December 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

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[alsa-devel] echo-cancellation plugin advice?
by Max Klein 27 Oct '23

27 Oct '23

> Well, asym wouldn't be the answer. Basically you'll need to open two > slave PCMs in one plugin. So, it's no simple filter-type plugin. > It's doable, but not easy as it sounds, I guess. Takashi, can you please further explain why asym is not the solution and what you see as a problem? I am also trying to figure out if it would be possible to create such a plugin for ALSA? I am still searching for some good starting points on how to write an ALSA plugin... Jean-Marc Valin (http://lists.linuxaudio.org/pipermail/linux-audio-dev/2005-June/012454.html) already released his work as part of speex but as far I know there is no plugin that has been made out of it? Thanks for any hint! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[alsa-devel] Help with wm8731 support for the Utilite (imx6q)
by Jonathan Bennett 06 Apr '16

06 Apr '16

Hey, I'm working with a compulab utilite, trying to get everything working with 3.17, with the eventual goal of getting better support for it upstream. Currently fighting the sound. It's a imx6q talking to a wm8731 over i2c/i2s. I am booting Fedora 21 arm Alpha, with my own dtb ( http://pastebin.com/4a4PBKwx) and driver (http://pastebin.com/1CsesuVV) It shows up in aplay -l, but trying to play anything to it gives me a div by 0 in a driver: [<c04f27e4>] (Ldiv0) from [<bf50512c>] (fsl_ssi_hw_params+0x194/0x364 [snd_soc_fsl_ssi]) This div0 happens in sound/soc/fsl/fsl_ssi.c, in the function fsl_ssi_set_bclk. My driver file is an almost direct copy of what compulab shipped with a 3.10 kernel. This is my first attempt at kernel hacking, so any help would be appreciated. ~Jonathan Bennett

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[alsa-devel] Trouble about Roland UA-11 UA-33 UA-55 and snd-usb-audio module
by FirebUntU 07 Mar '16

07 Mar '16

Hi guys, I write you about support of snd-usb-audio module and a relatively-new gamma of devices from Roland/Edirol: DUO CAPTURE, TRI CAPTURE and QUAD CAPTURE respectively: UA-11, UA-33, UA-55 These aren't not listed in the Alsa db. I have searched a solution for a week and I haven't found a trick to make UA-11 to work under Alsa 1.25.0 then I downloaded the driver for try to patch it and I found these devices is not listed in the tricks table In despair I have played with the switch button called "EXT * **" in the bottom of the unit of UA-11 and magically the device ID is transformed from * 0582:012b to ** 0582:0137 Image: https://dl.dropbox.com/u/32759898/output.jpg (Rear of the UA-11 unit, from the manual) *Before* (Recognized from udev - Unknown from alsa): http://www.alsa-project.org/db/?f=502ee68cf31882ba9ef23525fd66452c34d329e1 *After* (Recognized from udev - Recognized from alsa): http://www.alsa-project.org/db/?f=b2ae3e4331fb3fe613ba71e7aba1ad2e0e6dcd29 In the tricks table header of ALSA the 0582:0137 stay for PM-1000 in /TODO/ statement. I understand the EXT switch, undiscovered in the manual of the DUO CAPTURE is - as the fact - the Advanced Mode. Now the peripherical running both in playback and recording in 16 and 24 bit and without any mixer options in alsamixer. Can you update future releases with this gamma of devices? Awaiting for your positive response :) Thanks. Best regards, Leonardo

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[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

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[alsa-devel] [PATCH 0/4] ASoC: fix saif record unwork issue
by Dong Aisheng 16 Jan '16

16 Jan '16

This patch series mainly fixes the saif record unwork issue. The root cause consists of: 1) clk_prepare and clk_enable un-balanced 2) missed setting a proper initial clock rate for saif in dt 3) sgtl5000 dapm widget kcontrol does not work So can not test record on mx28evk. 4) missed setting clkmux mode for saif in dt With this patch series it can fix sgtl5000 widget kcontrol unwork and saif record unwork issue. BTW, just let people konw, i still met a few overruns during record, that's another issue and will be fixed in separte patches later. Patch based on sound-2.6/for-3.6. Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ARM: mx28evk: add missed clkmux select for saif arch/arm/mach-mxs/mach-mxs.c | 3 +++ sound/soc/codecs/sgtl5000.c | 2 -- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ 3 files changed, 27 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH] ARM: sun7i: Fix HDMI Audio driver's Alsa interface and do 8CH IEC60958 output
by thomas schorpp 13 Nov '15

13 Nov '15

Version 0.2 Users doc: All You need to get a HDMI- Pulseaudio Server on Your sunxi board to output Multichannel PCM is this line in default.pa, ttable pcms in asound.conf get ignored: +load-module module-alsa-sink device="hw:1,0" channels=8 channel_map="front-left,front-right,lfe,front-center,rear-left,rear-right,side-left,side-right" ### Automatically load driver modules depending on the hardware available .ifexists module-udev-detect.so load-module module-udev-detect Probe with speaker-test, You may need other channel mapping than I. If you run PA in system mode please read the security documentation before, see PA log for link. y tom

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[alsa-devel] Possible GPL-licensed header included in alsa-lib
by Clemens Lang 27 Jul '15

27 Jul '15

Hi, I recently reviewed the alsa-lib 1.0.27.2 source code from a licensing perspective to ensure compliant usage and distribution. It seems the intention is for the libraries to be covered by LGPL. I had a look at include/list.h, which seems to be licensed under GPL-2.0. It does not have a license header or any other copyright information, but it states it was taken from Linux 2.4.0: include/list.h, line 5: | This code was taken from the Linux 2.4.0 kernel. [jaroslav] I'm not sure one could argue that the header was trivial, especially since it does not only contain constants, structures and interface definitions, but also implementations of inline functions. I found one previous inquiry about a similar issue[1], but it does not cover the include/list.h header. Two questions: 1. Have the copyright holders of the code in question agreed to re-license under a different, LGPL-preserving license and, if yes, is there some documentation about that? 2. If the header is indeed covered by GPL-2.0, would you accept patches to replace it with a different implementation, e.g. Rusty Russell's MIT-licensed list from ccan[2]? [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-March/038267.html [2] http://ccodearchive.net/info/list.html Thank you for your time, Clemens Lang -- BMW Car IT GmbH Clemens Lang Spezialist Entwicklung Lise-Meitner-Straße 14 89081 Ulm Mail: clemens.lang(a)bmw-carit.de Web: http://www.bmw-carit.de -------------------------------------------------------------------- BMW Car IT GmbH Geschäftsführer: Michael Würtenberger und Reinhard Stolle Sitz und Registergericht: München HRB 134810 --------------------------------------------------------------------

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[alsa-devel] [V2 PATCH] ALSA: hda - Enable mute/mic-mute LEDs for more Thinkpads with Conexant codec
by Hui Wang 01 Jul '15

01 Jul '15

Most Thinkpad Edge series laptops use conexant codec, so far although the codecs have different minor Vendor Id and minor Subsystem Id, they all belong to the cxt5066 family, this change can make the mute/mic-mute LEDs support more generic among cxt_5066 family. This design refers to the similar solution for the realtek codec ALC269 family in the patch_realtek.c. Cc: Alex Hung <alex.hung(a)canonical.com> Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1..1f2717f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3244,9 +3244,29 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> +#include <acpi/acpi.h> static int (*led_set_func)(int, bool); +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + static void update_tpacpi_mute_led(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -3279,6 +3299,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3516,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} -- 1.8.1.2

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[alsa-devel] [ANNOUNCE] Audio Mini Summit 2014, Oct. 14, Düsseldorf, Germany
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This time it's along with Linux Plumbers conference, but for avoiding the conflicts with other LPC mini conferences, it'll be held on Tuesday October 14, at the same venue as LPC, Congress Center Düsseldorf, Germany. If anyone is interested, feel free to sign up in the attendee list below. It's a public accessible Google doc, so you can edit freely. http://goo.gl/VbXLPW (or https://docs.google.com/spreadsheets/d/1wBAiAyZSnMjVFJbxdXE5JoVdKGIBIWt-4tu…) The topics that have been currently raised are found in the URL below. If you have any more topics to discuss there, just put your favorite one in the URL below as well. http://goo.gl/JSasQ9 (or https://docs.google.com/spreadsheets/d/1cXzON65QYNrTXi31AIQHffUZ9NWwJuNEJAC…) Thanks! Takashi

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[alsa-devel] SoundBlaster X-fi HD SPDIF fix
by Richard Körber 05 Apr '15

05 Apr '15

Hi! I own a Creative USB SoundBlaster X-fi HD and tried to use its SPDIF optical output. However it didn't work with ALSA out of the box. The reason is that for SPDIF output device #1 must be used. The fix is: In /usr/share/alsa/cards/USB-Audio.conf, change the iec958_device block like this: USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 } Digital output works fine for me after that. BTW, I tried this line in /etc/asound.conf cards.USB-Audio.pcm.iec958_device."USB Sound Blaster HD" 1 but it didn't have any effect. Why was that? Thank you! Regards Richard Körber

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Re: [alsa-devel] ASoC: Intel: mrfld- add ACPI module
by Dan Carpenter 18 Mar '15

18 Mar '15

Hello Vinod Koul, The patch 336cfbb05edf: "ASoC: Intel: mrfld- add ACPI module" from Nov 11, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_acpi.c:313 sst_acpi_probe() info: ignoring unreachable code. sound/soc/intel/sst/sst_acpi.c 304 ret = sst_context_init(ctx); 305 if (ret < 0) 306 return ret; 307 308 /* need to save shim registers in BYT */ 309 ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), 310 GFP_KERNEL); 311 if (!ctx->shim_regs64) { 312 return -ENOMEM; ^^^^^^^^^^^^^^^ 313 goto do_sst_cleanup; ^^^^^^^^^^^^^^^^^^^^ 314 } 315 316 sst_configure_runtime_pm(ctx); 317 platform_set_drvdata(pdev, ctx); 318 return ret; 319 320 do_sst_cleanup: 321 sst_context_cleanup(ctx); 322 platform_set_drvdata(pdev, NULL); 323 dev_err(ctx->dev, "failed with %d\n", ret); 324 return ret; 325 } regards, dan carpenter

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[alsa-devel] [PATCH 0/20] ASoC: rsnd: add Synchronous SRC and U/O error salvage support
by Kuninori Morimoto 24 Feb '15

24 Feb '15

Hi Mark. These patches adds [1] Synchronous SRC, [2] under/over run error salvage support to Renesas sound driver. 1) - 16) : prepare cleanup/tidyup for 17) - 19) 17) : Synchronous SRC 18)- 19) : salvage support for under/over run error 20) : cleanup for 18) - 19) Current Renesas sound can use Asynchronous/Synchronous sampling rate convert. Asynchronous sampling rate convert is already supported on upstream. ("feature" is supported, "DT" bindings is not yet supported) 17) adds Synchronous sampling rate convert. Renesas sound has HW bug (?) which switches L/R channel if under/over run error had occurred. 18) - 19) adds salvage supported for it. 20) is cleanup patch which is based on 18)-19) patch Kuninori Morimoto (20): 1. ASoC: rsnd: add .fallback callback 2. ASoC: rsnd: add callback status check method 3. ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily 4. ASoC: rsnd: tidyup PIO/DMA mode settings method 5. ASoC: rsnd: tidyup SSI interrupt enable/disable method 6. ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() 7. ASoC: rsnd: clear status register when HW start 8. ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode 9. ASoC: rsnd: show master clock rate when ADG probe 10. ASoC: rsnd: move snd_kcontrol_new fucntions to core.c 11. ASoC: rsnd: tidyup rsnd_io_to_runtime() macro 12. ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() 13. ASoC: rsnd: initialize SRC on rsnd_src_init() 14. ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() 15. ASoC: rsnd: share SSI starting method between PIO/DMA mode 16. ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() 17. ASoC: rsnd: add Synchronous SRC mode 18. ASoC: rsnd: add salvage support for under/over flow error on SSI 19. ASoC: rsnd: add salvage support for under/over flow error on SRC 20. ASoC: rsnd: rename SSI function name of PIO include/sound/rcar_snd.h | 9 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 166 +++++++++++++++++++- sound/soc/sh/rcar/dvc.c | 177 ++-------------------- sound/soc/sh/rcar/gen.c | 15 ++ sound/soc/sh/rcar/rsnd.h | 94 +++++++++++- sound/soc/sh/rcar/src.c | 378 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 205 +++++++++++++++---------- 8 files changed, 736 insertions(+), 310 deletions(-)

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[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table
by Cyrus Harmon 03 Feb '15

03 Feb '15

Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

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[alsa-devel] [PATCH v2 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 27 Jan '15

27 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [Add driver data to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..6e3f175 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,32 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + .compatible = "wlf,wm8918", + .data = &wm8904_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

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[alsa-devel] [PATCH 0/3] ARM: edma: Correct header file usage
by Peter Ujfalusi 26 Jan '15

26 Jan '15

Hi, The linux/platform_data/edma.h file was used for API definition as well, which is not correct since the header should only contain platform data related structures, defines, etc. Mark: I think this series can go via arm since it is just changing include stuff under ASoC. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-evm: Do not include edma headers ARM: edma: Rename header file for dmaengine filter function definition ARM: edma: Split up header file to platform_data and API file arch/arm/common/edma.c | 3 +- arch/arm/mach-davinci/devices.c | 1 + arch/arm/mach-davinci/include/mach/da8xx.h | 1 + drivers/dma/edma.c | 2 +- drivers/mmc/host/davinci_mmc.c | 3 +- drivers/spi/spi-davinci.c | 2 +- include/linux/edma-dmaengine.h | 29 ++++++ include/linux/edma.h | 152 ++++++++++++++++++++++++++--- include/linux/platform_data/edma.h | 148 ++-------------------------- sound/soc/davinci/davinci-evm.c | 3 - sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/edma-pcm.c | 2 +- 12 files changed, 182 insertions(+), 165 deletions(-) create mode 100644 include/linux/edma-dmaengine.h -- 2.1.3

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[alsa-devel] [PATCH v2 1/1] alsa-lib: Add snd_pcm_start_at.
by Tim Cussins 19 Jan '15

19 Jan '15

Add notion of TSTAMP_CLASS (SYSTEM/AUDIO) as per Pierre's suggestion. Add snd_pcm_start_at() Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..c57edca 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -323,13 +323,25 @@ typedef enum _snd_pcm_tstamp { SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_ENABLE } snd_pcm_tstamp_t; +typedef enum _snd_pcm_tstamp_class { + SND_PCM_TSTAMP_CLASS_SYSTEM = 0, + SND_PCM_TSTAMP_CLASS_AUDIO, + SND_PCM_TSTAMP_CLASS_LAST = SND_PCM_TSTAMP_CLASS_AUDIO, +} snd_pcm_tstamp_class_t; + typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /** gettimeofday equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /** monotonic_raw (no NTP) */ SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +typedef enum _snd_pcm_audio_tstamp_type { + SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SND_PCM_AUDIO_TSTAMP_TYPE_LINK, + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK, +} snd_pcm_audio_tstamp_t; + /** Unsigned frames quantity */ typedef unsigned long snd_pcm_uframes_t; /** Signed frames quantity */ @@ -478,6 +490,7 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t* start_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f23cd6..1592160 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -466,12 +466,30 @@ struct snd_xfern { }; enum { + SNDRV_PCM_TSTAMP_CLASS_SYSTEM = 0, + SNDRV_PCM_TSTAMP_CLASS_AUDIO, + SNDRV_PCM_TSTAMP_CLASS_LAST = SNDRV_PCM_TSTAMP_CLASS_AUDIO, +}; + +enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, +}; + +struct snd_start_at { + int tstamp_class; + int tstamp_type; + struct timespec start_time; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -550,6 +568,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..40a4689 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,37 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param tstamp_class specifies the class of tstamp_type + * \param tstamp_type specifies the clock with which to interpret \p start_time + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec, tstamp_class or tstamp_type is invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * The timer is unconditionally cancelled upon any \a attempt to change the stream + * state e.g. drop, drain, start, start_at. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, tstamp_class, tstamp_type, start_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c34b766..0190787 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -620,6 +620,26 @@ static int snd_pcm_hw_start(snd_pcm_t *pcm) return 0; } +static int snd_pcm_hw_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time) +{ + snd_pcm_hw_t *hw = pcm->private_data; + int err; + + struct snd_start_at start_at = { + .tstamp_class = tstamp_class, + .tstamp_type = tstamp_type, + .start_time = *start_time + }; + + sync_ptr(hw, 0); + if (ioctl(hw->fd, SNDRV_PCM_IOCTL_START_AT, &start_at) < 0) { + err = -errno; + SYSMSG("SNDRV_PCM_IOCTL_START_AT failed (%i)", err); + return err; + } + return 0; +} + static int snd_pcm_hw_drop(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; @@ -1336,6 +1356,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .prepare = snd_pcm_hw_prepare, .reset = snd_pcm_hw_reset, .start = snd_pcm_hw_start, + .start_at = snd_pcm_hw_start_at, .drop = snd_pcm_hw_drop, .drain = snd_pcm_hw_drain, .pause = snd_pcm_hw_pause, diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 394505f..5cdfd3a 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -154,6 +154,7 @@ typedef struct { int (*prepare)(snd_pcm_t *pcm); int (*reset)(snd_pcm_t *pcm); int (*start)(snd_pcm_t *pcm); + int (*start_at)(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time); int (*drop)(snd_pcm_t *pcm); int (*drain)(snd_pcm_t *pcm); int (*pause)(snd_pcm_t *pcm, int enable); -- 1.7.10.4

3 10

[alsa-devel] [PATCH 1/2] ALSA: firewire-lib: remove rx_blocks_for_midi quirk
by Clemens Ladisch 16 Jan '15

16 Jan '15

There are several devices that expect to receive MIDI data only in the first eight data blocks of a packet. If the driver restricts the data rate to the allowed rate (as mandated by the specification, but not yet implemented by this driver), this happens naturally. Therefore, there is no reason to ever try to use more data packets with any device. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/firewire/amdtp.c | 10 +++++++--- sound/firewire/amdtp.h | 3 --- sound/firewire/bebob/bebob_stream.c | 7 ------- sound/firewire/fireworks/fireworks_stream.c | 5 ----- 4 files changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 3badc70..17548d8 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -21,6 +21,12 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* + * Several devices look only at the first eight data blocks. + * In any case, this is more than enough for the MIDI data rate. + */ +#define MAX_MIDI_RX_BLOCKS 8 + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 Âµs */ /* isochronous header parameters */ @@ -78,8 +84,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->callbacked = false; s->sync_slave = NULL; - s->rx_blocks_for_midi = UINT_MAX; - return 0; } EXPORT_SYMBOL(amdtp_stream_init); @@ -474,7 +478,7 @@ static void amdtp_fill_midi(struct amdtp_stream *s, b = (u8 *)&buffer[s->midi_position]; port = (s->data_block_counter + f) % 8; - if ((f >= s->rx_blocks_for_midi) || + if ((f >= MAX_MIDI_RX_BLOCKS) || (s->midi[port] == NULL) || (snd_rawmidi_transmit(s->midi[port], b + 1, 1) <= 0)) b[0] = 0x80; diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index e6e8926..cd4c4df 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -152,9 +152,6 @@ struct amdtp_stream { /* quirk: fixed interval of dbc between previos/current packets. */ unsigned int tx_dbc_interval; - /* quirk: the first count of data blocks in an rx packet for MIDI */ - unsigned int rx_blocks_for_midi; - bool callbacked; wait_queue_head_t callback_wait; struct amdtp_stream *sync_slave; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 1aab0a32..0ebcabf 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,13 +484,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) amdtp_stream_destroy(&bebob->rx_stream); destroy_both_connections(bebob); } - /* - * The firmware for these devices ignore MIDI messages in more than - * first 8 data blocks of an received AMDTP packet. - */ - if (bebob->spec == &maudio_fw410_spec || - bebob->spec == &maudio_special_spec) - bebob->rx_stream.rx_blocks_for_midi = 8; end: return err; } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index b985fc5..4f440e1 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -179,11 +179,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) destroy_stream(efw, &efw->tx_stream); goto end; } - /* - * Fireworks ignores MIDI messages in more than first 8 data - * blocks of an received AMDTP packet. - */ - efw->rx_stream.rx_blocks_for_midi = 8; /* set IEC61883 compliant mode (actually not fully compliant...) */ err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);

3 9

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration
by Peter Ujfalusi 14 Jan '15

14 Jan '15

In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha <jsarha(a)ti.com> (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- 2.1.0

3 6

[alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 14 Jan '15

14 Jan '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. Changes since previous version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (16): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 362 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 159 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

