cases when two inputs have the same label. BugLink: https://bugs.launchpad.net/bugs/923409 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_jack.c | 24 +++++++++++++++--------- 1 files changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d8a35da..9d819c4 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -282,7 +282,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl); static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, - const struct auto_pin_cfg *cfg) + const struct auto_pin_cfg *cfg, + char *lastname, int *lastidx) { unsigned int def_conf, conn; char name[44]; @@ -298,6 +299,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, return 0; snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx); + if (!strcmp(name, lastname) && idx == *lastidx) + idx++; + strncpy(lastname, name, 44); + *lastidx = idx; err = snd_hda_jack_add_kctl(codec, nid, name, idx); if (err < 0) return err; @@ -311,41 +316,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; - int i, err; + int i, err, lastidx = 0; + char lastname[44] = ""; for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0; i < cfg->num_inputs; i++) { - err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg); + err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx); if (err < 0) return err; } for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg); + err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx); if (err < 0) return err; } - err = add_jack_kctl(codec, cfg->dig_in_pin, cfg); + err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx); if (err < 0) return err; - err = add_jack_kctl(codec, cfg->mono_out_pin, cfg); + err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx); if (err < 0) return err; return 0; -- 1.7.8.3

3 8

[alsa-devel] [PATCH 0/5] Add device tree support of audio (ALC5632) on PAZ00 board
by Leon Romanovsky 01 Feb '12

01 Feb '12

The following patchset is going to: * add device tree support of audio (ALC5632) on PAZ00 board. * enable headset autodetection. * enable headset microphone. --- I will update the ALC5632 codec to routing names used in documentation, in the later patches, Leon Romanovsky (5): ASoC: Tegra+ALC5632: Implement device tree support in board file ASoC: ALC5632: Add device tree binding documentation ASoC: Tegra+ALC5632 machine: Add device tree binding documentation ASoC: dt: tegra: Enable headset autodetection on PAZ00 board. ASoC: tegra: Remove unused DAPM route structure. .../devicetree/bindings/sound/alc5632.txt | 24 ++++ .../bindings/sound/tegra-audio-alc5632.txt | 57 +++++++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + arch/arm/boot/dts/tegra-paz00.dts | 1 + sound/soc/tegra/tegra_alc5632.c | 129 +++++++++++++++---- 5 files changed, 185 insertions(+), 27 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt -- 1.7.3.4

4 5

[alsa-devel] [PATCH 5/5] ASoC: tegra: Remove unused DAPM route structure.
by Leon Romanovsky 01 Feb '12

01 Feb '12

All DAPM routes are configured via device tree, and there is no need in DAPM route structures in board file. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/tegra/tegra_alc5632.c | 16 ---------------- 1 files changed, 0 insertions(+), 16 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 1794139..c000f51 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -104,20 +104,6 @@ static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), }; -static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { - /* Internal Speaker */ - {"Int Spk", NULL, "SPKOUT"}, - {"Int Spk", NULL, "SPKOUTN"}, - - /* Headset Mic */ - {"MIC1", NULL, "MICBIAS1"}, - {"MICBIAS1", NULL, "Headset Mic"}, - - /* Headset Stereophone */ - {"Headset Stereophone", NULL, "HPR"}, - {"Headset Stereophone", NULL, "HPL"}, -}; - static const struct snd_kcontrol_new tegra_alc5632_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; @@ -172,8 +158,6 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = { .num_controls = ARRAY_SIZE(tegra_alc5632_controls), .dapm_widgets = tegra_alc5632_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), - .dapm_routes = tegra_alc5632_audio_map, - .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), .fully_routed = true, }; -- 1.7.3.4

1 1

[alsa-devel] [PATCH 4/5] ASoC: dt: tegra: Enable headset autodetection on PAZ00 board.
by Leon Romanovsky 01 Feb '12

01 Feb '12

This patch is adding support of headset autodetection on PAZ00 board. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- arch/arm/boot/dts/tegra-paz00.dts | 1 + sound/soc/tegra/tegra_alc5632.c | 39 +++++++++++++++++++++++++++++++++--- 2 files changed, 36 insertions(+), 4 deletions(-) diff --git a/arch/arm/boot/dts/tegra-paz00.dts b/arch/arm/boot/dts/tegra-paz00.dts index e6809b7..fa9dd9e 100644 --- a/arch/arm/boot/dts/tegra-paz00.dts +++ b/arch/arm/boot/dts/tegra-paz00.dts @@ -64,6 +64,7 @@ nvidia,audio-codec = <&alc5632>; nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */ }; serial@70006000 { diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c0ba1e4..1794139 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/gpio.h> +#include <linux/of_gpio.h> #include <sound/core.h> #include <sound/jack.h> @@ -34,9 +35,13 @@ #define DRV_NAME "tegra-alc5632" +#define GPIO_HP_DET BIT(0) + struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; struct platform_device *pcm_dev; + int gpio_requested; + int gpio_hp_det; }; static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, @@ -86,6 +91,13 @@ static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { }, }; +static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { + .name = "Headset detection", + .report = SND_JACK_HEADSET, + .debounce_time = 150, + .invert = 1, +}; + static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), @@ -114,6 +126,9 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + int ret; snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); @@ -121,6 +136,16 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(tegra_alc5632_hs_jack_pins), tegra_alc5632_hs_jack_pins); + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + 1, + &tegra_alc5632_hp_jack_gpio); + machine->gpio_requested |= GPIO_HP_DET; + } + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); return 0; @@ -239,13 +264,19 @@ err: static int __devexit tegra_alc5632_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); + + if (machine->gpio_requested & GPIO_HP_DET) + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, + 1, + &tegra_alc5632_hp_jack_gpio); + machine->gpio_requested = 0; snd_soc_unregister_card(card); - tegra_asoc_utils_fini(&alc5632->util_data); - if (!IS_ERR(alc5632->pcm_dev)) - platform_device_unregister(alc5632->pcm_dev); + tegra_asoc_utils_fini(&machine->util_data); + if (!IS_ERR(machine->pcm_dev)) + platform_device_unregister(machine->pcm_dev); return 0; } -- 1.7.3.4

1 1

[alsa-devel] [PATCH] ASoC: Add support for TI LM49453 Audio codec
by Reddy, MR Swami 31 Jan '12