2 24

[alsa-devel] "ak" vs "asahi-kasei" (was: Re: [PATCH 3/3 v3] ARM: shmobile: lager: add sound support on DT)
by Geert Uytterhoeven 13 Jan '15

13 Jan '15

Hi Morimoto-san, CC devicetree, alsa-devel On Wed, Jun 11, 2014 at 9:40 AM, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: >> On Wed, Jun 11, 2014 at 8:54 AM, Kuninori Morimoto >> <kuninori.morimoto.gx(a)renesas.com> wrote: >> > + compatible = "asahi-kasei,ak4643"; >> >> As Sergei already commented, >> Documentation/devicetree/bindings/vendor-prefixes.txt >> says >> >> ak Asahi Kasei Corp. > > I know, but, > > grep asahi-kasei sound/soc/codecs/* > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4554.c: { .compatible = "asahi-kasei,ak4554" }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > ak4642 driver can't probe by "ak"; That's bad. And all but one of these match entries have been added _after_ the addition of "ak" to vendor-prefixes.txt... Outside sound, there's also "asahi-kasei,ak8975" vs. "ak,ak8975". > My code is based on "code", not "document" Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

14 44

[alsa-devel] [PATCH] ALSA: compress_driver.h: include sound/core.h explicitly
by Qais Yousef 13 Jan '15

13 Jan '15

Fixes the following compilation error: include/sound/compress_driver.h: In function ‘snd_compr_drain_notify’: include/sound/compress_driver.h:177:2: error: implicit declaration of function ‘snd_BUG_ON’ [-Werror=implicit-function-declaration] if (snd_BUG_ON(!stream)) snd_BUG_ON() is defined in sound/core.h but the file is not included explicitly, so include it. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- include/sound/compress_driver.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 396e8f73670a..1e2531058b7e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -27,6 +27,7 @@ #include <linux/types.h> #include <linux/sched.h> +#include <sound/core.h> #include <sound/compress_offload.h> #include <sound/asound.h> #include <sound/pcm.h> -- 2.1.0

3 3

Re: [alsa-devel] [FFADO-devel] [RFC] libhinawa: a light-weight I/O library for status/transactions to ALSA FireWire devices
by Jonathan Woithe 12 Jan '15

12 Jan '15

Hi Takashi S On Tue, Sep 23, 2014 at 08:39:34PM +0900, Takashi Sakamoto wrote: > For some reasons (I describe later), I start to develop a light-weight > I/O library for status/transactions to ALSA FireWire devices. > https://github.com/takaswie/libhinawa > > This library gives ways to use ALSA FireWire driver functionality such as > lock/unlock kernel streaming, EFW transaction. This is one of my > solutions to some issues between ALSA/FFADO. As I understand it, this library will be a thin wrapper around whatever kernel interface is used to communicate these details out to userspace (I think byte-type mixer controls are the preferred avenue at present). In principle I have no problem with this approach so long as it remains lightweight. The situation to avoid is the one we had with libraw1394; when the new stack came about the preferred way to interact with it switched back to the native kernel interface since (as I understand the situation) it was felt that libraw1394 didn't really provide a worthwhile API abstraction. Out of interest, is there any signficance in the library's name "hinawa"? > I hope to merge this library into alsa-tools package in future, and this > is a first RFC for it. It's under development and I require your > comments about its APIs and structure ... I will try to find some time to look at the current proposal in the next week or so. > Dice notification is an actual example. Dice based devices transfer > notification to a specific address on host controller. The address space > is exclusive resources on an system. For Dice notification, ALSA Dice > driver gives a way to utilize it for applications As well as Fireworks > situation, some userspace stuffs are needed to use this functionality. For sure. One thing to keep in mind is that while this is an issue for DICE devices it may not apply to others. The library API should be structured to keep in mind that not all features will be needed across the board. We don't want to burden all subsequent firewire-audio streaming drivers with baggage which is only required for a small subset. Regards jonathan

2 2

[alsa-devel] [PATCH 1/2] lib: devres: add a helper function for ioremap_wc
by Abhilash Kesavan 12 Jan '15

12 Jan '15

Implement a resource managed writecombine ioremap function. Signed-off-by: Abhilash Kesavan <a.kesavan(a)samsung.com> --- Documentation/driver-model/devres.txt | 1 + include/linux/io.h | 2 ++ lib/devres.c | 28 ++++++++++++++++++++++++++++ 3 files changed, 31 insertions(+) diff --git a/Documentation/driver-model/devres.txt b/Documentation/driver-model/devres.txt index b5ab416..0f80cee 100644 --- a/Documentation/driver-model/devres.txt +++ b/Documentation/driver-model/devres.txt @@ -274,6 +274,7 @@ IOMAP devm_ioport_unmap() devm_ioremap() devm_ioremap_nocache() + devm_ioremap_wc() devm_ioremap_resource() : checks resource, requests memory region, ioremaps devm_iounmap() pcim_iomap() diff --git a/include/linux/io.h b/include/linux/io.h index fa02e55..42b33f0 100644 --- a/include/linux/io.h +++ b/include/linux/io.h @@ -64,6 +64,8 @@ void __iomem *devm_ioremap(struct device *dev, resource_size_t offset, resource_size_t size); void __iomem *devm_ioremap_nocache(struct device *dev, resource_size_t offset, resource_size_t size); +void __iomem *devm_ioremap_wc(struct device *dev, resource_size_t offset, + resource_size_t size); void devm_iounmap(struct device *dev, void __iomem *addr); int check_signature(const volatile void __iomem *io_addr, const unsigned char *signature, int length); diff --git a/lib/devres.c b/lib/devres.c index 0f1dd2e..e8e1738 100644 --- a/lib/devres.c +++ b/lib/devres.c @@ -72,6 +72,34 @@ void __iomem *devm_ioremap_nocache(struct device *dev, resource_size_t offset, EXPORT_SYMBOL(devm_ioremap_nocache); /** + * devm_ioremap_wc - Managed ioremap_wc() + * @dev: Generic device to remap IO address for + * @offset: BUS offset to map + * @size: Size of map + * + * Managed ioremap_wc(). Map is automatically unmapped on driver detach. + */ +void __iomem *devm_ioremap_wc(struct device *dev, resource_size_t offset, + resource_size_t size) +{ + void __iomem **ptr, *addr; + + ptr = devres_alloc(devm_ioremap_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return NULL; + + addr = ioremap_wc(offset, size); + if (addr) { + *ptr = addr; + devres_add(dev, ptr); + } else + devres_free(ptr); + + return addr; +} +EXPORT_SYMBOL(devm_ioremap_wc); + +/** * devm_iounmap - Managed iounmap() * @dev: Generic device to unmap for * @addr: Address to unmap -- 1.7.9.5

9 14

[alsa-devel] [PATCH v3 REPOST] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 08 Jan '15

08 Jan '15

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v2->v3: - drop .owner = THIS_MODULE from the struct platform_driver initialization; - rebase to 3.18-rc7. Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 675 +++++++++++++++++++++ 7 files changed, 706 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index ddb9ac8..f1eb40f 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10628,6 +10628,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7d5d644..dcc79aa 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -55,6 +55,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 865e090..5b3c8f6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -36,3 +36,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..1cfb19e --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,675 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

2 1

[alsa-devel] [PATCH 0/5] sanitize hda/i915 interface using the component fw
by Imre Deak 08 Jan '15

08 Jan '15

The current hda/i915 interface to enable/disable power wells and query the CD clock rate is based on looking up the relevant i915 module symbols from the hda driver. By using the component framework we can get rid of some global state tracking in the i915 driver and pave the way to fully decouple the two drivers: once support is added to enable/disable the HDMI functionality dynamically in the hda driver, it can bind/unbind itself from the i915 component master, without the need to keep a reference on the i915 module. This also gets rid of the problem that currently the i915 driver exposes the interface only on HSW and BDW, while it's also needed at least on VLV/CHV. Imre Deak (5): drm/i915: add dev_to_i915_priv helper drm/i915: add component support ALSA: hda: pass chip to all i915 interface functions ALSA: hda: add component support drm/i915: remove unused power_well/get_cdclk_freq api drivers/gpu/drm/i915/i915_dma.c | 80 ++++++++++++++++++++ drivers/gpu/drm/i915/i915_drv.c | 15 ++-- drivers/gpu/drm/i915/intel_drv.h | 8 ++ drivers/gpu/drm/i915/intel_runtime_pm.c | 56 -------------- include/drm/i915_component.h | 38 ++++++++++ include/drm/i915_powerwell.h | 37 ---------- sound/pci/hda/hda_i915.c | 126 +++++++++++++++++++++----------- sound/pci/hda/hda_i915.h | 12 +-- sound/pci/hda/hda_intel.c | 16 ++-- sound/pci/hda/hda_priv.h | 7 ++ 10 files changed, 238 insertions(+), 157 deletions(-) create mode 100644 include/drm/i915_component.h delete mode 100644 include/drm/i915_powerwell.h -- 1.8.4

6 39

[alsa-devel] Persisting background noise on HDA Intel PCH (ALC3232)
by Jens John 07 Jan '15

07 Jan '15

Hello all, On my Thinkpad x240, I am experiencing exactly the same bug as someone on the x230 [0], that is I get clearly audible background noise on headphone and speaker outputs unless I mute the Headphone/Speaker control or Master. In both cases, a Realtek ALC on a Thinkpad is involved. I have attached the alsa-info.sh output. The problems occur with plain ALSA output, Jack or Pulse are not involved. I have tried all suggestions mentioned in [0], and also tested all available ALC-related model= parameters to snd_hda_intel, but the issue persists. Muting all mixers on all cards didn't change anything either. Other Thinkpad laptops with the ALC3232 codec are also affected by this bug, for example [1]. The bug makes the audio output unuseable for any serious purpose. I would be very grateful if somebody could look into this (again?), and also assist as far as I can. Thank you & Best regards -Jens. -- [0]: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080039.html [1]: http://askubuntu.com/questions/541538/headphone-hiss-after-upgrade-to-14-10 PS. I'm sorry if this is not the correct place to report this, but the bug tracker linked on the ALSA project home page seems to be down.

3 3

[alsa-devel] [PATCH] ASoC: tegra: Add platform driver for rt5677 audio codec
by Anatol Pomozov 07 Jan '15

07 Jan '15

The driver supports NVIDIA Tegra Ryu board Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 347 +++++++++++++++++++++ 4 files changed, 426 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 sound/soc/tegra/tegra_rt5677.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 0000000..a4589cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,67 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- 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 RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7..a6768f8 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,13 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588c..9171655 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 0000000..e4cf978 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,347 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#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> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static int tegra_rt5677_asoc_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 *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai = { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = &tegra_rt5677_dai, + .num_links = 1, + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + 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_rt5677_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5677_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5677_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5677_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov <anatol(a)google.com>"); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH] ASoC: wm5102: Use put_unaligned_be16
by Vaishali Thakkar 07 Jan '15

07 Jan '15

This patch introduces the use of function put_unaligned_be16. This is done using Coccinelle and semantic patch used is as follows: @a@ typedef u16, __be16, uint16_t; {u16,__be16,uint16_t} e16; identifier tmp; expression ptr; expression y,e; type T; @@ - tmp = cpu_to_be16(y); <+... when != tmp ( - memcpy(ptr, (T)&tmp, $2\|sizeof(u16)\|sizeof(__be16)\|sizeof(uint16_t)\|sizeof(e16)$); + put_unaligned_be16(y,ptr); | - memcpy(ptr, (T)&tmp, ...); + put_unaligned_be16(y,ptr); ) ...+> ? tmp = e @@ type T; identifier a.tmp; @@ - T tmp; ...when != tmp Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/codecs/wm5102.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f602349..fec21f9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -28,6 +28,7 @@ #include <linux/mfd/arizona/core.h> #include <linux/mfd/arizona/registers.h> +#include <asm/unaligned.h> #include "arizona.h" #include "wm5102.h" @@ -617,11 +618,11 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - uint16_t data; mutex_lock(&codec->mutex); - data = cpu_to_be16(arizona->dac_comp_coeff); - memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + + put_unaligned_be16(arizona->dac_comp_coeff, + ucontrol->value.bytes.data); mutex_unlock(&codec->mutex); return 0; -- 1.9.1

3 9

[alsa-devel] endianness problems in fireworks/bebob_maudio
by Clemens Ladisch 07 Jan '15

07 Jan '15

Hi, sparse complains: sound/firewire/fireworks/fireworks_transaction.c:127:18: warning: restricted __be32 degrades to integer t = (struct snd_efw_transaction *)data; length = min_t(size_t, t->length * sizeof(t->length), length); 't->length' is still a big-endian value. This means that the driver ends up always using 'length'. sound/firewire/bebob/bebob_maudio.c:100:17: warning: incorrect type in initializer (different base types) sound/firewire/bebob/bebob_maudio.c:100:17: expected restricted __be32 sound/firewire/bebob/bebob_maudio.c:100:17: got int __be32 cues[3] = { MAUDIO_BOOTLOADER_CUE1, MAUDIO_BOOTLOADER_CUE2, MAUDIO_BOOTLOADER_CUE3 }; rcode = fw_run_transaction(device->card, TCODE_WRITE_BLOCK_REQUEST, device->node_id, device->generation, device->max_speed, BEBOB_ADDR_REG_REQ, cues, sizeof(cues)); The three MAUDIO_BOOTLOADER_CUEx values will end up as a different byte sequence on big-endian machines. The simplest way to have these twelve bytes unchanged on the bus is to have a twelve-byte array in the driver. Regards, Clemens

3 5

[alsa-devel] Intel HDA / ca0132: Multichannel + S/PDIF support
by Stefan Oltmanns 07 Jan '15

07 Jan '15

Hi, I own a Gigabyte G1.Sniper M3 Z77 mainboard with ca0132 onboard audio. Currently only stereo out and headphone out (at the back, always same signal) work by default. I was able to fix S/PDIF out by changing the headphone pin in function ca0132_config to something below 0x0d, like: spec->out_pins[1] = 0x0a; No idea why this fixes S/PDIF, but no other effects noticed. No idea if it´s going to break something on other devices. Original value is 0x10. In this function also 3 DACs are listed. The board has a total of 5 stereo outputs: -Frontpanel (HDA pin connector on mainboard) 4x 3,5mm output at backpanel: -Stereo out -Headphone out -Rear out -Center/LFE out I tried to increase the number of channels and create additional outputs and connected to the other DACs. The maximum I´m able to is to get different signals out of stereo out and front headphone (not useful, just some weird mixes) or same signal from all 3 outputs. Rear and c/lfe out always no signal. During initialization snd_hda_parse_pin_def_config is called and two outputs are found. I tried to override that and change spec->autocfg after that function is called, no success. Also no matter what I change there, a little later I get this kernel messages: kernel: [ 2.642972] input: HDA Intel PCH Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 kernel: [ 2.643142] input: HDA Intel PCH Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input7 kernel: [ 2.644932] input: HDA Intel PCH Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input6 kernel: [ 2.645050] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input5 I could not find the code where this messages are generated, where is that? Any ideas what I could/should try to get multichannel support working? Is there any kind of documentation on the ca0132 that could be helpful? Best regards Stefan

3 8

[alsa-devel] [PATCH] ASoC: tegra: Add platform driver for rt5677 audio codec
by Anatol Pomozov 06 Jan '15

06 Jan '15

The driver supports NVIDIA Tegra Ryu board Sponsored: Google ChromeOS Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++++ sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 351 +++++++++++++++++++++ 4 files changed, 430 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 sound/soc/tegra/tegra_rt5677.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt new file mode 100644 index 0000000..a4589cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt @@ -0,0 +1,67 @@ +NVIDIA Tegra audio complex, with RT5677 CODEC + +Required properties: +- compatible : "nvidia,tegra-audio-rt5677" +- clocks : Must contain an entry for each entry in clock-names. + See ../clocks/clock-bindings.txt for details. +- clock-names : Must include the following entries: + - pll_a + - pll_a_out0 + - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) +- 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 RT5677's pins (as documented in its binding), and the jacks + on the board: + + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + +- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's + connected to the CODEC. +- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding + assumes that AIF1 on the CODEC is connected to Tegra. + +Optional properties: +- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in +- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier +- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in +- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal + +Example: + +sound { + compatible = "nvidia,tegra-audio-rt5677-ryu", + "nvidia,tegra-audio-rt5677"; + nvidia,model = "NVIDIA Tegra Ryu"; + + nvidia,audio-routing = + "Headphone", "LOUT2", + "Headphone", "LOUT1", + "Headset Mic", "MICBIAS1", + "IN1P", "Headset Mic", + "IN1N", "Headset Mic", + "DMIC L1", "Internal Mic 1", + "DMIC R1", "Internal Mic 1", + "DMIC L2", "Internal Mic 2", + "DMIC R2", "Internal Mic 2", + "Speaker", "PDM1L", + "Speaker", "PDM1R"; + + nvidia,i2s-controller = <&tegra_i2s1>; + nvidia,audio-codec = <&rt5677>; + + nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; + nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; + nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; + + clocks = <&tegra_car TEGRA124_CLK_PLL_A>, + <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA124_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; +}; diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 31198cf7..a6768f8 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -128,3 +128,13 @@ config SND_SOC_TEGRA_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the MAX98090 codec, such as Venice2. + +config SND_SOC_TEGRA_RT5677 + tristate "SoC Audio support for Tegra boards using a RT5677 codec" + depends on SND_SOC_TEGRA && I2C && GPIOLIB + select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC + select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC + select SND_SOC_RT5677 + help + Say Y or M here if you want to add support for SoC audio on Tegra + boards using the RT5677 codec, such as Ryu. diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 5ae588c..9171655 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -19,6 +19,7 @@ obj-$(CONFIG_SND_SOC_TEGRA30_I2S) += snd-soc-tegra30-i2s.o # Tegra machine Support snd-soc-tegra-rt5640-objs := tegra_rt5640.o +snd-soc-tegra-rt5677-objs := tegra_rt5677.o snd-soc-tegra-wm8753-objs := tegra_wm8753.o snd-soc-tegra-wm8903-objs := tegra_wm8903.o snd-soc-tegra-wm9712-objs := tegra_wm9712.o @@ -27,6 +28,7 @@ snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o +obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o obj-$(CONFIG_SND_SOC_TEGRA_WM8753) += snd-soc-tegra-wm8753.o obj-$(CONFIG_SND_SOC_TEGRA_WM8903) += snd-soc-tegra-wm8903.o obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c new file mode 100644 index 0000000..6e98d10 --- /dev/null +++ b/sound/soc/tegra/tegra_rt5677.c @@ -0,0 +1,351 @@ +/* +* tegra_rt5677.c - Tegra machine ASoC driver for boards using RT5677 codec. + * + * Copyright (c) 2014, The Chromium OS Authors. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + * Based on code copyright/by: + * + * Copyright (C) 2010-2012 - NVIDIA, Inc. + * Copyright (C) 2011 The AC100 Kernel Team <ac100(a)lists.lauchpad.net> + * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd. + * Copyright 2007 Wolfson Microelectronics PLC. + */ + +#include <linux/module.h> +#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> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "../codecs/rt5677.h" + +#include "tegra_asoc_utils.h" + +#define DRV_NAME "tegra-snd-rt5677" + +struct tegra_rt5677 { + struct tegra_asoc_utils_data util_data; + int gpio_hp_det; + int gpio_hp_en; + int gpio_mic_present; + int gpio_dmic_clk_en; +}; + +static int tegra_rt5677_asoc_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 *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + int srate, mclk, err; + + srate = params_rate(params); + mclk = 256 * srate; + + err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk); + if (err < 0) { + dev_err(card->dev, "Can't configure clocks\n"); + return err; + } + + err = snd_soc_dai_set_sysclk(codec_dai, RT5677_SCLK_S_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (err < 0) { + dev_err(card->dev, "codec_dai clock not set\n"); + return err; + } + + return 0; +} + +static int tegra_rt5677_event_hp(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (!gpio_is_valid(machine->gpio_hp_en)) + return 0; + + gpio_set_value_cansleep(machine->gpio_hp_en, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static struct snd_soc_ops tegra_rt5677_ops = { + .hw_params = tegra_rt5677_asoc_hw_params, +}; + +static struct snd_soc_jack tegra_rt5677_hp_jack; + +static struct snd_soc_jack_pin tegra_rt5677_hp_jack_pins = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; +static struct snd_soc_jack_gpio tegra_rt5677_hp_jack_gpio = { + .name = "Headphone detection", + .report = SND_JACK_HEADPHONE, + .debounce_time = 150, +}; + +static struct snd_soc_jack tegra_rt5677_mic_jack; + +static struct snd_soc_jack_pin tegra_rt5677_mic_jack_pins = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static struct snd_soc_jack_gpio tegra_rt5677_mic_jack_gpio = { + .name = "Headset Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1 +}; + +static const struct snd_soc_dapm_widget tegra_rt5677_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Headphone", tegra_rt5677_event_hp), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic 1", NULL), + SND_SOC_DAPM_MIC("Internal Mic 2", NULL), +}; + +static const struct snd_kcontrol_new tegra_rt5677_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic 1"), + SOC_DAPM_PIN_SWITCH("Internal Mic 2"), +}; + +static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); + + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack); + snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_pins); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack); + snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_pins); + + if (gpio_is_valid(machine->gpio_mic_present)) { + tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; + snd_soc_jack_add_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + + return 0; +} + +static int tegra_rt5677_card_remove(struct snd_soc_card *card) +{ + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + if (gpio_is_valid(machine->gpio_hp_det)) { + snd_soc_jack_free_gpios(&tegra_rt5677_hp_jack, 1, + &tegra_rt5677_hp_jack_gpio); + } + + if (gpio_is_valid(machine->gpio_mic_present)) { + snd_soc_jack_free_gpios(&tegra_rt5677_mic_jack, 1, + &tegra_rt5677_mic_jack_gpio); + } + + return 0; +} + +static struct snd_soc_dai_link tegra_rt5677_dai = { + .name = "RT5677", + .stream_name = "RT5677 PCM", + .codec_dai_name = "rt5677-aif1", + .init = tegra_rt5677_asoc_init, + .ops = &tegra_rt5677_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card snd_soc_tegra_rt5677 = { + .name = "tegra-rt5677", + .owner = THIS_MODULE, + .remove = tegra_rt5677_card_remove, + .dai_link = &tegra_rt5677_dai, + .num_links = 1, + .controls = tegra_rt5677_controls, + .num_controls = ARRAY_SIZE(tegra_rt5677_controls), + .dapm_widgets = tegra_rt5677_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tegra_rt5677_dapm_widgets), + .fully_routed = true, +}; + +static int tegra_rt5677_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &snd_soc_tegra_rt5677; + struct tegra_rt5677 *machine; + int ret; + +#if defined(CONFIG_MODULES) && defined(MODULE) + request_module("snd_soc_tegra30_i2s"); +#endif + + machine = devm_kzalloc(&pdev->dev, + sizeof(struct tegra_rt5677), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (machine->gpio_hp_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_mic_present = of_get_named_gpio(np, + "nvidia,mic-present-gpios", 0); + if (machine->gpio_mic_present == -EPROBE_DEFER) + return -EPROBE_DEFER; + + machine->gpio_hp_en = of_get_named_gpio(np, "nvidia,hp-en-gpios", 0); + if (machine->gpio_hp_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_hp_en)) { + ret = devm_gpio_request_one(&pdev->dev, machine->gpio_hp_en, + GPIOF_OUT_INIT_LOW, "hp_en"); + if (ret) { + dev_err(card->dev, "cannot get hp_en gpio\n"); + return ret; + } + } + + machine->gpio_dmic_clk_en = of_get_named_gpio(np, + "nvidia,dmic-clk-en-gpios", 0); + if (machine->gpio_dmic_clk_en == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (gpio_is_valid(machine->gpio_dmic_clk_en)) { + ret = devm_gpio_request_one(&pdev->dev, + machine->gpio_dmic_clk_en, + GPIOF_OUT_INIT_HIGH, "dmic_clk_en"); + if (ret) { + dev_err(card->dev, "cannot get dmic_clk_en gpio\n"); + return ret; + } + } + + 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_rt5677_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); + if (!tegra_rt5677_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_rt5677_dai.cpu_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); + if (!tegra_rt5677_dai.cpu_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + tegra_rt5677_dai.platform_of_node = tegra_rt5677_dai.cpu_of_node; + + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); + if (ret) + goto err; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + goto err_fini_utils; + } + + return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&machine->util_data); +err: + return ret; +} + +static int tegra_rt5677_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + + tegra_asoc_utils_fini(&machine->util_data); + + return 0; +} + +static const struct of_device_id tegra_rt5677_of_match[] = { + { .compatible = "nvidia,tegra-audio-rt5677", }, + {}, +}; + +static struct platform_driver tegra_rt5677_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = tegra_rt5677_of_match, + }, + .probe = tegra_rt5677_probe, + .remove = tegra_rt5677_remove, +}; +module_platform_driver(tegra_rt5677_driver); + +MODULE_AUTHOR("Anatol Pomozov <anatol(a)google.com>"); +MODULE_DESCRIPTION("Tegra+RT5677 machine ASoC driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_rt5677_of_match); -- 2.2.0.rc0.207.ga3a616c

2 5

[alsa-devel] [PATCH 1/4] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 06 Jan '15

06 Jan '15

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK. When current bias_level is SND_SOC_BIAS_ON, it is preparing for a transition away from ON. In this case, disable the codec mclk. When current bias_level is not SND_SOC_BIAS_ON, it preparing for a transition to ON. In this case, enable the codec mclk. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..1a5f47b 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +118,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -618,6 +620,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -626,6 +629,20 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); + + if (!IS_ERR(wm8960->mclk)) { + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) + clk_disable_unprepare(wm8960->mclk); + else { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } + } break; case SND_SOC_BIAS_STANDBY: @@ -661,6 +678,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, /* Disable VMID and VREF, let them discharge */ snd_soc_write(codec, WM8960_POWER1, 0); + msleep(600); break; } @@ -674,7 +692,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg; + int reg, ret; switch (level) { case SND_SOC_BIAS_ON: @@ -715,9 +733,22 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", + ret); + return ret; + } + } break; case SND_SOC_BIAS_ON: + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -728,6 +759,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, /* Disable VMID and VREF */ snd_soc_update_bits(codec, WM8960_POWER1, WM8960_VREF | WM8960_VMID_MASK, 0); + break; case SND_SOC_BIAS_OFF: @@ -1002,6 +1034,12 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "codec_mclk"); + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); -- 1.9.1

3 7

[alsa-devel] [PATCH v3] ALSA: sound/atmel/ac97c.c: Add device tree support
by Alexander Stein 06 Jan '15

06 Jan '15

This adds device tree support for the AC97 controller. It uses the soc-ac97link bindings, but actually only ac97-reset is used. Signed-off-by: Alexander Stein <alexanders83(a)web.de> Acked-By Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Changes in v3: * Renamed the compatible string to atmel,at91sam9263-ac97c * Added 'Acked-By Alexandre Belloni <alexandre.belloni(a)free-electrons.com>' Changes in v2: * It now uses the soc-ac97link bindings (partly) instead of defining its own. This should ease the transition to ASoC once a new driver has been written. sound/atmel/ac97c.c | 45 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 42 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 4f6b14d..a0df1b1 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -22,6 +22,9 @@ #include <linux/gpio.h> #include <linux/types.h> #include <linux/io.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/initval.h> @@ -902,6 +905,40 @@ static void atmel_ac97c_reset(struct atmel_ac97c *chip) } } +#ifdef CONFIG_OF +static const struct of_device_id atmel_ac97c_dt_ids[] = { + { .compatible = "atmel,at91sam9263-ac97c", }, + { } +}; +MODULE_DEVICE_TABLE(of, atmel_ac97c_dt_ids); + +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + struct ac97c_platform_data *pdata; + struct device_node *node = dev->of_node; + const struct of_device_id *match; + + if (!node) { + dev_err(dev, "Device does not have associated DT data\n"); + return ERR_PTR(-EINVAL); + } + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return ERR_PTR(-ENOMEM); + + pdata->reset_pin = of_get_named_gpio(dev->of_node, "ac97-gpios", 2); + + return pdata; +} +#else +static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) +{ + dev_err(dev, "no platform data defined\n"); + return ERR_PTR(-ENXIO); +} +#endif + static int atmel_ac97c_probe(struct platform_device *pdev) { struct snd_card *card; @@ -922,10 +959,11 @@ static int atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pdata = pdev->dev.platform_data; + pdata = dev_get_platdata(&pdev->dev); if (!pdata) { - dev_dbg(&pdev->dev, "no platform data\n"); - return -ENXIO; + pdata = atmel_ac97c_probe_dt(&pdev->dev); + if (IS_ERR(pdata)) + return PTR_ERR(pdata); } irq = platform_get_irq(pdev, 0); @@ -1204,6 +1242,7 @@ static struct platform_driver atmel_ac97c_driver = { .driver = { .name = "atmel_ac97c", .pm = ATMEL_AC97C_PM_OPS, + .of_match_table = of_match_ptr(atmel_ac97c_dt_ids), }, }; module_platform_driver(atmel_ac97c_driver); -- 2.2.1

2 1

[alsa-devel] [PATCH v2 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 05 Jan '15

05 Jan '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: add info_ext field in hw_params and pcm_hardware ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info Documentation/sound/alsa/timestamping.txt | 176 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 71 +++++++++++- include/uapi/sound/asound.h | 31 +++++- sound/core/pcm_compat.c | 15 ++- sound/core/pcm_lib.c | 81 +++++++++----- sound/core/pcm_native.c | 41 ++++++- sound/pci/hda/hda_controller.c | 44 ++++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 417 insertions(+), 51 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

4 17

[alsa-devel] [PATCH] ASoC: adi: Add missing return statement.
by Andrew Jackson 05 Jan '15

05 Jan '15

From: Andrew Jackson <Andrew.Jackson(a)arm.com> The probe routine was disabling the clock even if the system was configured successfully. Add a return statement to leave clocks enabled. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- Spotted while reviewing clock preparation sound/soc/adi/axi-i2s.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7752860..4c23381 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -240,6 +240,8 @@ static int axi_i2s_probe(struct platform_device *pdev) if (ret) goto err_clk_disable; + return 0; + err_clk_disable: clk_disable_unprepare(i2s->clk); return ret; -- 1.7.1

3 2

[alsa-devel] [patch] ALSA: asihpi: fix an information leak in asihpi_hpi_ioctl()
by Dan Carpenter 05 Jan '15

05 Jan '15

So far as I can see "hr->h.size" is set to "res_max_size" which is a user controlled value between 12 and USHRT_MAX. If it's larger than sizeof(*hr), then that leads to an information leak. I am not very familiar with this code, my other question here is that on lines before we set "hr->h.size = sizeof(hr->h)". It think this is a bug. I also think this particular code is never executed and I added a comment to that effect. But we do it in earlier in the function as well: copy_to_user(puhr, hr, sizeof(hr->h)); It doesn't make sense to me. Anyway, I think my patch is safe and it seems to fix a real information leak. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 6aa677e..f88109a 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -283,6 +283,7 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) goto out; } + /* FIXME: isn't this a no-op? */ if (hr->h.size > res_max_size) { HPI_DEBUG_LOG(ERROR, "response too big %d %d\n", hr->h.size, res_max_size); @@ -290,6 +291,9 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) hr->h.specific_error = hr->h.size; hr->h.size = sizeof(hr->h); } + /* prevent an information leak */ + if (hr->h.size > sizeof(*hr)) + hr->h.size = sizeof(*hr); uncopied_bytes = copy_to_user(puhr, hr, hr->h.size); if (uncopied_bytes) {

3 6

[alsa-devel] DSD over HDMI: audio data packetization
by Xuebing Wang 05 Jan '15

05 Jan '15

Hi Community, I am trying to implement DSD (one bit audio) over HDMI, for Intel integrated graphics. I am referencing HDMI specification 1.3a. Regarding audio data packetization, I am referencing spec: 5.3.4 Audio Sample Packet 7.6 Audio Data Packetization Q1) Would you please point to me where is the source code to handle L-PCM audio sample packetization (to implement 5.3.4 Audio Sample Packet)? Q2) I admit that I did not carefully read HD-Audio specification. Is this possible that this L-PCM audio data packetization is done by *hardware* (e.g HDMI related widgets/pins)? Q3) Did we implement "7.2 Audio Sample Clock Capture and Regeneration"? Thanks. Xuebing Wang

3 8

[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

4 5

[alsa-devel] [PATCH v8 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 02 Jan '15

02 Jan '15

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set first adds an interface between a HDMI transmitter and the HDMI CODEC. The interface is then used by the TDA998x driver to describe its audio capabilities (DAIs), to give the audio constraints of the HDMI device (EDID) to the audio subsystem, and to connect the chosen audio source to the HDMI link. v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC: hdmi: Add a transmitter interface to the HDMI CODEC drm/i2c: tda998x: Use the HDMI audio CODEC interface .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 264 ++++++++++++++++++++- include/sound/hdmi.h | 20 ++ sound/soc/codecs/hdmi.c | 176 +++++++++++++- 5 files changed, 460 insertions(+), 19 deletions(-) create mode 100644 include/sound/hdmi.h -- 2.1.1

3 7

[alsa-devel] [PATCH] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 02 Jan '15

02 Jan '15

In soc_new_compress() when rtd->dai_link->daynmic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by removing this line of code since CAPTURE substream will always be NULL. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- Not sure if this is the correct fix but that's what I could come up with my limited knowledge. I think the more correct solution would be to use the value of rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture in the args of snd_pcm_new_internal() for playback_count and capture_count. sound/soc/soc-compress.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..7ab39f65384c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -669,7 +669,6 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

2 2

[alsa-devel] [PATCH v2] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Rickard Strandqvist 01 Jan '15