31 Jan '12

ASoC: Add support for TI LM49453 Audio codec Signed-off-by: MR Swami Reddy <MR.Swami.Reddy(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lm49453.c | 1508 ++++++++ sound/soc/codecs/lm49453.h | 384 ++ 4 files changed, 1898 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6943e24..ed5f66c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_LM4857 if I2C + select SND_SOC_LM49453 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C @@ -184,6 +185,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_LM49453 + tristate + config SND_SOC_MAX98088 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 379bc55..7bbfdb2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o +snd-soc-lm49453-objs := lm49453.o snd-soc-max98088-objs := max98088.o snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o @@ -106,6 +107,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_LM49453) += snd-soc-lm49453.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c new file mode 100644 index 0000000..f77647d --- /dev/null +++ b/sound/soc/codecs/lm49453.c @@ -0,0 +1,1508 @@ +/* + * lm49453.c - LM49453 ALSA Soc Audio driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * 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; version 2 of the License. + * + * Initially based on sound/soc/codecs/wm8350.c + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <asm/div64.h> +#include "lm49453.h" + +static struct reg_default lm49453_reg_defs[] = { + { 0, 0x00}, { 1, 0x00}, { 2, 0x00}, { 3, 0x00}, /*R3*/ + { 4, 0x00}, { 5, 0x00}, { 6, 0x00}, { 7, 0x00}, /*R7*/ + { 8, 0x00}, { 9, 0x00}, { 10, 0x00}, { 11, 0x00}, /*R11*/ + { 12, 0x00}, { 13, 0x00}, { 14, 0x00}, { 15, 0x00}, /*R15*/ + { 16, 0x00}, { 17, 0x00}, { 18, 0x00}, { 19, 0x00}, /*R19*/ + { 20, 0x00}, { 21, 0x00}, { 22, 0x00}, { 23, 0x00}, /*R23*/ + { 24, 0x00}, { 25, 0x00}, { 26, 0x00}, { 27, 0x00}, /*R27*/ + { 28, 0x00}, { 29, 0x00}, { 30, 0x00}, { 31, 0x00}, /*R31*/ + { 32, 0x00}, { 33, 0x00}, { 34, 0x00}, { 35, 0x00}, /*R35*/ + { 36, 0x00}, { 37, 0x00}, { 38, 0x00}, { 39, 0x00}, /*R39*/ + { 40, 0x00}, { 41, 0x00}, { 42, 0x00}, { 43, 0x00}, /*R43*/ + { 44, 0x00}, { 45, 0x00}, { 46, 0x00}, { 47, 0x00}, /*R47*/ + { 48, 0x00}, { 49, 0x00}, { 50, 0x00}, { 51, 0x00}, /*R51*/ + { 52, 0x00}, { 53, 0x00}, { 54, 0x00}, { 55, 0x00}, /*R55*/ + { 57, 0x00}, { 58, 0x00}, { 59, 0x00}, { 59, 0x00}, /*R59*/ + { 60, 0x00}, { 61, 0x00}, { 62, 0x00}, { 63, 0x00}, /*R63*/ + { 64, 0x00}, { 65, 0x00}, { 66, 0x00}, { 67, 0x00}, /*R67*/ + { 68, 0x00}, { 69, 0x00}, { 70, 0x00}, { 71, 0x00}, /*R71*/ + { 72, 0x00}, { 73, 0x00}, { 74, 0x00}, { 75, 0x00}, /*R75*/ + { 76, 0x00}, { 77, 0x00}, { 78, 0x00}, { 79, 0x00}, /*R79*/ + { 80, 0x00}, { 81, 0x00}, { 82, 0x00}, { 83, 0x00}, /*R83*/ + { 85, 0x00}, { 85, 0x00}, { 86, 0x00}, { 87, 0x00}, /*R87*/ + { 88, 0x00}, { 89, 0x00}, { 90, 0x00}, { 91, 0x00}, /*R91*/ + { 92, 0x00}, { 93, 0x00}, { 94, 0x00}, { 95, 0x00}, /*R95*/ + { 96, 0x01}, { 97, 0x00}, { 98, 0x00}, { 99, 0x00}, /*R99*/ + { 100, 0x00}, { 101, 0x00}, { 102, 0x00}, { 103, 0x01}, /*R103*/ + { 105, 0x01}, { 106, 0x00}, { 107, 0x01}, { 107, 0x00}, /*R107*/ + { 108, 0x00}, { 109, 0x00}, { 110, 0x00}, { 111, 0x02}, /*R111*/ + { 112, 0x02}, { 113, 0x00}, { 114, 0x00}, { 115, 0x00}, /*R115*/ + { 116, 0x00}, { 117, 0x00}, { 118, 0x00}, { 119, 0x00}, /*R119*/ + { 120, 0x00}, { 121, 0x80}, { 122, 0xBB}, { 123, 0x80}, /*R123*/ + { 124, 0xBB}, { 125, 0x00}, { 126, 0x00}, { 127, 0x00}, /*R127*/ + { 128, 0x00}, { 129, 0x00}, { 130, 0x00}, { 131, 0x00}, /*R131*/ + { 132, 0x00}, { 133, 0x0A}, { 134, 0x0A}, { 135, 0x0A}, /*R135*/ + { 136, 0x0F}, { 137, 0x00}, { 138, 0x73}, { 139, 0x33}, /*R139*/ + { 140, 0x73}, { 141, 0x33}, { 142, 0x73}, { 143, 0x33}, /*R143*/ + { 144, 0x73}, { 145, 0x33}, { 146, 0x73}, { 147, 0x33}, /*R147*/ + { 148, 0x73}, { 149, 0x33}, { 150, 0x73}, { 151, 0x33}, /*R151*/ + { 152, 0x00}, { 153, 0x00}, { 154, 0x00}, { 155, 0x00}, /*R155*/ + { 156, 0x00}, { 157, 0x00}, { 158, 0x00}, { 159, 0x00}, /*R159*/ + { 160, 0x00}, { 161, 0x00}, { 162, 0x00}, { 163, 0x00}, /*R163*/ + { 164, 0x00}, { 165, 0x00}, { 166, 0x00}, { 167, 0x00}, /*R167*/ + { 168, 0x00}, { 169, 0x00}, { 170, 0x00}, { 171, 0x00}, /*R171*/ + { 172, 0x00}, { 173, 0x00}, { 174, 0x00}, { 175, 0x00}, /*R175*/ + { 176, 0x00}, { 177, 0x00}, { 178, 0x00}, { 179, 0x00}, /*R179*/ + { 180, 0x00}, { 181, 0x00}, { 182, 0x00}, { 183, 0x00}, /*R183*/ + { 184, 0x00}, { 185, 0x00}, { 186, 0x00}, { 189, 0x00}, /*R187*/ + { 188, 0x00}, { 189, 0x00}, { 190, 0x00}, { 191, 0x00}, /*R191*/ + { 192, 0x00}, { 193, 0x00}, { 194, 0x00}, { 195, 0x00}, /*R195*/ + { 196, 0x00}, { 197, 0x00}, { 198, 0x00}, { 199, 0x00}, /*R199*/ + { 200, 0x00}, { 201, 0x00}, { 202, 0x00}, { 203, 0x00}, /*R203*/ + { 204, 0x00}, { 205, 0x00}, { 206, 0x00}, { 207, 0x00}, /*R207*/ + { 208, 0x00}, { 209, 0x00}, { 210, 0x00}, { 211, 0x00}, /*R211*/ + { 212, 0x00}, { 213, 0x00}, { 214, 0x00}, { 215, 0x00}, /*R215*/ + { 216, 0x00}, { 217, 0x00}, { 218, 0x00}, { 219, 0x00}, /*R219*/ + { 221, 0x00}, { 222, 0x00}, { 223, 0x00}, { 224, 0x00}, /*R223*/ + { 224, 0x00}, { 225, 0x00}, { 226, 0x00}, { 227, 0x00}, /*R227*/ + { 228, 0x00}, { 229, 0x00}, { 230, 0x13}, { 231, 0x00}, /*R231*/ + { 232, 0x80}, { 233, 0x0C}, { 234, 0xDD}, { 235, 0x00}, /*R235*/ + { 236, 0x04}, { 237, 0x00}, { 238, 0x00}, { 239, 0x00}, /*R239*/ + { 240, 0x00}, { 241, 0x00}, { 242, 0x00}, { 243, 0x00}, /*R243*/ + { 244, 0x00}, { 245, 0x00}, { 246, 0x00}, { 247, 0x00}, /*R247*/ + { 248, 0x00}, { 249, 0x00}, { 250, 0x00}, { 251, 0x00}, /*R251*/ + { 252, 0x00}, { 253, 0x00}, { 254, 0x00}, { 255, 0x00}, /*R255*/ +}; + +struct pll_state { + unsigned int in; + unsigned int out; +}; + +/* codec private data */ +struct lm49453_priv { + struct regmap *regmap; + struct pll_state pll1, pll2; + int fs_rate; + int sysclk; +}; + +/* capture path controls */ + +static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, + 5, lm49453_mic2mode_text); + +static const char *lm49453_dmic_cfg_text[] = {"DMIC1", "DMIC2"}; + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, + 7, lm49453_dmic_cfg_text); + +/* MUX Controls */ +static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; + +static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; + +static const struct soc_enum lm49453_adcl_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + ARRAY_SIZE(lm49453_adcl_mux_text), + lm49453_adcl_mux_text); + +static const struct soc_enum lm49453_adcr_enum = + SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + ARRAY_SIZE(lm49453_adcr_mux_text), + lm49453_adcr_mux_text); + +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 0, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_headset_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHPR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHPR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHPR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHPR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHPR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHPR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHPR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHPR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHPR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHPR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHPR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHPR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHPR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHPR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHPR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHPR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 1, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 2, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_speaker_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLSR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLSR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLSR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLSR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLSR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLSR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLSR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLSR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLSR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLSR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLSR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLSR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLSR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLSR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLSR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLSR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 3, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 4, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_haptic_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACHAR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACHAR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACHAR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACHAR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACHAR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACHAR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACHAR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACHAR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACHAR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACHAR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACHAR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACHAR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACHAR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACHAR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACHAR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACHAR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 5, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_left_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOL1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOL1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOL1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOL1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOL1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOL1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOL1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOL1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOL2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOL2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOL2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOL2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOL2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOL2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOL2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOL2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 6, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_lineout_right_mixer[] = { +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_DACLOR1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_DACLOR1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_DACLOR1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_DACLOR1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_DACLOR1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_DACLOR1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_DACLOR1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_DACLOR1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_DACLOR2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_DACLOR2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_DACLOR2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_DACLOR2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_DACLOR2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_DACLOR2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_DACLOR2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_DACLOR2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("Sidetone Switch", LM49453_P0_STN_SEL_REG, 7, 0, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT1_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT1_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT1_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT1_TX2_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx3_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX3_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX3_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX3_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX3_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX3_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX3_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_3 Switch", LM49453_P0_PORT1_TX3_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx4_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX4_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX4_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX4_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX4_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX4_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX4_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_4 Switch", LM49453_P0_PORT1_TX4_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx5_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX5_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX5_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX5_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX5_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX5_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX5_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_5 Switch", LM49453_P0_PORT1_TX5_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx6_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX6_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX6_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX6_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX6_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX6_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX6_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_6 Switch", LM49453_P0_PORT1_TX6_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx7_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX7_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX7_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX7_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX7_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX7_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX7_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_7 Switch", LM49453_P0_PORT1_TX7_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port1_tx8_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT1_TX8_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT1_TX8_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT1_TX8_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT1_TX8_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT1_TX8_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT1_TX8_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_8 Switch", LM49453_P0_PORT1_TX8_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx1_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX1_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX1_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX1_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX1_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX1_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_1 Switch", LM49453_P0_PORT2_TX1_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_1 Switch", LM49453_P0_PORT2_TX1_REG, 7, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_port2_tx2_mixer[] = { +SOC_DAPM_SINGLE("DMIC1L Switch", LM49453_P0_PORT2_TX2_REG, 0, 1, 0), +SOC_DAPM_SINGLE("DMIC1R Switch", LM49453_P0_PORT2_TX2_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DMIC2L Switch", LM49453_P0_PORT2_TX2_REG, 2, 1, 0), +SOC_DAPM_SINGLE("DMIC2R Switch", LM49453_P0_PORT2_TX2_REG, 3, 1, 0), +SOC_DAPM_SINGLE("ADCL Switch", LM49453_P0_PORT2_TX2_REG, 4, 1, 0), +SOC_DAPM_SINGLE("ADCR Switch", LM49453_P0_PORT2_TX2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("Port1_2 Switch", LM49453_P0_PORT2_TX2_REG, 6, 1, 0), +SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), +}; + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { +/* Sidetone supports mono only */ +SOC_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, + 0, 0x3F, 0, digital_tlv), +SOC_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, + 0, 0x3F, 0, digital_tlv), +SOC_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, + 0, 0x3F, 0, digital_tlv), +SOC_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, + 0, 0x3F, 0, digital_tlv), +SOC_DAPM_SINGLE("Sidetone Enable Switch", LM49453_P0_DMIX_CLK_SEL_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new lm49453_snd_controls[] = { + /* mic1 and mic2 supports mono only */ + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, + 0, digital_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, + 0, digital_tlv), + + SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, + LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, + LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), + SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), + SOC_DAPM_ENUM("DMIC34 SRC", lm49453_dmic34_cfg_enum), + + /* Capture path filter enable */ + SOC_DAPM_SINGLE("DMIC1 HPFilter Switch", + LM49453_P0_ADC_FX_ENABLES_REG, 0, 1, 0), + SOC_DAPM_SINGLE("DMIC2 HPFilter Switch", + LM49453_P0_ADC_FX_ENABLES_REG, 1, 1, 0), + SOC_DAPM_SINGLE("ADC HPFilter Switch", + LM49453_P0_ADC_FX_ENABLES_REG, 2, 1, 0), + + SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, + LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, + LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, + LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, + LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, + 0, 6, 0, digital_tlv), + + SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_2_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_3_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_4_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, + 6, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_5_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_6_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 2, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_7_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 4, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT1_8_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL2_REG, + 6, 3, 0, port_tlv), + + SOC_SINGLE_TLV("PORT2_1_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 0, 3, 0, port_tlv), + SOC_SINGLE_TLV("PORT2_2_RX_LVL Volume", LM49453_P0_PORT2_RX_LVL_REG, + 2, 3, 0, port_tlv), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget lm49453_dapm_widgets[] = { + + /* All end points HP,EP, LS, Lineout and Haptic */ + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("EPOUT"), + SND_SOC_DAPM_OUTPUT("LSOUTL"), + SND_SOC_DAPM_OUTPUT("LSOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTR"), + SND_SOC_DAPM_OUTPUT("LOOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTL"), + SND_SOC_DAPM_OUTPUT("HAOUTR"), + + SND_SOC_DAPM_INPUT("AMIC1"), + SND_SOC_DAPM_INPUT("AMIC2"), + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("DMIC2"), + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), + + SND_SOC_DAPM_PGA("PORT1_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_3_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_4_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_5_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_6_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_7_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT1_8_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_1_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PORT2_2_RX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0, NULL, 0), + + /* playback path driver enables */ + SND_SOC_DAPM_OUT_DRV("Headset Enable", + LM49453_P0_PMC_SETUP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earpiece Enable", + LM49453_P0_EP_REG, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Left Enable", + LM49453_P0_DIS_PKVL_FB_REG, 0, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Right Enable", + LM49453_P0_DIS_PKVL_FB_REG, 1, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Left Enable", + LM49453_P0_DIS_PKVL_FB_REG, 2, 1, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Haptic Right Enable", + LM49453_P0_DIS_PKVL_FB_REG, 3, 1, NULL, 0), + + /* DAC */ + SND_SOC_DAPM_DAC("HPL DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HPR DAC", "Headset", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSL DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LSR DAC", "Speaker", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAL DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("HAR DAC", "Haptic", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOL DAC", "Lineout", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("LOR DAC", "Lineout", SND_SOC_NOPM, 0, 0), + + + SND_SOC_DAPM_PGA("AUXL Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUXR Input", + LM49453_P0_ANALOG_MIXER_ADC_REG, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("Sidetone", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Capture", SND_SOC_NOPM, 1, 0), + SND_SOC_DAPM_ADC("ADC Right", "Capture", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("ADCL Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcl_mux_control), + SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &lm49453_adcr_mux_control), + + SND_SOC_DAPM_MUX("Mic1 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcl_mux_control), + + SND_SOC_DAPM_MUX("Mic2 Input", + SND_SOC_NOPM, 0, 0, &lm49453_adcr_mux_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("PORT1_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 2, 0), + SND_SOC_DAPM_AIF_IN("PORT2_SDI", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("PORT1_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 3, 0), + SND_SOC_DAPM_AIF_OUT("PORT2_SDO", NULL, 0, + LM49453_P0_PULL_CONFIG1_REG, 7, 0), + + /* Port1 TX controls */ + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Port2 TX controls */ + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Port1 Playback", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Port2 Playback", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 1, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("Port1 Capture", LM49453_P0_AUDIO_PORT1_BASIC_REG, + 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Port2 Capture", LM49453_P0_AUDIO_PORT2_BASIC_REG, + 2, 0, NULL, 0), + + /* Sidetone Mixer */ + SND_SOC_DAPM_MIXER("Sidetone Mixer", SND_SOC_NOPM, 0, 0, + &lm49453_sidetone_mixer_controls, + ARRAY_SIZE(lm49453_sidetone_mixer_controls)), + + /* DAC MIXERS */ + SND_SOC_DAPM_MIXER("HPL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_left_mixer, + ARRAY_SIZE(lm49453_headset_left_mixer)), + SND_SOC_DAPM_MIXER("HPR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_headset_right_mixer, + ARRAY_SIZE(lm49453_headset_right_mixer)), + SND_SOC_DAPM_MIXER("LOL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_left_mixer, + ARRAY_SIZE(lm49453_lineout_left_mixer)), + SND_SOC_DAPM_MIXER("LOR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_lineout_right_mixer, + ARRAY_SIZE(lm49453_lineout_right_mixer)), + SND_SOC_DAPM_MIXER("LSL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_left_mixer, + ARRAY_SIZE(lm49453_speaker_left_mixer)), + SND_SOC_DAPM_MIXER("LSR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_speaker_right_mixer, + ARRAY_SIZE(lm49453_speaker_right_mixer)), + SND_SOC_DAPM_MIXER("HAL Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_left_mixer, + ARRAY_SIZE(lm49453_haptic_left_mixer)), + SND_SOC_DAPM_MIXER("HAR Mixer", SND_SOC_NOPM, 0, 0, + lm49453_haptic_right_mixer, + ARRAY_SIZE(lm49453_haptic_right_mixer)), + + /* Capture Mixer */ + SND_SOC_DAPM_MIXER("Port1_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx1_mixer, + ARRAY_SIZE(lm49453_port1_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port1_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx2_mixer, + ARRAY_SIZE(lm49453_port1_tx2_mixer)), + SND_SOC_DAPM_MIXER("Port1_3 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx3_mixer, + ARRAY_SIZE(lm49453_port1_tx3_mixer)), + SND_SOC_DAPM_MIXER("Port1_4 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx4_mixer, + ARRAY_SIZE(lm49453_port1_tx4_mixer)), + SND_SOC_DAPM_MIXER("Port1_5 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx5_mixer, + ARRAY_SIZE(lm49453_port1_tx5_mixer)), + SND_SOC_DAPM_MIXER("Port1_6 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx6_mixer, + ARRAY_SIZE(lm49453_port1_tx6_mixer)), + SND_SOC_DAPM_MIXER("Port1_7 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx7_mixer, + ARRAY_SIZE(lm49453_port1_tx7_mixer)), + SND_SOC_DAPM_MIXER("Port1_8 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port1_tx8_mixer, + ARRAY_SIZE(lm49453_port1_tx8_mixer)), + + SND_SOC_DAPM_MIXER("Port2_1 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx1_mixer, + ARRAY_SIZE(lm49453_port2_tx1_mixer)), + SND_SOC_DAPM_MIXER("Port2_2 Mixer", SND_SOC_NOPM, 0, 0, + lm49453_port2_tx2_mixer, + ARRAY_SIZE(lm49453_port2_tx2_mixer)), +}; + +static const struct snd_soc_dapm_route lm49453_audio_map[] = { + /* Port SDI mapping */ + { "PORT1_1_RX", NULL, "PORT1_SDI" }, + { "PORT1_2_RX", NULL, "PORT1_SDI" }, + { "PORT1_3_RX", NULL, "PORT1_SDI" }, + { "PORT1_4_RX", NULL, "PORT1_SDI" }, + { "PORT1_5_RX", NULL, "PORT1_SDI" }, + { "PORT1_6_RX", NULL, "PORT1_SDI" }, + { "PORT1_7_RX", NULL, "PORT1_SDI" }, + { "PORT1_8_RX", NULL, "PORT1_SDI" }, + + { "PORT2_1_RX", NULL, "PORT2_SDI" }, + { "PORT2_2_RX", NULL, "PORT2_SDI" }, + + /* HP mapping */ + { "HPL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + { "HPL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPL Mixer", "ADCL Switch", "ADC Left" }, + { "HPL Mixer", "ADCR Switch", "ADC Right" }, + { "HPL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HPL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPL DAC", NULL, "HPL Mixer" }, + + { "HPR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HPR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HPR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HPR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HPR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HPR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HPR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HPR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HPR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HPR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HPR Mixer", "ADCL Switch", "ADC Left" }, + { "HPR Mixer", "ADCR Switch", "ADC Right" }, + { "HPR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HPR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HPR Mixer", "DMIC2L Switch", "DMIC2 Right" }, + { "HPR Mixer", "Sidetone Switch", "Sidetone" }, + + { "HPR DAC", NULL, "HPR Mixer" }, + + { "HPOUTL", "Headset Enable", "HPL DAC"}, + { "HPOUTR", "Headset Enable", "HPR DAC"}, + + /* EP map */ + { "EPOUT", "Earpiece Enable", "HPL DAC" }, + + /* Speaker map */ + { "LSL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSL Mixer", "ADCL Switch", "ADC Left" }, + { "LSL Mixer", "ADCR Switch", "ADC Right" }, + { "LSL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSL DAC", NULL, "LSL Mixer" }, + + { "LSR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LSR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LSR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LSR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LSR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LSR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LSR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LSR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LSR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LSR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LSR Mixer", "ADCL Switch", "ADC Left" }, + { "LSR Mixer", "ADCR Switch", "ADC Right" }, + { "LSR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LSR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LSR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LSR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LSR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LSR DAC", NULL, "LSR Mixer" }, + + { "LSOUTL", "Speaker Left Enable", "LSL DAC"}, + { "LSOUTR", "Speaker Left Enable", "LSR DAC"}, + + /* Haptic map */ + { "HAL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAL Mixer", "ADCL Switch", "ADC Left" }, + { "HAL Mixer", "ADCR Switch", "ADC Right" }, + { "HAL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAL Mixer", "Sidetone Switch", "Sidetone" }, + + { "HAL DAC", NULL, "HAL Mixer" }, + + { "HAR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "HAR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "HAR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "HAR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "HAR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "HAR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "HAR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "HAR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "HAR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "HAR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "HAR Mixer", "ADCL Switch", "ADC Left" }, + { "HAR Mixer", "ADCR Switch", "ADC Right" }, + { "HAR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "HAR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "HAR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "HAR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "HAR Mixer", "Sideton Switch", "Sidetone" }, + + { "HAR DAC", NULL, "HAR Mixer" }, + + { "HAOUTL", "Haptic Left Enable", "HAL DAC" }, + { "HAOUTR", "Haptic Right Enable", "HAR DAC" }, + + /* Lineout map */ + { "LOL Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOL Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOL Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOL Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOL Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOL Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOL Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOL Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOL Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOL Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOL Mixer", "ADCL Switch", "ADC Left" }, + { "LOL Mixer", "ADCR Switch", "ADC Right" }, + { "LOL Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOL Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOL Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOL Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOL Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOL DAC", NULL, "LOL Mixer" }, + + { "LOR Mixer", "Port1_1 Switch", "PORT1_1_RX" }, + { "LOR Mixer", "Port1_2 Switch", "PORT1_2_RX" }, + { "LOR Mixer", "Port1_3 Switch", "PORT1_3_RX" }, + { "LOR Mixer", "Port1_4 Switch", "PORT1_4_RX" }, + { "LOR Mixer", "Port1_5 Switch", "PORT1_5_RX" }, + { "LOR Mixer", "Port1_6 Switch", "PORT1_6_RX" }, + { "LOR Mixer", "Port1_7 Switch", "PORT1_7_RX" }, + { "LOR Mixer", "Port1_8 Switch", "PORT1_8_RX" }, + + /* Port 2 */ + { "LOR Mixer", "Port2_1 Switch", "PORT2_1_RX" }, + { "LOR Mixer", "Port2_2 Switch", "PORT2_2_RX" }, + + { "LOR Mixer", "ADCL Switch", "ADC Left" }, + { "LOR Mixer", "ADCR Switch", "ADC Right" }, + { "LOR Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "LOR Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "LOR Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "LOR Mixer", "DMIC2R Switch", "DMIC2 Right" }, + { "LOR Mixer", "Sidetone Switch", "Sidetone" }, + + { "LOR DAC", NULL, "LOR Mixer" }, + + { "LOOUTL", NULL, "LOL DAC" }, + { "LOOUTR", NULL, "LOR DAC" }, + + /* TX map */ + /* Port1 mappings */ + { "Port1_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_3 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_3 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_3 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_3 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_3 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_3 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_4 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_4 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_4 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_4 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_4 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_4 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_5 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_5 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_5 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_5 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_5 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_5 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_6 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_6 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_6 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_6 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_6 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_6 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_7 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_7 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_7 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_7 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_7 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_7 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port1_8 Mixer", "ADCL Switch", "ADC Left" }, + { "Port1_8 Mixer", "ADCR Switch", "ADC Right" }, + { "Port1_8 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port1_8 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port1_8 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port1_8 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_1 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_1 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_1 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_1 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_1 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_1 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "Port2_2 Mixer", "ADCL Switch", "ADC Left" }, + { "Port2_2 Mixer", "ADCR Switch", "ADC Right" }, + { "Port2_2 Mixer", "DMIC1L Switch", "DMIC1 Left" }, + { "Port2_2 Mixer", "DMIC1R Switch", "DMIC1 Right" }, + { "Port2_2 Mixer", "DMIC2L Switch", "DMIC2 Left" }, + { "Port2_2 Mixer", "DMIC2R Switch", "DMIC2 Right" }, + + { "P1_1_TX", NULL, "Port1_1 Mixer" }, + { "P1_2_TX", NULL, "Port1_2 Mixer" }, + { "P1_3_TX", NULL, "Port1_3 Mixer" }, + { "P1_4_TX", NULL, "Port1_4 Mixer" }, + { "P1_5_TX", NULL, "Port1_5 Mixer" }, + { "P1_6_TX", NULL, "Port1_6 Mixer" }, + { "P1_7_TX", NULL, "Port1_7 Mixer" }, + { "P1_8_TX", NULL, "Port1_8 Mixer" }, + + { "P2_1_TX", NULL, "Port2_1 Mixer" }, + { "P2_2_TX", NULL, "Port2_2 Mixer" }, + + { "PORT1_SDO", NULL, "P1_1_TX"}, + { "PORT1_SDO", NULL, "P1_2_TX"}, + { "PORT1_SDO", NULL, "P1_3_TX"}, + { "PORT1_SDO", NULL, "P1_4_TX"}, + { "PORT1_SDO", NULL, "P1_5_TX"}, + { "PORT1_SDO", NULL, "P1_6_TX"}, + { "PORT1_SDO", NULL, "P1_7_TX"}, + { "PORT1_SDO", NULL, "P1_8_TX"}, + + { "PORT2_SDO", NULL, "P2_1_TX"}, + { "PORT2_SDO", NULL, "P2_2_TX"}, + + { "Mic1 Input", NULL, "AMIC1" }, + { "Mic2 Input", NULL, "AMIC2" }, + + { "AUXL Input", NULL, "AUXL" }, + { "AUXR Input", NULL, "AUXR" }, + + /* AUX connections */ + { "ADCL Mux", "Aux_L", "AUXL Input" }, + { "ADCL Mux", "MIC1", "Mic1 Input" }, + + { "ADCR Mux", "Aux_R", "AUXR Input" }, + { "ADCR Mux", "MIC2", "Mic2 Input" }, + + /* ADC connection */ + { "ADC Left", NULL, "ADCL Mux"}, + { "ADC Right", NULL, "ADCR Mux"}, + + /* TX Paths */ + { "Port1 Capture", NULL, "P1_1_TX" }, + { "Port1 Capture", NULL, "P1_2_TX" }, + { "Port1 Capture", NULL, "P1_3_TX" }, + { "Port1 Capture", NULL, "P1_4_TX" }, + { "Port1 Capture", NULL, "P1_5_TX" }, + { "Port1 Capture", NULL, "P1_6_TX" }, + { "Port1 Capture", NULL, "P1_7_TX" }, + { "Port1 Capture", NULL, "P1_8_TX" }, + + { "Port2 Capture", NULL, "P2_1_TX" }, + { "Port2 Capture", NULL, "P2_2_TX" }, + + /* RX Paths */ + { "Port1 Playback", NULL, "P1_1_RX" }, + { "Port1 Playback", NULL, "P1_2_RX" }, + { "Port1 Playback", NULL, "P1_3_RX" }, + { "Port1 Playback", NULL, "P1_4_RX" }, + { "Port1 Playback", NULL, "P1_5_RX" }, + { "Port1 Playback", NULL, "P1_6_RX" }, + { "Port1 Playback", NULL, "P1_7_RX" }, + { "Port1 Playback", NULL, "P1_8_RX" }, + + { "Port2 Playback", NULL, "P2_1_RX" }, + { "Port2 Playback", NULL, "P2_2_RX" }, + + /* Sidetone map */ + { "Sidetone Mixer", "Sidetone Enable Switch", "ADC Left" }, + { "Sidetone Mixer", "Sidetone Enable Switch", "ADC Right" }, + { "Sidetone Mixer", "Sidetone Enable Switch", "DMIC1 Left" }, + { "Sidetone Mixer", "Sidetone Enable Switch", "DMIC1 Right" }, + { "Sidetone Mixer", "Sidetone Enable Switch", "DMIC2 Left" }, + { "Sidetone Mixer", "Sidetone Enable Switch", "DMIC2 Right" }, + + { "Sidetone", NULL, "Sidetone Mixer" }, +}; + +static int lm49453_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + struct pll_state *state; + unsigned int pwr_mask; + + switch (pll_id) { + case LM49453_PLL_1: + state = &lm49453->pll1; + pwr_mask = LM49453_PMC_SETUP_PLL_EN; + break; + case LM49453_PLL_2: + state = &lm49453->pll2; + pwr_mask = LM49453_PMC_SETUP_PLL_P2_EN; + break; + default: + return -EINVAL; + } + + state->in = freq_in; + state->out = freq_out; + + /* Always disable the PLL - it is not safe to leave it running + * while reprogramming it. + */ + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_PLL_EN, 0); + + if (!freq_in || !freq_out) + return 0; + + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQL_REG, (state->in & 0xff)); + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQH_REG, + ((state->in >> 8) & 0xff)); + + snd_soc_write(codec, LM49453_P0_VCO_TARGETLL_REG, + state->out & 0xff); + snd_soc_write(codec, LM49453_P0_VCO_TARGETLH_REG, + (state->out >> 8) & 0xff); + snd_soc_write(codec, LM49453_P0_VCO_TARGETHL_REG, + (state->out >> 16) & 0xff); + snd_soc_write(codec, LM49453_P0_VCO_TARGETHH_REG, + (state->out >> 24) & 0xff); + + /* All done, turn it on */ + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, pwr_mask, + pwr_mask); + + return 0; +} + +static int lm49453_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 clk_div = 0; + u16 pll_clk = 0; + + dev_dbg(codec->dev, "\n in lm49453_hw_params\n"); + + lm49453->fs_rate = params_rate(params); + + /* Setting DAC clock dividers based on substream sample rate. */ + switch (lm49453->fs_rate) { + case 8000: + case 16000: + case 32000: + case 24000: + case 48000: + clk_div = 256; + break; + case 11025: + case 22050: + case 44100: + clk_div = 216; + break; + case 96000: + clk_div = 127; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, clk_div); + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, clk_div); + + switch (lm49453->sysclk) { + case 12288000: + case 26000000: + case 19000000: + pll_clk = ~BIT(4); + break; + case 48000: + case 32576: + pll_clk = BIT(4); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), pll_clk); + + return 0; +} + +static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + int aif_val; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | + ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aif_val = ~LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + ~LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | + LM49453_AUDIO_PORT1_BASIC_SYNC_MS; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, + LM49453_AUDIO_PORT1_BASIC_FMT_MASK, aif_val); + + return 0; +} + +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 12288000: + case 26000000: + case 19200000: + case 48000: + case 32576: + lm49453->sysclk = freq; + return 0; + default: + return -EINVAL; + } +} + +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) +{ + if (mute) + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, + BIT(1)|BIT(0), BIT(1)|BIT(0)); + return 0; +} + +static int lm49453_lo_mute(struct snd_soc_dai *dai, int mute) +{ + if (mute) + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, + BIT(3)|BIT(2), BIT(3)|BIT(2)); + return 0; +} + +static int lm49453_ls_mute(struct snd_soc_dai *dai, int mute) +{ + if (mute) + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, + BIT(5)|BIT(4), BIT(5)|BIT(4)); + return 0; +} + +static int lm49453_ep_mute(struct snd_soc_dai *dai, int mute) +{ + if (mute) + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, + BIT(4), BIT(4)); + return 0; +} + +static int lm49453_ha_mute(struct snd_soc_dai *dai, int mute) +{ + if (mute) + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, + BIT(7)|BIT(6), BIT(7)|BIT(6)); + return 0; +} + +static int lm49453_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(lm49453->regmap); + break; + + case SND_SOC_BIAS_OFF: + reg = LM49453_PMC_SETUP_CHIP_EN; + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, + LM49453_PMC_SETUP_CHIP_EN, reg); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +/* Formates supported by LM49453 driver. */ +#define LM49453_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops lm49453_headset_dai_ops = { + .hw_params = lm49453_hw_params, + .set_sysclk = lm49453_set_dai_sysclk, + .set_pll = lm49453_set_dai_pll, + .set_fmt = lm49453_set_dai_fmt, + .digital_mute = lm49453_hp_mute, +}; + +static struct snd_soc_dai_ops lm49453_speaker_dai_ops = { + .digital_mute = lm49453_ls_mute, + .hw_params = lm49453_hw_params, +}; + +static struct snd_soc_dai_ops lm49453_haptic_dai_ops = { + .digital_mute = lm49453_ha_mute, + .hw_params = lm49453_hw_params, +}; + +static struct snd_soc_dai_ops lm49453_ep_dai_ops = { + .digital_mute = lm49453_ep_mute, + .hw_params = lm49453_hw_params, +}; + +static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { + .digital_mute = lm49453_lo_mute, + .hw_params = lm49453_hw_params, +}; + +/* LM49453 dai structure. */ +struct snd_soc_dai_driver lm49453_dai[] = { + { + .name = "LM49453 Headset", + .playback = { + .stream_name = "Headset", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 5, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_headset_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "LM49453 Speaker", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_speaker_dai_ops, + }, + { + .name = "LM49453 Haptic", + .playback = { + .stream_name = "Haptic", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_haptic_dai_ops, + }, + { + .name = "LM49453 Earpiece", + .playback = { + .stream_name = "Earpiece", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_ep_dai_ops, + }, + { + .name = "LM49453 line out", + .playback = { + .stream_name = "Lineout", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = LM49453_FORMATS, + }, + .ops = &lm49453_lineout_dai_ops, + }, +}; + +static int lm49453_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int lm49453_resume(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + + regcache_sync(lm49453->regmap); + + lm49453_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + lm49453_set_bias_level(codec, codec->dapm.suspend_bias_level); + + return 0; +} + +static int lm49453_probe(struct snd_soc_codec *codec) +{ + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + int reg; + + codec->control_data = lm49453->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Get the codec into a known state */ + reg = LM49453_RESET_REG_RST; + snd_soc_update_bits(codec, LM49453_P0_RESET_REG, BIT(0), reg); + if (ret != 0) { + dev_err(codec->dev, "Failed to reset codec: %d\n", ret); + return ret; + } + + return 0; +} + +/* power down chip */ +static int lm49453_remove(struct snd_soc_codec *codec) +{ + lm49453_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { + .probe = lm49453_probe, + .remove = lm49453_remove, + .suspend = lm49453_suspend, + .resume = lm49453_resume, + .set_bias_level = lm49453_set_bias_level, + .controls = lm49453_snd_controls, + .num_controls = ARRAY_SIZE(lm49453_snd_controls), + .dapm_widgets = lm49453_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm49453_dapm_widgets), + .dapm_routes = lm49453_audio_map, + .num_dapm_routes = ARRAY_SIZE(lm49453_audio_map), + .idle_bias_off = true, +}; + +static const struct regmap_config lm49453_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = LM49453_MAX_REGISTER, + .reg_defaults = lm49453_reg_defs, + .num_reg_defaults = ARRAY_SIZE(lm49453_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static __devinit int lm49453_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct lm49453_priv *lm49453; + int ret = 0; + + lm49453 = devm_kzalloc(&i2c->dev, sizeof(struct lm49453_priv), + GFP_KERNEL); + + if (lm49453 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, lm49453); + + lm49453->regmap = regmap_init_i2c(i2c, &lm49453_regmap_config); + if (IS_ERR(lm49453->regmap)) { + ret = PTR_ERR(lm49453->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_lm49453, + lm49453_dai, ARRAY_SIZE(lm49453_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(lm49453->regmap); + return ret; + } + + return ret; +} + +static int __devexit lm49453_i2c_remove(struct i2c_client *client) +{ + struct lm49453_priv *lm49453 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(lm49453->regmap); + return 0; +} + +static const struct i2c_device_id lm49453_i2c_id[] = { + { "lm49453", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, lm49453_i2c_id); + +static struct i2c_driver lm49453_i2c_driver = { + .driver = { + .name = "lm49453", + .owner = THIS_MODULE, + }, + .probe = lm49453_i2c_probe, + .remove = __devexit_p(lm49453_i2c_remove), + .id_table = lm49453_i2c_id, +}; + +module_i2c_driver(lm49453_i2c_driver); + +MODULE_DESCRIPTION("ASoC LM49453 driver"); +MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy(a)ti.com"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/lm49453.h b/sound/soc/codecs/lm49453.h new file mode 100644 index 0000000..093f626 --- /dev/null +++ b/sound/soc/codecs/lm49453.h @@ -0,0 +1,384 @@ +/* + * lm49453.h - LM49453 ALSA Soc Audio drive + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * 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; version 2 of the License. + * + */ + +#ifndef _LM49453_H +#define _LM49453_H + +#include <linux/bitops.h> + +#define LM49453_PLL_1 1 +#define LM49453_PLL_2 2 + +/* Page0 REGISTERS */ + +/* LM49453_P0 register space for page0 */ +#define LM49453_P0_PMC_SETUP_REG 0x00 +#define LM49453_P0_PLL_CLK_SEL1_REG 0x01 +#define LM49453_P0_PLL_CLK_SEL2_REG 0x02 +#define LM49453_P0_PMC_CLK_DIV_REG 0x03 +#define LM49453_P0_HSDET_CLK_DIV_REG 0x04 +#define LM49453_P0_DMIC_CLK_DIV_REG 0x05 +#define LM49453_P0_ADC_CLK_DIV_REG 0x06 +#define LM49453_P0_DAC_OT_CLK_DIV_REG 0x07 +#define LM49453_P0_PLL_HF_M_REG 0x08 +#define LM49453_P0_PLL_LF_M_REG 0x09 +#define LM49453_P0_PLL_NL_REG 0x0A +#define LM49453_P0_PLL_N_MODL_REG 0x0B +#define LM49453_P0_PLL_N_MODH_REG 0x0C +#define LM49453_P0_PLL_P1_REG 0x0D +#define LM49453_P0_PLL_P2_REG 0x0E +#define LM49453_P0_FLL_REF_FREQL_REG 0x0F +#define LM49453_P0_FLL_REF_FREQH_REG 0x10 +#define LM49453_P0_VCO_TARGETLL_REG 0x11 +#define LM49453_P0_VCO_TARGETLH_REG 0x12 +#define LM49453_P0_VCO_TARGETHL_REG 0x13 +#define LM49453_P0_VCO_TARGETHH_REG 0x14 +#define LM49453_P0_PLL_CONFIG_REG 0x15 +#define LM49453_P0_DAC_CLK_SEL_REG 0x16 +#define LM49453_P0_DAC_HP_CLK_DIV_REG 0x17 + +/* Analog Mixer Input Stages */ +#define LM49453_P0_MICL_REG 0x20 +#define LM49453_P0_MICR_REG 0x21 +#define LM49453_P0_EP_REG 0x24 +#define LM49453_P0_DIS_PKVL_FB_REG 0x25 + +/* Analog Mixer Output Stages */ +#define LM49453_P0_ANALOG_MIXER_ADC_REG 0x2E + +/*ADC or DAC */ +#define LM49453_P0_ADC_DSP_REG 0x30 +#define LM49453_P0_DAC_DSP_REG 0x31 + +/* EFFECTS ENABLES */ +#define LM49453_P0_ADC_FX_ENABLES_REG 0x33 + +/* GPIO */ +#define LM49453_P0_GPIO1_REG 0x38 +#define LM49453_P0_GPIO2_REG 0x39 +#define LM49453_P0_GPIO3_REG 0x3A +#define LM49453_P0_HAP_CTL_REG 0x3B +#define LM49453_P0_HAP_FREQ_PROG_LEFTL_REG 0x3C +#define LM49453_P0_HAP_FREQ_PROG_LEFTH_REG 0x3D +#define LM49453_P0_HAP_FREQ_PROG_RIGHTL_REG 0x3E +#define LM49453_P0_HAP_FREQ_PROG_RIGHTH_REG 0x3F + +/* DIGITAL MIXER */ +#define LM49453_P0_DMIX_CLK_SEL_REG 0x40 +#define LM49453_P0_PORT1_RX_LVL1_REG 0x41 +#define LM49453_P0_PORT1_RX_LVL2_REG 0x42 +#define LM49453_P0_PORT2_RX_LVL_REG 0x43 +#define LM49453_P0_PORT1_TX1_REG 0x44 +#define LM49453_P0_PORT1_TX2_REG 0x45 +#define LM49453_P0_PORT1_TX3_REG 0x46 +#define LM49453_P0_PORT1_TX4_REG 0x47 +#define LM49453_P0_PORT1_TX5_REG 0x48 +#define LM49453_P0_PORT1_TX6_REG 0x49 +#define LM49453_P0_PORT1_TX7_REG 0x4A +#define LM49453_P0_PORT1_TX8_REG 0x4B +#define LM49453_P0_PORT2_TX1_REG 0x4C +#define LM49453_P0_PORT2_TX2_REG 0x4D +#define LM49453_P0_STN_SEL_REG 0x4F +#define LM49453_P0_DACHPL1_REG 0x50 +#define LM49453_P0_DACHPL2_REG 0x51 +#define LM49453_P0_DACHPR1_REG 0x52 +#define LM49453_P0_DACHPR2_REG 0x53 +#define LM49453_P0_DACLOL1_REG 0x54 +#define LM49453_P0_DACLOL2_REG 0x55 +#define LM49453_P0_DACLOR1_REG 0x56 +#define LM49453_P0_DACLOR2_REG 0x57 +#define LM49453_P0_DACLSL1_REG 0x58 +#define LM49453_P0_DACLSL2_REG 0x59 +#define LM49453_P0_DACLSR1_REG 0x5A +#define LM49453_P0_DACLSR2_REG 0x5B +#define LM49453_P0_DACHAL1_REG 0x5C +#define LM49453_P0_DACHAL2_REG 0x5D +#define LM49453_P0_DACHAR1_REG 0x5E +#define LM49453_P0_DACHAR2_REG 0x5F + +/* AUDIO PORT 1 (TDM) */ +#define LM49453_P0_AUDIO_PORT1_BASIC_REG 0x60 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN1_REG 0x61 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN2_REG 0x62 +#define LM49453_P0_AUDIO_PORT1_CLK_GEN3_REG 0x63 +#define LM49453_P0_AUDIO_PORT1_SYNC_RATE_REG 0x64 +#define LM49453_P0_AUDIO_PORT1_SYNC_SDO_SETUP_REG 0x65 +#define LM49453_P0_AUDIO_PORT1_DATA_WIDTH_REG 0x66 +#define LM49453_P0_AUDIO_PORT1_RX_MSB_REG 0x67 +#define LM49453_P0_AUDIO_PORT1_TX_MSB_REG 0x68 +#define LM49453_P0_AUDIO_PORT1_TDM_CHANNELS_REG 0x69 + +/* AUDIO PORT 2 */ +#define LM49453_P0_AUDIO_PORT2_BASIC_REG 0x6A +#define LM49453_P0_AUDIO_PORT2_CLK_GEN1_REG 0x6B +#define LM49453_P0_AUDIO_PORT2_CLK_GEN2_REG 0x6C +#define LM49453_P0_AUDIO_PORT2_SYNC_GEN_REG 0x6D +#define LM49453_P0_AUDIO_PORT2_DATA_WIDTH_REG 0x6E +#define LM49453_P0_AUDIO_PORT2_RX_MODE_REG 0x6F +#define LM49453_P0_AUDIO_PORT2_TX_MODE_REG 0x70 + +/* SAMPLE RATE */ +#define LM49453_P0_PORT1_SR_LSB_REG 0x79 +#define LM49453_P0_PORT1_SR_MSB_REG 0x7A +#define LM49453_P0_PORT2_SR_LSB_REG 0x7B +#define LM49453_P0_PORT2_SR_MSB_REG 0x7C + +/* EFFECTS - HPFs */ +#define LM49453_P0_HPF_REG 0x80 + +/* EFFECTS ADC ALC */ +#define LM49453_P0_ADC_ALC1_REG 0x82 +#define LM49453_P0_ADC_ALC2_REG 0x83 +#define LM49453_P0_ADC_ALC3_REG 0x84 +#define LM49453_P0_ADC_ALC4_REG 0x85 +#define LM49453_P0_ADC_ALC5_REG 0x86 +#define LM49453_P0_ADC_ALC6_REG 0x87 +#define LM49453_P0_ADC_ALC7_REG 0x88 +#define LM49453_P0_ADC_ALC8_REG 0x89 +#define LM49453_P0_DMIC1_LEVELL_REG 0x8A +#define LM49453_P0_DMIC1_LEVELR_REG 0x8B +#define LM49453_P0_DMIC2_LEVELL_REG 0x8C +#define LM49453_P0_DMIC2_LEVELR_REG 0x8D +#define LM49453_P0_ADC_LEVELL_REG 0x8E +#define LM49453_P0_ADC_LEVELR_REG 0x8F +#define LM49453_P0_DAC_HP_LEVELL_REG 0x90 +#define LM49453_P0_DAC_HP_LEVELR_REG 0x91 +#define LM49453_P0_DAC_LO_LEVELL_REG 0x92 +#define LM49453_P0_DAC_LO_LEVELR_REG 0x93 +#define LM49453_P0_DAC_LS_LEVELL_REG 0x94 +#define LM49453_P0_DAC_LS_LEVELR_REG 0x95 +#define LM49453_P0_DAC_HA_LEVELL_REG 0x96 +#define LM49453_P0_DAC_HA_LEVELR_REG 0x97 +#define LM49453_P0_SOFT_MUTE_REG 0x98 +#define LM49453_P0_DMIC_MUTE_CFG_REG 0x99 +#define LM49453_P0_ADC_MUTE_CFG_REG 0x9A +#define LM49453_P0_DAC_MUTE_CFG_REG 0x9B + +/*DIGITAL MIC1 */ +#define LM49453_P0_DIGITAL_MIC1_CONFIG_REG 0xB0 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYL_REG 0xB1 +#define LM49453_P0_DIGITAL_MIC1_DATA_DELAYR_REG 0xB2 + +/*DIGITAL MIC2 */ +#define LM49453_P0_DIGITAL_MIC2_CONFIG_REG 0xB3 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYL_REG 0xB4 +#define LM49453_P0_DIGITAL_MIC2_DATA_DELAYR_REG 0xB5 + +/* ADC DECIMATOR */ +#define LM49453_P0_ADC_DECIMATOR_REG 0xB6 + +/* DAC CONFIGURE */ +#define LM49453_P0_DAC_CONFIG_REG 0xB7 + +/* SIDETONE */ +#define LM49453_P0_STN_VOL_ADCL_REG 0xB8 +#define LM49453_P0_STN_VOL_ADCR_REG 0xB9 +#define LM49453_P0_STN_VOL_DMIC1L_REG 0xBA +#define LM49453_P0_STN_VOL_DMIC1R_REG 0xBB +#define LM49453_P0_STN_VOL_DMIC2L_REG 0xBC +#define LM49453_P0_STN_VOL_DMIC2R_REG 0xBD + +/* ADC/DAC CLIPPING MONITORS (Read Only/Write to Clear) */ +#define LM49453_P0_ADC_DEC_CLIP_REG 0xC2 +#define LM49453_P0_ADC_HPF_CLIP_REG 0xC3 +#define LM49453_P0_ADC_LVL_CLIP_REG 0xC4 +#define LM49453_P0_DAC_LVL_CLIP_REG 0xC5 + +/* ADC ALC EFFECT MONITORS (Read Only) */ +#define LM49453_P0_ADC_LVLMONL_REG 0xC8 +#define LM49453_P0_ADC_LVLMONR_REG 0xC9 +#define LM49453_P0_ADC_ALCMONL_REG 0xCA +#define LM49453_P0_ADC_ALCMONR_REG 0xCB +#define LM49453_P0_ADC_MUTED_REG 0xCC +#define LM49453_P0_DAC_MUTED_REG 0xCD + +/* HEADSET DETECT */ +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITL_REG 0xD0 +#define LM49453_P0_HSD_PPB_LONG_CNT_LIMITR_REG 0xD1 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITL_REG 0xD2 +#define LM49453_P0_HSD_PIN3_4_EX_LOOP_CNT_LIMITH_REG 0xD3 +#define LM49453_P0_HSD_TIMEOUT1_REG 0xD4 +#define LM49453_P0_HSD_TIMEOUT2_REG 0xD5 +#define LM49453_P0_HSD_TIMEOUT3_REG 0xD6 +#define LM49453_P0_HSD_PIN3_4_CFG_REG 0xD7 +#define LM49453_P0_HSD_IRQ1_REG 0xD8 +#define LM49453_P0_HSD_IRQ2_REG 0xD9 +#define LM49453_P0_HSD_IRQ3_REG 0xDA +#define LM49453_P0_HSD_IRQ4_REG 0xDB +#define LM49453_P0_HSD_IRQ_MASK1_REG 0xDC +#define LM49453_P0_HSD_IRQ_MASK2_REG 0xDD +#define LM49453_P0_HSD_IRQ_MASK3_REG 0xDE +#define LM49453_P0_HSD_R_HPLL_REG 0xE0 +#define LM49453_P0_HSD_R_HPLH_REG 0xE1 +#define LM49453_P0_HSD_R_HPLU_REG 0xE2 +#define LM49453_P0_HSD_R_HPRL_REG 0xE3 +#define LM49453_P0_HSD_R_HPRH_REG 0xE4 +#define LM49453_P0_HSD_R_HPRU_REG 0xE5 +#define LM49453_P0_HSD_VEL_L_FINALL_REG 0xE6 +#define LM49453_P0_HSD_VEL_L_FINALH_REG 0xE7 +#define LM49453_P0_HSD_VEL_L_FINALU_REG 0xE8 +#define LM49453_P0_HSD_RO_FINALL_REG 0xE9 +#define LM49453_P0_HSD_RO_FINALH_REG 0xEA +#define LM49453_P0_HSD_RO_FINALU_REG 0xEB +#define LM49453_P0_HSD_VMIC_BIAS_FINALL_REG 0xEC +#define LM49453_P0_HSD_VMIC_BIAS_FINALH_REG 0xED +#define LM49453_P0_HSD_VMIC_BIAS_FINALU_REG 0xEE +#define LM49453_P0_HSD_PIN_CONFIG_REG 0xEF +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS1_REG 0xF1 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS2_REG 0xF2 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATUS3_REG 0xF3 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEL_REG 0xF4 +#define LM49453_P0_HSD_PLUG_DETECT_BB_IRQ_STATEH_REG 0xF5 + +/* I/O PULLDOWN CONFIG */ +#define LM49453_P0_PULL_CONFIG1_REG 0xF8 +#define LM49453_P0_PULL_CONFIG2_REG 0xF9 +#define LM49453_P0_PULL_CONFIG3_REG 0xFA + +/* RESET */ +#define LM49453_P0_RESET_REG 0xFE + +/* PAGE */ +#define LM49453_PAGE_REG 0xFF + +#define LM49453_MAX_REGISTER 0xFF + +/* LM49453_P0_PMC_SETUP_REG (0x00h) */ +#define LM49453_PMC_SETUP_CHIP_EN (BIT(1)|BIT(0)) +#define LM49453_PMC_SETUP_PLL_EN BIT(2) +#define LM49453_PMC_SETUP_PLL_P2_EN BIT(3) +#define LM49453_PMC_SETUP_PLL_FLL BIT(4) +#define LM49453_PMC_SETUP_MCLK_OVER BIT(5) +#define LM49453_PMC_SETUP_RTC_CLK_OVER BIT(6) +#define LM49453_PMC_SETUP_CHIP_ACTIVE BIT(7) + +/* Chip Enable bits */ +#define LM49453_CHIP_EN_SHUTDOWN 0x00 +#define LM49453_CHIP_EN_NO_HSD_DETECT 0x01 +#define LM49453_CHIP_EN_HSD_DETECT 0x02 +#define LM49453_CHIP_EN_INVALID_HSD 0x03 + +/* LM49453_P0_PLL_CLK_SEL1_REG (0x01h) */ +#define LM49453_CLK_SEL1_MCLK_SEL 0x11 +#define LM49453_CLK_SEL1_RTC_SEL 0x11 +#define LM49453_CLK_SEL1_PORT1_SEL 0x10 +#define LM49453_CLK_SEL1_PORT2_SEL 0x11 + +/* LM49453_P0_PLL_CLK_SEL2_REG (0x02h) */ +#define LM49453_CLK_SEL2_ADC_CLK_SEL 0x38 + +/* LM49453_P0_FLL_REF_FREQL_REG (0x0F) */ +#define LM49453_FLL_REF_FREQ_VAL 0x8ca0001 + +/* LM49453_P0_VCO_TARGETLL_REG (0x11) */ +#define LM49453_VCO_TARGET_VAL 0x8ca0001 + +/* LM49453_P0_ADC_DSP_REG (0x30h) */ +#define LM49453_ADC_DSP_ADC_MUTEL BIT(0) +#define LM49453_ADC_DSP_ADC_MUTER BIT(1) +#define LM49453_ADC_DSP_DMIC1_MUTEL BIT(2) +#define LM49453_ADC_DSP_DMIC1_MUTER BIT(3) +#define LM49453_ADC_DSP_DMIC2_MUTEL BIT(4) +#define LM49453_ADC_DSP_DMIC2_MUTER BIT(5) +#define LM49453_ADC_DSP_MUTE_ALL 0x3F + +/* LM49453_P0_DAC_DSP_REG (0x31h) */ +#define LM49453_DAC_DSP_MUTE_ALL 0xFF + +/* LM49453_P0_AUDIO_PORT1_BASIC_REG (0x60h) */ +#define LM49453_AUDIO_PORT1_BASIC_FMT_MASK (BIT(4)|BIT(3)) +#define LM49453_AUDIO_PORT1_BASIC_CLK_MS BIT(3) +#define LM49453_AUDIO_PORT1_BASIC_SYNC_MS BIT(4) + +/* LM49453_P0_RESET_REG (0xFEh) */ +#define LM49453_RESET_REG_RST BIT(0) + +/* Page select register bits (0xFF) */ +#define LM49453_PAGE0_SELECT 0x0 +#define LM49453_PAGE1_SELECT 0x1 + +/* LM49453_P0_HSD_PIN3_4_CFG_REG (Jack Pin config - 0xD7) */ +#define LM49453_JACK_DISABLE 0x00 +#define LM49453_JACK_CONFIG1 0x01 +#define LM49453_JACK_CONFIG2 0x02 +#define LM49453_JACK_CONFIG3 0x03 +#define LM49453_JACK_CONFIG4 0x04 +#define LM49453_JACK_CONFIG5 0x05 + +/* Page 1 REGISTERS */ + +/* SIDETONE */ +#define LM49453_P1_SIDETONE_SA0L_REG 0x80 +#define LM49453_P1_SIDETONE_SA0H_REG 0x81 +#define LM49453_P1_SIDETONE_SAB0U_REG 0x82 +#define LM49453_P1_SIDETONE_SB0L_REG 0x83 +#define LM49453_P1_SIDETONE_SB0H_REG 0x84 +#define LM49453_P1_SIDETONE_SH0L_REG 0x85 +#define LM49453_P1_SIDETONE_SH0H_REG 0x86 +#define LM49453_P1_SIDETONE_SH0U_REG 0x87 +#define LM49453_P1_SIDETONE_SA1L_REG 0x88 +#define LM49453_P1_SIDETONE_SA1H_REG 0x89 +#define LM49453_P1_SIDETONE_SAB1U_REG 0x8A +#define LM49453_P1_SIDETONE_SB1L_REG 0x8B +#define LM49453_P1_SIDETONE_SB1H_REG 0x8C +#define LM49453_P1_SIDETONE_SH1L_REG 0x8D +#define LM49453_P1_SIDETONE_SH1H_REG 0x8E +#define LM49453_P1_SIDETONE_SH1U_REG 0x8F +#define LM49453_P1_SIDETONE_SA2L_REG 0x90 +#define LM49453_P1_SIDETONE_SA2H_REG 0x91 +#define LM49453_P1_SIDETONE_SAB2U_REG 0x92 +#define LM49453_P1_SIDETONE_SB2L_REG 0x93 +#define LM49453_P1_SIDETONE_SB2H_REG 0x94 +#define LM49453_P1_SIDETONE_SH2L_REG 0x95 +#define LM49453_P1_SIDETONE_SH2H_REG 0x96 +#define LM49453_P1_SIDETONE_SH2U_REG 0x97 +#define LM49453_P1_SIDETONE_SA3L_REG 0x98 +#define LM49453_P1_SIDETONE_SA3H_REG 0x99 +#define LM49453_P1_SIDETONE_SAB3U_REG 0x9A +#define LM49453_P1_SIDETONE_SB3L_REG 0x9B +#define LM49453_P1_SIDETONE_SB3H_REG 0x9C +#define LM49453_P1_SIDETONE_SH3L_REG 0x9D +#define LM49453_P1_SIDETONE_SH3H_REG 0x9E +#define LM49453_P1_SIDETONE_SH3U_REG 0x9F +#define LM49453_P1_SIDETONE_SA4L_REG 0xA0 +#define LM49453_P1_SIDETONE_SA4H_REG 0xA1 +#define LM49453_P1_SIDETONE_SAB4U_REG 0xA2 +#define LM49453_P1_SIDETONE_SB4L_REG 0xA3 +#define LM49453_P1_SIDETONE_SB4H_REG 0xA4 +#define LM49453_P1_SIDETONE_SH4L_REG 0xA5 +#define LM49453_P1_SIDETONE_SH4H_REG 0xA6 +#define LM49453_P1_SIDETONE_SH4U_REG 0xA7 +#define LM49453_P1_SIDETONE_SA5L_REG 0xA8 +#define LM49453_P1_SIDETONE_SA5H_REG 0xA9 +#define LM49453_P1_SIDETONE_SAB5U_REG 0xAA +#define LM49453_P1_SIDETONE_SB5L_REG 0xAB +#define LM49453_P1_SIDETONE_SB5H_REG 0xAC +#define LM49453_P1_SIDETONE_SH5L_REG 0xAD +#define LM49453_P1_SIDETONE_SH5H_REG 0xAE +#define LM49453_P1_SIDETONE_SH5U_REG 0xAF + +/* CHARGE PUMP CONFIG */ +#define LM49453_P1_CP_CONFIG1_REG 0xB0 +#define LM49453_P1_CP_CONFIG2_REG 0xB1 +#define LM49453_P1_CP_CONFIG3_REG 0xB2 +#define LM49453_P1_CP_CONFIG4_REG 0xB3 +#define LM49453_P1_CP_LA_VTH1L_REG 0xB4 +#define LM49453_P1_CP_LA_VTH1M_REG 0xB5 +#define LM49453_P1_CP_LA_VTH2L_REG 0xB6 +#define LM49453_P1_CP_LA_VTH2M_REG 0xB7 +#define LM49453_P1_CP_LA_VTH3L_REG 0xB8 +#define LM49453_P1_CP_LA_VTH3H_REG 0xB9 +#define LM49453_P1_CP_CLK_DIV_REG 0xBA + +/* DAC */ +#define LM49453_P1_DAC_CHOP_REG 0xC0 + +#endif