01 Jan '15

Removes some functions that are not used anywhere: hpi_stream_response_to_legacy() hpi_outstream_host_buffer_free() hpi_outstream_host_buffer_get_info() hpi_outstream_host_buffer_allocate() hpi_outstream_set_time_scale() hpi_outstream_ancillary_read() hpi_outstream_ancillary_get_info() hpi_outstream_ancillary_reset() hpi_outstream_set_punch_in_out() hpi_outstream_set_velocity() hpi_outstream_set_format() hpi_outstream_sinegen() hpi_outstream_wait_start() hpi_stream_estimate_buffer_size() hpi_vox_get_threshold() hpi_vox_set_threshold() hpi_volume_query_auto_fade_profile() hpi_volume_auto_fade() hpi_adapter_enumerate_property() hpi_volume_query_channels() hpi_pad_get_rdsPI() hpi_pad_get_program_type() hpi_pad_get_comment() hpi_pad_get_title() hpi_pad_get_artist() hpi_pad_get_channel_name() hpi_tuner_get_rds() hpi_tuner_set_hd_radio_signal_blend() hpi_tuner_get_hd_radio_signal_blend() hpi_tuner_get_hd_radio_signal_quality() hpi_tuner_get_mode() hpi_tuner_set_mode() hpi_tuner_get_status() hpi_tuner_get_hd_radio_sdk_version() hpi_tuner_get_hd_radio_dsp_version() hpi_tuner_get_program() hpi_tuner_set_program() hpi_tuner_query_program() hpi_tuner_get_deemphasis() hpi_tuner_set_deemphasis() hpi_tuner_query_deemphasis() hpi_tuner_get_raw_rf_level() hpi_tuner_get_rf_level() hpi_silence_detector_get_threshold() hpi_silence_detector_set_threshold() hpi_silence_detector_get_delay() hpi_silence_detector_set_delay() hpi_silence_detector_get_event_enable() hpi_silence_detector_set_event_enable() hpi_silence_detector_get_enable() hpi_silence_detector_set_enable() hpi_silence_detector_get_state() hpi_tone_detector_get_threshold() hpi_tone_detector_set_threshold() hpi_tone_detector_get_event_enable() hpi_tone_detector_set_event_enable() hpi_tone_detector_get_enable() hpi_volume_auto_fade_profile() hpi_tone_detector_set_enable() hpi_tone_detector_get_state() hpi_tone_detector_get_frequency() hpi_sample_clock_get_local_rate_lock() hpi_sample_clock_set_local_rate_lock() hpi_sample_clock_get_auto() hpi_sample_clock_set_auto() hpi_parametric_eq_get_coeffs() hpi_parametric_eq_set_band() hpi_parametric_eq_get_band() hpi_parametric_eq_set_state() hpi_parametric_eq_get_info() hpi_adapter_get_module_by_index() hpi_microphone_get_phantom_power() hpi_microphone_set_phantom_power() hpi_meter_get_peak_ballistics() hpi_meter_set_peak_ballistics() hpi_meter_get_rms_ballistics() hpi_meter_set_rms_ballistics() hpi_meter_get_rms() hpi_compander_get_ratio() hpi_compander_set_ratio() hpi_compander_get_threshold() hpi_compander_set_threshold() hpi_compander_get_decay_time_constant() hpi_compander_set_decay_time_constant() hpi_compander_get_attack_time_constant() hpi_compander_set_attack_time_constant() hpi_compander_get_makeup_gain() hpi_compander_set_makeup_gain() hpi_compander_get_enable() hpi_compander_set_enable() hpi_adapter_get_mode() hpi_cobranet_get_macaddress() hpi_cobranet_set_static_ip_address() hpi_cobranet_get_static_ip_address() hpi_cobranet_set_ip_address() hpi_cobranet_get_ip_address() hpi_cobranet_hmi_get_status() hpi_adapter_set_mode() hpi_bitstream_get_activity() hpi_bitstream_set_data_polarity() hpi_bitstream_set_clock_edge() hpi_aesebu_transmitter_query_format() hpi_aesebu_transmitter_get_channel_status() hpi_aesebu_transmitter_set_channel_status() hpi_aesebu_transmitter_set_user_data() hpi_aesebu_transmitter_set_sample_rate() hpi_aesebu_receiver_get_channel_status() hpi_aesebu_receiver_get_user_data() hpi_aesebu_receiver_get_sample_rate() hpi_adapter_close() hpi_aesebu_receiver_query_format() hpi_adapter_open() hpi_mixer_store() hpi_mixer_close() hpi_subsys_get_adapter() hpi_instream_host_buffer_get_info() hpi_subsys_get_num_adapters() hpi_instream_host_buffer_allocate() hpi_instream_ancillary_write() hpi_instream_ancillary_get_info() hpi_instream_ancillary_reset() hpi_adapter_set_mode_ex() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/hpi.h | 304 ------- sound/pci/asihpi/hpi_internal.h | 7 - sound/pci/asihpi/hpifunc.c | 1693 ++++----------------------------------- 3 files changed, 136 insertions(+), 1868 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..fa9c607 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1152,13 +1152,6 @@ struct hpi_async_event { #pragma pack(pop) #endif -/*****************/ -/* HPI FUNCTIONS */ -/*****************/ - -/* Stream */ -u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, - u32 host_polling_rate_in_milli_seconds, u32 *recommended_buffer_size); /*************/ /* SubSystem */ @@ -1166,34 +1159,14 @@ u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, u16 hpi_subsys_get_version_ex(u32 *pversion_ex); -u16 hpi_subsys_get_num_adapters(int *pn_num_adapters); - -u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, - u16 *pw_adapter_type); - /***********/ /* Adapter */ /***********/ -u16 hpi_adapter_open(u16 adapter_index); - -u16 hpi_adapter_close(u16 adapter_index); - u16 hpi_adapter_get_info(u16 adapter_index, u16 *pw_num_outstreams, u16 *pw_num_instreams, u16 *pw_version, u32 *pserial_number, u16 *pw_adapter_type); -u16 hpi_adapter_get_module_by_index(u16 adapter_index, u16 module_index, - u16 *pw_num_outputs, u16 *pw_num_inputs, u16 *pw_version, - u32 *pserial_number, u16 *pw_module_type, u32 *ph_module); - -u16 hpi_adapter_set_mode(u16 adapter_index, u32 adapter_mode); - -u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, - u16 query_or_set); - -u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode); - u16 hpi_adapter_get_assert2(u16 adapter_index, u16 *p_assert_count, char *psz_assert, u32 *p_param1, u32 *p_param2, u32 *p_dsp_string_addr, u16 *p_processor_id); @@ -1213,8 +1186,6 @@ u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 paramter1, u16 hpi_adapter_get_property(u16 adapter_index, u16 property, u16 *pw_paramter1, u16 *pw_paramter2); -u16 hpi_adapter_enumerate_property(u16 adapter_index, u16 index, - u16 what_to_enumerate, u16 property_index, u32 *psetting); /*************/ /* OutStream */ /*************/ @@ -1232,38 +1203,12 @@ u16 hpi_outstream_write_buf(u32 h_outstream, const u8 *pb_write_buf, u16 hpi_outstream_start(u32 h_outstream); -u16 hpi_outstream_wait_start(u32 h_outstream); - u16 hpi_outstream_stop(u32 h_outstream); -u16 hpi_outstream_sinegen(u32 h_outstream); - u16 hpi_outstream_reset(u32 h_outstream); u16 hpi_outstream_query_format(u32 h_outstream, struct hpi_format *p_format); -u16 hpi_outstream_set_format(u32 h_outstream, struct hpi_format *p_format); - -u16 hpi_outstream_set_punch_in_out(u32 h_outstream, u32 punch_in_sample, - u32 punch_out_sample); - -u16 hpi_outstream_set_velocity(u32 h_outstream, short velocity); - -u16 hpi_outstream_ancillary_reset(u32 h_outstream, u16 mode); - -u16 hpi_outstream_ancillary_get_info(u32 h_outstream, u32 *pframes_available); - -u16 hpi_outstream_ancillary_read(u32 h_outstream, - struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_read); - -u16 hpi_outstream_set_time_scale(u32 h_outstream, u32 time_scaleX10000); - -u16 hpi_outstream_host_buffer_allocate(u32 h_outstream, u32 size_in_bytes); - -u16 hpi_outstream_host_buffer_free(u32 h_outstream); - u16 hpi_outstream_group_add(u32 h_outstream, u32 h_stream); u16 hpi_outstream_group_get_map(u32 h_outstream, u32 *poutstream_map, @@ -1299,18 +1244,6 @@ u16 hpi_instream_get_info_ex(u32 h_instream, u16 *pw_state, u32 *pbuffer_size, u32 *pdata_recorded, u32 *psamples_recorded, u32 *pauxiliary_data_recorded); -u16 hpi_instream_ancillary_reset(u32 h_instream, u16 bytes_per_frame, - u16 mode, u16 alignment, u16 idle_bit); - -u16 hpi_instream_ancillary_get_info(u32 h_instream, u32 *pframe_space); - -u16 hpi_instream_ancillary_write(u32 h_instream, - const struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_write); - -u16 hpi_instream_host_buffer_allocate(u32 h_instream, u32 size_in_bytes); - u16 hpi_instream_host_buffer_free(u32 h_instream); u16 hpi_instream_group_add(u32 h_instream, u32 h_stream); @@ -1325,8 +1258,6 @@ u16 hpi_instream_group_reset(u32 h_instream); /*********/ u16 hpi_mixer_open(u16 adapter_index, u32 *ph_mixer); -u16 hpi_mixer_close(u32 h_mixer); - u16 hpi_mixer_get_control(u32 h_mixer, u16 src_node_type, u16 src_node_type_index, u16 dst_node_type, u16 dst_node_type_index, u16 control_type, u32 *ph_control); @@ -1335,8 +1266,6 @@ u16 hpi_mixer_get_control_by_index(u32 h_mixer, u16 control_index, u16 *pw_src_node_type, u16 *pw_src_node_index, u16 *pw_dst_node_type, u16 *pw_dst_node_index, u16 *pw_control_type, u32 *ph_control); -u16 hpi_mixer_store(u32 h_mixer, enum HPI_MIXER_STORE_COMMAND command, - u16 index); /************/ /* Controls */ /************/ @@ -1358,18 +1287,6 @@ u16 hpi_volume_get_mute(u32 h_control, u32 *mute); u16 hpi_volume_query_range(u32 h_control, short *min_gain_01dB, short *max_gain_01dB, short *step_gain_01dB); -u16 hpi_volume_query_channels(const u32 h_control, u32 *p_channels); - -u16 hpi_volume_auto_fade(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms); - -u16 hpi_volume_auto_fade_profile(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms, - u16 profile); - -u16 hpi_volume_query_auto_fade_profile(const u32 h_control, const u32 i, - u16 *profile); - /*****************/ /* Level control */ /*****************/ @@ -1392,17 +1309,6 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peak0_01dB_out[HPI_MAX_CHANNELS] ); -u16 hpi_meter_get_rms(u32 h_control, short an_peak0_01dB_out[HPI_MAX_CHANNELS] - ); - -u16 hpi_meter_set_peak_ballistics(u32 h_control, u16 attack, u16 decay); - -u16 hpi_meter_set_rms_ballistics(u32 h_control, u16 attack, u16 decay); - -u16 hpi_meter_get_peak_ballistics(u32 h_control, u16 *attack, u16 *decay); - -u16 hpi_meter_get_rms_ballistics(u32 h_control, u16 *attack, u16 *decay); - /************************/ /* ChannelMode control */ /************************/ @@ -1429,106 +1335,31 @@ u16 hpi_tuner_set_frequency(u32 h_control, u32 freq_ink_hz); u16 hpi_tuner_get_frequency(u32 h_control, u32 *pw_freq_ink_hz); -u16 hpi_tuner_get_rf_level(u32 h_control, short *pw_level); - -u16 hpi_tuner_get_raw_rf_level(u32 h_control, short *pw_level); - u16 hpi_tuner_query_gain(const u32 h_tuner, const u32 index, u16 *pw_gain); u16 hpi_tuner_set_gain(u32 h_control, short gain); u16 hpi_tuner_get_gain(u32 h_control, short *pn_gain); -u16 hpi_tuner_get_status(u32 h_control, u16 *pw_status_mask, u16 *pw_status); - -u16 hpi_tuner_set_mode(u32 h_control, u32 mode, u32 value); - -u16 hpi_tuner_get_mode(u32 h_control, u32 mode, u32 *pn_value); - -u16 hpi_tuner_get_rds(u32 h_control, char *p_rds_data); - -u16 hpi_tuner_query_deemphasis(const u32 h_tuner, const u32 index, - const u16 band, u32 *pdeemphasis); - -u16 hpi_tuner_set_deemphasis(u32 h_control, u32 deemphasis); -u16 hpi_tuner_get_deemphasis(u32 h_control, u32 *pdeemphasis); - -u16 hpi_tuner_query_program(const u32 h_tuner, u32 *pbitmap_program); - -u16 hpi_tuner_set_program(u32 h_control, u32 program); - -u16 hpi_tuner_get_program(u32 h_control, u32 *pprogram); - -u16 hpi_tuner_get_hd_radio_dsp_version(u32 h_control, char *psz_dsp_version, - const u32 string_size); - -u16 hpi_tuner_get_hd_radio_sdk_version(u32 h_control, char *psz_sdk_version, - const u32 string_size); - -u16 hpi_tuner_get_hd_radio_signal_quality(u32 h_control, u32 *pquality); - -u16 hpi_tuner_get_hd_radio_signal_blend(u32 h_control, u32 *pblend); - -u16 hpi_tuner_set_hd_radio_signal_blend(u32 h_control, const u32 blend); - /***************/ /* PAD control */ /***************/ -u16 hpi_pad_get_channel_name(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_artist(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_title(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_comment(u32 h_control, char *psz_string, - const u32 string_length); - -u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY); - -u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI); - u16 hpi_pad_get_program_type_string(u32 h_control, const u32 data_type, const u32 pTY, char *psz_string, const u32 string_length); /****************************/ /* AES/EBU Receiver control */ /****************************/ -u16 hpi_aesebu_receiver_query_format(const u32 h_aes_rx, const u32 index, - u16 *pw_format); - u16 hpi_aesebu_receiver_set_format(u32 h_control, u16 source); u16 hpi_aesebu_receiver_get_format(u32 h_control, u16 *pw_source); -u16 hpi_aesebu_receiver_get_sample_rate(u32 h_control, u32 *psample_rate); - -u16 hpi_aesebu_receiver_get_user_data(u32 h_control, u16 index, u16 *pw_data); - -u16 hpi_aesebu_receiver_get_channel_status(u32 h_control, u16 index, - u16 *pw_data); - u16 hpi_aesebu_receiver_get_error_status(u32 h_control, u16 *pw_error_data); /*******************************/ /* AES/EBU Transmitter control */ /*******************************/ -u16 hpi_aesebu_transmitter_set_sample_rate(u32 h_control, u32 sample_rate); - -u16 hpi_aesebu_transmitter_set_user_data(u32 h_control, u16 index, u16 data); - -u16 hpi_aesebu_transmitter_set_channel_status(u32 h_control, u16 index, - u16 data); - -u16 hpi_aesebu_transmitter_get_channel_status(u32 h_control, u16 index, - u16 *pw_data); - -u16 hpi_aesebu_transmitter_query_format(const u32 h_aes_tx, const u32 index, - u16 *pw_format); - u16 hpi_aesebu_transmitter_set_format(u32 h_control, u16 output_format); u16 hpi_aesebu_transmitter_get_format(u32 h_control, u16 *pw_output_format); @@ -1545,22 +1376,6 @@ u16 hpi_multiplexer_get_source(u32 h_control, u16 *source_node_type, u16 hpi_multiplexer_query_source(u32 h_control, u16 index, u16 *source_node_type, u16 *source_node_index); -/***************/ -/* Vox control */ -/***************/ -u16 hpi_vox_set_threshold(u32 h_control, short an_gain0_01dB); - -u16 hpi_vox_get_threshold(u32 h_control, short *an_gain0_01dB); - -/*********************/ -/* Bitstream control */ -/*********************/ -u16 hpi_bitstream_set_clock_edge(u32 h_control, u16 edge_type); - -u16 hpi_bitstream_set_data_polarity(u32 h_control, u16 polarity); - -u16 hpi_bitstream_get_activity(u32 h_control, u16 *pw_clk_activity, - u16 *pw_data_activity); /***********************/ /* SampleClock control */ @@ -1589,72 +1404,6 @@ u16 hpi_sample_clock_set_local_rate(u32 h_control, u32 sample_rate); u16 hpi_sample_clock_get_local_rate(u32 h_control, u32 *psample_rate); -u16 hpi_sample_clock_set_auto(u32 h_control, u32 enable); - -u16 hpi_sample_clock_get_auto(u32 h_control, u32 *penable); - -u16 hpi_sample_clock_set_local_rate_lock(u32 h_control, u32 lock); - -u16 hpi_sample_clock_get_local_rate_lock(u32 h_control, u32 *plock); - -/***********************/ -/* Microphone control */ -/***********************/ -u16 hpi_microphone_set_phantom_power(u32 h_control, u16 on_off); - -u16 hpi_microphone_get_phantom_power(u32 h_control, u16 *pw_on_off); - -/********************************/ -/* Parametric Equalizer control */ -/********************************/ -u16 hpi_parametric_eq_get_info(u32 h_control, u16 *pw_number_of_bands, - u16 *pw_enabled); - -u16 hpi_parametric_eq_set_state(u32 h_control, u16 on_off); - -u16 hpi_parametric_eq_set_band(u32 h_control, u16 index, u16 type, - u32 frequency_hz, short q100, short gain0_01dB); - -u16 hpi_parametric_eq_get_band(u32 h_control, u16 index, u16 *pn_type, - u32 *pfrequency_hz, short *pnQ100, short *pn_gain0_01dB); - -u16 hpi_parametric_eq_get_coeffs(u32 h_control, u16 index, short coeffs[5] - ); - -/*******************************/ -/* Compressor Expander control */ -/*******************************/ - -u16 hpi_compander_set_enable(u32 h_control, u32 on); - -u16 hpi_compander_get_enable(u32 h_control, u32 *pon); - -u16 hpi_compander_set_makeup_gain(u32 h_control, short makeup_gain0_01dB); - -u16 hpi_compander_get_makeup_gain(u32 h_control, short *pn_makeup_gain0_01dB); - -u16 hpi_compander_set_attack_time_constant(u32 h_control, u32 index, - u32 attack); - -u16 hpi_compander_get_attack_time_constant(u32 h_control, u32 index, - u32 *pw_attack); - -u16 hpi_compander_set_decay_time_constant(u32 h_control, u32 index, - u32 decay); - -u16 hpi_compander_get_decay_time_constant(u32 h_control, u32 index, - u32 *pw_decay); - -u16 hpi_compander_set_threshold(u32 h_control, u32 index, - short threshold0_01dB); - -u16 hpi_compander_get_threshold(u32 h_control, u32 index, - short *pn_threshold0_01dB); - -u16 hpi_compander_set_ratio(u32 h_control, u32 index, u32 ratio100); - -u16 hpi_compander_get_ratio(u32 h_control, u32 index, u32 *pw_ratio100); - /********************/ /* Cobranet control */ /********************/ @@ -1664,59 +1413,6 @@ u16 hpi_cobranet_hmi_write(u32 h_control, u32 hmi_address, u32 byte_count, u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count, u32 *pbyte_count, u8 *pb_data); -u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus, - u32 *preadable_size, u32 *pwriteable_size); - -u16 hpi_cobranet_get_ip_address(u32 h_control, u32 *pdw_ip_address); - -u16 hpi_cobranet_set_ip_address(u32 h_control, u32 dw_ip_address); - -u16 hpi_cobranet_get_static_ip_address(u32 h_control, u32 *pdw_ip_address); - -u16 hpi_cobranet_set_static_ip_address(u32 h_control, u32 dw_ip_address); - -u16 hpi_cobranet_get_macaddress(u32 h_control, u32 *p_mac_msbs, - u32 *p_mac_lsbs); - -/*************************/ -/* Tone Detector control */ -/*************************/ -u16 hpi_tone_detector_get_state(u32 hC, u32 *state); - -u16 hpi_tone_detector_set_enable(u32 hC, u32 enable); - -u16 hpi_tone_detector_get_enable(u32 hC, u32 *enable); - -u16 hpi_tone_detector_set_event_enable(u32 hC, u32 event_enable); - -u16 hpi_tone_detector_get_event_enable(u32 hC, u32 *event_enable); - -u16 hpi_tone_detector_set_threshold(u32 hC, int threshold); - -u16 hpi_tone_detector_get_threshold(u32 hC, int *threshold); - -u16 hpi_tone_detector_get_frequency(u32 hC, u32 index, u32 *frequency); - -/****************************/ -/* Silence Detector control */ -/****************************/ -u16 hpi_silence_detector_get_state(u32 hC, u32 *state); - -u16 hpi_silence_detector_set_enable(u32 hC, u32 enable); - -u16 hpi_silence_detector_get_enable(u32 hC, u32 *enable); - -u16 hpi_silence_detector_set_event_enable(u32 hC, u32 event_enable); - -u16 hpi_silence_detector_get_event_enable(u32 hC, u32 *event_enable); - -u16 hpi_silence_detector_set_delay(u32 hC, u32 delay); - -u16 hpi_silence_detector_get_delay(u32 hC, u32 *delay); - -u16 hpi_silence_detector_set_threshold(u32 hC, int threshold); - -u16 hpi_silence_detector_get_threshold(u32 hC, int *threshold); /*********************/ /* Utility functions */ /*********************/ diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index bc86cb7..7c11994 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1404,12 +1404,6 @@ void hpi_send_recv(struct hpi_message *phm, struct hpi_response *phr); u16 hpi_subsys_create_adapter(const struct hpi_resource *p_resource, u16 *pw_adapter_index); -u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status); - -u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status); - u16 hpi_adapter_restart(u16 adapter_index); /* @@ -1421,7 +1415,6 @@ compatibility. void hpi_format_to_msg(struct hpi_msg_format *pMF, const struct hpi_format *pF); -void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR); /*////////////////////////////////////////////////////////////////////////// */ /* declarations for individual HPI entry points */ diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..5d4d2f6 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -80,13 +80,6 @@ static void hpi_msg_to_format(struct hpi_format *pF, pF->unused = 0; } -void hpi_stream_response_to_legacy(struct hpi_stream_res *pSR) -{ - pSR->u.legacy_stream_info.auxiliary_data_available = - pSR->u.stream_info.auxiliary_data_available; - pSR->u.legacy_stream_info.state = pSR->u.stream_info.state; -} - static inline void hpi_send_recvV1(struct hpi_message_header *m, struct hpi_response_header *r) { @@ -105,93 +98,6 @@ u16 hpi_subsys_get_version_ex(u32 *pversion_ex) return hr.error; } -u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_NUM_ADAPTERS); - hpi_send_recv(&hm, &hr); - *pn_num_adapters = (int)hr.u.s.num_adapters; - return hr.error; -} - -u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, - u16 *pw_adapter_type) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_ADAPTER); - hm.obj_index = (u16)iterator; - hpi_send_recv(&hm, &hr); - *padapter_index = (int)hr.u.s.adapter_index; - *pw_adapter_type = hr.u.s.adapter_type; - - return hr.error; -} - -u16 hpi_adapter_open(u16 adapter_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_OPEN); - hm.adapter_index = adapter_index; - - hpi_send_recv(&hm, &hr); - - return hr.error; - -} - -u16 hpi_adapter_close(u16 adapter_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_CLOSE); - hm.adapter_index = adapter_index; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_adapter_set_mode(u16 adapter_index, u32 adapter_mode) -{ - return hpi_adapter_set_mode_ex(adapter_index, adapter_mode, - HPI_ADAPTER_MODE_SET); -} - -u16 hpi_adapter_set_mode_ex(u16 adapter_index, u32 adapter_mode, - u16 query_or_set) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_SET_MODE); - hm.adapter_index = adapter_index; - hm.u.ax.mode.adapter_mode = adapter_mode; - hm.u.ax.mode.query_or_set = query_or_set; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_adapter_get_mode(u16 adapter_index, u32 *padapter_mode) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_GET_MODE); - hm.adapter_index = adapter_index; - hpi_send_recv(&hm, &hr); - if (padapter_mode) - *padapter_mode = hr.u.ax.mode.adapter_mode; - return hr.error; -} - u16 hpi_adapter_get_info(u16 adapter_index, u16 *pw_num_outstreams, u16 *pw_num_instreams, u16 *pw_version, u32 *pserial_number, u16 *pw_adapter_type) @@ -212,30 +118,6 @@ u16 hpi_adapter_get_info(u16 adapter_index, u16 *pw_num_outstreams, return hr.error; } -u16 hpi_adapter_get_module_by_index(u16 adapter_index, u16 module_index, - u16 *pw_num_outputs, u16 *pw_num_inputs, u16 *pw_version, - u32 *pserial_number, u16 *pw_module_type, u32 *ph_module) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ADAPTER, - HPI_ADAPTER_MODULE_INFO); - hm.adapter_index = adapter_index; - hm.u.ax.module_info.index = module_index; - - hpi_send_recv(&hm, &hr); - - *pw_module_type = hr.u.ax.info.adapter_type; - *pw_num_outputs = hr.u.ax.info.num_outstreams; - *pw_num_inputs = hr.u.ax.info.num_instreams; - *pw_version = hr.u.ax.info.version; - *pserial_number = hr.u.ax.info.serial_number; - *ph_module = 0; - - return hr.error; -} - u16 hpi_adapter_set_property(u16 adapter_index, u16 property, u16 parameter1, u16 parameter2) { @@ -274,12 +156,6 @@ u16 hpi_adapter_get_property(u16 adapter_index, u16 property, return hr.error; } -u16 hpi_adapter_enumerate_property(u16 adapter_index, u16 index, - u16 what_to_enumerate, u16 property_index, u32 *psetting) -{ - return 0; -} - u16 hpi_format_create(struct hpi_format *p_format, u16 channels, u16 format, u32 sample_rate, u32 bit_rate, u32 attributes) { @@ -375,52 +251,6 @@ u16 hpi_format_create(struct hpi_format *p_format, u16 channels, u16 format, return err; } -u16 hpi_stream_estimate_buffer_size(struct hpi_format *p_format, - u32 host_polling_rate_in_milli_seconds, u32 *recommended_buffer_size) -{ - - u32 bytes_per_second; - u32 size; - u16 channels; - struct hpi_format *pF = p_format; - - channels = pF->channels; - - switch (pF->format) { - case HPI_FORMAT_PCM16_BIGENDIAN: - case HPI_FORMAT_PCM16_SIGNED: - bytes_per_second = pF->sample_rate * 2L * channels; - break; - case HPI_FORMAT_PCM24_SIGNED: - bytes_per_second = pF->sample_rate * 3L * channels; - break; - case HPI_FORMAT_PCM32_SIGNED: - case HPI_FORMAT_PCM32_FLOAT: - bytes_per_second = pF->sample_rate * 4L * channels; - break; - case HPI_FORMAT_PCM8_UNSIGNED: - bytes_per_second = pF->sample_rate * 1L * channels; - break; - case HPI_FORMAT_MPEG_L1: - case HPI_FORMAT_MPEG_L2: - case HPI_FORMAT_MPEG_L3: - bytes_per_second = pF->bit_rate / 8L; - break; - case HPI_FORMAT_DOLBY_AC2: - - bytes_per_second = 256000L / 8L; - break; - default: - return HPI_ERROR_INVALID_FORMAT; - } - size = (bytes_per_second * host_polling_rate_in_milli_seconds * 2) / - 1000L; - - *recommended_buffer_size = - roundup_pow_of_two(((size + 4095L) & ~4095L)); - return 0; -} - u16 hpi_outstream_open(u16 adapter_index, u16 outstream_index, u32 *ph_outstream) { @@ -527,12 +357,12 @@ u16 hpi_outstream_start(u32 h_outstream) return hr.error; } -u16 hpi_outstream_wait_start(u32 h_outstream) +u16 hpi_outstream_stop(u32 h_outstream) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_WAIT_START); + HPI_OSTREAM_STOP); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; @@ -541,12 +371,12 @@ u16 hpi_outstream_wait_start(u32 h_outstream) return hr.error; } -u16 hpi_outstream_stop(u32 h_outstream) +u16 hpi_outstream_reset(u32 h_outstream) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_STOP); + HPI_OSTREAM_RESET); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; @@ -555,432 +385,231 @@ u16 hpi_outstream_stop(u32 h_outstream) return hr.error; } -u16 hpi_outstream_sinegen(u32 h_outstream) +u16 hpi_outstream_query_format(u32 h_outstream, struct hpi_format *p_format) { struct hpi_message hm; struct hpi_response hr; + hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SINEGEN); + HPI_OSTREAM_QUERY_FORMAT); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_format_to_msg(&hm.u.d.u.data.format, p_format); + hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_reset(u32 h_outstream) +u16 hpi_outstream_group_add(u32 h_outstream, u32 h_stream) { struct hpi_message hm; struct hpi_response hr; + u16 adapter; + char c_obj_type; + hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_RESET); + HPI_OSTREAM_GROUP_ADD); + if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); + if (hpi_handle_indexes(h_stream, &adapter, + &hm.u.d.u.stream.stream_index)) + return HPI_ERROR_INVALID_HANDLE; + + c_obj_type = hpi_handle_object(h_stream); + switch (c_obj_type) { + case HPI_OBJ_OSTREAM: + case HPI_OBJ_ISTREAM: + hm.u.d.u.stream.object_type = c_obj_type; + break; + default: + return HPI_ERROR_INVALID_OBJ; + } + if (adapter != hm.adapter_index) + return HPI_ERROR_NO_INTERADAPTER_GROUPS; + hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_query_format(u32 h_outstream, struct hpi_format *p_format) +u16 hpi_outstream_group_get_map(u32 h_outstream, u32 *poutstream_map, + u32 *pinstream_map) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_QUERY_FORMAT); + HPI_OSTREAM_GROUP_GETMAP); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - hpi_send_recv(&hm, &hr); + if (poutstream_map) + *poutstream_map = hr.u.d.u.group_info.outstream_group_map; + if (pinstream_map) + *pinstream_map = hr.u.d.u.group_info.instream_group_map; + return hr.error; } -u16 hpi_outstream_set_format(u32 h_outstream, struct hpi_format *p_format) +u16 hpi_outstream_group_reset(u32 h_outstream) { struct hpi_message hm; struct hpi_response hr; hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_FORMAT); + HPI_OSTREAM_GROUP_RESET); if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - hpi_send_recv(&hm, &hr); - return hr.error; } -u16 hpi_outstream_set_velocity(u32 h_outstream, short velocity) +u16 hpi_instream_open(u16 adapter_index, u16 instream_index, u32 *ph_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_VELOCITY); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.velocity = velocity; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_OPEN); + hm.adapter_index = adapter_index; + hm.obj_index = instream_index; hpi_send_recv(&hm, &hr); + if (hr.error == 0) + *ph_instream = + hpi_indexes_to_handle(HPI_OBJ_ISTREAM, adapter_index, + instream_index); + else + *ph_instream = 0; + return hr.error; } -u16 hpi_outstream_set_punch_in_out(u32 h_outstream, u32 punch_in_sample, - u32 punch_out_sample) +u16 hpi_instream_close(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_PUNCHINOUT); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_HOSTBUFFER_FREE); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_send_recv(&hm, &hr); - hm.u.d.u.pio.punch_in_sample = punch_in_sample; - hm.u.d.u.pio.punch_out_sample = punch_out_sample; + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_GROUP_RESET); + hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); + hpi_send_recv(&hm, &hr); + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_CLOSE); + hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_ancillary_reset(u32 h_outstream, u16 mode) +u16 hpi_instream_query_format(u32 h_instream, + const struct hpi_format *p_format) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_ANC_RESET); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_QUERY_FORMAT); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.format.channels = mode; + hpi_format_to_msg(&hm.u.d.u.data.format, p_format); + hpi_send_recv(&hm, &hr); + return hr.error; } -u16 hpi_outstream_ancillary_get_info(u32 h_outstream, u32 *pframes_available) +u16 hpi_instream_set_format(u32 h_instream, const struct hpi_format *p_format) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_ANC_GET_INFO); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_SET_FORMAT); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_format_to_msg(&hm.u.d.u.data.format, p_format); + hpi_send_recv(&hm, &hr); - if (hr.error == 0) { - if (pframes_available) - *pframes_available = - hr.u.d.u.stream_info.data_available / - sizeof(struct hpi_anc_frame); - } + return hr.error; } -u16 hpi_outstream_ancillary_read(u32 h_outstream, - struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_read) +u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_data, u32 bytes_to_read) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_ANC_READ); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_READ); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.pb_data = (u8 *)p_anc_frame_buffer; - hm.u.d.u.data.data_size = - number_of_ancillary_frames_to_read * - sizeof(struct hpi_anc_frame); - if (hm.u.d.u.data.data_size <= anc_frame_buffer_size_in_bytes) - hpi_send_recv(&hm, &hr); - else - hr.error = HPI_ERROR_INVALID_DATASIZE; + hm.u.d.u.data.data_size = bytes_to_read; + hm.u.d.u.data.pb_data = pb_data; + + hpi_send_recv(&hm, &hr); + return hr.error; } -u16 hpi_outstream_set_time_scale(u32 h_outstream, u32 time_scale) +u16 hpi_instream_start(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_SET_TIMESCALE); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_START); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.time_scale = time_scale; - hpi_send_recv(&hm, &hr); return hr.error; } -u16 hpi_outstream_host_buffer_allocate(u32 h_outstream, u32 size_in_bytes) +u16 hpi_instream_wait_start(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_HOSTBUFFER_ALLOC); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_WAIT_START); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.data_size = size_in_bytes; + hpi_send_recv(&hm, &hr); + return hr.error; } -u16 hpi_outstream_host_buffer_get_info(u32 h_outstream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status) +u16 hpi_instream_stop(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_HOSTBUFFER_GET_INFO); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) + hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, + HPI_ISTREAM_STOP); + if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; + hpi_send_recv(&hm, &hr); - if (hr.error == 0) { - if (pp_buffer) - *pp_buffer = hr.u.d.u.hostbuffer_info.p_buffer; - if (pp_status) - *pp_status = hr.u.d.u.hostbuffer_info.p_status; - } return hr.error; } -u16 hpi_outstream_host_buffer_free(u32 h_outstream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_HOSTBUFFER_FREE); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_outstream_group_add(u32 h_outstream, u32 h_stream) -{ - struct hpi_message hm; - struct hpi_response hr; - u16 adapter; - char c_obj_type; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_GROUP_ADD); - - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - if (hpi_handle_indexes(h_stream, &adapter, - &hm.u.d.u.stream.stream_index)) - return HPI_ERROR_INVALID_HANDLE; - - c_obj_type = hpi_handle_object(h_stream); - switch (c_obj_type) { - case HPI_OBJ_OSTREAM: - case HPI_OBJ_ISTREAM: - hm.u.d.u.stream.object_type = c_obj_type; - break; - default: - return HPI_ERROR_INVALID_OBJ; - } - if (adapter != hm.adapter_index) - return HPI_ERROR_NO_INTERADAPTER_GROUPS; - - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_outstream_group_get_map(u32 h_outstream, u32 *poutstream_map, - u32 *pinstream_map) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_GROUP_GETMAP); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - - if (poutstream_map) - *poutstream_map = hr.u.d.u.group_info.outstream_group_map; - if (pinstream_map) - *pinstream_map = hr.u.d.u.group_info.instream_group_map; - - return hr.error; -} - -u16 hpi_outstream_group_reset(u32 h_outstream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_OSTREAM, - HPI_OSTREAM_GROUP_RESET); - if (hpi_handle_indexes(h_outstream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_instream_open(u16 adapter_index, u16 instream_index, u32 *ph_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_OPEN); - hm.adapter_index = adapter_index; - hm.obj_index = instream_index; - - hpi_send_recv(&hm, &hr); - - if (hr.error == 0) - *ph_instream = - hpi_indexes_to_handle(HPI_OBJ_ISTREAM, adapter_index, - instream_index); - else - *ph_instream = 0; - - return hr.error; -} - -u16 hpi_instream_close(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_HOSTBUFFER_FREE); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_GROUP_RESET); - hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); - hpi_send_recv(&hm, &hr); - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_CLOSE); - hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index); - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_query_format(u32 h_instream, - const struct hpi_format *p_format) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_QUERY_FORMAT); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_set_format(u32 h_instream, const struct hpi_format *p_format) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_SET_FORMAT); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_format_to_msg(&hm.u.d.u.data.format, p_format); - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_data, u32 bytes_to_read) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_READ); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.data_size = bytes_to_read; - hm.u.d.u.data.pb_data = pb_data; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_wait_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_WAIT_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_stop(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_STOP); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_instream_reset(u32 h_instream) +u16 hpi_instream_reset(u32 h_instream) { struct hpi_message hm; struct hpi_response hr; @@ -1022,98 +651,6 @@ u16 hpi_instream_get_info_ex(u32 h_instream, u16 *pw_state, u32 *pbuffer_size, return hr.error; } -u16 hpi_instream_ancillary_reset(u32 h_instream, u16 bytes_per_frame, - u16 mode, u16 alignment, u16 idle_bit) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_ANC_RESET); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.format.attributes = bytes_per_frame; - hm.u.d.u.data.format.format = (mode << 8) | (alignment & 0xff); - hm.u.d.u.data.format.channels = idle_bit; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_instream_ancillary_get_info(u32 h_instream, u32 *pframe_space) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_ANC_GET_INFO); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - if (pframe_space) - *pframe_space = - (hr.u.d.u.stream_info.buffer_size - - hr.u.d.u.stream_info.data_available) / - sizeof(struct hpi_anc_frame); - return hr.error; -} - -u16 hpi_instream_ancillary_write(u32 h_instream, - const struct hpi_anc_frame *p_anc_frame_buffer, - u32 anc_frame_buffer_size_in_bytes, - u32 number_of_ancillary_frames_to_write) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_ANC_WRITE); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.pb_data = (u8 *)p_anc_frame_buffer; - hm.u.d.u.data.data_size = - number_of_ancillary_frames_to_write * - sizeof(struct hpi_anc_frame); - if (hm.u.d.u.data.data_size <= anc_frame_buffer_size_in_bytes) - hpi_send_recv(&hm, &hr); - else - hr.error = HPI_ERROR_INVALID_DATASIZE; - return hr.error; -} - -u16 hpi_instream_host_buffer_allocate(u32 h_instream, u32 size_in_bytes) -{ - - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_HOSTBUFFER_ALLOC); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.d.u.data.data_size = size_in_bytes; - hpi_send_recv(&hm, &hr); - return hr.error; -} - -u16 hpi_instream_host_buffer_get_info(u32 h_instream, u8 **pp_buffer, - struct hpi_hostbuffer_status **pp_status) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_HOSTBUFFER_GET_INFO); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hpi_send_recv(&hm, &hr); - - if (hr.error == 0) { - if (pp_buffer) - *pp_buffer = hr.u.d.u.hostbuffer_info.p_buffer; - if (pp_status) - *pp_status = hr.u.d.u.hostbuffer_info.p_status; - } - return hr.error; -} - u16 hpi_instream_host_buffer_free(u32 h_instream) { @@ -1215,19 +752,6 @@ u16 hpi_mixer_open(u16 adapter_index, u32 *ph_mixer) return hr.error; } -u16 hpi_mixer_close(u32 h_mixer) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_MIXER, HPI_MIXER_CLOSE); - if (hpi_handle_indexes(h_mixer, &hm.adapter_index, NULL)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - return hr.error; -} - u16 hpi_mixer_get_control(u32 h_mixer, u16 src_node_type, u16 src_node_type_index, u16 dst_node_type, u16 dst_node_type_index, u16 control_type, u32 *ph_control) @@ -1289,20 +813,6 @@ u16 hpi_mixer_get_control_by_index(u32 h_mixer, u16 control_index, return hr.error; } -u16 hpi_mixer_store(u32 h_mixer, enum HPI_MIXER_STORE_COMMAND command, - u16 index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_MIXER, HPI_MIXER_STORE); - if (hpi_handle_indexes(h_mixer, &hm.adapter_index, NULL)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.mx.store.command = command; - hm.u.mx.store.index = index; - hpi_send_recv(&hm, &hr); - return hr.error; -} - static u16 hpi_control_param_set(const u32 h_control, const u16 attrib, const u32 param1, const u32 param2) @@ -1465,17 +975,6 @@ static u16 hpi_control_get_string(const u32 h_control, const u16 attribute, return err; } -u16 hpi_aesebu_receiver_query_format(const u32 h_aes_rx, const u32 index, - u16 *pw_format) -{ - u32 qr; - u16 err; - - err = hpi_control_query(h_aes_rx, HPI_AESEBURX_FORMAT, index, 0, &qr); - *pw_format = (u16)qr; - return err; -} - u16 hpi_aesebu_receiver_set_format(u32 h_control, u16 format) { return hpi_control_param_set(h_control, HPI_AESEBURX_FORMAT, format, @@ -1494,49 +993,6 @@ u16 hpi_aesebu_receiver_get_format(u32 h_control, u16 *pw_format) return err; } -u16 hpi_aesebu_receiver_get_sample_rate(u32 h_control, u32 *psample_rate) -{ - return hpi_control_param1_get(h_control, HPI_AESEBURX_SAMPLERATE, - psample_rate); -} - -u16 hpi_aesebu_receiver_get_user_data(u32 h_control, u16 index, u16 *pw_data) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_AESEBURX_USERDATA; - hm.u.c.param1 = index; - - hpi_send_recv(&hm, &hr); - - if (pw_data) - *pw_data = (u16)hr.u.c.param2; - return hr.error; -} - -u16 hpi_aesebu_receiver_get_channel_status(u32 h_control, u16 index, - u16 *pw_data) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_AESEBURX_CHANNELSTATUS; - hm.u.c.param1 = index; - - hpi_send_recv(&hm, &hr); - - if (pw_data) - *pw_data = (u16)hr.u.c.param2; - return hr.error; -} - u16 hpi_aesebu_receiver_get_error_status(u32 h_control, u16 *pw_error_data) { u32 error_data = 0; @@ -1549,42 +1005,6 @@ u16 hpi_aesebu_receiver_get_error_status(u32 h_control, u16 *pw_error_data) return err; } -u16 hpi_aesebu_transmitter_set_sample_rate(u32 h_control, u32 sample_rate) -{ - return hpi_control_param_set(h_control, HPI_AESEBUTX_SAMPLERATE, - sample_rate, 0); -} - -u16 hpi_aesebu_transmitter_set_user_data(u32 h_control, u16 index, u16 data) -{ - return hpi_control_param_set(h_control, HPI_AESEBUTX_USERDATA, index, - data); -} - -u16 hpi_aesebu_transmitter_set_channel_status(u32 h_control, u16 index, - u16 data) -{ - return hpi_control_param_set(h_control, HPI_AESEBUTX_CHANNELSTATUS, - index, data); -} - -u16 hpi_aesebu_transmitter_get_channel_status(u32 h_control, u16 index, - u16 *pw_data) -{ - return HPI_ERROR_INVALID_OPERATION; -} - -u16 hpi_aesebu_transmitter_query_format(const u32 h_aes_tx, const u32 index, - u16 *pw_format) -{ - u32 qr; - u16 err; - - err = hpi_control_query(h_aes_tx, HPI_AESEBUTX_FORMAT, index, 0, &qr); - *pw_format = (u16)qr; - return err; -} - u16 hpi_aesebu_transmitter_set_format(u32 h_control, u16 output_format) { return hpi_control_param_set(h_control, HPI_AESEBUTX_FORMAT, @@ -1598,39 +1018,9 @@ u16 hpi_aesebu_transmitter_get_format(u32 h_control, u16 *pw_output_format) err = hpi_control_param1_get(h_control, HPI_AESEBUTX_FORMAT, &param); if (!err && pw_output_format) - *pw_output_format = (u16)param; - - return err; -} - -u16 hpi_bitstream_set_clock_edge(u32 h_control, u16 edge_type) -{ - return hpi_control_param_set(h_control, HPI_BITSTREAM_CLOCK_EDGE, - edge_type, 0); -} - -u16 hpi_bitstream_set_data_polarity(u32 h_control, u16 polarity) -{ - return hpi_control_param_set(h_control, HPI_BITSTREAM_DATA_POLARITY, - polarity, 0); -} - -u16 hpi_bitstream_get_activity(u32 h_control, u16 *pw_clk_activity, - u16 *pw_data_activity) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_BITSTREAM_ACTIVITY; - hpi_send_recv(&hm, &hr); - if (pw_clk_activity) - *pw_clk_activity = (u16)hr.u.c.param1; - if (pw_data_activity) - *pw_data_activity = (u16)hr.u.c.param2; - return hr.error; + *pw_output_format = (u16)param; + + return