2 1

[alsa-devel] [PATCH 3/5] ASoC: Tegra+ALC5632 machine: Add device tree binding documentation
by Leon Romanovsky 31 Jan '12

31 Jan '12

Document device tree binding for the tegra board with ALC5632 codec according to datasheet functional block description. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- .../bindings/sound/tegra-audio-alc5632.txt | 57 ++++++++++++++++++++ 1 files changed, 57 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt new file mode 100644 index 0000000..300f2da --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt @@ -0,0 +1,57 @@ +NVIDIA Tegra audio complex + +Required properties: +- compatible : "nvidia,tegra-audio-alc5632" +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the ALC5632's pins: + + ALC5632 pins: + + * SPK_OUTP + * SPK_OUTN + * HP_OUT_L + * HP_OUT_R + * AUX_OUT_P + * AUX_OUT_N + * LINE_IN_L + * LINE_IN_R + * PHONE_P + * PHONE_N + * MIC1_P + * MIC1_N + * MIC2_P + * MIC2_N + * MICBIAS1 + + Board connectors: + + * Headset Stereophone + * Int Spk + * Headset Mic + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller +- nvidia,audio-codec : The phandle of the ALC5632 audio codec + +Example: + +sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = + "Int Spk", "SPK_OUTP", + "Int Spk", "SPK_OUTN", + "Headset Mic","MICBIAS1", + "MIC1_N", "Headset Mic", + "MIC1_P", "Headset Mic", + "Headset Stereophone", "HP_OUT_R", + "Headset Stereophone", "HP_OUT_L"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&alc5632>; +}; -- 1.7.3.4

3 6

[alsa-devel] [PATCH 2/5] ASoC: ALC5632: Add device tree binding documentation
by Leon Romanovsky 31 Jan '12

31 Jan '12

Document the device tree binding for the ALC5632 codec and update vendor specific prefix for the Realtek. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- .../devicetree/bindings/sound/alc5632.txt | 24 ++++++++++++++++++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + 2 files changed, 25 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt new file mode 100644 index 0000000..8608f74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -0,0 +1,24 @@ +ALC5632 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "realtek,alc5632" + + - reg : the I2C address of the device. + + - gpio-controller : Indicates this device is a GPIO controller. + + - #gpio-cells : Should be two. The first cell is the pin number and the + second cell is used to specify optional parameters (currently unused). + +Example: + +alc5632: alc5632@1e { + compatible = "realtek,alc5632"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; +}; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index ecc6a6c..ca8ee14 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -33,6 +33,7 @@ nxp NXP Semiconductors powervr Imagination Technologies qcom Qualcomm, Inc. ramtron Ramtron International +realtek Realtek Semiconductor Corp. samsung Samsung Semiconductor sbs Smart Battery System schindler Schindler -- 1.7.3.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: Tegra+ALC5632: Implement device tree support in board file
by Leon Romanovsky 31 Jan '12

31 Jan '12

This patch implements device tree support for tagra board with ALC5632 codec. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/tegra/tegra_alc5632.c | 78 ++++++++++++++++++++++++++++++++++---- 1 files changed, 69 insertions(+), 9 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 4a0e805..c0ba1e4 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -36,6 +36,7 @@ struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; + struct platform_device *pcm_dev; }; static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, @@ -128,9 +129,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", - .codec_name = "alc5632.0-001e", .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "alc5632-hifi", .init = tegra_alc5632_asoc_init, .ops = &tegra_alc5632_asoc_ops, @@ -163,26 +162,78 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) sizeof(struct tegra_alc5632), GFP_KERNEL); if (!alc5632) { dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err; } - ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); - if (ret) - return ret; - card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); + alc5632->pcm_dev = ERR_PTR(-EINVAL); + + if (!(pdev->dev.of_node)) { + dev_err(&pdev->dev, "Must be instantiated using device tree\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_alc5632_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + + if (!tegra_alc5632_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_alc5632_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + alc5632->pcm_dev = platform_device_register_simple( + "tegra-pcm-audio", -1, NULL, 0); + if (IS_ERR(alc5632->pcm_dev)) { + dev_err(&pdev->dev, + "Can't instantiate tegra-pcm-audio\n"); + ret = PTR_ERR(alc5632->pcm_dev); + goto err; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + goto err_unregister; + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - tegra_asoc_utils_fini(&alc5632->util_data); - return ret; + goto err_fini_utils; } return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&alc5632->util_data); +err_unregister: + if (!IS_ERR(alc5632->pcm_dev)) + platform_device_unregister(alc5632->pcm_dev); +err: + return ret; } static int __devexit tegra_alc5632_remove(struct platform_device *pdev) @@ -193,15 +244,23 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&alc5632->util_data); + if (!IS_ERR(alc5632->pcm_dev)) + platform_device_unregister(alc5632->pcm_dev); return 0; } +static const struct of_device_id tegra_alc5632_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-alc5632", }, + {}, +}; + static struct platform_driver tegra_alc5632_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = tegra_alc5632_of_match, }, .probe = tegra_alc5632_probe, .remove = __devexit_p(tegra_alc5632_remove), @@ -212,3 +271,4 @@ MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_alc5632_of_match); -- 1.7.3.4

3 2

[alsa-devel] [PATCH] ASoC: wm_hubs: Push check for idle_bias_off out into drivers
by Mark Brown 31 Jan '12

31 Jan '12

For later wm_hubs devices we have much less need to keep the biases up even when using single ended line outputs so flag idle_bias_off for everything except the WM8993 and WM8994. --- sound/soc/codecs/wm8993.c | 6 ++++++ sound/soc/codecs/wm8994.c | 8 ++++++++ sound/soc/codecs/wm_hubs.c | 6 ------ 3 files changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 474dc72..db51007 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1622,6 +1622,12 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, wm8993->pdata.lineout2_diff); + /* If the line outputs are differential then we aren't presenting + * VMID as an output and can disable it. + */ + if (wm8993->pdata.lineout1_diff && wm8993->pdata.lineout2_diff) + codec->dapm.idle_bias_off = 1; + return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ca2db79..09641d7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3368,10 +3368,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); + /* By default use idle_bias_off, will override for WM8994 */ + codec->dapm.idle_bias_off = 1; + /* Set revision-specific configuration */ wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION); switch (control->type) { case WM8994: + /* Single ended line outputs should have VMID on. */ + if (!wm8994->pdata->lineout1_diff || + !wm8994->pdata->lineout2_diff) + codec->dapm.idle_bias_off = 0; + switch (wm8994->revision) { case 2: case 3: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c1a3f8c..57514dd2 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -958,12 +958,6 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); - /* If the line outputs are differential then we aren't presenting - * VMID as an output and can disable it. - */ - if (lineout1_diff && lineout2_diff) - codec->dapm.idle_bias_off = 1; - if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: core: Better support for idle_bias_off suspend ignores
by Mark Brown 31 Jan '12

31 Jan '12

If an idle_bias_off device is in any state other than off then it is still active for some reason (typically a low power function such as accessory detection). This wasn't an issue when the feature was implemented as we always went to _ON for any active function, subsequent power improvements have changed things. With the modern way of doing things we should overhaul the infrastructure to allow devices to explicitly take references for these functions but that's a much more invasive change and will require driver updates to deploy, this will bring the framework into line with the existing driver set before we do that work. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 11 +++++++++++ 1 files changed, 11 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 53fd9d2..09fa8a6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -567,6 +567,17 @@ int snd_soc_suspend(struct device *dev) if (!codec->suspended && codec->driver->suspend) { switch (codec->dapm.bias_level) { case SND_SOC_BIAS_STANDBY: + /* + * If the CODEC is capable of idle + * bias off then being in STANDBY + * means it's doing something, + * otherwise fall through. + */ + if (codec->dapm.idle_bias_off) { + dev_dbg(codec->dev, + "idle_bias_off CODEC on over suspend\n"); + break; + } case SND_SOC_BIAS_OFF: codec->driver->suspend(codec); codec->suspended = 1; -- 1.7.9.rc1

2 1

[alsa-devel] [PATCH] ASoC: core: Support suspend to disk
by Mark Brown 31 Jan '12

31 Jan '12

Use the same pm_ops for all system suspend and resume paths. This isn't ideal for suspend to disk with older CODECs as we'll suspend and then resume the CODEC before powering off all of which takes a long time due to VMID ramps but it's very simple to implement and for modern CODECs the overhead should be minimal. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e25d69c..53fd9d2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1663,8 +1663,7 @@ int snd_soc_poweroff(struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_poweroff); const struct dev_pm_ops snd_soc_pm_ops = { - .suspend = snd_soc_suspend, - .resume = snd_soc_resume, + SET_SYSTEM_SLEEP_PM_OPS(snd_soc_suspend, snd_soc_resume) .poweroff = snd_soc_poweroff, }; EXPORT_SYMBOL_GPL(snd_soc_pm_ops); -- 1.7.9.rc1

2 1

[alsa-devel] [PATCH] ASoC: wm8993: Convert to use a regmap patch
by Mark Brown 31 Jan '12

31 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 18 +++++++++++++----- 1 files changed, 13 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index dd687c3..474dc72 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1077,11 +1077,6 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(wm8993->regmap, false); regcache_sync(wm8993->regmap); - /* Tune DC servo configuration */ - snd_soc_write(codec, 0x44, 3); - snd_soc_write(codec, 0x56, 3); - snd_soc_write(codec, 0x44, 0); - /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | @@ -1691,6 +1686,13 @@ static int wm8993_resume(struct snd_soc_codec *codec) #define wm8993_resume NULL #endif +/* Tune DC servo configuration */ +static struct reg_default wm8993_regmap_patch[] = { + { 0x44, 3 }, + { 0x56, 3 }, + { 0x44, 0 }, +}; + static const struct regmap_config wm8993_regmap = { .reg_bits = 8, .val_bits = 16, @@ -1769,6 +1771,12 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_enable; + ret = regmap_register_patch(wm8993->regmap, wm8993_regmap_patch, + ARRAY_SIZE(wm8993_regmap_patch)); + if (ret != 0) + dev_warn(wm8993->dev, "Failed to apply regmap patch: %d\n", + ret); + if (i2c->irq) { /* Put GPIO1 into interrupt mode (only GPIO1 can output IRQ) */ ret = regmap_update_bits(wm8993->regmap, WM8993_GPIO1, -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: wm8994: Remove ASoC level register cache sync
by Mark Brown 31 Jan '12