err; } u16 hpi_channel_mode_query_mode(const u32 h_mode, const u32 index, @@ -1723,243 +1113,6 @@ u16 hpi_cobranet_hmi_read(u32 h_control, u32 hmi_address, u32 max_byte_count, return hr.h.error; } -u16 hpi_cobranet_hmi_get_status(u32 h_control, u32 *pstatus, - u32 *preadable_size, u32 *pwriteable_size) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hm.u.c.attribute = HPI_COBRANET_GET_STATUS; - - hpi_send_recv(&hm, &hr); - if (!hr.error) { - if (pstatus) - *pstatus = hr.u.cu.cobranet.status.status; - if (preadable_size) - *preadable_size = - hr.u.cu.cobranet.status.readable_size; - if (pwriteable_size) - *pwriteable_size = - hr.u.cu.cobranet.status.writeable_size; - } - return hr.error; -} - -u16 hpi_cobranet_get_ip_address(u32 h_control, u32 *pdw_ip_address) -{ - u32 byte_count; - u32 iP; - u16 err; - - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_currentIP, 4, &byte_count, - (u8 *)&iP); - - *pdw_ip_address = - ((iP & 0xff000000) >> 8) | ((iP & 0x00ff0000) << 8) | ((iP & - 0x0000ff00) >> 8) | ((iP & 0x000000ff) << 8); - - if (err) - *pdw_ip_address = 0; - - return err; - -} - -u16 hpi_cobranet_set_ip_address(u32 h_control, u32 dw_ip_address) -{ - u32 iP; - u16 err; - - iP = ((dw_ip_address & 0xff000000) >> 8) | ((dw_ip_address & - 0x00ff0000) << 8) | ((dw_ip_address & 0x0000ff00) >> - 8) | ((dw_ip_address & 0x000000ff) << 8); - - err = hpi_cobranet_hmi_write(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_currentIP, 4, (u8 *)&iP); - - return err; - -} - -u16 hpi_cobranet_get_static_ip_address(u32 h_control, u32 *pdw_ip_address) -{ - u32 byte_count; - u32 iP; - u16 err; - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_staticIP, 4, &byte_count, - (u8 *)&iP); - - *pdw_ip_address = - ((iP & 0xff000000) >> 8) | ((iP & 0x00ff0000) << 8) | ((iP & - 0x0000ff00) >> 8) | ((iP & 0x000000ff) << 8); - - if (err) - *pdw_ip_address = 0; - - return err; - -} - -u16 hpi_cobranet_set_static_ip_address(u32 h_control, u32 dw_ip_address) -{ - u32 iP; - u16 err; - - iP = ((dw_ip_address & 0xff000000) >> 8) | ((dw_ip_address & - 0x00ff0000) << 8) | ((dw_ip_address & 0x0000ff00) >> - 8) | ((dw_ip_address & 0x000000ff) << 8); - - err = hpi_cobranet_hmi_write(h_control, - HPI_COBRANET_HMI_cobra_ip_mon_staticIP, 4, (u8 *)&iP); - - return err; - -} - -u16 hpi_cobranet_get_macaddress(u32 h_control, u32 *p_mac_msbs, - u32 *p_mac_lsbs) -{ - u32 byte_count; - u16 err; - u32 mac; - - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_if_phy_address, 4, &byte_count, - (u8 *)&mac); - - if (!err) { - *p_mac_msbs = - ((mac & 0xff000000) >> 8) | ((mac & 0x00ff0000) << 8) - | ((mac & 0x0000ff00) >> 8) | ((mac & 0x000000ff) << - 8); - - err = hpi_cobranet_hmi_read(h_control, - HPI_COBRANET_HMI_cobra_if_phy_address + 1, 4, - &byte_count, (u8 *)&mac); - } - - if (!err) { - *p_mac_lsbs = - ((mac & 0xff000000) >> 8) | ((mac & 0x00ff0000) << 8) - | ((mac & 0x0000ff00) >> 8) | ((mac & 0x000000ff) << - 8); - } else { - *p_mac_msbs = 0; - *p_mac_lsbs = 0; - } - - return err; -} - -u16 hpi_compander_set_enable(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_ENABLE, enable, - 0); -} - -u16 hpi_compander_get_enable(u32 h_control, u32 *enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_ENABLE, enable); -} - -u16 hpi_compander_set_makeup_gain(u32 h_control, short makeup_gain0_01dB) -{ - return hpi_control_log_set2(h_control, HPI_COMPANDER_MAKEUPGAIN, - makeup_gain0_01dB, 0); -} - -u16 hpi_compander_get_makeup_gain(u32 h_control, short *makeup_gain0_01dB) -{ - return hpi_control_log_get2(h_control, HPI_COMPANDER_MAKEUPGAIN, - makeup_gain0_01dB, NULL); -} - -u16 hpi_compander_set_attack_time_constant(u32 h_control, unsigned int index, - u32 attack) -{ - return hpi_control_param_set(h_control, HPI_COMPANDER_ATTACK, attack, - index); -} - -u16 hpi_compander_get_attack_time_constant(u32 h_control, unsigned int index, - u32 *attack) -{ - return hpi_control_param_get(h_control, HPI_COMPANDER_ATTACK, 0, - index, attack, NULL); -} - -u16 hpi_compander_set_decay_time_constant(u32 h_control, unsigned int index, - u32 decay) -{ - return hpi_control_param_set(h_control, HPI_COMPANDER_DECAY, decay, - index); -} - -u16 hpi_compander_get_decay_time_constant(u32 h_control, unsigned int index, - u32 *decay) -{ - return hpi_control_param_get(h_control, HPI_COMPANDER_DECAY, 0, index, - decay, NULL); - -} - -u16 hpi_compander_set_threshold(u32 h_control, unsigned int index, - short threshold0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_COMPANDER_THRESHOLD; - hm.u.c.param2 = index; - hm.u.c.an_log_value[0] = threshold0_01dB; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_compander_get_threshold(u32 h_control, unsigned int index, - short *threshold0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_COMPANDER_THRESHOLD; - hm.u.c.param2 = index; - - hpi_send_recv(&hm, &hr); - *threshold0_01dB = hr.u.c.an_log_value[0]; - - return hr.error; -} - -u16 hpi_compander_set_ratio(u32 h_control, u32 index, u32 ratio100) -{ - return hpi_control_param_set(h_control, HPI_COMPANDER_RATIO, ratio100, - index); -} - -u16 hpi_compander_get_ratio(u32 h_control, u32 index, u32 *ratio100) -{ - return hpi_control_param_get(h_control, HPI_COMPANDER_RATIO, 0, index, - ratio100, NULL); -} - u16 hpi_level_query_range(u32 h_control, short *min_gain_01dB, short *max_gain_01dB, short *step_gain_01dB) { @@ -2030,233 +1183,48 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peakdB[HPI_MAX_CHANNELS] else for (i = 0; i < HPI_MAX_CHANNELS; i++) an_peakdB[i] = HPI_METER_MINIMUM; - return hr.error; -} - -u16 hpi_meter_get_rms(u32 h_control, short an_rmsdB[HPI_MAX_CHANNELS] - ) -{ - short i = 0; - - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_METER_RMS; - - hpi_send_recv(&hm, &hr); - - if (!hr.error) - memcpy(an_rmsdB, hr.u.c.an_log_value, - sizeof(short) * HPI_MAX_CHANNELS); - else - for (i = 0; i < HPI_MAX_CHANNELS; i++) - an_rmsdB[i] = HPI_METER_MINIMUM; - - return hr.error; -} - -u16 hpi_meter_set_rms_ballistics(u32 h_control, u16 attack, u16 decay) -{ - return hpi_control_param_set(h_control, HPI_METER_RMS_BALLISTICS, - attack, decay); -} - -u16 hpi_meter_get_rms_ballistics(u32 h_control, u16 *pn_attack, u16 *pn_decay) -{ - u32 attack; - u32 decay; - u16 error; - - error = hpi_control_param2_get(h_control, HPI_METER_RMS_BALLISTICS, - &attack, &decay); - - if (pn_attack) - *pn_attack = (unsigned short)attack; - if (pn_decay) - *pn_decay = (unsigned short)decay; - - return error; -} - -u16 hpi_meter_set_peak_ballistics(u32 h_control, u16 attack, u16 decay) -{ - return hpi_control_param_set(h_control, HPI_METER_PEAK_BALLISTICS, - attack, decay); -} - -u16 hpi_meter_get_peak_ballistics(u32 h_control, u16 *pn_attack, - u16 *pn_decay) -{ - u32 attack; - u32 decay; - u16 error; - - error = hpi_control_param2_get(h_control, HPI_METER_PEAK_BALLISTICS, - &attack, &decay); - - if (pn_attack) - *pn_attack = (short)attack; - if (pn_decay) - *pn_decay = (short)decay; - - return error; -} - -u16 hpi_microphone_set_phantom_power(u32 h_control, u16 on_off) -{ - return hpi_control_param_set(h_control, HPI_MICROPHONE_PHANTOM_POWER, - (u32)on_off, 0); -} - -u16 hpi_microphone_get_phantom_power(u32 h_control, u16 *pw_on_off) -{ - u16 error = 0; - u32 on_off = 0; - error = hpi_control_param1_get(h_control, - HPI_MICROPHONE_PHANTOM_POWER, &on_off); - if (pw_on_off) - *pw_on_off = (u16)on_off; - return error; -} - -u16 hpi_multiplexer_set_source(u32 h_control, u16 source_node_type, - u16 source_node_index) -{ - return hpi_control_param_set(h_control, HPI_MULTIPLEXER_SOURCE, - source_node_type, source_node_index); -} - -u16 hpi_multiplexer_get_source(u32 h_control, u16 *source_node_type, - u16 *source_node_index) -{ - u32 node, index; - u16 err = hpi_control_param2_get(h_control, - HPI_MULTIPLEXER_SOURCE, &node, - &index); - if (source_node_type) - *source_node_type = (u16)node; - if (source_node_index) - *source_node_index = (u16)index; - return err; -} - -u16 hpi_multiplexer_query_source(u32 h_control, u16 index, - u16 *source_node_type, u16 *source_node_index) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_MULTIPLEXER_QUERYSOURCE; - hm.u.c.param1 = index; - - hpi_send_recv(&hm, &hr); - - if (source_node_type) - *source_node_type = (u16)hr.u.c.param1; - if (source_node_index) - *source_node_index = (u16)hr.u.c.param2; - return hr.error; -} - -u16 hpi_parametric_eq_get_info(u32 h_control, u16 *pw_number_of_bands, - u16 *pw_on_off) -{ - u32 oB = 0; - u32 oO = 0; - u16 error = 0; - - error = hpi_control_param2_get(h_control, HPI_EQUALIZER_NUM_FILTERS, - &oO, &oB); - if (pw_number_of_bands) - *pw_number_of_bands = (u16)oB; - if (pw_on_off) - *pw_on_off = (u16)oO; - return error; -} - -u16 hpi_parametric_eq_set_state(u32 h_control, u16 on_off) -{ - return hpi_control_param_set(h_control, HPI_EQUALIZER_NUM_FILTERS, - on_off, 0); -} - -u16 hpi_parametric_eq_get_band(u32 h_control, u16 index, u16 *pn_type, - u32 *pfrequency_hz, short *pnQ100, short *pn_gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_EQUALIZER_FILTER; - hm.u.c.param2 = index; - - hpi_send_recv(&hm, &hr); - - if (pfrequency_hz) - *pfrequency_hz = hr.u.c.param1; - if (pn_type) - *pn_type = (u16)(hr.u.c.param2 >> 16); - if (pnQ100) - *pnQ100 = hr.u.c.an_log_value[1]; - if (pn_gain0_01dB) - *pn_gain0_01dB = hr.u.c.an_log_value[0]; - - return hr.error; -} - -u16 hpi_parametric_eq_set_band(u32 h_control, u16 index, u16 type, - u32 frequency_hz, short q100, short gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hm.u.c.param1 = frequency_hz; - hm.u.c.param2 = (index & 0xFFFFL) + ((u32)type << 16); - hm.u.c.an_log_value[0] = gain0_01dB; - hm.u.c.an_log_value[1] = q100; - hm.u.c.attribute = HPI_EQUALIZER_FILTER; + return hr.error; +} - hpi_send_recv(&hm, &hr); +u16 hpi_multiplexer_set_source(u32 h_control, u16 source_node_type, + u16 source_node_index) +{ + return hpi_control_param_set(h_control, HPI_MULTIPLEXER_SOURCE, + source_node_type, source_node_index); +} - return hr.error; +u16 hpi_multiplexer_get_source(u32 h_control, u16 *source_node_type, + u16 *source_node_index) +{ + u32 node, index; + u16 err = hpi_control_param2_get(h_control, + HPI_MULTIPLEXER_SOURCE, &node, + &index); + if (source_node_type) + *source_node_type = (u16)node; + if (source_node_index) + *source_node_index = (u16)index; + return err; } -u16 hpi_parametric_eq_get_coeffs(u32 h_control, u16 index, short coeffs[5] - ) +u16 hpi_multiplexer_query_source(u32 h_control, u16 index, + u16 *source_node_type, u16 *source_node_index) { struct hpi_message hm; struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, HPI_CONTROL_GET_STATE); if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_EQUALIZER_COEFFICIENTS; - hm.u.c.param2 = index; + hm.u.c.attribute = HPI_MULTIPLEXER_QUERYSOURCE; + hm.u.c.param1 = index; hpi_send_recv(&hm, &hr); - coeffs[0] = (short)hr.u.c.an_log_value[0]; - coeffs[1] = (short)hr.u.c.an_log_value[1]; - coeffs[2] = (short)hr.u.c.param1; - coeffs[3] = (short)(hr.u.c.param1 >> 16); - coeffs[4] = (short)hr.u.c.param2; - + if (source_node_type) + *source_node_type = (u16)hr.u.c.param1; + if (source_node_index) + *source_node_index = (u16)hr.u.c.param2; return hr.error; } @@ -2360,130 +1328,6 @@ u16 hpi_sample_clock_get_sample_rate(u32 h_control, u32 *psample_rate) return err; } -u16 hpi_sample_clock_set_auto(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_SAMPLECLOCK_AUTO, enable, - 0); -} - -u16 hpi_sample_clock_get_auto(u32 h_control, u32 *penable) -{ - return hpi_control_param1_get(h_control, HPI_SAMPLECLOCK_AUTO, - penable); -} - -u16 hpi_sample_clock_set_local_rate_lock(u32 h_control, u32 lock) -{ - return hpi_control_param_set(h_control, HPI_SAMPLECLOCK_LOCAL_LOCK, - lock, 0); -} - -u16 hpi_sample_clock_get_local_rate_lock(u32 h_control, u32 *plock) -{ - return hpi_control_param1_get(h_control, HPI_SAMPLECLOCK_LOCAL_LOCK, - plock); -} - -u16 hpi_tone_detector_get_frequency(u32 h_control, u32 index, u32 *frequency) -{ - return hpi_control_param_get(h_control, HPI_TONEDETECTOR_FREQUENCY, - index, 0, frequency, NULL); -} - -u16 hpi_tone_detector_get_state(u32 h_control, u32 *state) -{ - return hpi_control_param1_get(h_control, HPI_TONEDETECTOR_STATE, - state); -} - -u16 hpi_tone_detector_set_enable(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_ENABLE, enable, - 0); -} - -u16 hpi_tone_detector_get_enable(u32 h_control, u32 *enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_ENABLE, enable); -} - -u16 hpi_tone_detector_set_event_enable(u32 h_control, u32 event_enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_EVENT_ENABLE, - (u32)event_enable, 0); -} - -u16 hpi_tone_detector_get_event_enable(u32 h_control, u32 *event_enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_EVENT_ENABLE, - event_enable); -} - -u16 hpi_tone_detector_set_threshold(u32 h_control, int threshold) -{ - return hpi_control_param_set(h_control, HPI_TONEDETECTOR_THRESHOLD, - (u32)threshold, 0); -} - -u16 hpi_tone_detector_get_threshold(u32 h_control, int *threshold) -{ - return hpi_control_param1_get(h_control, HPI_TONEDETECTOR_THRESHOLD, - (u32 *)threshold); -} - -u16 hpi_silence_detector_get_state(u32 h_control, u32 *state) -{ - return hpi_control_param1_get(h_control, HPI_SILENCEDETECTOR_STATE, - state); -} - -u16 hpi_silence_detector_set_enable(u32 h_control, u32 enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_ENABLE, enable, - 0); -} - -u16 hpi_silence_detector_get_enable(u32 h_control, u32 *enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_ENABLE, enable); -} - -u16 hpi_silence_detector_set_event_enable(u32 h_control, u32 event_enable) -{ - return hpi_control_param_set(h_control, HPI_GENERIC_EVENT_ENABLE, - event_enable, 0); -} - -u16 hpi_silence_detector_get_event_enable(u32 h_control, u32 *event_enable) -{ - return hpi_control_param1_get(h_control, HPI_GENERIC_EVENT_ENABLE, - event_enable); -} - -u16 hpi_silence_detector_set_delay(u32 h_control, u32 delay) -{ - return hpi_control_param_set(h_control, HPI_SILENCEDETECTOR_DELAY, - delay, 0); -} - -u16 hpi_silence_detector_get_delay(u32 h_control, u32 *delay) -{ - return hpi_control_param1_get(h_control, HPI_SILENCEDETECTOR_DELAY, - delay); -} - -u16 hpi_silence_detector_set_threshold(u32 h_control, int threshold) -{ - return hpi_control_param_set(h_control, HPI_SILENCEDETECTOR_THRESHOLD, - threshold, 0); -} - -u16 hpi_silence_detector_get_threshold(u32 h_control, int *threshold) -{ - return hpi_control_param1_get(h_control, - HPI_SILENCEDETECTOR_THRESHOLD, (u32 *)threshold); -} - u16 hpi_tuner_query_band(const u32 h_tuner, const u32 index, u16 *pw_band) { u32 qr; @@ -2554,196 +1398,6 @@ u16 hpi_tuner_get_gain(u32 h_control, short *pn_gain) return error; } -u16 hpi_tuner_get_rf_level(u32 h_control, short *pw_level) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.cu.attribute = HPI_TUNER_LEVEL_AVG; - hpi_send_recv(&hm, &hr); - if (pw_level) - *pw_level = hr.u.cu.tuner.s_level; - return hr.error; -} - -u16 hpi_tuner_get_raw_rf_level(u32 h_control, short *pw_level) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.cu.attribute = HPI_TUNER_LEVEL_RAW; - hpi_send_recv(&hm, &hr); - if (pw_level) - *pw_level = hr.u.cu.tuner.s_level; - return hr.error; -} - -u16 hpi_tuner_query_deemphasis(const u32 h_tuner, const u32 index, - const u16 band, u32 *pdeemphasis) -{ - return hpi_control_query(h_tuner, HPI_TUNER_DEEMPHASIS, index, band, - pdeemphasis); -} - -u16 hpi_tuner_set_deemphasis(u32 h_control, u32 deemphasis) -{ - return hpi_control_param_set(h_control, HPI_TUNER_DEEMPHASIS, - deemphasis, 0); -} - -u16 hpi_tuner_get_deemphasis(u32 h_control, u32 *pdeemphasis) -{ - return hpi_control_param1_get(h_control, HPI_TUNER_DEEMPHASIS, - pdeemphasis); -} - -u16 hpi_tuner_query_program(const u32 h_tuner, u32 *pbitmap_program) -{ - return hpi_control_query(h_tuner, HPI_TUNER_PROGRAM, 0, 0, - pbitmap_program); -} - -u16 hpi_tuner_set_program(u32 h_control, u32 program) -{ - return hpi_control_param_set(h_control, HPI_TUNER_PROGRAM, program, - 0); -} - -u16 hpi_tuner_get_program(u32 h_control, u32 *pprogram) -{ - return hpi_control_param1_get(h_control, HPI_TUNER_PROGRAM, pprogram); -} - -u16 hpi_tuner_get_hd_radio_dsp_version(u32 h_control, char *psz_dsp_version, - const u32 string_size) -{ - return hpi_control_get_string(h_control, - HPI_TUNER_HDRADIO_DSP_VERSION, psz_dsp_version, string_size); -} - -u16 hpi_tuner_get_hd_radio_sdk_version(u32 h_control, char *psz_sdk_version, - const u32 string_size) -{ - return hpi_control_get_string(h_control, - HPI_TUNER_HDRADIO_SDK_VERSION, psz_sdk_version, string_size); -} - -u16 hpi_tuner_get_status(u32 h_control, u16 *pw_status_mask, u16 *pw_status) -{ - u32 status = 0; - u16 error = 0; - - error = hpi_control_param1_get(h_control, HPI_TUNER_STATUS, &status); - if (pw_status) { - if (!error) { - *pw_status_mask = (u16)(status >> 16); - *pw_status = (u16)(status & 0xFFFF); - } else { - *pw_status_mask = 0; - *pw_status = 0; - } - } - return error; -} - -u16 hpi_tuner_set_mode(u32 h_control, u32 mode, u32 value) -{ - return hpi_control_param_set(h_control, HPI_TUNER_MODE, mode, value); -} - -u16 hpi_tuner_get_mode(u32 h_control, u32 mode, u32 *pn_value) -{ - return hpi_control_param_get(h_control, HPI_TUNER_MODE, mode, 0, - pn_value, NULL); -} - -u16 hpi_tuner_get_hd_radio_signal_quality(u32 h_control, u32 *pquality) -{ - return hpi_control_param1_get(h_control, - HPI_TUNER_HDRADIO_SIGNAL_QUALITY, pquality); -} - -u16 hpi_tuner_get_hd_radio_signal_blend(u32 h_control, u32 *pblend) -{ - return hpi_control_param1_get(h_control, HPI_TUNER_HDRADIO_BLEND, - pblend); -} - -u16 hpi_tuner_set_hd_radio_signal_blend(u32 h_control, const u32 blend) -{ - return hpi_control_param_set(h_control, HPI_TUNER_HDRADIO_BLEND, - blend, 0); -} - -u16 hpi_tuner_get_rds(u32 h_control, char *p_data) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_TUNER_RDS; - hpi_send_recv(&hm, &hr); - if (p_data) { - *(u32 *)&p_data[0] = hr.u.cu.tuner.rds.data[0]; - *(u32 *)&p_data[4] = hr.u.cu.tuner.rds.data[1]; - *(u32 *)&p_data[8] = hr.u.cu.tuner.rds.bLER; - } - return hr.error; -} - -u16 hpi_pad_get_channel_name(u32 h_control, char *psz_string, - const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_CHANNEL_NAME, - psz_string, data_length); -} - -u16 hpi_pad_get_artist(u32 h_control, char *psz_string, const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_ARTIST, psz_string, - data_length); -} - -u16 hpi_pad_get_title(u32 h_control, char *psz_string, const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_TITLE, psz_string, - data_length); -} - -u16 hpi_pad_get_comment(u32 h_control, char *psz_string, - const u32 data_length) -{ - return hpi_control_get_string(h_control, HPI_PAD_COMMENT, psz_string, - data_length); -} - -u16 hpi_pad_get_program_type(u32 h_control, u32 *ppTY) -{ - return hpi_control_param1_get(h_control, HPI_PAD_PROGRAM_TYPE, ppTY); -} - -u16 hpi_pad_get_rdsPI(u32 h_control, u32 *ppI) -{ - return hpi_control_param1_get(h_control, HPI_PAD_PROGRAM_ID, ppI); -} - -u16 hpi_volume_query_channels(const u32 h_volume, u32 *p_channels) -{ - return hpi_control_query(h_volume, HPI_VOLUME_NUM_CHANNELS, 0, 0, - p_channels); -} - u16 hpi_volume_set_gain(u32 h_control, short an_log_gain[HPI_MAX_CHANNELS] ) { @@ -2794,78 +1448,3 @@ u16 hpi_volume_query_range(u32 h_control, short *min_gain_01dB, *step_gain_01dB = (short)hr.u.c.param1; return hr.error; } - -u16 hpi_volume_auto_fade_profile(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms, - u16 profile) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - memcpy(hm.u.c.an_log_value, an_stop_gain0_01dB, - sizeof(short) * HPI_MAX_CHANNELS); - - hm.u.c.attribute = HPI_VOLUME_AUTOFADE; - hm.u.c.param1 = duration_ms; - hm.u.c.param2 = profile; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_volume_auto_fade(u32 h_control, - short an_stop_gain0_01dB[HPI_MAX_CHANNELS], u32 duration_ms) -{ - return hpi_volume_auto_fade_profile(h_control, an_stop_gain0_01dB, - duration_ms, HPI_VOLUME_AUTOFADE_LOG); -} - -u16 hpi_volume_query_auto_fade_profile(const u32 h_volume, const u32 i, - u16 *profile) -{ - u16 e; - u32 u; - e = hpi_control_query(h_volume, HPI_VOLUME_AUTOFADE, i, 0, &u); - *profile = (u16)u; - return e; -} - -u16 hpi_vox_set_threshold(u32 h_control, short an_gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_SET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_VOX_THRESHOLD; - - hm.u.c.an_log_value[0] = an_gain0_01dB; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - -u16 hpi_vox_get_threshold(u32 h_control, short *an_gain0_01dB) -{ - struct hpi_message hm; - struct hpi_response hr; - hpi_init_message_response(&hm, &hr, HPI_OBJ_CONTROL, - HPI_CONTROL_GET_STATE); - if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - hm.u.c.attribute = HPI_VOX_THRESHOLD; - - hpi_send_recv(&hm, &hr); - - *an_gain0_01dB = hr.u.c.an_log_value[0]; - - return hr.error; -} -- 1.7.10.4

3 9

[alsa-devel] Regression for Lenovo X240 sound in 3.18
by Jan Engelhardt 01 Jan '15

01 Jan '15

I have a X240 (without any dock), and in Linux 3.17.4, I used to have - "Speaker" with switch and volume bar (in alsamixer), - "PCM" with switch and volume bar, - the headphone output volume would be controlled by both Master and PCM. With 3.18.1, bisected down to below commit, this all went bonkers. - "Speaker" has only a switch but no volume bar, - speakers do not output _any_ sound whatsoever anymore - changing the PCM volume does not affect either Headphone or Speaker. commit 7c21539c45eace48e9f6fbdd59347d8416960af5 Author: Lukas Bossard <mr.bobukas(a)gmail.com> Date: Wed Oct 29 18:31:07 2014 +0100 ALSA: hda - Add ultra dock support for Thinkpad X240. Adding ultra doch support for Lenovo Thinkpad X240 (17aa:2214). [Actually replaced the entry with ALC292_FIXUP_TPT440_DOCK -- tiwai] Reverting this commit on top of v3.18.1 restores the sought functionality.

3 7

[alsa-devel] [PATCH] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 01 Jan '15

01 Jan '15

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 10 +++++++++- 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3652c94..bf32cea 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -106,6 +106,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd6..0e88e7a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -268,6 +268,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ #define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d957..a5fe336 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -420,7 +420,8 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, hw = &substream->runtime->hw; if (!params->info) { - params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; + params->info = hw->info & ~(SNDRV_PCM_INFO_FIFO_IN_FRAMES | + SNDRV_PCM_INFO_DRAIN_TRIGGER); if (!hw_support_mmap(substream)) params->info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); @@ -1566,6 +1567,13 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) snd_pcm_post_stop(substream, new_state); } } + + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + substream->runtime->trigger_master == substream && + (substream->runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER)) + return substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH 1/3] ALSA: Fix handling of multiple msbits constraints on the same runtime
by Lars-Peter Clausen 31 Dec '14

31 Dec '14

If the sound card is made up of discrete components, each with their own driver (e.g. like in the ASoC case), we might end up with multiple msbits constraint rules installed. Currently this will result in msbits being set to whatever the last rule set it to. This patch updates the behavior of the rule to choose the minimum (other than zero) of all the installed rules. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ec9e786..b0c1535 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1300,7 +1300,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, unsigned int msbits = l >> 16; struct snd_interval *i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); if (snd_interval_single(i) && snd_interval_value(i) == width) - params->msbits = msbits; + params->msbits = min_not_zero(params->msbits, msbits); return 0; } -- 1.8.0

3 6

[alsa-devel] [PATCH] ASoC: dwc: Remove unnecessary conditional compilation
by Andrew Jackson 31 Dec '14

31 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> of_match_ptr is already conditionally compiled based on CONFIG_OF so further conditional compilation is not required. Remove conditional compilation surrounding of_match_ptr. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 99cf64b..a3e97b4 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -624,9 +624,7 @@ static struct platform_driver dw_i2s_driver = { .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", -#ifdef CONFIG_OF .of_match_table = of_match_ptr(dw_i2s_of_match), -#endif }, }; -- 1.7.1

1 0

Re: [alsa-devel] [patch v2] ALSA: asihpi: fix an information leak in asihpi_hpi_ioctl()
by Takashi Iwai 31 Dec '14

31 Dec '14

At Wed, 31 Dec 2014 23:48:32 +1300, Eliot Blennerhassett wrote: > > Add missing limits to keep copied data within allocated buffer. > > Signed-off-by: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> > --- > Regenerated, this should apply cleanly to for-next OK, applied now. Thanks. Takashi > > sound/pci/asihpi/hpi6000.c | 7 +++++-- > sound/pci/asihpi/hpioctl.c | 2 ++ > 2 files changed, 7 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c > index 2414d7a..2d63648 100644 > --- a/sound/pci/asihpi/hpi6000.c > +++ b/sound/pci/asihpi/hpi6000.c > @@ -47,7 +47,7 @@ > > /* operational/messaging errors */ > #define HPI6000_ERROR_MSG_RESP_IDLE_TIMEOUT 901 > - > +#define HPI6000_ERROR_RESP_GET_LEN 902 > #define HPI6000_ERROR_MSG_RESP_GET_RESP_ACK 903 > #define HPI6000_ERROR_MSG_GET_ADR 904 > #define HPI6000_ERROR_RESP_GET_ADR 905 > @@ -1365,7 +1365,10 @@ static short hpi6000_message_response_sequence(struct hpi_adapter_obj *pao, > length = hpi_read_word(pdo, HPI_HIF_ADDR(length)); > } while (hpi6000_check_PCI2040_error_flag(pao, H6READ) && --timeout); > if (!timeout) > - length = sizeof(struct hpi_response); > + return HPI6000_ERROR_RESP_GET_LEN; > + > + if (length > phr->size) > + return HPI_ERROR_RESPONSE_BUFFER_TOO_SMALL; > > /* get the response */ > p_data = (u32 *)phr; > diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c > index 6aa677e..72af66b 100644 > --- a/sound/pci/asihpi/hpioctl.c > +++ b/sound/pci/asihpi/hpioctl.c > @@ -153,6 +153,8 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) > goto out; > } > > + res_max_size = min_t(size_t, res_max_size, sizeof(*hr)); > + > switch (hm->h.function) { > case HPI_SUBSYS_CREATE_ADAPTER: > case HPI_ADAPTER_DELETE: > -- > 1.9.1 > >

1 0

[alsa-devel] [PATCH] pcm: fix buffer overflow in snd_pcm_chmap_print()
by Anssi Hannula 31 Dec '14

31 Dec '14

The size argument is wrong for one of the snprintf() calls in snd_pcm_chmap_print(), allowing an overflow to happen (the user-provided buffer may be written data up to 2x its actual size). Seen in an user report here: http://trac.kodi.tv/ticket/15641 Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..e74e02f 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7621,7 +7621,7 @@ int snd_pcm_chmap_print(const snd_pcm_chmap_t *map, size_t maxlen, char *buf) return -ENOMEM; } if (map->pos[i] & SND_CHMAP_DRIVER_SPEC) - len += snprintf(buf + len, maxlen, "%d", p); + len += snprintf(buf + len, maxlen - len, "%d", p); else { const char *name = chmap_names[p]; if (name) -- 1.8.4.5

2 1

[alsa-devel] [PATCH] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 31 Dec '14

31 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/pcm_native.c | 10 +++++++++- 3 files changed, 11 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3652c94..bf32cea 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -106,6 +106,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd6..0e88e7a 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -268,6 +268,7 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ #define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ typedef int __bitwise snd_pcm_state_t; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d957..bd57f6e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -420,7 +420,8 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, hw = &substream->runtime->hw; if (!params->info) { - params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES; + params->info = hw->info & ~SNDRV_PCM_INFO_FIFO_IN_FRAMES + & ~SNDRV_PCM_INFO_DRAIN_TRIGGER; if (!hw_support_mmap(substream)) params->info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); @@ -1566,6 +1567,13 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) snd_pcm_post_stop(substream, new_state); } } + + if (runtime->status->state == SNDRV_PCM_STATE_DRAINING && + substream->runtime->trigger_master == substream && + (substream->runtime->hw.info & SNDRV_PCM_INFO_DRAIN_TRIGGER)) + return substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH v4] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 30 Dec '14

30 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..aff24a7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +119,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -1002,6 +1005,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "codec_mclk"); + + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); @@ -1041,6 +1051,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1066,38 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1115,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 6

[alsa-devel] [PATCH v5 0/6] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 30 Dec '14

30 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v4->v5 + Remove all clk_put calls [Andrew Jackson] + Call clk_disable_unprepare [Andrew Jackson] Changes v3->v4 + Drop applied patches + Use combined function to prepare clock [Mark Brown] + Use managed clock resources to avoid clk_put [Mark Brown] + Read configuration prameters from hardware for both platform data and device tree [Mark Brown] + Re-order patch sequence [Mark Brown] + Change union name to avoid future collisions [Mark Brown] + Check return code from clk_set_rate [Mark Brown] + Check parametsrs read from hardware agaist array limits and add further comments [Mark Brown] + Add of_match_ptr [Mark Brown] Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (6): ASoC: dwc: Switch to managed clock resource ASoC: dwc: Prepare clock before use ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Register components with managed interface ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S .../devicetree/bindings/sound/designware-i2s.txt | 31 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 283 ++++++++++++++++---- 3 files changed, 263 insertions(+), 52 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

3 11

[alsa-devel] [PATCH] ASoC: mxs-saif: Do not store the irq number in the private structure
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The irq number is only used inside the probe function, so there is really no need to store it in the private structure. Use a local 'irq' variable to hold the the irq number instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 10 +++++----- sound/soc/mxs/mxs-saif.h | 1 - 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d986508..c866ade 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -710,7 +710,7 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores; struct mxs_saif *saif; - int ret = 0; + int irq, ret = 0; struct device_node *master; if (!np) @@ -763,16 +763,16 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - saif->irq = platform_get_irq(pdev, 0); - if (saif->irq < 0) { - ret = saif->irq; + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + ret = irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); return ret; } saif->dev = &pdev->dev; - ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0, dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index fbaf7ba..9a4c0b2 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -116,7 +116,6 @@ struct mxs_saif { unsigned int mclk; unsigned int mclk_in_use; void __iomem *base; - int irq; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Support for combined tx/rx interrupt line
by Peter Ujfalusi 30 Dec '14

30 Dec '14

Some SoC, like da850/OMAP-L138 uses one common interrupt request for TX/RX events. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 30b94d4f9c5d..95eef582331b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -364,6 +364,20 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) return IRQ_RETVAL(handled_mask); } +static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + irqreturn_t ret = IRQ_NONE; + + if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]) + ret = davinci_mcasp_tx_irq_handler(irq, data); + + if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]) + ret |= davinci_mcasp_rx_irq_handler(irq, data); + + return ret; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -1441,6 +1455,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "common"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_common_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "common IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", -- 2.2.1

2 1

[alsa-devel] [RFC] ASoC: wm8962: Allow codec to work when CONFIG_PM=n
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently the codec does not work in the CONFIG_PM=n case because the regulators and clock are kept disabled. Allow the codec to be functional by calling wm8962_runtime_resume() when CONFIG_PM=n. Tested on a imx6sl-evk in the CONFIG_PM=y and CONFIG_PM=n cases. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Probably there is a better solution, hence I am sending it as RFC. Thanks sound/soc/codecs/wm8962.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d32d554..0b041c9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3546,6 +3546,9 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, return 0; } +static int wm8962_runtime_resume(struct device *dev); +static int wm8962_runtime_suspend(struct device *dev); + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3771,6 +3774,9 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, /* The drivers should power up as needed */ regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); + if (!IS_ENABLED(CONFIG_PM)) + wm8962_runtime_resume(&i2c->dev); + return 0; err_enable: @@ -3781,11 +3787,12 @@ err: static int wm8962_i2c_remove(struct i2c_client *client) { + if (!IS_ENABLED(CONFIG_PM)) + wm8962_runtime_suspend(&client->dev); snd_soc_unregister_codec(&client->dev); return 0; } -#ifdef CONFIG_PM static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); @@ -3860,7 +3867,6 @@ static int wm8962_runtime_suspend(struct device *dev) return 0; } -#endif static struct dev_pm_ops wm8962_pm = { SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) -- 1.9.1

2 1

[alsa-devel] [PATCH 1/9] ALSA: pcm_params: Remove unused add/sub functions
by Lars-Peter Clausen 30 Dec '14

30 Dec '14

Those two functions are not used anywhere and also their name is a bit to generic to be in a global header, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm_params.h | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 6b1c78f..15760f5 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -325,20 +325,6 @@ static inline int snd_interval_eq(const struct snd_interval *i1, const struct sn i1->max == i2->max && i1->openmax == i2->openmax; } -static inline unsigned int add(unsigned int a, unsigned int b) -{ - if (a >= UINT_MAX - b) - return UINT_MAX; - return a + b; -} - -static inline unsigned int sub(unsigned int a, unsigned int b) -{ - if (a > b) - return a - b; - return 0; -} - #define params_access(p) ((__force snd_pcm_access_t)\ snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_ACCESS))) #define params_format(p) ((__force snd_pcm_format_t)\ -- 1.8.0

2 9

[alsa-devel] [GIT PULL] ASoC updates for v3.20
by Mark Brown 30 Dec '14

30 Dec '14

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-rc2 for you to fetch changes up to 1b9f1ae8907cb13d3d2538d98385b2744c9490d5: Merge remote-tracking branches 'asoc/topic/samsung' and 'asoc/topic/wm8750' into asoc-next (2014-12-24 12:56:48 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.20 Nothing too exciting here yet, a small optimization for DAPM from Lars-Peter and a few small bits and pieces for drivers but nothing that really stands out. ---------------------------------------------------------------- Andrew Jackson (4): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels Bard Liao (3): ASoC: rt5670: set platform data by dmi ASoC: rt5670: Add runtime PM support ASoC: rt5670: Keep sysclk on if JD func is used Fang, Yang A (1): ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic Jarkko Nikula (4): ASoC: Intel: Add I2C dependency to two new machines ASoC: Intel: Fix BYTCR firmware name ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS ASoC: pcm: Fix vague codec and cpu DAI prepare error messages Jianqun Xu (3): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: fix maxburst of dma data to 4 ASoC: rockchip: i2s: set TDL and RDL to 16 samples Jie Yang (2): ASoC: Intel: correct the fixed free block allocation ASoC: Intel: Store the entry_point read from FW file Lars-Peter Clausen (4): ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget ASoC: wm8750: Cleanup manual bias level transitions ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected ASoC: dapm: Simplify fully route card handling Mark Brown (9): ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/intel' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/rockchip', 'asoc/topic/rt5670' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung' and 'asoc/topic/wm8750' into asoc-next Oder Chiou (3): ASoC: rt5677: Revise the filter powers ASoC: rt5677: Add the ASRC support ASoC: rt5677: Adjust the routing of "PLL1" Peter Rosin (1): ASoC: pcm512x: Also support PCM514x devices Takashi Iwai (1): ASoC: intel: Remove superfluous backslash in Kconfig Wei Yongjun (1): ASoC: Intel: fix sparse non static symbol warnings Wolfram Sang (4): ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers kbuild test robot (2): ASoC: Intel: fix platform_no_drv_owner.cocci warnings ASoC: Intel: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/pcm512x.txt | 3 +- include/sound/soc-dapm.h | 1 - sound/soc/codecs/pcm512x-i2c.c | 4 + sound/soc/codecs/pcm512x-spi.c | 4 + sound/soc/codecs/rt5670.c | 38 ++++ sound/soc/codecs/rt5677.c | 204 ++++++++++++++++++--- sound/soc/codecs/wm8750.c | 2 +- sound/soc/dwc/designware_i2s.c | 135 +++++++------- sound/soc/intel/Kconfig | 8 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 3 +- sound/soc/intel/cht_bsw_rt5672.c | 1 - sound/soc/intel/sst-firmware.c | 7 +- sound/soc/intel/sst/sst_acpi.c | 7 +- sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/pxa/spitz.c | 1 - sound/soc/rockchip/rockchip_i2s.c | 8 +- sound/soc/rockchip/rockchip_i2s.h | 2 +- sound/soc/samsung/arndale_rt5631.c | 1 - sound/soc/soc-core.c | 17 +- sound/soc/soc-dapm.c | 105 ++--------- sound/soc/soc-pcm.c | 7 +- 21 files changed, 349 insertions(+), 210 deletions(-)

2 1

[alsa-devel] ASoC fixes for v3.19
by Mark Brown 30 Dec '14

30 Dec '14

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.19-rc2 for you to fetch changes up to f2ec1d3b25776051a9205289dab12e9401d33f8a: Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus (2014-12-24 12:56:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 A few fixes for v3.19, a few driver specifics and one core fix which fixes a boot crash on OMAP if deferred probing kicks in due to attempting to modify static data. ---------------------------------------------------------------- Andrew Jackson (2): ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels Fang, Yang A (1): ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic Jarkko Nikula (3): ASoC: Intel: Add I2C dependency to two new machines ASoC: Intel: Fix BYTCR firmware name ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS Jianqun Xu (2): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: fix maxburst of dma data to 4 Jie Yang (1): ASoC: Intel: correct the fixed free block allocation Mark Brown (3): ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus sound/soc/codecs/rt5677.c | 9 ++++--- sound/soc/dwc/designware_i2s.c | 49 +++++++++++++++++++++++-------------- sound/soc/intel/Kconfig | 4 +-- sound/soc/intel/bytcr_dpcm_rt5640.c | 2 +- sound/soc/intel/sst-firmware.c | 6 ++++- sound/soc/intel/sst/sst_acpi.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 4 +-- sound/soc/rockchip/rockchip_i2s.h | 2 +- sound/soc/soc-core.c | 14 ++++------- 9 files changed, 52 insertions(+), 40 deletions(-)

2 1

[alsa-devel] [PATCH 0/2 v7] Digidesign Mbox 1 enhancements
by Damien Zammit 30 Dec '14

30 Dec '14

Revision 7. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 41 +++++++++++---- 3 files changed, 170 insertions(+), 9 deletions(-) -- 1.9.1

3 5

[alsa-devel] [PATCH 1/2] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 30 Dec '14

30 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + sound/core/pcm_native.c | 5 +++++ 2 files changed, 6 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8bb00a2..bb77aa3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -109,6 +109,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 166d59c..15fe682 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1517,6 +1517,11 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + if (substream->runtime->trigger_master == substream) + substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + switch (runtime->status->state) { case SNDRV_PCM_STATE_PREPARED: /* start playback stream if possible */ -- 1.9.1

3 8

[alsa-devel] [PATCH] mfd: wm8994: set mfd id-base for regulator devs creation to avoid conflicts
by Inha Song 30 Dec '14

30 Dec '14

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. If not, duplicate filename error was reported. - sysfs: cannot create duplicate filename '/devices/.../wm8994-ldo' Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..47cd3cc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,12 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, + .id = 0, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, + .id = 1, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +344,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, 0, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

1 0

[alsa-devel] [PATCH] mfd: wm8994: use PLATFORM_DEVID_AUTO for regulator devs creation to avoid conflicts
by Inha Song 30 Dec '14

30 Dec '14

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. but, if we use PLATFORM_DEVID_AUTO flag, cell ids are automatically allocated and managed without cell id setting. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..0505e45 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,10 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +342,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, PLATFORM_DEVID_AUTO, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

2 2

[alsa-devel] [PATCH] ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers
by Nicolin Chen 30 Dec '14

30 Dec '14

The xDC field should have 5 bit width according to Reference Manual. Thus this patch fixes it. Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- The patch was originally submitted by Aurelien BOUIN while in an informal way. And he hasn't finished any re-submitting during the past two weeks. Thus I create a new patch with his signed-off included so as to fix the probelm as soon as possible, not sure whether it's decent or not though. --Nicolin sound/soc/fsl/fsl_esai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