31 Jan '12

Now we've switched over to regmap the ASoC level cache sync will be ineffectual and potentially harmful as there is no longer an ASoC level cache. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ae05e29..ca2db79 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2736,11 +2736,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Toshiba Qosmio G50
by David Henningsson 31 Jan '12

31 Jan '12

The user reports that model=auto works better than current handling on a 3.2 based kernel (with jack detection patches backported). Since model=auto is what we prefer these days anyway, the quirk should be removed. Alsa-info for the relevant machine: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/923316/+attachment/270… BugLink: https://bugs.launchpad.net/bugs/923316 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 ------------- 1 files changed, 0 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42b6a01..a8e82be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4747,7 +4747,6 @@ enum { ALC262_FIXUP_FSC_H270, ALC262_FIXUP_HP_Z200, ALC262_FIXUP_TYAN, - ALC262_FIXUP_TOSHIBA_RX1, ALC262_FIXUP_LENOVO_3000, ALC262_FIXUP_BENQ, ALC262_FIXUP_BENQ_T31, @@ -4777,16 +4776,6 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, - [ALC262_FIXUP_TOSHIBA_RX1] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x90170110 }, /* speaker */ - { 0x15, 0x0421101f }, /* HP */ - { 0x1a, 0x40f000f0 }, /* N/A */ - { 0x1b, 0x40f000f0 }, /* N/A */ - { 0x1e, 0x40f000f0 }, /* N/A */ - } - }, [ALC262_FIXUP_LENOVO_3000] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -4819,8 +4808,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ), SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_FIXUP_TOSHIBA_RX1), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ), -- 1.7.8.3

2 1

[alsa-devel] [PATCH] ASoC: wm_hubs: Fix routing of input PGAs to line output mixer
by Mark Brown 31 Jan '12

31 Jan '12

IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1 and IN2 to LINEOUT2. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ea26724..c1a3f8c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -586,8 +586,8 @@ SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0), }; static const struct snd_kcontrol_new line2_mix[] = { -SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0), -SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0), +SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER2, 2, 1, 0), +SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER2, 1, 1, 0), SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0), }; @@ -848,8 +848,8 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = { }; static const struct snd_soc_dapm_route lineout2_diff_routes[] = { - { "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" }, - { "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" }, + { "LINEOUT2 Mixer", "IN1L Switch", "IN1L PGA" }, + { "LINEOUT2 Mixer", "IN1R Switch", "IN1R PGA" }, { "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" }, -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: max9768: Fix up review comments in max9768 driver
by Mark Brown 31 Jan '12

31 Jan '12

The Mute Switch should be called Playback Switch to match the volume and the regmap core has been updated so we don't need to mark all the registers as non-volatile. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/max9768.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 6348995..79e9901 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -115,7 +115,7 @@ static const struct snd_kcontrol_new max9768_volume[] = { }; static const struct snd_kcontrol_new max9768_mute[] = { - SOC_SINGLE_BOOL_EXT("Mute Switch", 0, max9768_get_gpio, max9768_set_gpio), + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; static int max9768_probe(struct snd_soc_codec *codec) @@ -150,18 +150,12 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .num_controls = ARRAY_SIZE(max9768_volume), }; -static bool max9768_always_false(struct device *dev, unsigned int reg) -{ - return false; -} - static const struct regmap_config max9768_i2c_regmap_config = { .reg_bits = 2, .val_bits = 6, .max_register = 3, .reg_defaults = max9768_default_regs, .num_reg_defaults = ARRAY_SIZE(max9768_default_regs), - .volatile_reg = max9768_always_false, .cache_type = REGCACHE_RBTREE, }; -- 1.7.9.rc1

2 1

[alsa-devel] [PATCH 1/3] regmap: Properly round reg_bytes and val_bytes
by Wolfram Sang 30 Jan '12

30 Jan '12

For the upcoming 2/6-format, we don't see debugfs output otherwise, since the current division results in 0. I'd think 10/14 is broken currently, too. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- drivers/base/regmap/regmap.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 6555803..8bd0232 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -159,8 +159,8 @@ struct regmap *regmap_init(struct device *dev, mutex_init(&map->lock); map->format.buf_size = (config->reg_bits + config->val_bits) / 8; - map->format.reg_bytes = config->reg_bits / 8; - map->format.val_bytes = config->val_bits / 8; + map->format.reg_bytes = DIV_ROUND_UP(config->reg_bits, 8); + map->format.val_bytes = DIV_ROUND_UP(config->val_bits, 8); map->dev = dev; map->bus = bus; map->max_register = config->max_register; -- 1.7.8.3

2 10

[alsa-devel] [PATCH -next] sound: fix include/sound/core.h warning
by Randy Dunlap 30 Jan '12

30 Jan '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix build warning by providing a struct stub since no fields of the struct are used: include/sound/core.h:454:56: warning: 'struct pci_dev' declared inside parameter list Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- include/sound/core.h | 2 ++ 1 file changed, 2 insertions(+) --- linux-next-20120130.orig/include/sound/core.h +++ linux-next-20120130/include/sound/core.h @@ -450,6 +450,8 @@ struct snd_pci_quirk { {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)} #endif +struct pci_dev; + const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list);

2 1

[alsa-devel] ASoC updates for 3.3
by Mark Brown 30 Jan '12

30 Jan '12

The following changes since commit c83f1d7e71625801c72f4013291194e09b6f0a6e: ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on error (2012-01-24 11:34:19 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.3 for you to fetch changes up to 1ae5cbc52e7c6619a3f44b87809fd25370df31bb: ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine (2012-01-30 13:22:13 +0000) ---------------------------------------------------------------- A bunch of small driver specific fixes - nothing terribly exciting here, all of this will only affect people using particular devices and then usually only in some use cases. ---------------------------------------------------------------- Denis 'GNUtoo' Carikli (1): ASoC: neo1973_wm8753: remove references to the neo1973-gta01 machine Mark Brown (6): ASoC: wm5100: Make sure we switch to button reporting mode ASoC: wm5100: Fix microphone configuration ASoC: wm8996: Mark register cache as dirty when regulators are disabled ASoC: wm8962: Mark register cache as dirty when regulators are disabled ASoC: wm5100: Mark register cache as dirty when regulators are disabled ASoC: wm_hubs: Enable line out VMID buffer for single ended line outputs UK KIM (1): ASoC: wm_hubs: fix wrong bits for LINEOUT2 N/P mixer sound/soc/codecs/wm5100.c | 12 ++++++- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm_hubs.c | 10 ++++- sound/soc/samsung/neo1973_wm8753.c | 65 +----------------------------------- 5 files changed, 22 insertions(+), 69 deletions(-)

2 3

[alsa-devel] [PATCH 0/3] Support U8500/AB8500 ASoC
by Ola Lilja 30 Jan '12

30 Jan '12

First try to upload the ASoC-driver for ST-Ericsson U8500-platform. The audio system in the Ux500 platform consists of an external mixed signal ASIC called AB8500: http://www.stericsson.com/developers/CD00291561_UM1031_AB8500_user_manual-r… The audio portions of the AB8500 are configured over I2C and the raw data streams are sent using I2S/TDM. For this reason the I2S/TDM drivers for the DB8500 chip are included in this submission. Two of the patches mainly concern adding proper DMA channels and regulators to the Ux500 platform and may be treated individually, but preferably all patches shall be merged through the ASoC tree. Since we have more codecs for related platforms named U5500 (etc) the soc directory is name ux500. Ola Lilja (3): ARM: ux500: Add placeholder for clk_set_parent ARM: ux500: Add DMA-channels for MSP ASoC: Ux500: Add driver for Ux500/AB8500 platform/codec arch/arm/mach-ux500/Makefile | 3 +- arch/arm/mach-ux500/board-mop500-msp.c | 195 ++ arch/arm/mach-ux500/board-mop500-regulators.c | 28 + arch/arm/mach-ux500/board-mop500.c | 7 +- arch/arm/mach-ux500/board-mop500.h | 1 + arch/arm/mach-ux500/clock.c | 15 +- arch/arm/mach-ux500/clock.h | 1 + arch/arm/mach-ux500/devices-common.h | 2 +- arch/arm/mach-ux500/devices-db8500.c | 6 + arch/arm/mach-ux500/include/mach/hardware.h | 2 + arch/arm/mach-ux500/include/mach/msp.h | 1030 +++++++++ include/sound/ux500_ab8500.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/Makefile.rej | 10 + sound/soc/codecs/ab8500_audio.c | 2928 +++++++++++++++++++++++++ sound/soc/codecs/ab8500_audio.h | 673 ++++++ sound/soc/ux500/Kconfig | 33 + sound/soc/ux500/Makefile | 25 + sound/soc/ux500/u8500.c | 150 ++ sound/soc/ux500/ux500_ab8500.c | 790 +++++++ sound/soc/ux500/ux500_msp_dai.c | 998 +++++++++ sound/soc/ux500/ux500_msp_dai.h | 83 + sound/soc/ux500/ux500_msp_i2s.c | 1004 +++++++++ sound/soc/ux500/ux500_msp_i2s.h | 40 + sound/soc/ux500/ux500_pcm.c | 428 ++++ sound/soc/ux500/ux500_pcm.h | 44 + 29 files changed, 8535 insertions(+), 9 deletions(-) create mode 100644 arch/arm/mach-ux500/board-mop500-msp.c create mode 100644 arch/arm/mach-ux500/include/mach/msp.h create mode 100644 include/sound/ux500_ab8500.h create mode 100644 sound/soc/codecs/Makefile.rej create mode 100644 sound/soc/codecs/ab8500_audio.c create mode 100644 sound/soc/codecs/ab8500_audio.h create mode 100644 sound/soc/ux500/Kconfig create mode 100644 sound/soc/ux500/Makefile create mode 100644 sound/soc/ux500/u8500.c create mode 100644 sound/soc/ux500/ux500_ab8500.c create mode 100644 sound/soc/ux500/ux500_msp_dai.c create mode 100644 sound/soc/ux500/ux500_msp_dai.h create mode 100644 sound/soc/ux500/ux500_msp_i2s.c create mode 100644 sound/soc/ux500/ux500_msp_i2s.h create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h -- 1.7.8.3

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[alsa-devel] [Patch v2] ASOC: neo1973_wm8753: remove references to the neo1973-gta01 machine
by Denis 'GNUtoo' Carikli 30 Jan '12

30 Jan '12

The Openmoko GTA01 machine has been removed from the machine ID database, so we need to remove references to it as well. Without that fix we have: sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’: sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’ Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo(a)no-log.org> --- sound/soc/samsung/neo1973_wm8753.c | 65 +----------------------------------- 1 files changed, 1 insertions(+), 64 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 1bf0c32..d23b19a 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -230,8 +230,6 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { /* GTA02 specific routes and controls */ -#ifdef CONFIG_MACH_NEO1973_GTA02 - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -311,10 +309,6 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) return 0; } -#else -static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; } -#endif - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; @@ -322,10 +316,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) int ret; /* set up NC codec pins */ - if (machine_is_neo1973_gta01()) { - snd_soc_dapm_nc_pin(dapm, "LOUT2"); - snd_soc_dapm_nc_pin(dapm, "ROUT2"); - } snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "OUT4"); snd_soc_dapm_nc_pin(dapm, "LINE1"); @@ -370,50 +360,6 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return 0; } -/* GTA01 specific controls */ - -#ifdef CONFIG_MACH_NEO1973_GTA01 - -static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = { - {"Amp IN", NULL, "ROUT1"}, - {"Amp IN", NULL, "LOUT1"}, - - {"Handset Spk", NULL, "Amp EP"}, - {"Stereo Out", NULL, "Amp LS"}, - {"Headphone", NULL, "Amp HP"}, -}; - -static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), -}; - -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) -{ - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets, - ARRAY_SIZE(neo1973_lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes, - ARRAY_SIZE(neo1973_lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Headphone"); - - return 0; -} - -#else -static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; }; -#endif - static struct snd_soc_dai_link neo1973_dai[] = { { /* Hifi Playback - for similatious use with voice below */ .name = "WM8753", @@ -440,11 +386,6 @@ static struct snd_soc_aux_dev neo1973_aux_devs[] = { .name = "dfbmcs320", .codec_name = "dfbmcs320.0", }, - { - .name = "lm4857", - .codec_name = "lm4857.0-007c", - .init = neo1973_lm4857_init, - }, }; static struct snd_soc_codec_conf neo1973_codec_conf[] = { @@ -454,14 +395,10 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -#ifdef CONFIG_MACH_NEO1973_GTA02 static const struct gpio neo1973_gta02_gpios[] = { { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, }; -#else -static const struct gpio neo1973_gta02_gpios[] = {}; -#endif static struct snd_soc_card neo1973 = { .name = "neo1973", @@ -480,7 +417,7 @@ static int __init neo1973_init(void) { int ret; - if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02()) + if (!machine_is_neo1973_gta02()) return -ENODEV; if (machine_is_neo1973_gta02()) { -- 1.7.4.1

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[alsa-devel] [PATCH v2] ALSA: Add #ifdef CONFIG_PCI guard for snd_pci_quirk_* functions
by Axel Lin 30 Jan '12

30 Jan '12

This fixes below build warning when CONFIG_PCI is not set. CC sound/sound_core.o In file included from sound/sound_core.c:15: include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: begin the ifdef including the definition of struct snd_pci_quirk. include/sound/core.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 5ab255f..cea1b54 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -417,6 +417,7 @@ static inline int __snd_bug_on(int cond) #define gameport_get_port_data(gp) (gp)->port_data #endif +#ifdef CONFIG_PCI /* PCI quirk list helper */ struct snd_pci_quirk { unsigned short subvendor; /* PCI subvendor ID */ @@ -456,5 +457,6 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); const struct snd_pci_quirk * snd_pci_quirk_lookup_id(u16 vendor, u16 device, const struct snd_pci_quirk *list); +#endif #endif /* __SOUND_CORE_H */ -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ALSA: Add #ifdef CONFIG_PCI guard for snd_pci_quirk_lookup* functions
by Axel Lin 30 Jan '12

30 Jan '12

This fixes below build warning when CONFIG_PCI is not set. CC sound/sound_core.o In file included from sound/sound_core.c:15: include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- include/sound/core.h | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 5ab255f..fce4b6f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -450,11 +450,13 @@ struct snd_pci_quirk { {_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)} #endif +#ifdef CONFIG_PCI const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); const struct snd_pci_quirk * snd_pci_quirk_lookup_id(u16 vendor, u16 device, const struct snd_pci_quirk *list); +#endif #endif /* __SOUND_CORE_H */ -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: Include linux/pci.h in sound/core.h
by Axel Lin 30 Jan '12

30 Jan '12

commit d1458279b "ALSA: Add snd_pci_quirk_lookup_id()" introduces below build waring: CC sound/sound_core.o In file included from sound/sound_core.c:15: include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want Include linux/pci.h to fix the build warning. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- include/sound/core.h | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 5ab255f..4ea14f2 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -28,6 +28,7 @@ #include <linux/pm.h> /* pm_message_t */ #include <linux/device.h> #include <linux/stringify.h> +#include <linux/pci.h> /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS -- 1.7.5.4

1 1

[alsa-devel] question application playback
by Ricardo Barbosa 30 Jan '12

30 Jan '12

Hi all. I am new to development using the alsa API and follow the link code http://equalarea.com/paul/alsa-audio.html insert the following code. FILE *fp; fp = fopen("file.mp3", "r"); fclose(fp); and changed variable buf by the variable fp, but by running the file file.mp3 he does not play according to the way it interferes with the function snd_pcm_writei ? someone could help me in developing this playback, I thank Regards.

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[alsa-devel] kernel panic with gta02 sound driver
by Denis 'GNUtoo' Carikli 30 Jan '12

30 Jan '12

hi, The patch contained in that mail fixes the dai link of the gta02 machine, it worked (better) on a 3.2.1 kernel (it didn't made the card work but only permitted the dai link to happen). but with linux-next it produces a kernel panic: [ 0.985000] Alignment trap: not handling swp instruction [ 0.985000] Alignment trap: not handling instruction e1003092 at [<c03b0244>] [ 0.990000] Unhandled fault: alignment exception (0x803) at 0x37386d77 [ 0.995000] Internal error: : 803 [#1] [ 0.995000] CPU: 0 Not tainted (3.3.0-rc1-next-20120127+ #55) [ 0.995000] PC is at mutex_lock+0x8/0x14 [ 0.995000] LR is at regmap_write+0x14/0x38 [ 0.995000] pc : [<c03b0248>] lr : [<c01bf7ac>] psr: 60000013 [ 0.995000] sp : c7821e28 ip : 00000000 fp : 00000000 [ 0.995000] r10: c79a59c0 r9 : 00000001 r8 : c052fd5c [ 0.995000] r7 : 00000001 r6 : 0000001f r5 : 00000000 r4 : 37386d77 [ 0.995000] r3 : 00000000 r2 : 00000000 r1 : 0000001f r0 : 37386d77 [ 0.995000] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 0.995000] Control: c000717f Table: 30004000 DAC: 00000017 [ 0.995000] Process swapper (pid: 1, stack limit = 0xc7820270) [ 0.995000] Stack: (0xc7821e28 to 0xc7822000) [ 0.995000] 1e20: c79bfa00 00000000 c79a5a40 c024e408 000000c0 c0257838 [ 0.995000] 1e40: c79bfa00 c052fb80 c052efb8 c024e7e4 c041e348 c0013658 c052fb80 c79bf800 [ 0.995000] 1e60: c79bfa00 00000000 c79bee60 00000000 c7894240 c024efb4 c052fb90 3ab5e842 [ 0.995000] 1e80: c79bec80 00000000 c052ed50 c052fce8 c052fc3c c052fba0 c052fc54 c052fc34 [ 0.995000] 1ea0: c052fc64 c052ebe8 00000000 c052fb80 c052fb98 c052eb60 00000000 c052ebe8 [ 0.995000] 1ec0: 00000000 00000000 00000000 c0250484 c79bfc08 c79bfc08 00000000 00000000 [ 0.995000] 1ee0: c052eb8c c025053c c052eb8c c79bfc08 c05842d4 c01b5edc c01b5ec8 c01b4c70 [ 0.995000] 1f00: c79bfc08 c01b4e10 00000000 00000000 c05842b0 c01b3bc0 c7840c1c c79a60f4 [ 0.995000] 1f20: c79bfc08 c79bfc3c c79bfc08 c01b4ee0 c79bfc08 c0527a98 c79bfc08 c01b3db4 [ 0.995000] 1f40: c79bfc08 c79bfc10 c0527958 c01b26d0 c79bfc10 00000000 00000021 c0165884 [ 0.995000] 1f60: c7821f88 c79bfc00 c79bfc08 c79bfc00 c79bfc08 00000000 00000013 00000000 [ 0.995000] 1f80: 00000000 00000000 00000000 c01b646c c79bec20 c052fb80 00000000 c05854f8 [ 0.995000] 1fa0: 00000013 c04efdb8 c05023b0 00000000 c04efd5c c04da800 c0009850 60000013 [ 0.995000] 1fc0: 00000000 c0501f28 c050269c c05023b0 c050269c c0009850 00000013 00000000 [ 0.995000] 1fe0: 00000000 c04da960 00000000 00000000 c04da8d0 c0009850 00000000 00000000 [ 0.995000] [<c03b0248>] (mutex_lock+0x8/0x14) from [<c01bf7ac>] (regmap_write+0x14/0x38) [ 0.995000] [<c01bf7ac>] (regmap_write+0x14/0x38) from [<c024e408>] (snd_soc_write+0xc/0x10) [ 0.995000] [<c024e408>] (snd_soc_write+0xc/0x10) from [<c0257838>] (wm8753_probe+0x74/0x1c4) [ 0.995000] [<c0257838>] (wm8753_probe+0x74/0x1c4) from [<c024e7e4>] (soc_probe_codec+0x124/0x208) [ 0.995000] [<c024e7e4>] (soc_probe_codec+0x124/0x208) from [<c024efb4>] (snd_soc_instantiate_cards+0x664/0xd5c) [ 0.995000] [<c024efb4>] (snd_soc_instantiate_cards+0x664/0xd5c) from [<c0250484>] (snd_soc_register_card+0x240/0x2d4) [ 0.995000] [<c0250484>] (snd_soc_register_card+0x240/0x2d4) from [<c025053c>] (soc_probe+0x24/0x40) [ 0.995000] [<c025053c>] (soc_probe+0x24/0x40) from [<c01b5edc>] (platform_drv_probe+0x14/0x18) [ 0.995000] [<c01b5edc>] (platform_drv_probe+0x14/0x18) from [<c01b4c70>] (driver_probe_device+0xa0/0x1b0) [ 0.995000] [<c01b4c70>] (driver_probe_device+0xa0/0x1b0) from [<c01b3bc0>] (bus_for_each_drv+0x64/0x8c) [ 0.995000] [<c01b3bc0>] (bus_for_each_drv+0x64/0x8c) from [<c01b4ee0>] (device_attach+0x8c/0xa4) [ 0.995000] [<c01b4ee0>] (device_attach+0x8c/0xa4) from [<c01b3db4>] (bus_probe_device+0x88/0xac) [ 0.995000] [<c01b3db4>] (bus_probe_device+0x88/0xac) from [<c01b26d0>] (device_add+0x4a0/0x570) [ 0.995000] [<c01b26d0>] (device_add+0x4a0/0x570) from [<c01b646c>] (platform_device_add+0xf0/0x18c) [ 0.995000] [<c01b646c>] (platform_device_add+0xf0/0x18c) from [<c04efdb8>] (neo1973_init+0x5c/0x9c) [ 0.995000] [<c04efdb8>] (neo1973_init+0x5c/0x9c) from [<c04da800>] (do_one_initcall+0xa4/0x174) [ 0.995000] [<c04da800>] (do_one_initcall+0xa4/0x174) from [<c04da960>] (kernel_init+0x90/0x128) [ 0.995000] [<c04da960>] (kernel_init+0x90/0x128) from [<c0009850>] (kernel_thread_exit+0x0/0x8) [ 0.995000] Code: e28dd010 e8bd81f0 e3a02000 e1003092 (e3530001) [ 1.000000] ---[ end trace 81e43b55a6684d02 ]--- What could be the cause? (the CPU is an armv4). Denis.

1 1

[alsa-devel] [PATCH] Sound, SoC, WM9090: Remove unused variable 'wm9090' from wm9090_i2c_remove()
by Jesper Juhl 29 Jan '12

29 Jan '12

The variable 'wm9090' is assigned to but never used, so let's just remove it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/wm9090.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) compile teste only diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 41ebe0d..db1ba30 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -655,10 +655,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { - struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&i2c->dev); - return 0; } -- 1.7.8.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] [PATCH v4 - resend 00/13] OMAP4: ASoC: Support for PandaBoard family
by Peter Ujfalusi 28 Jan '12

28 Jan '12

Hello, the following series will add ASoC support for PandaBoards. PandaBoards have different audio routings compared to SDP4430/Blaze boards, but the differences not that big to justify a new ASoC machine driver. The v3 series did not made it to 3.3 kernel. This set contains all the patches needed to enable the audio support on the PandaBoard family. The machine driver has been converted to a generic OMAP4 driver which can support wide range of machines using OMAP4 with twl6040 codec. I have remove the hardcoded MCLK clock frequency use in the machine driver (there can be devices with different MCLK configuration). This changed patch 04, 05, and 12. Since this does not affect the functionality I kept the acks for these patches. Patch 11 did not existed in the v3 series. Regards, Peter --- Peter Ujfalusi (13): ASoC: sdp4430: Correct author e-mail address ASoC: OMAP4: Rename the sdp4430 machine driver ASoC: omap-abe-twl6040: Correct internal prefix, Kconfig entry include: platform_data: Platform data header for OMAP4 ASoC audio OMAP4: 4430sdp: Register platform device for OMAP4 audio ASoC: omap-abe-twl6040: Convert to platform deriver ASoC: twl6040: Convert MICBIAS to SUPPLY widget ASoC: omap-abe-twl6040: Add complete DAPM routing ASoC: omap-abe-twl6040: DAI link selection based on platform data ASoC: omap-abe-twl6040: Configure card according to platform data ASoC: omap-abe-twl6040: Use provided MCLK frequency from pdata OMAP4: omap4panda: Enable audio support ASoC: Kconfig: OMAP4: Enable support for PandaBoards arch/arm/mach-omap2/board-4430sdp.c | 29 ++ arch/arm/mach-omap2/board-omap4panda.c | 60 ++++- include/linux/platform_data/omap-abe-twl6040.h | 49 ++++ sound/soc/codecs/twl6040.c | 16 +- sound/soc/omap/Kconfig | 13 +- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-abe-twl6040.c | 349 ++++++++++++++++++++++++ sound/soc/omap/sdp4430.c | 279 ------------------- 8 files changed, 504 insertions(+), 295 deletions(-) create mode 100644 include/linux/platform_data/omap-abe-twl6040.h create mode 100644 sound/soc/omap/omap-abe-twl6040.c delete mode 100644 sound/soc/omap/sdp4430.c -- 1.7.8.3

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[alsa-devel] [PATCH] ASoC: wm8962: Don't automatically enable and disable FLL
by Mark Brown 27 Jan '12

27 Jan '12

Only enable and disable the FLL when explicitly told to, supporting some additional use cases and making the driver behaviour more standard. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 72 ++++----------------------------------------- 1 files changed, 6 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 58afbde..de74af7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1832,65 +1832,6 @@ SOC_SINGLE_TLV("SPKOUTR Mixer DACR Volume", WM8962_SPEAKER_MIXER_5, 4, 1, 0, inmix_tlv), }; -static int sysclk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - unsigned long timeout; - int src; - int fll; - - /* Ignore attempts to run the event during startup */ - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - return 0; - - src = snd_soc_read(codec, WM8962_CLOCKING2) & WM8962_SYSCLK_SRC_MASK; - - switch (src) { - case 0: /* MCLK */ - fll = 0; - break; - case 0x200: /* FLL */ - fll = 1; - break; - default: - dev_err(codec->dev, "Unknown SYSCLK source %x\n", src); - return -EINVAL; - } - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - if (fll) { - try_wait_for_completion(&wm8962->fll_lock); - - snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, - WM8962_FLL_ENA, WM8962_FLL_ENA); - - timeout = msecs_to_jiffies(5); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); - - if (wm8962->irq && timeout == 0) - dev_err(codec->dev, - "Timed out starting FLL\n"); - } - break; - - case SND_SOC_DAPM_POST_PMD: - if (fll) - snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, - WM8962_FLL_ENA, 0); - break; - - default: - BUG(); - return -EINVAL; - } - - return 0; -} - static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2176,8 +2117,7 @@ SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0), -SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), @@ -2880,8 +2820,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct _fll_div fll_div; unsigned long timeout; int ret; - int fll1 = snd_soc_read(codec, WM8962_FLL_CONTROL_1) & WM8962_FLL_ENA; - int sysclk = snd_soc_read(codec, WM8962_CLOCKING2) & WM8962_SYSCLK_ENA; + int fll1 = 0; /* Any change? */ if (source == wm8962->fll_src && Fref == wm8962->fll_fref && @@ -2904,6 +2843,9 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (ret != 0) return ret; + /* Parameters good, disable so we can reprogram */ + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, 0); + switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: @@ -2942,12 +2884,10 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, try_wait_for_completion(&wm8962->fll_lock); - if (sysclk) - fll1 |= WM8962_FLL_ENA; snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | - WM8962_FLL_ENA, fll1); + WM8962_FLL_ENA, fll1 | WM8962_FLL_ENA); dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Enable idle_bias_off
by Mark Brown 27 Jan '12

27 Jan '12

The WM8962 has ground referenced and class D outputs so can happily go down to BIAS_OFF without a large startup time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 679af29..58afbde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3661,6 +3661,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .set_pll = wm8962_set_fll, .reg_cache_size = WM8962_MAX_REGISTER, .volatile_register = wm8962_soc_volatile, + .idle_bias_off = true, }; /* Improve power consumption for IN4 DC measurement mode */ -- 1.7.9.rc1

1 0

[alsa-devel] Why does snd_seq_drain_output() need lots of time to execute?
by Rafał Cieślak 27 Jan '12

27 Jan '12

Hello, I have a serious problem using the ALSA sequencer interface. I am writing an application that makes use of it's queues, and have got into troubles which I can't resolve at any cost. The main problem is that although the documentation states that snd_seq_drain_output() returns immediately, it turns out that it occasionally needs lots of time (even a second or two), and blocks my application, as I call it. According to my research, it looks like a possible bug in ALSA, but I'd rather ask you if it is indeed unintended. I won't post the code unless you ask me for it, as it is quite a complex application, and that would unnecessarily mess up this post. So let me describe what I've found about this problem. Most of time snd_seq_drain_output() works as it should. I call it from time to time after putting a bunch of MIDI events on a queue. I use it to schedule lots of notes, MIDI controller change messages, and several custom events. It seems I use it more or less appropriately, for the outputting data is scheduled as I wish it to be. I have followed tutorials and examples to learn how to use ALSA sequencer interface, for the documentation does not mention a word on how one should actually use these calls. However, I have compared my snd_seq_* calls to many other open-source MIDI applications, and it seems I got everything right. And, indeed, most of time the app works perfectly. Problems start when I schedule a big amount of events on the queue. There is no exact number that triggers the problem, but it's usually in between 900 and 1100. In that case snd_seq_drain_output() freezes for a very long time. The exact time depends on the tempo applied to the queue (which I find very weird), but is about 0,6 sec for tempo of 60bpm, and greater for slower tempos. I tried also scheduling notes using snd_seq_event_output_direct() to avoid calling snd_seq_drain_output(), but that caused snd_seq_event_output_direct() to lag similarly after such large amount of notes gets scheduled on the queue. And before you ask, I've set the output buffer to some huge amount, that is much lager then the amount of notes I am trying to schedule. It is likely that it's my fault, for I can't fully understand how these calls should be used. In this case could anyone please explain me in which cases snd_seq_drain_output() may block the calling app? I will check if my code might trigger this behaviour. Otherwise, is there any workaround for this lag? Or maybe I have missed something? I need help a lot, because this problem makes me completely unable to continue development of my application. Please let me know in case you need me to provide additional information. Regards, Rafał Cieślak

3 6

[alsa-devel] [PATCH 1/3] ARM: dt: tegra: Enable device tree audio on PAZ00 board.
by Leon Romanovsky 27 Jan '12

27 Jan '12

This patch adds initial device tree support of ALC5632 sound codec and machine driver for PAZ00 board. The implementation is based on the WM8903 codec. Signed-off-by: Marc Dietrich <marvin24(a)gmx.de> Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- .../devicetree/bindings/sound/alc5632.txt | 24 ++++++ .../bindings/sound/tegra-audio-alc5632.txt | 55 ++++++++++++++ arch/arm/boot/dts/tegra-paz00.dts | 29 ++++++-- sound/soc/tegra/tegra_alc5632.c | 78 ++++++++++++++++++-- 4 files changed, 173 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt diff --git a/Documentation/devicetree/bindings/sound/alc5632.txt b/Documentation/devicetree/bindings/sound/alc5632.txt new file mode 100644 index 0000000..8608f74 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/alc5632.txt @@ -0,0 +1,24 @@ +ALC5632 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "realtek,alc5632" + + - reg : the I2C address of the device. + + - gpio-controller : Indicates this device is a GPIO controller. + + - #gpio-cells : Should be two. The first cell is the pin number and the + second cell is used to specify optional parameters (currently unused). + +Example: + +alc5632: alc5632@1e { + compatible = "realtek,alc5632"; + reg = <0x1a>; + + gpio-controller; + #gpio-cells = <2>; +}; diff --git a/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt new file mode 100644 index 0000000..315213f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt @@ -0,0 +1,55 @@ +NVIDIA Tegra audio complex + +Required properties: +- compatible : "nvidia,tegra-audio-alc5632" +- nvidia,model : The user-visible name of this sound complex. +- nvidia,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the ALC5632's pins: + + ALC5632 pins: + + * SPKOUT + * SPKOUTN + * HPL + * HPR + * AUXOUT + * LINEINL + * LINEINR + * PHONEP + * PHONEP + * MIC1 + * MIC2 + * MICBIAS1 + * MICBIAS2 + + Board connectors: + + * Headset Stereophone + * Int Spk + * MIC1 + * MICBIAS1 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller +- nvidia,audio-codec : The phandle of the ALC5632 audio codec + +Example: + +sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = + "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "MIC1", "MICBIAS1", + "MICBIAS1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&alc5632>; +}; diff --git a/arch/arm/boot/dts/tegra-paz00.dts b/arch/arm/boot/dts/tegra-paz00.dts index 4d1bcdc..692d20c 100644 --- a/arch/arm/boot/dts/tegra-paz00.dts +++ b/arch/arm/boot/dts/tegra-paz00.dts @@ -12,6 +12,13 @@ i2c@7000c000 { clock-frequency = <400000>; + + alc5632: alc5632@1e { + compatible = "realtek,alc5632"; + reg = <0x1e>; + gpio-controller; + #gpio-cells = <2>; + }; }; i2c@7000c400 { @@ -37,16 +44,26 @@ clock-frequency = <400000>; }; - i2s@70002800 { - status = "disable"; - }; - i2s@70002a00 { status = "disable"; }; - das@70000c00 { - status = "disable"; + sound { + compatible = "nvidia,tegra-audio-alc5632-paz00", + "nvidia,tegra-audio-alc5632"; + + nvidia,model = "Compal PAZ00"; + + nvidia,audio-routing = + "Int Spk", "SPKOUT", + "Int Spk", "SPKOUTN", + "MIC1", "MICBIAS1", + "MICBIAS1", "Headset Mic", + "Headset Stereophone", "HPR", + "Headset Stereophone", "HPL"; + + nvidia,audio-codec = <&alc5632>; + nvidia,i2s-controller = <&tegra_i2s1>; }; serial@70006000 { diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 4a0e805..22348e8 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -36,6 +36,7 @@ struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; + struct platform_device *pcm_dev; }; static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, @@ -163,26 +164,80 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) sizeof(struct tegra_alc5632), GFP_KERNEL); if (!alc5632) { dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err; } - ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); - if (ret) - return ret; - card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); + alc5632->pcm_dev = ERR_PTR(-EINVAL); + + if (!(pdev->dev.of_node)) { + dev_err(&pdev->dev, "No platform data supplied\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_alc5632_dai.codec_name = NULL; + tegra_alc5632_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + + if (!tegra_alc5632_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_alc5632_dai.cpu_dai_name = NULL; + tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_alc5632_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + alc5632->pcm_dev = platform_device_register_simple( + "tegra-pcm-audio", -1, NULL, 0); + if (IS_ERR(alc5632->pcm_dev)) { + dev_err(&pdev->dev, + "Can't instantiate tegra-pcm-audio\n"); + ret = PTR_ERR(alc5632->pcm_dev); + goto err; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + goto err_unregister; + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - tegra_asoc_utils_fini(&alc5632->util_data); - return ret; + goto err_fini_utils; } return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&alc5632->util_data); +err_unregister: + if (!IS_ERR(alc5632->pcm_dev)) + platform_device_unregister(alc5632->pcm_dev); +err: + return ret; } static int __devexit tegra_alc5632_remove(struct platform_device *pdev) @@ -193,15 +248,23 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&alc5632->util_data); + if (!IS_ERR(alc5632->pcm_dev)) + platform_device_unregister(alc5632->pcm_dev); return 0; } +static const struct of_device_id tegra_alc5632_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-alc5632", }, + {}, +}; + static struct platform_driver tegra_alc5632_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = tegra_alc5632_of_match, }, .probe = tegra_alc5632_probe, .remove = __devexit_p(tegra_alc5632_remove), @@ -212,3 +275,4 @@ MODULE_AUTHOR("Leon Romanovsky <leon(a)leon.nu>"); MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_alc5632_of_match); -- 1.7.3.4

2 4

Re: [alsa-devel] [PATCH] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 27 Jan '12

27 Jan '12

On Fri, Jan 27, 2012 at 05:46:02AM -0800, Reddy, MR Swami wrote: Overall this is a substantial improvement on your last version but there's still quite a few issues here. Looking at some of the issues I'm concerned about how you're testing this, there are some things here that I'd expect the core to complain about when you load the driver. > +/* Register default values for LM49453 river. */ > +static const u8 lm49453_reg_defs[LM49453_MAX_REGISTER] = { Please use the regmap API as previously requested. If anything this is going backwards as you've used a big table rather than [reg] = value but really, use the regmap API. The long term goal is to remove the register cache from ASoC completely. > +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { > +SOC_DAPM_SINGLE("Port1_1 HPL Switch", LM49453_P0_DACHPL1_REG, 0, 1, 0), These controls are all going to end up called "HPL Mixer Port1_1 HPL Switch" and so on which probably isn't what you want. You shouldn't have the "HPL" in any of the individual mixer controls, the prefixing is enough to let the user know this is affecting the HPL mixer. The same issue applies to all your mixer controls. > +static const struct snd_kcontrol_new lm49453_snd_controls[] = { > + SOC_DAPM_ENUM("Mic2Mode Capture Route", lm49453_mic2mode_enum), > + SOC_DAPM_ENUM("DMIC12 Capture Route", lm49453_dmic12_cfg_enum), > + SOC_DAPM_ENUM("DMIC34 Capture Route", lm49453_dmic34_cfg_enum), Why are these appearing as regular controls? DAPM controls should be attached to widgets... > + /* Sidetone supports mono only */ > + SOC_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, > + 0, 0x3F, 0, digital_tlv), These should probably line up with the relevant mixer controls. > + SND_SOC_DAPM_INPUT("AMIC1"), /* Headset mic */ Why is this specifically the headset mic? > + /* HP mapping */ > + { "HPL Mixer", "PORT1_1_RX_LVL Volume", "PORT1_1_RX" }, > + { "HPL Mixer", "PORT1_2_RX_LVL Volume", "PORT1_2_RX" }, > + { "HPL Mixer", "PORT1_3_RX_LVL Volume", "PORT1_3_RX"}, > + { "HPL Mixer", "PORT1_4_RX_LVL Volume", "PORT1_4_RX" }, > + { "HPL Mixer", "PORT1_5_RX_LVL Volume", "PORT1_5_RX" }, > + { "HPL Mixer", "PORT1_6_RX_LVL Volume", "PORT1_6_RX" }, > + { "HPL Mixer", "PORT1_7_RX_LVL Volume", "PORT1_7_RX" }, > + { "HPL Mixer", "PORT1_8_RX_LVL Volume", "PORT1_8_RX" }, This doesn't line up with the switches that are in the DAPM controls for the mixers, the Volume controls are nothing to do with DAPM. > + { "Port2_1 Mixer", "Port1_1 Port2_1 Switch", "PORT1_1_RX_LVL Volume" }, PORT1_1_RX_LVL Volume is not a DAPM widget. I'm very surprised this driver manages to register without generating errors, how have you tested this? > + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQL_REG, > + LM49453_FLL_REF_FREQ_VAL & 0xff); > + snd_soc_write(codec, LM49453_P0_FLL_REF_FREQH_REG, > + (LM49453_FLL_REF_FREQ_VAL >> 8) & 0xff); > + > + snd_soc_write(codec, LM49453_P0_VCO_TARGETLL_REG, > + LM49453_VCO_TARGET_VAL & 0xff); > + snd_soc_write(codec, LM49453_P0_VCO_TARGETLH_REG, > + (LM49453_VCO_TARGET_VAL >> 8) & 0xff); > + snd_soc_write(codec, LM49453_P0_VCO_TARGETHL_REG, > + (LM49453_VCO_TARGET_VAL >> 16) & 0xff); > + snd_soc_write(codec, LM49453_P0_VCO_TARGETHH_REG, > + (LM49453_VCO_TARGET_VAL >> 24) & 0xff); This appears to completely ignore the parameters the user passed in, how can this ever work? > +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, > + unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); > + > + switch (freq) { > + case 12288000: /* Platform System Clock */ What does this comment mean? > +static int lm49453_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *codec_dai) > +{ > + unsigned int startup_mask; > + struct snd_soc_codec *codec = codec_dai->codec; > + dev_dbg(codec->dev, "lm49453_startup\n"); > + > + startup_mask = LM49453_PMC_SETUP_CHIP_EN | LM49453_PMC_SETUP_PLL_EN; > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, startup_mask, > + startup_mask); This looks like it should be DAPM or bias level manaement, and indeed CHIP_EN is disabled by bias management. > +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) > +{ > + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1)|BIT(0), > + BIT(1)|BIT(0)); > + return 0; > +} This completely ignores the mute parameter. > +static int __init lm49453_modinit(void) > +{ > + int ret; > + ret = i2c_add_driver(&lm49453_i2c_driver); > + if (ret != 0) > + pr_err("Failed to register LM49453 I2C driver: %d\n", ret); > + > + return ret; > +} > +module_init(lm49453_modinit); Use module_i2c_driver.

1 0

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Asus N53Jq
by David Henningsson 27 Jan '12

27 Jan '12

The user reports that he needs to add model=auto for audio to work properly. In fact, since node 0x15 is not even a pin node, the existing fixup is definitely wrong. Relevant information can be found in the buglink below. Cc: stable(a)kernel.org (3.2+) BugLink: https://bugs.launchpad.net/bugs/918254 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0db1dc4..a7f17be 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5377,7 +5377,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC), -- 1.7.8.3

2 1

[alsa-devel] ASoC davinci: playing mono sound doubles all samples on the i2s (pcm) bus except the first one
by Jan Pohanka 27 Jan '12

27 Jan '12

Hello, I'm working with a board with TI dm365 processor and using McBSP pheripheral to send sound samples to a GSM module. The kernel version is 2.6.32.17 and the ASoC driver state is based on this commit http://arago-project.org/git/projects/?p=linux-davinci.git;a=commit;h=a4c8e… I have configured the hw interface to send two words in a frame on 8000Hz, this is working fine. When playing mono waveforms I noticed that all samples are doubled (it should be correct because snd_pcm_hardware has SNDRV_PCM_INFO_INTERLEAVED flag) but except the first one, that is send only once. This leads to a state when one sample in each frame is delayed. Unfortunately I'm not an expert on ALSA and ASoC and I'm not sure whether this problem rises in the core of ALSA or only in the driver for davinci dm365 (probably somewhere in the dma handling). Could please someone give me some hint where to find the source of the problem? I hope I described the situation enough... best regards Jan -- Tato zpráva byla vytvořena převratným poštovním klientem Opery: http://www.opera.com/mail/

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[alsa-devel] [PATCH] ASoC: Set idle_bias_off flag in snd_soc_codec_driver
by Axel Lin 27 Jan '12

27 Jan '12

Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver for devices can unconditionally support idle_bias_off. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Harsha Priya <priya.harsha(a)intel.com> Cc: Vladimir Barinov <vova.barinov(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Steve Sakoman <steve(a)sakoman.com> Cc: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/adau1373.c | 3 +-- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/sn95031.c | 3 +-- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/wm8770.c | 3 +-- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 4 +--- 11 files changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 971ba45..facda33 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1244,8 +1244,6 @@ static int adau1373_probe(struct snd_soc_codec *codec) return ret; } - codec->dapm.idle_bias_off = true; - if (pdata) { if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) return -EINVAL; @@ -1340,6 +1338,7 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), .reg_cache_default = adau1373_default_regs, .reg_word_size = sizeof(uint8_t), diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 6b325ea..78e9ce4 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -457,7 +457,6 @@ static int adau1701_probe(struct snd_soc_codec *codec) { int ret; - codec->dapm.idle_bias_off = 1; codec->control_data = to_i2c_client(codec->dev); ret = adau1701_load_firmware(codec); @@ -473,6 +472,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver adau1701_codec_drv = { .probe = adau1701_probe, .set_bias_level = adau1701_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ADAU1701_NUM_REGS, .reg_word_size = sizeof(u16), diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index f99baa0..aa03923 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -827,8 +827,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - codec->dapm.idle_bias_off = 1; - /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -891,6 +889,7 @@ struct snd_soc_codec_driver sn95031_codec = { .read = sn95031_read, .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, + .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 492f22f..285b7a2 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1377,7 +1377,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); if (ret != 0) { @@ -1471,6 +1470,7 @@ static int aic3x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(aic3x_reg), .reg_word_size = sizeof(u8), .reg_cache_default = aic3x_reg, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c06c3e4..2c957c8 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1395,7 +1395,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->dapm.idle_bias_off = 1; dac33->codec = codec; /* Read the tlv320dac33 ID registers */ @@ -1476,6 +1475,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, .set_bias_level = dac33_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(dac33_reg), .reg_word_size = sizeof(u8), .reg_cache_default = dac33_reg, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a193f5f..3039ba2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2227,7 +2227,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_audio_get_mclk() / 1000; - codec->dapm.idle_bias_off = 1; twl4030_init_chip(codec); @@ -2253,6 +2252,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, + .idle_bias_off = true, .reg_cache_size = sizeof(twl4030_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl4030_reg, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19374a9..bd60f84 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,8 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - codec->dapm.idle_bias_off = 1; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -679,6 +677,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .suspend = wm8770_suspend, .resume = wm8770_resume, .set_bias_level = wm8770_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(wm8770_reg_defs), .reg_word_size = sizeof (u16), .reg_cache_default = wm8770_reg_defs diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 8abe375..7ee8dcf 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -549,7 +549,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); - codec->dapm.idle_bias_off = 1; codec->control_data = wm8804->regmap; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); @@ -678,6 +677,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .suspend = wm8804_suspend, .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, + .idle_bias_off = true, .controls = wm8804_snd_controls, .num_controls = ARRAY_SIZE(wm8804_snd_controls), diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 14afc11..37079ea 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2088,7 +2088,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) int ret, i; codec->cache_sync = 1; - codec->dapm.idle_bias_off = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2237,6 +2236,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .suspend = wm8904_suspend, .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, + .idle_bias_off = true, }; static const struct regmap_config wm8904_regmap = { diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index c8aada5..89a8642 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2047,7 +2047,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) int i; int ret; - codec->dapm.idle_bias_off = 1; wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; @@ -2241,6 +2240,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .suspend = wm8995_suspend, .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, + .idle_bias_off = true, }; static struct regmap_config wm8995_regmap = { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c70b7e6..f41e17a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2805,7 +2805,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) int ret; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - struct snd_soc_dapm_context *dapm = &codec->dapm; int i, irq_flags; wm8996->codec = codec; @@ -2813,8 +2812,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->dcs_done); init_completion(&wm8996->fll_lock); - dapm->idle_bias_off = true; - codec->control_data = wm8996->regmap; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); @@ -3068,6 +3065,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .probe = wm8996_probe, .remove = wm8996_remove, .set_bias_level = wm8996_set_bias_level, + .idle_bias_off = true, .seq_notifier = wm8996_seq_notifier, .controls = wm8996_snd_controls, .num_controls = ARRAY_SIZE(wm8996_snd_controls), -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: Convert at73c213 to module_spi_driver
by Axel Lin 27 Jan '12

27 Jan '12

This patch converts at73c213 to use the module_spi_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/spi/at73c213.c | 12 +----------- 1 files changed, 1 insertions(+), 11 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd051b..c6500d0 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1112,17 +1112,7 @@ static struct spi_driver at73c213_driver = { .remove = __devexit_p(snd_at73c213_remove), }; -static int __init at73c213_init(void) -{ - return spi_register_driver(&at73c213_driver); -} -module_init(at73c213_init); - -static void __exit at73c213_exit(void) -{ - spi_unregister_driver(&at73c213_driver); -} -module_exit(at73c213_exit); +module_spi_driver(at73c213_driver); MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt(a)samfundet.no>"); MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: aoa: Convert onyx and tas codec drivers to module_i2c_driver
by Axel Lin 27 Jan '12

27 Jan '12

This patch converts onyx and tas codec drivers to use the module_i2c_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/aoa/codecs/onyx.c | 13 +------------ sound/aoa/codecs/tas.c | 13 +------------ 2 files changed, 2 insertions(+), 24 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 762af68..270790d 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1132,15 +1132,4 @@ static struct i2c_driver onyx_driver = { .id_table = onyx_i2c_id, }; -static int __init onyx_init(void) -{ - return i2c_add_driver(&onyx_driver); -} - -static void __exit onyx_exit(void) -{ - i2c_del_driver(&onyx_driver); -} - -module_init(onyx_init); -module_exit(onyx_exit); +module_i2c_driver(onyx_driver); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index fd2188c..8e63d1f 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -1026,15 +1026,4 @@ static struct i2c_driver tas_driver = { .id_table = tas_i2c_id, }; -static int __init tas_init(void) -{ - return i2c_add_driver(&tas_driver); -} - -static void __exit tas_exit(void) -{ - i2c_del_driver(&tas_driver); -} - -module_init(tas_init); -module_exit(tas_exit); +module_i2c_driver(tas_driver); -- 1.7.5.4

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[alsa-devel] Problems with rate plugin and writing sizes other than the period size
by Trent Piepho 27 Jan '12

27 Jan '12

This is a problem I've been able to duplicate on multiple systems with different hardware. If one sets up rate conversion, from 22050 Hz to 48000 Hz, which is a non-integer ratio, then a hw params with two periods doesn't work if you write greater than the period size. An audio underrun occurs after each period. Tracing with both strace and inside the driver reveals what's going on. The first snd_pcm_writei() goes through and the driver receives the appropriate number of rate converted frames. On the next snd_pcm_writei() alsa-lib calls poll() to wait for space to be available in the buffer. avail_min is set to one period, and there is less than one period left in the buffer, since the first write wrote more than one period. Eventually avail is greater than avail_min and poll returns. So far as expected. Now here's the problem, rather than writing a period alsa-lib just calls poll() again. Which returns immediately since there is still greater than avail_min available space in the buffer. And this continues, with alsa-lib in a tight loop calling poll() and poll() returning immediately, until the buffer empties and underflow occurs. If the number of periods is increased the same thing happens, without the underflow each period. Alsa-lib will wait until there are TWO empty periods in the buffer, in a tight loop calling poll() which returns immediately while avail is greater avail_min and less than two periods worth. It's basically like the avail_min setting has been changed to period_size*2 as far as alsa-lib is concerned, so of course two periods will always cause underruns. More than two work without certain underruns, but alsa-lib is sitting in a tight loop because it keeps calling poll() even though avail > avail_min. Anyone know what's going on here? Seems like a bug in alsa-lib, but I find it hard to believe no one would have noticed this before.

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[alsa-devel] [PATCH 1/3] ASoC: wm8996: Handle failures to determine accessory polarity
by Mark Brown 26 Jan '12

26 Jan '12

If we get an indeterminate impedance with both headset polarities then give up and report the accessory as a headphone rather than continually retrying. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 38 ++++++++++++++++++++++++++------------ 1 files changed, 26 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index fc4f8fc..ab1fb0c 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -89,6 +89,7 @@ struct wm8996_priv { struct snd_soc_jack *jack; bool detecting; bool jack_mic; + int jack_flips; wm8996_polarity_fn polarity_cb; #ifdef CONFIG_GPIOLIB @@ -2427,6 +2428,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8996->jack = jack; wm8996->detecting = true; wm8996->polarity_cb = polarity_cb; + wm8996->jack_flips = 0; if (wm8996->polarity_cb) wm8996->polarity_cb(codec, 0); @@ -2542,6 +2544,19 @@ static void wm8996_hpdet_start(struct snd_soc_codec *codec) WM8996_HP_POLL, WM8996_HP_POLL); } +static void wm8996_report_headphone(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); + + /* Increase the detection rate a bit for responsiveness. */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 7 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2561,6 +2576,7 @@ static void wm8996_micd(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Jack removal detected\n"); wm8996->jack_mic = false; wm8996->detecting = true; + wm8996->jack_flips = 0; snd_soc_jack_report(wm8996->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2601,9 +2617,17 @@ static void wm8996_micd(struct snd_soc_codec *codec) /* If we detected a lower impedence during initial startup * then we probably have the wrong polarity, flip it. Don't * do this for the lowest impedences to speed up detection of - * plain headphones. + * plain headphones. If both polarities report a low + * impedence then give up and report headphones. */ if (wm8996->detecting && (val & 0x3f0)) { + wm8996->jack_flips++; + + if (wm8996->jack_flips > 1) { + wm8996_report_headphone(codec); + return; + } + reg = snd_soc_read(codec, WM8996_ACCESSORY_DETECT_MODE_2); reg ^= WM8996_HPOUT1FB_SRC | WM8996_MICD_SRC | WM8996_MICD_BIAS_SRC; @@ -2630,17 +2654,7 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm8996->detecting) { - dev_dbg(codec->dev, "Headphone detected\n"); - wm8996_hpdet_start(codec); - - /* Increase the detection rate a bit for - * responsiveness. - */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK | - WM8996_MICD_BIAS_STARTTIME_MASK, - 7 << WM8996_MICD_RATE_SHIFT | - 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); + wm8996_report_headphone(codec); } } } -- 1.7.9.rc1

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[alsa-devel] [PATCH] ASoC: wm5100: Use pm_runtime for powerdown managment
by Mark Brown 26 Jan '12

26 Jan '12

Using pm_runtime to decide if the device should go into full power down has the dual advantage of allowing easier integration with non-DAPM reasons to power on the device (like the FLL) and allowing userspace to control the final power down which is useful for tuning retention of DSP firmware. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 116 ++++++++++++++++++++++----------------------- 1 files changed, 56 insertions(+), 60 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 39de946..c6c3821 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -18,6 +18,7 @@ #include <linux/gcd.h> #include <linux/gpio.h> #include <linux/i2c.h> +#include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <linux/regulator/fixed.h> #include <linux/slab.h> @@ -1261,54 +1262,6 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_3_19, 1 }, }; -static int wm5100_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int ret; - - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - break; - - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - if (ret != 0) { - dev_err(codec->dev, - "Failed to enable supplies: %d\n", - ret); - return ret; - } - - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, - 1); - msleep(2); - } - - regcache_cache_only(wm5100->regmap, false); - regcache_sync(wm5100->regmap); - } - break; - - case SND_SOC_BIAS_OFF: - regcache_cache_only(wm5100->regmap, true); - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - break; - } - codec->dapm.bias_level = level; - - return 0; -} - static int wm5100_dai_to_base(struct snd_soc_dai *dai) { switch (dai->id) { @@ -1836,6 +1789,8 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (!Fout) { dev_dbg(codec->dev, "FLL%d disabled", fll_id); + if (fll->fout) + pm_runtime_put(codec->dev); fll->fout = 0; snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); return 0; @@ -1880,6 +1835,8 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Clear any pending completions */ try_wait_for_completion(&fll->lock); + pm_runtime_get_sync(codec->dev); + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, WM5100_FLL1_ENA); if (i2c->irq) @@ -1914,6 +1871,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } if (i == timeout) { dev_err(codec->dev, "FLL%d lock timed out\n", fll_id); + pm_runtime_put(codec->dev); return -ETIMEDOUT; } @@ -2377,9 +2335,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(wm5100->regmap, true); - - for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, WM5100_OUT_VU); @@ -2405,14 +2360,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) } } - /* We'll get woken up again when the system has something useful - * for us to do. - */ - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - return 0; err_gpio: @@ -2444,7 +2391,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, - .set_bias_level = wm5100_set_bias_level, .idle_bias_off = 1, .reg_cache_size = WM5100_MAX_REGISTER, .volatile_register = wm5100_soc_volatile, @@ -2661,6 +2607,10 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, } } + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); @@ -2714,6 +2664,51 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm5100_runtime_suspend(struct device *dev) +{ + struct wm5100_priv *wm5100 = dev_get_drvdata(dev); + + regcache_cache_only(wm5100->regmap, true); + regcache_mark_dirty(wm5100->regmap); + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return 0; +} + +static int wm5100_runtime_resume(struct device *dev) +{ + struct wm5100_priv *wm5100 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 1); + msleep(2); + } + + regcache_cache_only(wm5100->regmap, false); + regcache_sync(wm5100->regmap); + + return 0; +} +#endif + +static struct dev_pm_ops wm5100_pm = { + SET_RUNTIME_PM_OPS(wm5100_runtime_suspend, wm5100_runtime_resume, + NULL) +}; + static const struct i2c_device_id wm5100_i2c_id[] = { { "wm5100", 0 }, { } @@ -2724,6 +2719,7 @@ static struct i2c_driver wm5100_i2c_driver = { .driver = { .name = "wm5100", .owner = THIS_MODULE, + .pm = &wm5100_pm, }, .probe = wm5100_i2c_probe, .remove = __devexit_p(wm5100_i2c_remove), -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: wm5100: Convert to devm_regulator_bulk_get()
by Mark Brown 26 Jan '12

26 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 12 ++++-------- 1 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index c1c8bdb..39de946 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2504,8 +2504,9 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2517,7 +2518,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_core; + goto err_regmap; } if (wm5100->pdata.ldo_ena) { @@ -2686,9 +2687,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_core: - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); err_regmap: regmap_exit(wm5100->regmap); err: @@ -2711,8 +2709,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); regmap_exit(wm5100->regmap); return 0; -- 1.7.9.rc1

1 0

[alsa-devel] ASoC updates for 3.4
by Mark Brown 26 Jan '12

26 Jan '12

The following changes since commit dcd6c92267155e70a94b3927bce681ce74b80d1f: Linux 3.3-rc1 (2012-01-19 15:04:48 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.4 for you to fetch changes up to 5509f2f80c711add6bbcec9af7f4bbba2e2cc22b: ASoC: wm5100: Fix warnings from recent patches (2012-01-24 19:51:34 +0000) ---------------------------------------------------------------- A random collection of updates, pretty much all driver specific cleanups of various kinds which have accumilated over the merge window. There's one feature (support for announcing msbits in the core) and a merge over of a new regmap feature (register patches) the initial users for which are a few ASoC drivers. ---------------------------------------------------------------- (I'm also trying this shiny new git signed tags foo, which may well not actually accomplish anything if you're not using a shiny new git 1.7.9 prerelease, though the regmap merge did the right thing at least. If the asoc-3.4 tag can't be pulled correctly then the for-3.4 branch is still there as normal.) Axel Lin (2): ASoC: jz4740: Convert qi_lb60 to use snd_soc_register_card() ASoC: Add __devinit/__devexit annotations at necessary places Dimitris Papastamos (1): ASoC: wm8983: Remove useless snd_kcontrol Fabio Estevam (2): ASoC: sgtl5000: Print revision number in hex ASoC: sgtl5000: Convert to table based DAPM and control init Javier Martin (2): ASoC: Route Mic Bias in Visstrim_M10 board. ASoC: Add external amplifier controls for Visstrim_M10. Joachim Eastwood (2): ASoC: use proper defines for stream directions in pcm engines ASoC: replace 0xffffffff with DMA_BIT_MASK macro Liam Girdwood (3): ASoC: core - Improve card registration error messaging for large DAI links. ASoC: twl6040 - add method to query HS DC offset step size in mV ASoC: utils - Add support for a dummy codec driver. Manjunath Hadli (1): ASoC: CQ93VC: remove machine specific header file inclusion from codec driver Mark Brown (66): ASoC: Mark WM5100 register map cache only when going into BIAS_OFF ASoC: Don't go through cache when applying WM5100 rev A updates ASoC: Disable register synchronisation for low frequency WM8996 SYSCLK ASoC: Make WM8978 I2C usage unconditional ASoC: Remove unused control type from wm8978 driver ASoC: Convert wm8978 to table based DAPM and control init ASoC: Convert wm8978 to devm_kzalloc() ASoC: Use standard cache sync for WM8978 ASoC: Convert wm8978 to direct regmap API usage ASoC: Push wm8978 reset down into the I2C probe ASoC: Convert WM9090 to use regmap directly ASoC: Move WM9090 device identification and reset to I2C probe ASoC: Convert wm8993 to direct regmap API usage ASoC: Move WM8993 resource acquisition and device reset to bus probe ASoC: Convert WM8753 to direct regmap API usage ASoC: Make WM8971 I2C usage unconditional ASoC: Convert wm8971 to table based DAPM and control init ASoC: Convert wm8971 to devm_kzalloc() ASoC: Make wm8974 I2C usage unconditional ASoC: Convert wm8974 to table based DAPM and control init ASoC: Remove wm8974 private data ASoC: Use standard cache sync for wm8804 ASoC: Convert wm8804 to devm_kzalloc() ASoC: Convert wm8804 to direct regmap API usage ASoC: Make WM8904 I2C usage unconditional ASoC: Convert wm8904 to devm_kzalloc() ASoC: Use standard cache sync for WM8904 ASoC: Convert WM8904 to direct regmap API usage ASoC: Make I2C usage unconditional in WM8940 ASoC: Convert WM8940 to devm_kzalloc() ASoC: Make I2C usage unconditional in WM8955 ASoC: Convert WM8955 to devm_kzalloc() ASoC: Use standard register cache sync for WM8955 ASoC: Convert WM8955 to direct regmap API usage ASoC: Convert WM8955 to table based DAPM and control init ASoC: Optimise performance of WM8904 ADC 128fs OSR mode ASoC: Make WM8962 I2C usage unconditional ASoC: Convert WM8962 register access map to modern style ASoC: Add WM8962 DAC and ADC L/R swap support ASoC: Allow drivers to specify how many bits are significant on a DAI ASoC: 24 bits are significant on the WM8996 audio interfaces ASoC: Implement basic WM8993 interrupt support ASoC: Make WM8993 I2C usage unconditional ASoC: 24 bits are significant on wm_hubs DAIs ASoC: When releasing WM5100 put /RESET into reset ASoC: Say we can't read WM5100 ID register ASoC: Remove redundant set_bias_level() from WM5100 remove() ASoC: Make WM5100 interrupt path use regmap directly ASoC: Push WM5100 interrupt request into I2C probe ASoC: Don't tell applications about msbits unless we're ignoring input ASoC: Use regmap update bits operation for drivers using regmap ASoC: Fix build of tlv320dac33 ASoC: wm8985: Convert to devm_kzalloc() ASoC: wm8985: Convert to table based DAPM and control init ASoC: wm8985: Use standard cache sync implementation ASoC: wm8985 Don't directly reference the cache data structure ASoC: wm8985: Convert to direct regmap API usage ASoC: wm8988: Convert to table based DAPM and control init ASoC: wm8988: Convert to devm_kzalloc() ASoC: sgtl5000: It's sgtl5000 not sgtl500 regmap: Support register patch sets ASoC: Convert WM8731 to direct regmap API usage ASoC: wm5100: Fix mismerge of IRQ frees Merge tag 'regmap-patch-initial' of git://git.kernel.org/.../broonie/regmap into for-3.4 ASoC: Convert the WM5100 revision A updates to a regmap patch ASoC: wm5100: Fix warnings from recent patches Peter Ujfalusi (4): ASoC: tlv320dac33: Use core to set the msbits constraint ASoC: twl4030: Use core to set the msbits constraint ASoC: omap-dmic: Use core to set the msbits constraint ASoC: omap-mcpdm: Set 24msbits constraint Ryan Mallon (1): ep93xx: Don't use system controller defines in audio drivers Wolfram Sang (2): ASoC: tlv320aic32x4: always enable dividers ASoC: tlv320aic32x4: always enable analouge block Zeng Zhaoming (1): ASoC: sgtl5000: Fix wrong register name in restore arch/arm/mach-ep93xx/core.c | 19 +- arch/arm/mach-ep93xx/include/mach/platform.h | 2 +- arch/mips/jz4740/board-qi_lb60.c | 6 + drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache.c | 11 + drivers/base/regmap/regmap.c | 58 + include/linux/regmap.h | 3 + include/sound/soc.h | 2 + sound/soc/atmel/atmel-pcm.c | 4 +- sound/soc/codecs/alc5623.c | 4 +- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 2 - sound/soc/codecs/sgtl5000.c | 36 +- sound/soc/codecs/tlv320aic32x4.c | 110 +- sound/soc/codecs/tlv320dac33.c | 6 +- sound/soc/codecs/twl4030.c | 7 +- sound/soc/codecs/twl6040.c | 13 + sound/soc/codecs/twl6040.h | 1 + sound/soc/codecs/wm5100.c | 356 +++--- sound/soc/codecs/wm8731.c | 109 ++- sound/soc/codecs/wm8753.c | 186 ++- sound/soc/codecs/wm8804.c | 146 +-- sound/soc/codecs/wm8904.c | 838 ++++-------- sound/soc/codecs/wm8904.h | 11 + sound/soc/codecs/wm8940.c | 14 +- sound/soc/codecs/wm8955.c | 247 ++-- sound/soc/codecs/wm8962.c | 1819 +++++++++----------------- sound/soc/codecs/wm8971.c | 37 +- sound/soc/codecs/wm8974.c | 45 +- sound/soc/codecs/wm8978.c | 185 ++- sound/soc/codecs/wm8978.h | 2 + sound/soc/codecs/wm8983.c | 5 - sound/soc/codecs/wm8985.c | 315 +++-- sound/soc/codecs/wm8988.c | 29 +- sound/soc/codecs/wm8993.c | 599 ++++++--- sound/soc/codecs/wm8994.c | 8 +- sound/soc/codecs/wm8996.c | 8 + sound/soc/codecs/wm8996.h | 4 + sound/soc/codecs/wm9090.c | 266 ++-- sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/ep93xx/edb93xx.c | 4 +- sound/soc/ep93xx/ep93xx-pcm.c | 4 +- sound/soc/ep93xx/snappercl15.c | 4 +- sound/soc/fsl/fsl_dma.c | 10 +- sound/soc/fsl/mpc5200_dma.c | 16 +- sound/soc/imx/mx27vis-aic32x4.c | 104 ++ sound/soc/jz4740/qi_lb60.c | 56 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/omap/omap-dmic.c | 7 +- sound/soc/omap/omap-mcpdm.c | 2 + sound/soc/s6000/s6000-pcm.c | 5 +- sound/soc/samsung/dma.c | 2 +- sound/soc/soc-core.c | 33 +- sound/soc/soc-dapm.c | 27 +- sound/soc/soc-io.c | 1 + sound/soc/soc-pcm.c | 38 + sound/soc/soc-utils.c | 20 +- sound/soc/tegra/tegra_pcm.c | 2 +- 58 files changed, 2917 insertions(+), 2944 deletions(-)

2 9

Re: [alsa-devel] soc-dsp programming model for loopbacks
by Mark Brown 26 Jan '12

26 Jan '12

On Wed, Jan 25, 2012 at 12:03:42PM -0800, Pierre-Louis Bossart wrote: > > That's what the CODEC<->CODEC stuff is all about, hiding the link from > > userspace. > I can see how this would simplify the loopback part, but doesn't it make the > 'regular' playback/capture functionality more complex? If my DSP is seen as > a CODEC, how do I represent the CPU-DSP interface (DMA or shared memory)? You'd need a DAI for that interface but you'll need that no matter what you do, it'll just be a very simple DAI that needs no configuration for things like AIF format. With the soc-pcm setup you also need the front and back end DAIs so it's about the same, the benefit comes at runtime. My primary focus for the CODEC<->CODEC links is actually things like basebands rather than on-SoC DSPs, it's just that if the hardware can be thought of in the same way it should simplify things a little.

3 5

[alsa-devel] [PATCH 3/3] ASoC: dt: tegra: Enable headphone autodetection on PAZ00 board.
by Leon Romanovsky 26 Jan '12

26 Jan '12

This patch is adding support of headphone autodetection on PAZ00 board. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- arch/arm/boot/dts/tegra-paz00.dts | 1 + sound/soc/tegra/tegra_alc5632.c | 40 +++++++++++++++++++++++++++++++++--- 2 files changed, 37 insertions(+), 4 deletions(-) diff --git a/arch/arm/boot/dts/tegra-paz00.dts b/arch/arm/boot/dts/tegra-paz00.dts index 692d20c..f95bade 100644 --- a/arch/arm/boot/dts/tegra-paz00.dts +++ b/arch/arm/boot/dts/tegra-paz00.dts @@ -64,6 +64,7 @@ nvidia,audio-codec = <&alc5632>; nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */ }; serial@70006000 { diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 22348e8..ce211af 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <linux/gpio.h> +#include <linux/of_gpio.h> #include <sound/core.h> #include <sound/jack.h> @@ -34,9 +35,14 @@ #define DRV_NAME "tegra-alc5632" +#define GPIO_HP_DET BIT(1) + + struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; struct platform_device *pcm_dev; + int gpio_requested; + int gpio_hp_det; }; static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, @@ -86,6 +92,13 @@ static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { }, }; +static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { + .name = "Headphone detect", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, + .invert = 1, +}; + static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), @@ -114,6 +127,9 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + int ret; snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); @@ -121,6 +137,16 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(tegra_alc5632_hs_jack_pins), tegra_alc5632_hs_jack_pins); + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + 1, + &tegra_alc5632_hp_jack_gpio); + machine->gpio_requested |= GPIO_HP_DET; + } + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); return 0; @@ -243,13 +269,19 @@ err: static int __devexit tegra_alc5632_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); + + if (machine->gpio_requested & GPIO_HP_DET) + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, + 1, + &tegra_alc5632_hp_jack_gpio); + machine->gpio_requested = 0; snd_soc_unregister_card(card); - tegra_asoc_utils_fini(&alc5632->util_data); - if (!IS_ERR(alc5632->pcm_dev)) - platform_device_unregister(alc5632->pcm_dev); + tegra_asoc_utils_fini(&machine->util_data); + if (!IS_ERR(machine->pcm_dev)) + platform_device_unregister(machine->pcm_dev); return 0; } -- 1.7.3.4

3 2

[alsa-devel] E-mu 1616 PCIe support.
by Сергей Королёв 26 Jan '12

26 Jan '12

Hello, This is my first email to developers' mailing list so I apologise for inconvinience. Yesterday 1.0.25 was released and I was hoping that small patch that fixes E-mu 1616 PCIe support has made it's way up to the final release. It didn't happen. Patch that should be applied to sound/pci/emu10k1/emu10k1_main.c is here - https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5250 So I'm here just to remind you that there are bunch of E-mu users who will be very glad if you will free them from hard burden of recompiling the kernel and adding patch after every new kernel release. Thank you.

1 0

[alsa-devel] [PATCH 2/3] ARM: tegra: Enable the headphone detection gpio on PAZ00 board.
by Leon Romanovsky 26 Jan '12

26 Jan '12

Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- arch/arm/mach-tegra/board-paz00-pinmux.c | 1 + arch/arm/mach-tegra/board-paz00.h | 3 +++ 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-tegra/board-paz00-pinmux.c b/arch/arm/mach-tegra/board-paz00-pinmux.c index c775572..4e261d3 100644 --- a/arch/arm/mach-tegra/board-paz00-pinmux.c +++ b/arch/arm/mach-tegra/board-paz00-pinmux.c @@ -152,6 +152,7 @@ static struct tegra_gpio_table gpio_table[] = { { .gpio = TEGRA_WIFI_PWRN, .enable = true }, { .gpio = TEGRA_WIFI_RST, .enable = true }, { .gpio = TEGRA_WIFI_LED, .enable = true }, + { .gpio = TEGRA_HP_DET, .enable = true }, }; static struct tegra_board_pinmux_conf conf = { diff --git a/arch/arm/mach-tegra/board-paz00.h b/arch/arm/mach-tegra/board-paz00.h index ffa83f5..77c8d83 100644 --- a/arch/arm/mach-tegra/board-paz00.h +++ b/arch/arm/mach-tegra/board-paz00.h @@ -32,6 +32,9 @@ #define TEGRA_WIFI_RST TEGRA_GPIO_PD1 #define TEGRA_WIFI_LED TEGRA_GPIO_PD0 +/* Audio */ +#define TEGRA_HP_DET TEGRA_GPIO_PW2 + /* WakeUp */ #define TEGRA_GPIO_POWERKEY TEGRA_GPIO_PJ7 -- 1.7.3.4

2 2

[alsa-devel] [PATCH] ASoC: dapm: Ignore isolated signal generators for power purposes
by Mark Brown 25 Jan '12

25 Jan '12

A signal generator has no power control itself and so shouldn't cause a power up of the device. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ff9939c..15c0821 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1435,9 +1435,13 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* Supplies and micbiases only bring the * context up to STANDBY as unless something * else is active and passing audio they - * generally don't require full power. + * generally don't require full power. Signal + * generators are virtual pins and have no + * power impact themselves. */ switch (w->id) { + case snd_soc_dapm_siggen: + break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_micbias: -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: wm5100: Mark register cache as dirty when regulators are disabled
by Mark Brown 25 Jan '12

25 Jan '12

Otherwise we won't resync later. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 51b281b..273093f 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1376,6 +1376,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: regcache_cache_only(wm5100->regmap, true); + regcache_mark_dirty(wm5100->regmap); for (i = 0; i < ARRAY_SIZE(wm5100->adsp1); i++) wm_adsp1_reinit(wm5100->adsp1[i]); regcache_cache_only(wm5100->regmap, true); -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: dapm: Drop runtime PM references asynchronously
by Mark Brown 25 Jan '12

25 Jan '12

We don't really care if any action is taken immediately so let the PM core defer things if it wants to. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 131508a..ff9939c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1285,7 +1285,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); if (d->dev) - pm_runtime_put_sync(d->dev); + pm_runtime_put(d->dev); } /* If we just powered up then move to active bias */ -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 1/3] ASoC: wm8996: Mark register cache as dirty when regulators are disabled
by Mark Brown 25 Jan '12

25 Jan '12

Otherwise we won't resync later. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e4fa104..4062679 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -104,7 +104,7 @@ static int wm8996_regulator_event_##n(struct notifier_block *nb, \ struct wm8996_priv *wm8996 = container_of(nb, struct wm8996_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - regcache_cache_only(wm8996->regmap, true); \ + regcache_mark_dirty(wm8996->regmap); \ } \ return 0; \ } -- 1.7.9.rc1

1 2

[alsa-devel] [PATCH] ASoC: fsi: Remove unneeded empty runtime PM callbacks
by Mark Brown 25 Jan '12

25 Jan '12

The runtime PM core no longer requires any callbacks so don't provide empty ones for it any more. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/sh/fsi.c | 14 -------------- 1 files changed, 0 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index db6c89a..3241e5b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1408,23 +1408,9 @@ static int fsi_resume(struct device *dev) return 0; } -static int fsi_runtime_nop(struct device *dev) -{ - /* Runtime PM callback shared between ->runtime_suspend() - * and ->runtime_resume(). Simply returns success. - * - * This driver re-initializes all registers after - * pm_runtime_get_sync() anyway so there is no need - * to save and restore registers here. - */ - return 0; -} - static struct dev_pm_ops fsi_pm_ops = { .suspend = fsi_suspend, .resume = fsi_resume, - .runtime_suspend = fsi_runtime_nop, - .runtime_resume = fsi_runtime_nop, }; static struct fsi_core fsi1_core = { -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH 0/3] Add device tree support of audio (ALC5632) on PAZ00 board and enable headphones autodetection.
by Leon Romanovsky 25 Jan '12

25 Jan '12

The following patchset is going to add device tree support of audio (ALC5632) on PAZ00 board and enable headphones autodetection. Leon Romanovsky (3): ARM: dt: tegra: Enable device tree audio on PAZ00 board. ARM: tegra: Enable the headphone detection gpio on PAZ00 board. ASoC: dt: tegra: Enable headphone autodetection on PAZ00 board. .../devicetree/bindings/sound/alc5632.txt | 24 ++++ .../bindings/sound/tegra-audio-alc5632.txt | 55 ++++++++++ arch/arm/boot/dts/tegra-paz00.dts | 30 ++++- arch/arm/mach-tegra/board-paz00-pinmux.c | 1 + arch/arm/mach-tegra/board-paz00.h | 3 + sound/soc/tegra/tegra_alc5632.c | 114 ++++++++++++++++++-- 6 files changed, 212 insertions(+), 15 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5632.txt create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-alc5632.txt -- 1.7.3.4

1 0

Re: [alsa-devel] soc-dsp programming model for loopbacks
by Mark Brown 25 Jan '12

25 Jan '12

On Tue, Jan 24, 2012 at 01:36:18PM -0800, Pierre-Louis Bossart wrote: > Now if the FM radio routing is handled with a digital loopback on the > application processor audio dsp (omap-abe, Intel SST, etc), then soc-dsp > will need to be used. And for a simple FM playback, I need to Note that the use of soc-dsp isn't required for on-SoC DSPs - it really depends on how the hardware looks. If the DSP external format is very strongly tied to the format used to DMA to and from main memory (or other interfaces) then soc-dsp is what you need, if the DSP is able to rewrite formats more flexibly then it should be more direct to model it as a CODEC device with whatever number of DAIs and routing between them.

2 4

[alsa-devel] [PATCH] ASoC: wm5100: Fix microphone configuration
by Mark Brown 25 Jan '12

25 Jan '12

We need to write the configuration for each microphone to a different register. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 0c82f38..51b281b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2610,6 +2610,13 @@ static const struct regmap_config wm5100_regmap = { .cache_type = REGCACHE_RBTREE, }; +static const unsigned int wm5100_mic_ctrl_reg[] = { + WM5100_IN1L_CONTROL, + WM5100_IN2L_CONTROL, + WM5100_IN3L_CONTROL, + WM5100_IN4L_CONTROL, +}; + static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2740,7 +2747,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, } for (i = 0; i < ARRAY_SIZE(wm5100->pdata.in_mode); i++) { - regmap_update_bits(wm5100->regmap, WM5100_IN1L_CONTROL, + regmap_update_bits(wm5100->regmap, wm5100_mic_ctrl_reg[i], WM5100_IN1_MODE_MASK | WM5100_IN1_DMIC_SUP_MASK, (wm5100->pdata.in_mode[i] << -- 1.7.9.rc1

1 0

[alsa-devel] salsa-lib-0.1.2 alsa/pcm_macros.h (on Linux, at least)
by John P. Hartmann 25 Jan '12

25 Jan '12

The salsa macros do not compile with -Wall -Werror. The culprit is the use of "access", which is also a UNIX function. This function is defined unistd.h, which is included by stddef.h. And stddef.h is one of the files mangled into the belly of GCC. So it is a royal mess, and nobody's fault in particular, but could we please have the header file updated, for example to use "Acc" rather than "access" for the parameter? Thanks,

2 1

[alsa-devel] [PATCH] ASoC: wm5100: Make sure we switch to button reporting mode
by Mark Brown 25 Jan '12

25 Jan '12

When we have identified an accessory make sure we've flagged that we've done so in order to make sure we always report buttons and don't continue to polarity flip. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 6a04ee2..0c82f38 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2161,6 +2161,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) if (wm5100->jack_detecting) { dev_dbg(wm5100->dev, "Microphone detected\n"); wm5100->jack_mic = true; + wm5100->jack_detecting = false; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADSET, SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2199,6 +2200,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { dev_dbg(wm5100->dev, "Headphone detected\n"); + wm5100->jack_detecting = false; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); -- 1.7.9.rc1

1 0

[alsa-devel] [PATCH] ASoC: soc-pcm: msbits constraint: Drop 8 and 16 bit sample sizes
by Peter Ujfalusi 25 Jan '12

25 Jan '12

As per discussion we can safely ignore the 8 and 16 bit sample sizes when applying the msbits constraint. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, As we discussed [1] let's remove the 8 and 16 bit size from the array. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2012-January/048276.ht… Regards, Peter sound/soc/soc-pcm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3268901..93be95b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -68,7 +68,7 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, * like the DAC/ADC resolution to use but there isn't right now. */ static int sample_sizes[] = { - 8, 16, 24, 32, + 24, 32, }; static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, -- 1.7.8.3

2 1

[alsa-devel] ALSA release version 1.0.25
by Jaroslav Kysela 25 Jan '12

25 Jan '12

Hello all, the ALSA release version 1.0.25 is available for download at http://www.alsa-project.org . All packages were updated. The full changelog can be obtained from: http://www.alsa-project.org/main/index.php/Changes_v1.0.24_v1.0.25 Have fun with it, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

1 0

[alsa-devel] soc-dsp programming model for loopbacks
by Pierre-Louis Bossart 24 Jan '12

24 Jan '12

Hi, I've been doing some work on audio loopback (FM radio, BT, Modem -> audio codec), and I am somewhat confused by the soc-dsp programming model. Let me take the FM radio example: If the FM radio is directly connected to the audio codec, my understanding is that changing the routing with codec controls will trigger the DAPM logic, which will turn on everything that needs to be on. Very simple, no matter if the FM-codec link is analog or digital. Now if the FM radio routing is handled with a digital loopback on the application processor audio dsp (omap-abe, Intel SST, etc), then soc-dsp will need to be used. And for a simple FM playback, I need to 1. configure the audio codec routing for output selection 2. open a virtual front-end for FM capture 3. configure the DSP routing to link capture front-end to I2S1 backend (FM interface) 4. open a virtual front-end for FM playback 5. configure the DSP routing to link playback front-end to I2S2 backend (codec interface) That seems complicated. To some extent, having the ability to have two back-ends connected together would make more sense, and would simplify the programming model a great deal. User-space code would be similar for loopbacks internal to the codecs or handled on the application processor. This would apply to Bluetooth and modem connections as well. Without this capability, we will end-up with multiple 'virtual' front-ends (6 in my case), making user-space code quite complex. Looking at the current soc-dsp code, I saw that each back-end is supposed to have at least one front-end client, and the impact of my proposal seems fairly important. Before I start looking further into code changes, I wanted to see if my understanding is correct and if there are other ideas to simplify loopbacks. Thanks for your feedback, -Pierre

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Optimise power consumption for IN4 DC measurement usage
by Mark Brown 24 Jan '12

24 Jan '12

When the hardware is configured with one or both of the IN4 inputs used for DC measurement (with no DC blocking capacitor connected) then we can improve power consumption slightly in idle modes by applying a register write sequence. Provide platform data to enable this, implemented using a regmap patch. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm8962.h | 6 ++++++ sound/soc/codecs/wm8962.c | 18 ++++++++++++++++++ 2 files changed, 24 insertions(+), 0 deletions(-) diff --git a/include/sound/wm8962.h b/include/sound/wm8962.h index f131cd2..39477a7 100644 --- a/include/sound/wm8962.h +++ b/include/sound/wm8962.h @@ -50,6 +50,12 @@ struct wm8962_pdata { bool irq_edge; bool spk_mono; /* Speaker outputs tied together as mono */ + + /** + * This flag should be set if one or both IN4 inputs is wired + * in a DC measurement configuration. + */ + bool in4_dc_measure; }; #endif diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 9faef60..f1fd630 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3668,6 +3668,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .volatile_register = wm8962_soc_volatile, }; +/* Improve power consumption for IN4 DC measurement mode */ +static const struct reg_default wm8962_dc_measure[] = { + { 0xfd, 0x1 }, + { 0xcc, 0x40 }, + { 0xfd, 0 }, +}; + static const struct regmap_config wm8962_regmap = { .reg_bits = 16, .val_bits = 16, @@ -3683,6 +3690,7 @@ static const struct regmap_config wm8962_regmap = { static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8962_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm8962_priv *wm8962; unsigned int reg; int ret, i; @@ -3761,6 +3769,16 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, goto err_regmap; } + if (pdata && pdata->in4_dc_measure) { + ret = regmap_register_patch(wm8962->regmap, + wm8962_dc_measure, + ARRAY_SIZE(wm8962_dc_measure)); + if (ret != 0) + dev_err(&i2c->dev, + "Failed to configure for DC mesurement: %d\n", + ret); + } + regcache_cache_only(wm8962->regmap, true); ret = snd_soc_register_codec(&i2c->dev, -- 1.7.9.rc1

1 0

[alsa-devel] default is working, but hw:0,0 is do not...
by Dmitrij K 24 Jan '12

24 Jan '12

Hi all. I have problem with openning a device for playback. this is working code: snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); but this is not working: snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); I get this is: Device or resource busy PS: I have been started the `phonon' (KDE sound daemon), and the amarok (audio player). My question: what I need to do for correct open "hw:0,0" for multiply playback ? Thanks. The Best Regards.

2 2

[alsa-devel] ASoC update for 3.3
by Mark Brown 24 Jan '12

24 Jan '12

The following changes since commit dcd6c92267155e70a94b3927bce681ce74b80d1f: Linux 3.3-rc1 (2012-01-19 15:04:48 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-3.3-20120124 for you to fetch changes up to c83f1d7e71625801c72f4013291194e09b6f0a6e: ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on error (2012-01-24 11:34:19 +0000) ---------------------------------------------------------------- ASoC: Updates for 3.3 A bunch of small updates for drivers that accumilated during the merge window. ---------------------------------------------------------------- Fabio Estevam (1): ASoC: mxs: Fix mxs-saif timeout Jesper Juhl (2): ASoC: wm8958: Use correct format string in dev_err() call ASoC: wm2000: Fix use-after-free - don't release_firmware() twice on error Mark Brown (4): ASoC: Mark WM5100 register map cache only when going into BIAS_OFF ASoC: Don't go through cache when applying WM5100 rev A updates ASoC: Disable register synchronisation for low frequency WM8996 SYSCLK ASoC: wm8996: Call _POST_PMU callback for CPVDD Wolfram Sang (2): ASoC: tlv320aic32x4: always enable dividers ASoC: tlv320aic32x4: always enable analouge block Zeng Zhaoming (1): ASoC: sgtl5000: Fix wrong register name in restore sound/soc/codecs/sgtl5000.c | 17 +++++-- sound/soc/codecs/tlv320aic32x4.c | 110 +++++++++++++++++-------------------- sound/soc/codecs/wm2000.c | 31 +++++------ sound/soc/codecs/wm5100.c | 3 + sound/soc/codecs/wm8958-dsp2.c | 2 +- sound/soc/codecs/wm8996.c | 7 ++- sound/soc/codecs/wm8996.h | 4 ++ sound/soc/mxs/mxs-saif.c | 5 ++ 8 files changed, 96 insertions(+), 83 deletions(-)

1 0

[PATCH] sound/soc/mxs/mxs-saif.c: use devm_ functions
by Julia Lawall 24 Jan '12

24 Jan '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> The various devm_ functions allocate memory that is released when a driver detaches. This patch uses these functions for data that is allocated in the probe function of a platform device and is only freed in the remove function. By removing the need for iounmap, this also eliminates a missing iounmap problem, in the the original error handling code following a successful call to ioremap. Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- sound/soc/mxs/mxs-saif.c | 43 ++++++++++++++----------------------------- 1 file changed, 14 insertions(+), 29 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index f204dba..7d755e0 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -630,7 +630,7 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - saif = kzalloc(sizeof(*saif), GFP_KERNEL); + saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -655,7 +655,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", ret); - goto failed_clk; + return ret; } iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -666,18 +666,19 @@ static int mxs_saif_probe(struct platform_device *pdev) goto failed_get_resource; } - if (!request_mem_region(iores->start, resource_size(iores), - "mxs-saif")) { + if (!devm_request_mem_region(&pdev->dev, iores->start, + resource_size(iores), "mxs-saif")) { dev_err(&pdev->dev, "request_mem_region failed\n"); ret = -EBUSY; goto failed_get_resource; } - saif->base = ioremap(iores->start, resource_size(iores)); + saif->base = devm_ioremap(&pdev->dev, iores->start, + resource_size(iores)); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; - goto failed_ioremap; + goto failed_get_resource; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -685,7 +686,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_ioremap; + goto failed_get_resource; } saif->dma_param.chan_num = dmares->start; @@ -694,14 +695,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_irq1; + goto failed_get_resource; } saif->dev = &pdev->dev; - ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif); + ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_irq1; + goto failed_get_resource; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -709,7 +711,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_irq2; + goto failed_get_resource; } platform_set_drvdata(pdev, saif); @@ -717,7 +719,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_register; + goto failed_get_resource; } saif->soc_platform_pdev = platform_device_alloc( @@ -740,36 +742,19 @@ failed_pdev_add: platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_register: -failed_get_irq2: - free_irq(saif->irq, saif); -failed_get_irq1: - iounmap(saif->base); -failed_ioremap: - release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); -failed_clk: - kfree(saif); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct mxs_saif *saif = platform_get_drvdata(pdev); platform_device_unregister(saif->soc_platform_pdev); - snd_soc_unregister_dai(&pdev->dev); - - iounmap(saif->base); - release_mem_region(res->start, resource_size(res)); - free_irq(saif->irq, saif); - clk_put(saif->clk); - kfree(saif); return 0; }

1 0

[alsa-devel] [PATCH] ALSA: hda: set mute led polarity for laptops with buggy BIOS based on SSID
by Gustavo Maciel Dias Vieira 24 Jan '12

24 Jan '12

HP laptop models with buggy BIOS are apparently frequent, including machines with different codecs. Set the polarity of the mute led based on the SSID and include an entry for the HP Mini 110-3100. Signed-off-by: Gustavo Maciel Dias Vieira <gustavo(a)sagui.org> --- Takashi, is this ok? Predrag, can you test this patch? Abraços, Gustavo sound/pci/hda/patch_sigmatel.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 336cfcd..948f0be 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4870,7 +4870,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) /* BIOS bug: unfilled OEM string */ if (strstr(dev->name, "HP_Mute_LED_P_G")) { set_hp_led_gpio(codec); - spec->gpio_led_polarity = 1; + switch (codec->subsystem_id) { + case 0x103c148a: + spec->gpio_led_polarity = 0; + break; + default: + spec->gpio_led_polarity = 1; + break; + } return 1; } } -- 1.7.7.5

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[alsa-devel] [PATCH] ALSA Au88x0 - Implement subdevice volume controls
by Raymond Yau 24 Jan '12

24 Jan '12

[PATCH] ALSA Au88x0 - Implement subdevice volume controls - add "PCM Playback Volume" controls for 16 playback subdevices This allow application to change the volume of each subdevice by using hardware mixer of au88x0 and default is zero gain/attenunation.

2 1

[alsa-devel] [PATCH] ALSA ymfpci - Don't create rear device, 4ch device and 4ch duplication switch when AC97 codec does not has SDAC bit
by Raymond Yau 24 Jan '12

24 Jan '12

[PATCH] ALSA ymfpci - Don't create rear device, 4ch device and 4ch duplication switch when AC97 codec does not has SDAC bit - check SDAC bit of AC97 primary codec when create "rear" device 3, "4ch" device 2 and "4ch Duplication" switch as the card need a four channels AC97 codec to support surround40.

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[alsa-devel] [PATCH] sound: Fix memory leak on error in snd_compr_set_params()
by Jesper Juhl 24 Jan '12

24 Jan '12

If copy_from_user() does not return 0 we'll leak the memory we allocated for 'params' when that variable goes out of scope. Also a small CodingStyle cleanup: Use braces on both branches of if/else when one branch needs it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/core/compress_offload.c | 13 ++++++++----- 1 files changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index dac3633..a68aed7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) return -ENOMEM; - if (copy_from_user(params, (void __user *)arg, sizeof(*params))) - return -EFAULT; + if (copy_from_user(params, (void __user *)arg, sizeof(*params))) { + retval = -EFAULT; + goto out; + } retval = snd_compr_allocate_buffer(stream, params); if (retval) { - kfree(params); - return -ENOMEM; + retval = -ENOMEM; + goto out; } retval = stream->ops->set_params(stream, params); if (retval) goto out; stream->runtime->state = SNDRV_PCM_STATE_SETUP; - } else + } else { return -EPERM; + } out: kfree(params); return retval; -- 1.7.8.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] Hang up at .prepare( )
by Tomoya MORINAGA 24 Jan '12

24 Jan '12

Hi I've started debugging our ASoC drivers. I've faced issue when installing these drivers, system becomes hang up at async_synchronize_full_domain() of dapm_power_widgets() in .prepare(). Do you have any information about this hang up ? I show details below. ======Details============ * Procedure install codec-driver install platform-driver install machine driver echo ml26124 0x1A > /sys/bus/i2c/devices/i2c-1/new_device After this command, occur system hang up within a few seconds * Call Trees platform-driver:.open() machine-driver:.hw_params() codec-driver:.hw_params() platform-driver:.dai_hw_params() platform-driver:.hw_params() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.open() machine-driver:.hw_params() codec-driver:.hw_params() platform-driver:.dai_hw_params() platform-driver:.hw_params() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.close() platform-driver:.open() machine-driver:.hw_params() codec-driver:.hw_params() platform-driver:.dai_hw_params() platform-driver:.hw_params() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.close() platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.close() platform-driver:.dai_digital_mute() platform-driver:.open() machine-driver:.hw_params() codec-driver:.hw_params() platform-driver:.dai_hw_params() platform-driver:.hw_params() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.open() machine-driver:.hw_params() codec-driver:.hw_params() platform-driver:.dai_hw_params() platform-driver:.hw_params() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.prepare() platform-driver:.dai_digital_mute() platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.hw_free platform-driver:.dai_hw_free platform-driver:.close() platform-driver:.open() machine-driver:.hw_params() codec-driver:.hw_params() platform-driver:.dai_hw_params() platform-driver:.hw_params() platform-driver:.prepare() ==> Occur system hang up. According to system log, the hang up seems occuring at async_synchronize_full_domain(), which is at soc_pcm_prepare -> soc_dapm_stream_event -> dapm_power_widgets -> async_synchronize_full_domain [2nd call] * Test condition CPU=Intel Atom E6xx series IOH=LAPIS Semiconductor ML7213 Fedora15 Kernel=3.2 thanks in advance. tomoya

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[alsa-devel] ASoC: Fix use-after-free in wm2000 - don't release_firmware() twice
by Jesper Juhl 24 Jan '12

24 Jan '12

In wm2000_i2c_probe(), if we take the true branch in " ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0); if (ret != 0) goto err_fw; " then we'll release_firmware(fw) at the 'err_fw' label. But we've already done that just a few lines above. That's a use-after-free bug. This patch restructures the code so that we always call release_firmware(fw) before leaving the function, but only ever call it once. This means that we have to initialize 'fw' to NULL since some paths may now end up calling it without having called request_firmware(), but since request_firmware() deals gracefully with NULL pointers, we are fine if we just NULL initialize it. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/wm2000.c | 31 +++++++++++++------------------ 1 files changed, 13 insertions(+), 18 deletions(-) note: Since I have no real way to test this, it is compile tested only. I believe the logic is identical, but please check my work before applying it. diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index c288090..a75c376 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, struct wm2000_priv *wm2000; struct wm2000_platform_data *pdata; const char *filename; - const struct firmware *fw; - int reg, ret; + const struct firmware *fw = NULL; + int ret; + int reg; u16 id; wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv), @@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm2000->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + goto out; } /* Verify that this is a WM2000 */ @@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (id != 0x2000) { dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id); ret = -ENODEV; - goto err_regmap; + goto out_regmap_exit; } reg = wm2000_read(i2c, WM2000_REG_REVISON); @@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, ret = request_firmware(&fw, filename, &i2c->dev); if (ret != 0) { dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret); - goto err_regmap; + goto out_regmap_exit; } /* Pre-cook the concatenation of the register address onto the image */ @@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, if (wm2000->anc_download == NULL) { dev_err(&i2c->dev, "Out of memory\n"); ret = -ENOMEM; - goto err_fw; + goto out_regmap_exit; } wm2000->anc_download[0] = 0x80; wm2000->anc_download[1] = 0x00; memcpy(wm2000->anc_download + 2, fw->data, fw->size); - release_firmware(fw); - wm2000->anc_eng_ena = 1; wm2000->anc_active = 1; wm2000->spk_ena = 1; @@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c, wm2000_reset(wm2000); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, - NULL, 0); - if (ret != 0) - goto err_fw; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0); + if (!ret) + goto out; - return 0; - -err_fw: - release_firmware(fw); -err_regmap: +out_regmap_exit: regmap_exit(wm2000->regmap); -err: +out: + release_firmware(fw); return ret; } -- 1.7.8.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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Re: [alsa-devel] [PATCH v4 1/5] OMAP4: hwmod: Add names for DMIC memory address space
by Peter Ujfalusi 24 Jan '12

24 Jan '12

Hi Paul, Tony, On 12/19/2011 11:51 PM, Paul Walmsley wrote: > Hi > > On Fri, 16 Dec 2011, Cousson, Benoit wrote: > >> On 12/16/2011 7:28 AM, Paul Walmsley wrote: >>> On Mon, 28 Nov 2011, Peter Ujfalusi wrote: >>> >>>> To be able to get the memory resources by name from >>>> the DMIC driver (for MPU and for DMA). >>> >>> Any comments on this one? Looks like we'd need to add these names as >>> hints in the script. >> >> Maybe not, in theory we should be able to detect the dual mapping and add the >> proper name. I did that for some other IP in the ABE domain, so it should be >> doable for that one too. >> >>>> Signed-off-by: Peter Ujfalusi<peter.ujfalusi(a)ti.com> >> >> Acked-by: Benoit Cousson <b-cousson(a)ti.com> > > Cool. Queued for when Tony wants to pull it. This patch seams to be missing from 3.3-rc1. The rest of the series is there, but because of the missing patch audio does not probe on 4430SDP/Blaze. Can you queue this as a fix for Linus? Thanks, Péter

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[alsa-devel] [PATCH 1/6] hda_intel: Add Lynx Point HD Audio Controller DeviceIDs
by Seth Heasley 24 Jan '12

24 Jan '12

This patch adds the HD Audio DeviceIDs for the Intel Lynx Point PCH. Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> --- sound/pci/hda/hda_intel.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0852e20..adb4fa7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -148,6 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, PCH}," "{Intel, CPT}," "{Intel, PPT}," + "{Intel, LPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -2989,6 +2990,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE}, + /* Lynx Point */ + { PCI_DEVICE(0x8086, 0x8c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- 1.7.4.4

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[alsa-devel] [PATCH V2 1/4] ARM: dt: tegra: Add Tegra APB DMA device tree binding
by Stephen Warren 24 Jan '12

24 Jan '12

Document binding, and add the node to tegra*.dtsi. The driver isn't actually instantiated from this node yet, but the I2S binding will rely on being able to refer to the APB DMA node using a phandle. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- v2: Added this to tegra30.dtsi too. Fixed interrupt IDs. Fixed reg to include all the per-channel registers. --- .../devicetree/bindings/dma/tegra20-apbdma.txt | 30 ++++++++++++++++ arch/arm/boot/dts/tegra20.dtsi | 21 +++++++++++ arch/arm/boot/dts/tegra30.dtsi | 37 ++++++++++++++++++++ 3 files changed, 88 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/tegra20-apbdma.txt diff --git a/Documentation/devicetree/bindings/dma/tegra20-apbdma.txt b/Documentation/devicetree/bindings/dma/tegra20-apbdma.txt new file mode 100644 index 0000000..90fa7da --- /dev/null +++ b/Documentation/devicetree/bindings/dma/tegra20-apbdma.txt @@ -0,0 +1,30 @@ +* NVIDIA Tegra APB DMA controller + +Required properties: +- compatible: Should be "nvidia,<chip>-apbdma" +- reg: Should contain DMA registers location and length. This shuld include + all of the per-channel registers. +- interrupts: Should contain all of the per-channel DMA interrupts. + +Examples: + +apbdma: dma@6000a000 { + compatible = "nvidia,tegra20-apbdma"; + reg = <0x6000a000 0x1200>; + interrupts = < 0 136 0x04 + 0 137 0x04 + 0 138 0x04 + 0 139 0x04 + 0 140 0x04 + 0 141 0x04 + 0 142 0x04 + 0 143 0x04 + 0 144 0x04 + 0 145 0x04 + 0 146 0x04 + 0 147 0x04 + 0 148 0x04 + 0 149 0x04 + 0 150 0x04 + 0 151 0x04 >; +}; diff --git a/arch/arm/boot/dts/tegra20.dtsi b/arch/arm/boot/dts/tegra20.dtsi index 853a5c6..0a5d092 100644 --- a/arch/arm/boot/dts/tegra20.dtsi +++ b/arch/arm/boot/dts/tegra20.dtsi @@ -12,6 +12,27 @@ < 0x50040100 0x0100 >; }; + apbdma: dma@6000a000 { + compatible = "nvidia,tegra20-apbdma"; + reg = <0x6000a000 0x1200>; + interrupts = < 0 104 0x04 + 0 105 0x04 + 0 106 0x04 + 0 107 0x04 + 0 108 0x04 + 0 109 0x04 + 0 110 0x04 + 0 111 0x04 + 0 112 0x04 + 0 113 0x04 + 0 114 0x04 + 0 115 0x04 + 0 116 0x04 + 0 117 0x04 + 0 118 0x04 + 0 119 0x04 >; + }; + i2c@7000c000 { #address-cells = <1>; #size-cells = <0>; diff --git a/arch/arm/boot/dts/tegra30.dtsi b/arch/arm/boot/dts/tegra30.dtsi index 368cbb3..537953a 100644 --- a/arch/arm/boot/dts/tegra30.dtsi +++ b/arch/arm/boot/dts/tegra30.dtsi @@ -12,6 +12,43 @@ < 0x50040100 0x0100 >; }; + apbdma: dma@6000a000 { + compatible = "nvidia,tegra30-apbdma", "nvidia,tegra20-apbdma"; + reg = <0x6000a000 0x1400>; + interrupts = < 0 104 0x04 + 0 105 0x04 + 0 106 0x04 + 0 107 0x04 + 0 108 0x04 + 0 109 0x04 + 0 110 0x04 + 0 111 0x04 + 0 112 0x04 + 0 113 0x04 + 0 114 0x04 + 0 115 0x04 + 0 116 0x04 + 0 117 0x04 + 0 118 0x04 + 0 119 0x04 + 0 128 0x04 + 0 129 0x04 + 0 130 0x04 + 0 131 0x04 + 0 132 0x04 + 0 133 0x04 + 0 134 0x04 + 0 135 0x04 + 0 136 0x04 + 0 137 0x04 + 0 138 0x04 + 0 139 0x04 + 0 140 0x04 + 0 141 0x04 + 0 142 0x04 + 0 143 0x04 >; + }; + i2c@7000c000 { #address-cells = <1>; #size-cells = <0>; -- 1.7.0.4

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Re: [alsa-devel] [REGRESSION][IDT 92HD88B1] Reversed mute key LED status after update to 3.2
by Gustavo Maciel Dias Vieira 24 Jan '12

24 Jan '12

Em Dom, 2012-01-08 às 01:39 +0100, Predrag Ivanovic escreveu: > With kernel-3.2, LED on a mute button reversed its function. > When sound is muted,it is off, when sound is unmuted, it is on. > With 3.1.6, works as it should. > > It seems that commit > https://github.com/torvalds/linux/commit/e2ef36c688d5844a986424c3194bd33212… > > (originated from "[PATCH] ALSA: hda: invert mute led polarity for HP > laptops" > http://mailman.alsa-project.org/pipermail/alsa-devel/2011-December/046862.h…) > > broke it, because when I reverted it, and rebuilt the modules, LED > works as expected. > > Link to Bugzilla issue I have opened : > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5505 > > Output of alsa-info.sh is attached there, and some more info. > > If you need more information, I'll be happy to provide it. > Would output of dmidecode be useful in any way? > Please cc me, I am not subscribed to the list. The output of dmidecode would definitely help. Abraços, Gustavo

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[alsa-devel] [PATCH] Sound WM8958 DSP2: Use correct format string in dev_err() call
by Jesper Juhl 23 Jan '12

23 Jan '12

To print a value of type size_t one should use %zd, not %d. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/wm8958-dsp2.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 8d4ea43..40ac888 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -55,7 +55,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, return 0; if (fw->size < 32) { - dev_err(codec->dev, "%s: firmware too short (%d bytes)\n", + dev_err(codec->dev, "%s: firmware too short (%zd bytes)\n", name, fw->size); goto err; } -- 1.7.8.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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[alsa-devel] Request for compilation test - v1.0.25
by Jaroslav Kysela 23 Jan '12

23 Jan '12

Hi all, I would like to ask for a package compilation test in your distribution for latest ALSA package snapshots available at: http://www.alsa-project.org/snapshot/ At this time, these snapshots are available: alsa-driver 1.0.24.110.g22743.2525.ge60f5 alsa-lib 1.0.24.1.53.g98d4e alsa-utils 1.0.24.2.26.g7d33a alsa-firmware 1.0.24.1.3.g3c38d alsa-tools 1.0.24.1.51.g234fa alsa-oss 1.0.17.2.g1dfe1 alsa-plugins 1.0.24.11.gb3fb5 alsa-python 1.0.24.2.g4c452 Please, report any compilation/installation issues to this list (you may add CC to me). I am going to make 1.0.25 release on Wednesday. The alsa-driver package will not support the kernels bellow 2.6.18 (released Sep 20 2006), from README: - Vanilla 3.0 to 3.2 - Vanilla 2.6.18 to 2.6.39 I kept the 2.4 build code, but it is not maintained and definitely, it will be removed in next alsa-driver versions. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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Re: [alsa-devel] Request for compilation test - v1.0.25
by Jaroslav Kysela 23 Jan '12

23 Jan '12

Date 23.1.2012 17:32, Takashi Iwai wrote: > At Mon, 23 Jan 2012 16:41:05 +0100, > Jaroslav Kysela wrote: >> >> Date 23.1.2012 15:16, Takashi Iwai wrote: >>> At Mon, 23 Jan 2012 12:11:26 +0100, >>> Jaroslav Kysela wrote: >>>> >>>> The alsa-driver package will not support the kernels bellow 2.6.18 >>>> (released Sep 20 2006), from README: >>>> >>>> - Vanilla 3.0 to 3.2 >>>> - Vanilla 2.6.18 to 2.6.39 >>>> >>>> I kept the 2.4 build code, but it is not maintained and definitely, it >>>> will be removed in next alsa-driver versions. >>> >>> 2.4 kernel won't be built since many driver codes need >>> KBUILD_MODNAME. >>> >>> OTOH, 2.6 kernels below 2.6.18 should be still OK. I'm building the >>> drivers regularly with 2.6.9 and 2.6.16 kernels for SLE10 and SLE11. >> >> I tried vanilla kernels, but there are many things to resolve for recent >> gcc and make - the 2.6.18 vanilla was the first kernel which works. I >> believe that modified distribution specific kernels - which includes >> some newer code/interface - might work ok. It was the reason to avoid a >> hard-coded kernel version check in configure.in. > > Yeah, running an old gcc is nasty, so I'm OK to mention the support > status in the way above you stated. OTOH, would be good to mention > that the builds are OK with the older 2.6 kernels. Yep, added. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ALSA: HDA: Use model=auto for Thinkpad T510
by David Henningsson 23 Jan '12

23 Jan '12

The user reports that model=auto works fine for him. Using model=auto bring in new features such as jack detection notification to userspace. Alsa info is available at http://paste.ubuntu.com/805351/ Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8a32a69..c3e7952 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3027,7 +3027,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), -- 1.7.8.3

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Re: [alsa-devel] [PATCH] - HDA-Models.txt: Update model names2
by Dave Lentz 23 Jan '12

23 Jan '12

Update Conexant model names and add Realtek ALC665. Signed-off-by: Dave Lentz <dtl131 at gmail.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index c8c5454..a2a2865 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -53,7 +53,7 @@ ALC269 laptop-amic Laptops with analog-mic input laptop-dmic Laptops with digital-mic input -ALC662/663/272 +ALC662/663/665/272 ============== asus-mode1 ASUS asus-mode2 ASUS @@ -192,7 +192,7 @@ AD1988/AD1988B/AD1989A/AD1989B laptop-dig ditto with SPDIF auto auto-config reading BIOS (default) -Conexant 5045 +Conexant 5045 Venice/CX20549 ============= laptop-hpsense Laptop with HP sense (old model laptop) laptop-micsense Laptop with Mic sense (old model fujitsu) @@ -203,7 +203,7 @@ Conexant 5045 can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y -Conexant 5047 +Conexant 5047 Waikiki/CX20551 ============= laptop Basic Laptop config laptop-hp Laptop config for some HP models (subdevice 30A5) @@ -212,7 +212,7 @@ Conexant 5047 can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y -Conexant 5051 +Conexant 5051 Hermosa/CX20561 ============= laptop Basic Laptop config (default) hp HP Spartan laptop @@ -221,7 +221,7 @@ Conexant 5051 ideapad Lenovo IdeaPad laptop toshiba Toshiba Satellite M300 -Conexant 5066 +Conexant 5066 Pebble/CX20582 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60

2 1

[alsa-devel] alsa-utils-1.0.24.2 won't compile on x86_64 - CoreFoundation error
by John Rigg 23 Jan '12

23 Jan '12

Apologies if this should have gone to alsa-user instead, but I've just tried compiling alsa-utils-1.0.24.2 on a Debian testing amd64 system, and configure gave a list of errors related to CoreFoundation. There were several similar to this in config.log: ... configure:4589: checking for CFPreferencesCopyAppValue configure:4607: gcc -o conftest -g -O2 conftest.c -Wl,-framework -Wl,CoreFoundation >&5 conftest.c:11:42: fatal error: CoreFoundation/CFPreferences.h: No such file or directory compilation terminated. ... I ran ./configure with default settings. Kernel is 3.2.1 compiled for k8 from kernel.org source. alsa-lib-1.0.24.1 compiled and installed sucessfully. Possibly a stupid question, but why is it checking for CoreFoundation on a Linux system? John

3 3

[alsa-devel] [PATCH 1/3] mfd: Add __devinit and __devexit annotations in wm8994
by Mark Brown 23 Jan '12

23 Jan '12

The wm8994_device_init() and wm8994_device_exit() functions were not annotated as device init and exit functions, meaning they shouldn't reference __devinitdata. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- This is mostly just a warning fix for the second patch and doesn't seem terribly critical otherwise so it should probably get merged along with it. drivers/mfd/wm8994-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index a04b3c1..a215b15 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -362,7 +362,7 @@ static int wm8994_ldo_in_use(struct wm8994_pdata *pdata, int ldo) /* * Instantiate the generic non-control parts of the device. */ -static int wm8994_device_init(struct wm8994 *wm8994, int irq) +static __devinit int wm8994_device_init(struct wm8994 *wm8994, int irq) { struct wm8994_pdata *pdata = wm8994->dev->platform_data; struct regmap_config *regmap_config; @@ -594,7 +594,7 @@ err_regmap: return ret; } -static void wm8994_device_exit(struct wm8994 *wm8994) +static __devexit void wm8994_device_exit(struct wm8994 *wm8994) { pm_runtime_disable(wm8994->dev); mfd_remove_devices(wm8994->dev); -- 1.7.7.3

1 2

[alsa-devel] [PATCH] - HDA-Models.txt: Update model names2
by Dave Lentz 23 Jan '12

23 Jan '12

Update Conexant model names and add Realtek ALC665. Signed-off-by: Dave Lentz <dtl131 at gmail.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index c8c5454..a2a2865 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -53,7 +53,7 @@ ALC269 laptop-amic Laptops with analog-mic input laptop-dmic Laptops with digital-mic input -ALC662/663/272 +ALC662/663/665/272 ============== asus-mode1 ASUS asus-mode2 ASUS @@ -192,7 +192,7 @@ AD1988/AD1988B/AD1989A/AD1989B laptop-dig ditto with SPDIF auto auto-config reading BIOS (default) -Conexant 5045 +Conexant 5045 Venice/CX20549 ============= laptop-hpsense Laptop with HP sense (old model laptop) laptop-micsense Laptop with Mic sense (old model fujitsu) @@ -203,7 +203,7 @@ Conexant 5045 can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y -Conexant 5047 +Conexant 5047 Waikiki/CX20551 ============= laptop Basic Laptop config laptop-hp Laptop config for some HP models (subdevice 30A5) @@ -212,7 +212,7 @@ Conexant 5047 can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y -Conexant 5051 +Conexant 5051 Hermosa/CX20561 ============= laptop Basic Laptop config (default) hp HP Spartan laptop @@ -221,7 +221,7 @@ Conexant 5051 ideapad Lenovo IdeaPad laptop toshiba Toshiba Satellite M300 -Conexant 5066 +Conexant 5066 Pebble/CX20582 ============= laptop Basic Laptop config (default) hp-laptop HP laptops, e g G60

2 1

[alsa-devel] [PATCH v2] ASoC: da7210: Add support for PLL and SRM
by Ashish Chavan 23 Jan '12

23 Jan '12

Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. This patch has been tested on DA7210 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v1: - Removed unnecessary function to configure clock dividers - Changed data layout and code related to PLL dividers. Tried making it more readable and robust --- sound/soc/codecs/da7210.c | 259 +++++++++++++++++++++++++++++++++++++-------- 1 files changed, 216 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab38e93..4d7eec4 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,6 +145,7 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) #define DA7210_PLL_BYP (1 << 6) @@ -161,12 +162,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -205,6 +210,57 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* PLL out frequency values */ +#define FOUT_2822400 2822400 +#define FOUT_3072000 3072000 + +/* Index fields for PLL divider array */ +#define FREF_IDX (0 << 0) +#define DIV1_IDX (1 << 0) +#define DIV2_IDX (2 << 0) +#define DIV3_IDX (3 << 0) + +/* Number of PLL input frequencies supported */ +#define FREF_CNT (7 << 0) +/* Number of PLL dividers for each frequency */ +#define DIV_CNT (3 << 0) + +/* PLL Master mode dividers */ +static const u32 da7210_fout_2822400_div[][DIV_CNT + 1] = { + { 12000000, 0xE8, 0x6C, 0x2, }, /* MCLK=12Mhz Fs=44.1Khz */ + { 13000000, 0xDF, 0x28, 0xC, }, /* MCLK=13Mhz Fs=44.1Khz */ + { 13500000, 0xDB, 0x0A, 0xD, }, /* MCLK=13.5Mhz Fs=44.1Khz */ + { 14400000, 0xD4, 0x5A, 0x2, }, /* MCLK=14.4Mhz Fs=44.1Khz */ + { 19200000, 0xBB, 0x43, 0x9, }, /* MCLK=19.2Mhz Fs=44.1Khz */ + { 19680000, 0xB9, 0x6D, 0xA, }, /* MCLK=19.68Mhz Fs=44.1Khz */ + { 19800000, 0xB8, 0xFB, 0xB, }, /* MCLK=19.8Mhz Fs=44.1Khz */ +}; + +static const u32 da7210_fout_3072000_div[][DIV_CNT + 1] = { + { 12000000, 0xF3, 0x12, 0x7, }, /* MCLK=12Mhz Fs=48Khz */ + { 13000000, 0xE8, 0xFD, 0x5, }, /* MCLK=13Mhz Fs=48Khz */ + { 13500000, 0xE4, 0x82, 0x3, }, /* MCLK=13.5Mhz Fs=48Khz */ + { 14400000, 0xDD, 0x3A, 0x0, }, /* MCLK=14.4Mhz Fs=48Khz */ + { 19200000, 0xC1, 0xEB, 0x8, }, /* MCLK=19.2Mhz Fs=48Khz */ + { 19680000, 0xBF, 0xEC, 0x0, }, /* MCLK=19.68Mhz Fs=48Khz */ + { 19800000, 0xBF, 0x70, 0x0, }, /* MCLK=19.8Mhz Fs=48Khz */ +}; + +/* PLL SRM mode dividers */ +static const u32 da7210_pll_srm_div[][DIV_CNT + 1] = { + { 12000000, 0xED, 0xBF, 0x5, }, /* MCLK=12Mhz */ + { 13000000, 0xE4, 0x13, 0x0, }, /* MCLK=13Mhz */ + { 13500000, 0xDF, 0xC6, 0x8, }, /* MCLK=13.5Mhz */ + { 14400000, 0xD8, 0xCA, 0x1, }, /* MCLK=14.4Mhz */ + { 19200000, 0xBE, 0x97, 0x9, }, /* MCLK=19.2Mhz */ + { 19680000, 0xBC, 0xAC, 0xD, }, /* MCLK=19.68Mhz */ + { 19800000, 0xBC, 0x35, 0xE, }, /* MCLK=19.8Mhz */ +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -627,6 +683,8 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; + unsigned int mclk_rate; + int master; }; /* @@ -673,8 +731,9 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -707,43 +766,33 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; break; default: return -EINVAL; @@ -753,8 +802,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + } else { + /* PLL bypass mode, enable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -768,6 +824,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; @@ -776,9 +833,11 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -830,10 +889,148 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3, row_idx; + + if (da7210->master) { + /* In PLL master mode, use master mode PLL dividers */ + if (fout == FOUT_2822400) { + for (row_idx = 0; row_idx < FREF_CNT; row_idx++) { + if (fref == da7210_fout_2822400_div[row_idx] + [FREF_IDX]) { + pll_div1 = + da7210_fout_2822400_div[row_idx] + [DIV1_IDX]; + pll_div2 = + da7210_fout_2822400_div[row_idx] + [DIV2_IDX]; + pll_div3 = + da7210_fout_2822400_div[row_idx] + [DIV3_IDX]; + break; + } + } + } else if (fout == FOUT_3072000) { + for (row_idx = 0; row_idx < FREF_CNT; row_idx++) { + if (fref == da7210_fout_3072000_div[row_idx] + [FREF_IDX]) { + pll_div1 = + da7210_fout_3072000_div[row_idx] + [DIV1_IDX]; + pll_div2 = + da7210_fout_3072000_div[row_idx] + [DIV2_IDX]; + pll_div3 = + da7210_fout_3072000_div[row_idx] + [DIV3_IDX]; + break; + } + } + } else { + dev_err(codec_dai->dev, + "Unsupported PLL output frequency %d\n", fout); + return -EINVAL; + } + if (row_idx >= FREF_CNT) + goto err; + } else { + /* In PLL slave mode, use SRM mode PLL dividers */ + for (row_idx = 0; row_idx < FREF_CNT; row_idx++) { + if (fref == da7210_pll_srm_div[row_idx][FREF_IDX]) { + pll_div1 = + da7210_pll_srm_div[row_idx][DIV1_IDX]; + pll_div2 = + da7210_pll_srm_div[row_idx][DIV2_IDX]; + pll_div3 = + da7210_pll_srm_div[row_idx][DIV3_IDX]; + break; + } + } + if (row_idx >= FREF_CNT) + goto err; + } + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + if (da7210->master) { + /* In master mode, no need to enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, + DA7210_PLL_EN); + } else { + /* In slave mode, enable SRM and PLL */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +err: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", fref); + return -EINVAL; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -872,24 +1069,13 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ + + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); - /* - * make sure that DA7210 use bypass mode before start up - */ snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* * ADC settings @@ -964,24 +1150,11 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Enable Aux2 */ snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Set PLL Master clock range 10-20 MHz */ + snd_soc_write(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_10_20_MHZ); /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* As suggested by Dialog */ snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); -- 1.7.1

2 3

[alsa-devel] [PATCH] ASoC: Add __devinit/__devexit annotations at necessary places
by Axel Lin 23 Jan '12

23 Jan '12

Fix below build warning when CONFIG_HOTPLUG is not set. CC sound/soc/codecs/alc5623.o sound/soc/codecs/alc5623.c:1062: warning: ‘alc5623_i2c_remove’ defined but not used CC sound/soc/codecs/alc5632.o sound/soc/codecs/alc5632.c:1112: warning: ‘alc5632_i2c_remove’ defined but not used Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 4 ++-- sound/soc/codecs/alc5632.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 3feee56..08f2419 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -992,7 +992,7 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = { * low = 0x1a * high = 0x1b */ -static int alc5623_i2c_probe(struct i2c_client *client, +static __devinit int alc5623_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct alc5623_platform_data *pdata; @@ -1059,7 +1059,7 @@ static int alc5623_i2c_probe(struct i2c_client *client, return ret; } -static int alc5623_i2c_remove(struct i2c_client *client) +static __devexit int alc5623_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); return 0; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 390e437..af9c27a 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1109,7 +1109,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, return ret; } -static int alc5632_i2c_remove(struct i2c_client *client) +static __devexit int alc5632_i2c_remove(struct i2c_client *client) { struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); -- 1.7.5.4

3 2

[alsa-devel] [PATCH v2] ASoC: sgtl5000: Convert to table based DAPM and control init
by Fabio Estevam 23 Jan '12

23 Jan '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Convert to table based DAPM and control init. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fixed typo in codec name (500 instead of 5000) sound/soc/codecs/sgtl5000.c | 17 +++++++---------- 1 files changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7f4ba81..d6f76cf 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -227,7 +227,7 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { }; /* routes for sgtl5000 */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ @@ -1353,15 +1353,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_add_controls(codec, sgtl5000_snd_controls, - ARRAY_SIZE(sgtl5000_snd_controls)); - - snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, - ARRAY_SIZE(sgtl5000_dapm_widgets)); - - snd_soc_dapm_add_routes(&codec->dapm, audio_map, - ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(&codec->dapm); return 0; @@ -1402,6 +1393,12 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .reg_cache_step = 2, .reg_cache_default = sgtl5000_regs, .volatile_register = sgtl5000_volatile_register, + .controls = sgtl5000_snd_controls, + .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), + .dapm_widgets = sgtl5000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sgtl5000_dapm_widgets), + .dapm_routes = sgtl5000_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), }; static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, -- 1.7.1

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[alsa-devel] alsa doubt
by Debashis Bhowmick 23 Jan '12

23 Jan '12

Hi, Can anybody please explain the below doubt. Acccording to my knowledge for any audio play scenarios using alsa. ALSA has to set the data path first and once the data path is set the data flow begins. Now my question is how alsa know when the data path is set I mean how it is implemented? Thanks

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[alsa-devel] [PATCH] ASoC: Use framework for WM8996 HPOUT RMV_SHORT management
by Mark Brown 23 Jan '12

23 Jan '12

The framework is now perfactly capable of implementing the RMV_SHORT bits for us and doing so actually improves performance somewhat due to being able to combine register writes better. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 96 ++++----------------------------------------- 1 files changed, 8 insertions(+), 88 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e4b062d..e4fa104 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -68,9 +68,6 @@ struct wm8996_priv { u16 dcs_pending; struct completion dcs_done; - u16 hpout_ena; - u16 hpout_pending; - struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; int bg_ena; @@ -805,27 +802,6 @@ static int cp_event(struct snd_soc_dapm_widget *w, return 0; } -static int rmv_short_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(w->codec); - - /* Record which outputs we enabled */ - switch (event) { - case SND_SOC_DAPM_PRE_PMD: - wm8996->hpout_pending &= ~w->shift; - break; - case SND_SOC_DAPM_PRE_PMU: - wm8996->hpout_pending |= w->shift; - break; - default: - BUG(); - return -EINVAL; - } - - return 0; -} - static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) { struct i2c_client *i2c = to_i2c_client(codec->dev); @@ -864,7 +840,6 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, struct snd_soc_codec *codec = container_of(dapm, struct snd_soc_codec, dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - u16 val, mask; /* Complete any pending DC servo starts */ if (wm8996->dcs_pending) { @@ -877,57 +852,6 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, wm8996->dcs_pending = 0; } - - if (wm8996->hpout_pending != wm8996->hpout_ena) { - dev_dbg(codec->dev, "Applying RMV_SHORTs %x->%x\n", - wm8996->hpout_ena, wm8996->hpout_pending); - - val = 0; - mask = 0; - if (wm8996->hpout_pending & HPOUT1L) { - val |= WM8996_HPOUT1L_RMV_SHORT; - mask |= WM8996_HPOUT1L_RMV_SHORT; - } else { - mask |= WM8996_HPOUT1L_RMV_SHORT | - WM8996_HPOUT1L_OUTP | - WM8996_HPOUT1L_DLY; - } - - if (wm8996->hpout_pending & HPOUT1R) { - val |= WM8996_HPOUT1R_RMV_SHORT; - mask |= WM8996_HPOUT1R_RMV_SHORT; - } else { - mask |= WM8996_HPOUT1R_RMV_SHORT | - WM8996_HPOUT1R_OUTP | - WM8996_HPOUT1R_DLY; - } - - snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, mask, val); - - val = 0; - mask = 0; - if (wm8996->hpout_pending & HPOUT2L) { - val |= WM8996_HPOUT2L_RMV_SHORT; - mask |= WM8996_HPOUT2L_RMV_SHORT; - } else { - mask |= WM8996_HPOUT2L_RMV_SHORT | - WM8996_HPOUT2L_OUTP | - WM8996_HPOUT2L_DLY; - } - - if (wm8996->hpout_pending & HPOUT2R) { - val |= WM8996_HPOUT2R_RMV_SHORT; - mask |= WM8996_HPOUT2R_RMV_SHORT; - } else { - mask |= WM8996_HPOUT2R_RMV_SHORT | - WM8996_HPOUT2R_OUTP | - WM8996_HPOUT2R_DLY; - } - - snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_2, mask, val); - - wm8996->hpout_ena = wm8996->hpout_pending; - } } static int dcs_start(struct snd_soc_dapm_widget *w, @@ -1226,36 +1150,32 @@ SND_SOC_DAPM_PGA_S("HPOUT2L_DLY", 1, WM8996_ANALOGUE_HP_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_DCS", 2, WM8996_DC_SERVO_1, 2, 0, dcs_start, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HPOUT2L_OUTP", 3, WM8996_ANALOGUE_HP_2, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPOUT2L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2L, 0, - rmv_short_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPOUT2L_RMV_SHORT", 3, WM8996_ANALOGUE_HP_2, 7, 0, + NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R PGA", 0, WM8996_POWER_MANAGEMENT_1, 6, 0,NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_DLY", 1, WM8996_ANALOGUE_HP_2, 1, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_DCS", 2, WM8996_DC_SERVO_1, 3, 0, dcs_start, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HPOUT2R_OUTP", 3, WM8996_ANALOGUE_HP_2, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPOUT2R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2R, 0, - rmv_short_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPOUT2R_RMV_SHORT", 3, WM8996_ANALOGUE_HP_2, 3, 0, + NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L PGA", 0, WM8996_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_DLY", 1, WM8996_ANALOGUE_HP_1, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_DCS", 2, WM8996_DC_SERVO_1, 0, 0, dcs_start, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HPOUT1L_OUTP", 3, WM8996_ANALOGUE_HP_1, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPOUT1L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1L, 0, - rmv_short_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPOUT1L_RMV_SHORT", 3, WM8996_ANALOGUE_HP_1, 7, 0, + NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1R PGA", 0, WM8996_POWER_MANAGEMENT_1, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1R_DLY", 1, WM8996_ANALOGUE_HP_1, 1, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1R_DCS", 2, WM8996_DC_SERVO_1, 1, 0, dcs_start, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HPOUT1R_OUTP", 3, WM8996_ANALOGUE_HP_1, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPOUT1R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1R, 0, - rmv_short_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPOUT1R_RMV_SHORT", 3, WM8996_ANALOGUE_HP_1, 3, 0, + NULL, 0), SND_SOC_DAPM_OUTPUT("HPOUT1L"), SND_SOC_DAPM_OUTPUT("HPOUT1R"), -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: sgtl5000: Convert to table based DAPM and control init
by Fabio Estevam 22 Jan '12

22 Jan '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Convert to table based DAPM and control init. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 17 +++++++---------- 1 files changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7f4ba81..d6f76cf 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -227,7 +227,7 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { }; /* routes for sgtl5000 */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ @@ -1353,15 +1353,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_add_controls(codec, sgtl5000_snd_controls, - ARRAY_SIZE(sgtl5000_snd_controls)); - - snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, - ARRAY_SIZE(sgtl5000_dapm_widgets)); - - snd_soc_dapm_add_routes(&codec->dapm, audio_map, - ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(&codec->dapm); return 0; @@ -1402,6 +1393,12 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .reg_cache_step = 2, .reg_cache_default = sgtl5000_regs, .volatile_register = sgtl5000_volatile_register, + .controls = sgtl5000_snd_controls, + .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), + .dapm_widgets = sgtl500_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sgtl500_dapm_widgets), + .dapm_routes = sgtl500_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sgtl500_dapm_routes), }; static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, -- 1.7.1

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