2 2

[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Use dev_name() for registering the irq
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'name' array is currently stored inside the fsl_spdif_priv private structure only for registering the interrupt name. This can be simplified by registering it with dev_name() instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index af042942..73da1f0 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -90,7 +90,6 @@ struct spdif_mixer_control { * @sysclk: system clock for rx clock rate measurement * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel - * @name: driver name */ struct fsl_spdif_priv { struct spdif_mixer_control fsl_spdif_control; @@ -109,12 +108,8 @@ struct fsl_spdif_priv { struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; - - /* The name space will be allocated dynamically */ - char name[0]; }; - /* DPLL locked and lock loss interrupt handler */ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) { @@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev) if (!np) return -ENODEV; - spdif_priv = devm_kzalloc(&pdev->dev, - sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, - GFP_KERNEL); + spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) return -ENOMEM; - strcpy(spdif_priv->name, np->name); - spdif_priv->pdev = pdev; /* Initialize this copy of the CPU DAI driver structure */ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); - spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1203,7 +1194,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, - spdif_priv->name, spdif_priv); + dev_name(&pdev->dev), spdif_priv); if (ret) { dev_err(&pdev->dev, "could not claim irq %u\n", irq); return ret; -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: fsl_esai: Use dev_name() for registering the irq
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'name' array is currently stored inside the fsl_esai private structure only for registering the interrupt name. This can be simplified by registering it with dev_name() instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 1c08ab1..dbe9204 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -44,7 +44,6 @@ * @sck_div: if using PSR/PM dividers for SCKx clock * @slave_mode: if fully using DAI slave mode * @synchronous: if using tx/rx synchronous mode - * @name: driver name */ struct fsl_esai { struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -63,7 +62,6 @@ struct fsl_esai { bool sck_div[2]; bool slave_mode; bool synchronous; - char name[32]; }; static irqreturn_t esai_isr(int irq, void *devid) @@ -739,7 +737,6 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -779,7 +776,7 @@ static int fsl_esai_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, - esai_priv->name, esai_priv); + dev_name(&pdev->dev), esai_priv); if (ret) { dev_err(&pdev->dev, "failed to claim irq %u\n", irq); return ret; -- 1.9.1

3 3

[alsa-devel] [PATCH v5 0/4] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 29 Dec '14

29 Dec '14

This patch set adds multi-CODEC support to the simple card. v5: accept only "sound-dai" as the dai link property (Arnd Bergmann) v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (4): ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: simple-card: add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 18 +--- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 87 ++++++++-------- sound/soc/soc-core.c | 110 ++++++++++++++++++--- 4 files changed, 149 insertions(+), 69 deletions(-) -- 2.1.3

2 7

[alsa-devel] [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 29 Dec '14

29 Dec '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a Codec DAI driver that controls the SOC external pins - a CPU DAI driver for controlling the LPASS-LPAIF block - a PCM MI2S platform driver These drivers, together, are used by simple-audio-card to complete the audio implementation. Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. == Updates from my previous post: [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://thread.gmane.org/gmane.linux.ports.arm.msm/10989 - Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. - Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. - Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. - Removed machine driver and tied DAI drivers to simple-audio-card. - Packaged the build files and Maxim codec files together in one change. - Removed QCOM as vendor from Maxim code and documentation. - Separated the SOC and board definitions into the correct DTS files. - Update device tree documentation to reflect changes. - General code cleanup. == Additional patch series upon which this patch series has device-tree dependencies (lcc) [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../bindings/sound/qcom,lpass-cpu-mi2s.txt | 42 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 136 ++++++ sound/soc/qcom/Kconfig | 27 ++ sound/soc/qcom/Makefile | 6 + sound/soc/qcom/lpass-cpu-mi2s.c | 378 ++++++++++++++++ sound/soc/qcom/lpass-lpaif-reg.h | 155 +++++++ sound/soc/qcom/lpass-mi2s.h | 34 ++ sound/soc/qcom/lpass-pcm-mi2s.c | 486 +++++++++++++++++++++ 15 files changed, 1316 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-mi2s.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.c create mode 100644 sound/soc/qcom/lpass-lpaif-reg.h create mode 100644 sound/soc/qcom/lpass-mi2s.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c -- 1.8.2.1

2 15

[alsa-devel] minos-one live DVD with Advanced Gtk+ Sequencer
by Joël Krähemann 28 Dec '14

28 Dec '14

Hi out there, I just built a http://linuxfromscratch.org based live DVD called minos-one http://sourceforge.net/projects/ags/files/minos-one.iso Cheers Joël

1 0

[alsa-devel] [PATCH v3] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 27 Dec '14

27 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Changes in v3: - added my SoB sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..e458957c5f52 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() 0 #endif #include "ac97c.h" -- 2.1.0

2 1

[alsa-devel] [PATCH v2] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 26 Dec '14

26 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html --- sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..83975f8d5947 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() 0 #endif #include "ac97c.h" -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: wm5102: Use put_unaligned_be16
by Vaishali Thakkar 26 Dec '14

26 Dec '14

This patch introduces the use of function put_unaligned_le16. This is done using Coccinelle and semantic patch used is as follows: @a@ typedef u16, __be16, uint16_t; {u16,__be16,uint16_t} e16; identifier tmp; expression ptr; expression y,e; type T; @@ - tmp = cpu_to_be16(y); <+... when != tmp ( - memcpy(ptr, (T)&tmp, $2\|sizeof(u16)\|sizeof(__be16)\|sizeof(uint16_t)\|sizeof(e16)$); + put_unaligned_be16(y,ptr); | - memcpy(ptr, (T)&tmp, ...); + put_unaligned_be16(y,ptr); ) ...+> ? tmp = e @@ type T; identifier a.tmp; @@ - T tmp; ...when != tmp Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/codecs/wm5102.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f602349..fec21f9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -28,6 +28,7 @@ #include <linux/mfd/arizona/core.h> #include <linux/mfd/arizona/registers.h> +#include <asm/unaligned.h> #include "arizona.h" #include "wm5102.h" @@ -617,11 +618,11 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - uint16_t data; mutex_lock(&codec->mutex); - data = cpu_to_be16(arizona->dac_comp_coeff); - memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + + put_unaligned_be16(arizona->dac_comp_coeff, + ucontrol->value.bytes.data); mutex_unlock(&codec->mutex); return 0; -- 1.9.1

1 0

[alsa-devel] [PATCH] sound: pci: hda: patch_analog.c: Remove some unused functions
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Removes some functions that are not used anywhere: ad198x_ch_mode_get() ad198x_ch_mode_info() ad198x_ch_mode_info() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 33 --------------------------------- 1 file changed, 33 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..0e76dcc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -697,39 +697,6 @@ static int patch_ad1981(struct hda_codec *codec) * E/F quad mic array */ -#ifdef ENABLE_AD_STATIC_QUIRKS -static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, spec->multiout.max_channels); -} - -static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return err; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ - static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { -- 1.7.10.4

2 1

[alsa-devel] [PATCH] sound: pci: hda: patch_analog.c: Remove unused function
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Remove the function ad198x_ch_mode_get() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..9d5257b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -707,15 +707,6 @@ static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, spec->num_channel_mode); } -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, spec->multiout.max_channels); -} - static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { -- 1.7.10.4

3 3

[alsa-devel] [PATCH] sound: isa: gus: gus_instr.c: Remove unused function
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Remove the function snd_gus_simple_remove_sample() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/isa/gus/gus_instr.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c index 4dc9caf..2d21408 100644 --- a/sound/isa/gus/gus_instr.c +++ b/sound/isa/gus/gus_instr.c @@ -162,11 +162,3 @@ int snd_gus_simple_get_sample(void *private_data, struct simple_instrument *inst return snd_gus_dram_read(gus, data, instr->address.memory, instr->size, 0); } - -int snd_gus_simple_remove_sample(void *private_data, struct simple_instrument *instr, - int atomic) -{ - struct snd_gus_card *gus = private_data; - - return snd_gf1_mem_free(&gus->gf1.mem_alloc, instr->address.memory); -} -- 1.7.10.4

2 1

[alsa-devel] [PATCH] sound: pci: ice1712: wm8776.c: Remove some unused functions
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Removes some functions that are not used anywhere: snd_wm8776_set_master_mode() snd_wm8776_set_adc_if() snd_wm8776_set_dac_if() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8776.c | 15 --------------- sound/pci/ice1712/wm8776.h | 3 --- 2 files changed, 18 deletions(-) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index e66c0da..ebd2fe4 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -452,21 +452,6 @@ void snd_wm8776_resume(struct snd_wm8776 *wm) snd_wm8776_write(wm, i, wm->regs[i]); } -void snd_wm8776_set_dac_if(struct snd_wm8776 *wm, u16 dac) -{ - snd_wm8776_write(wm, WM8776_REG_DACIFCTRL, dac); -} - -void snd_wm8776_set_adc_if(struct snd_wm8776 *wm, u16 adc) -{ - snd_wm8776_write(wm, WM8776_REG_ADCIFCTRL, adc); -} - -void snd_wm8776_set_master_mode(struct snd_wm8776 *wm, u16 mode) -{ - snd_wm8776_write(wm, WM8776_REG_MSTRCTRL, mode); -} - void snd_wm8776_set_power(struct snd_wm8776 *wm, u16 power) { snd_wm8776_write(wm, WM8776_REG_PWRDOWN, power); diff --git a/sound/pci/ice1712/wm8776.h b/sound/pci/ice1712/wm8776.h index 93a2d69..42acef0 100644 --- a/sound/pci/ice1712/wm8776.h +++ b/sound/pci/ice1712/wm8776.h @@ -216,9 +216,6 @@ struct snd_wm8776 { void snd_wm8776_init(struct snd_wm8776 *wm); void snd_wm8776_resume(struct snd_wm8776 *wm); -void snd_wm8776_set_dac_if(struct snd_wm8776 *wm, u16 dac); -void snd_wm8776_set_adc_if(struct snd_wm8776 *wm, u16 adc); -void snd_wm8776_set_master_mode(struct snd_wm8776 *wm, u16 mode); void snd_wm8776_set_power(struct snd_wm8776 *wm, u16 power); void snd_wm8776_volume_restore(struct snd_wm8776 *wm); int snd_wm8776_build_controls(struct snd_wm8776 *wm); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ALSA: pcm_dmaengine: correct the error handler of dmaengine_prep_dma_cyclic
by Barry Song 25 Dec '14

25 Dec '14

From: Barry Song <Baohua.Song(a)csr.com> preparing cyclic DMA description can fail either due to DMA desc list is full(-ENOMEM), or due to the coming DMA configuration is illegal or not supported by the acting DMA hardware(other ERR codes). Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/core/pcm_dmaengine.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c40..5cac7e4 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -163,6 +163,8 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) if (!desc) return -ENOMEM; + if (IS_ERR(desc)) + return PTR_ERR(desc); desc->callback = dmaengine_pcm_dma_complete; desc->callback_param = substream; -- 2.2.0

2 5

[alsa-devel] Can not rmmod snd_hda_codec_hdmi
by Xuebing Wang 24 Dec '14

24 Dec '14

Hi community, I am trying to implement the function of streaming SACD music over HDMI, my PC is Intel Haswell based CPU (Intel i7 4770). When I am trying to 'rmmod snd_hda_codec_hdmi', I get the prompt that this module is in use (detailed message as below) rmmod: ERROR: Module snd_hda_codec_hdmi is in use Is it possible that workqueue [hd-audio] is using this kernel module (snd_hda_codec_hdmi)? Can I use rmmod / insmod as the way to debug this module? Thanks. Xuebing Wang

5 9

[alsa-devel] [PATCH 0/2] ASoC: rockchip: i2s: fix TDL and RDL to 16 bits
by Jianqun Xu 24 Dec '14

24 Dec '14

To make Bus DMA works more efficiency, DMA may work in burst mode, we set the watermark of transmit and receive reach to 16 bits. Jianqun Xu (2): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: set TDL and RDL to 16 bits sound/soc/rockchip/rockchip_i2s.c | 4 ++++ sound/soc/rockchip/rockchip_i2s.h | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) -- 1.9.1

4 9

[alsa-devel] [PATCH 1/3 v2] ASoC: rt5677: Revise the filter powers
by Oder Chiou 24 Dec '14

24 Dec '14

Add the filter powers of the dac mono3 and mono4, and remove the connection of dac stereo1 filter that connect to DAC1 MIX. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..a15e39d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2645,10 +2645,18 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), @@ -3455,10 +3463,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXL", NULL, "dac stereo1 filter" }, { "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXR", NULL, "dac stereo1 filter" }, { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, @@ -3530,30 +3536,34 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, { "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, { "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" }, - { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXL", NULL, "dac mono2 left filter" }, { "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" }, { "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, { "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, { "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" }, - { "Mono DAC MIXR", NULL, "dac mono right filter" }, + { "Mono DAC MIXR", NULL, "dac mono2 right filter" }, { "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXL", NULL, "dac mono3 left filter" }, { "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXR", NULL, "dac mono3 right filter" }, { "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXL", NULL, "dac mono4 left filter" }, { "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXR", NULL, "dac mono4 right filter" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXL" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXR" }, -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: Intel: Store the entry_point read from FW file
by Jie Yang 24 Dec '14

24 Dec '14

To enable some modules from other than base FW, according to FW interface spec, we need pass the correct entry point param to FW, so here store the entry_point read from FW file for later usage. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index ef2e8b5..50d6925 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -497,6 +497,7 @@ struct sst_module *sst_module_new(struct sst_fw *sst_fw, sst_module->sst_fw = sst_fw; sst_module->scratch_size = template->scratch_size; sst_module->persistent_size = template->persistent_size; + sst_module->entry = template->entry; INIT_LIST_HEAD(&sst_module->block_list); INIT_LIST_HEAD(&sst_module->runtime_list); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: correct the fixed free block allocation
by Jie Yang 24 Dec '14

24 Dec '14

For block span more than 1 section, when allocate it from a free block, we need allocate the remain buffers within the block, and then continue alloc the rest of needed size buffer. Here also make sure this free block is moved from free list to used list, and add it to block_list which may be used for power gating disabling later. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index 4a5bde9..ef2e8b5 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -763,8 +763,12 @@ static int block_alloc_fixed(struct sst_dsp *dsp, struct sst_block_allocator *ba /* does block span more than 1 section */ if (ba->offset >= block->offset && ba->offset < block_end) { + /* add block */ + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, block_list); /* align ba to block boundary */ - ba->offset = block->offset; + ba->size -= block_end - ba->offset; + ba->offset = block_end; err = block_alloc_contiguous(dsp, ba, block_list); if (err < 0) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: pcm: Fix vague codec and cpu DAI prepare error messages
by Jarkko Nikula 24 Dec '14

24 Dec '14

Both codec and cpu DAI prepare print the same error message making it a bit more difficult to grep quickly from sources. Fix this by telling it explicitly. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-pcm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index eb87d96e2cf0..0ae0e2a9eed7 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -746,7 +746,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) codec_dai); if (ret < 0) { dev_err(codec_dai->dev, - "ASoC: DAI prepare error: %d\n", ret); + "ASoC: codec DAI prepare error: %d\n", + ret); goto out; } } @@ -755,8 +756,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", - ret); + dev_err(cpu_dai->dev, + "ASoC: cpu DAI prepare error: %d\n", ret); goto out; } } -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic
by yang.a.fang＠intel.com 24 Dec '14

24 Dec '14

From: "Fang, Yang A" <yang.a.fang(a)intel.com> snd_kcontrol_chip should return snd_soc_component instead of snd_soc_codec Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- sound/soc/codecs/rt5677.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..c0fbe18 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -784,8 +784,8 @@ static unsigned int bst_tlv[] = { static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); ucontrol->value.integer.value[0] = rt5677->dsp_vad_en; @@ -795,8 +795,9 @@ static int rt5677_dsp_vad_get(struct snd_kcontrol *kcontrol, static int rt5677_dsp_vad_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5677_priv *rt5677 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); rt5677->dsp_vad_en = !!ucontrol->value.integer.value[0]; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 24 Dec '14

24 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v3->v4 + Drop applied patches + Use combined function to prepare clock [Mark Brown] + Use managed clock resources to avoid clk_put [Mark Brown] + Read configuration prameters from hardware for both platform data and device tree [Mark Brown] + Re-order patch sequence [Mark Brown] + Change union name to avoid future collisions [Mark Brown] + Check return code from clk_set_rate [Mark Brown] + Check parametsrs read from hardware agaist array limits and add further comments [Mark Brown] + Add of_match_ptr [Mark Brown] Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (5): ASoC: dwc: Prepare clock before use ASoC: dwc: Switch to managed clock resource ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S .../devicetree/bindings/sound/designware-i2s.txt | 31 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 275 ++++++++++++++++---- 3 files changed, 260 insertions(+), 47 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

1 7

Re: [alsa-devel] [PATCH] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Eliot Blennerhassett 23 Dec '14

23 Dec '14

On 22/12/14 05:50, Rickard Strandqvist wrote: > Removes some functions that are not used anywhere: NAK - because hpi_adapter_set_property() is now used in for-next branch of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git Otherwise I have no problem with the patch.

2 2

[alsa-devel] [PATCH v2 1/2] ASoC: rt5670: Add runtime PM support
by Bard Liao 22 Dec '14

22 Dec '14

This patch adds runtime PM support on rt5670 codec. Signed-off-by: Lin Mengdong <mengdong.lin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Remove the unused runtime suspend/resume code. We will switch the sysclk source on another patch. --- sound/soc/codecs/rt5670.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 09cf100..a8a471b 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/acpi.h> @@ -2734,18 +2735,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) goto err; + pm_runtime_put(&i2c->dev); + return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Revise the filter powers
by Oder Chiou 22 Dec '14

22 Dec '14

Revise the filter powers. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 81fe146..a15e39d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2645,10 +2645,18 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { /* DAC Mixer */ SND_SOC_DAPM_SUPPLY("dac stereo1 filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_S1F_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono left filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 left filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("dac mono right filter", RT5677_PWR_DIG2, + SND_SOC_DAPM_SUPPLY("dac mono2 right filter", RT5677_PWR_DIG2, RT5677_PWR_DAC_M2F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono3 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M3F_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 left filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("dac mono4 right filter", RT5677_PWR_DIG2, + RT5677_PWR_DAC_M4F_R_BIT, 0, NULL, 0), SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, rt5677_sto1_dac_l_mix, ARRAY_SIZE(rt5677_sto1_dac_l_mix)), @@ -3455,10 +3463,8 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "DAC1 MIXL", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXL", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXL", NULL, "dac stereo1 filter" }, { "DAC1 MIXR", "Stereo ADC Switch", "ADDA1 Mux" }, { "DAC1 MIXR", "DAC1 Switch", "DAC1 Mux" }, - { "DAC1 MIXR", NULL, "dac stereo1 filter" }, { "DAC1 FS", NULL, "DAC1 MIXL" }, { "DAC1 FS", NULL, "DAC1 MIXR" }, @@ -3530,30 +3536,34 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "Mono DAC MIXL", "DAC1 L Switch", "DAC1 MIXL" }, { "Mono DAC MIXL", "DAC2 L Switch", "DAC2 L Mux" }, { "Mono DAC MIXL", "DAC2 R Switch", "DAC2 R Mux" }, - { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXL", NULL, "dac mono2 left filter" }, { "Mono DAC MIXR", "ST R Switch", "Sidetone Mux" }, { "Mono DAC MIXR", "DAC1 R Switch", "DAC1 MIXR" }, { "Mono DAC MIXR", "DAC2 R Switch", "DAC2 R Mux" }, { "Mono DAC MIXR", "DAC2 L Switch", "DAC2 L Mux" }, - { "Mono DAC MIXR", NULL, "dac mono right filter" }, + { "Mono DAC MIXR", NULL, "dac mono2 right filter" }, { "DD1 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD1 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD1 MIXL", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXL", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXL", NULL, "dac mono3 left filter" }, { "DD1 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD1 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD1 MIXR", "DAC3 L Switch", "DAC3 L Mux" }, { "DD1 MIXR", "DAC3 R Switch", "DAC3 R Mux" }, + { "DD1 MIXR", NULL, "dac mono3 right filter" }, { "DD2 MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, { "DD2 MIXL", "Mono DAC Mix L Switch", "Mono DAC MIXL" }, { "DD2 MIXL", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXL", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXL", NULL, "dac mono4 left filter" }, { "DD2 MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, { "DD2 MIXR", "Mono DAC Mix R Switch", "Mono DAC MIXR" }, { "DD2 MIXR", "DAC4 L Switch", "DAC4 L Mux" }, { "DD2 MIXR", "DAC4 R Switch", "DAC4 R Mux" }, + { "DD2 MIXR", NULL, "dac mono4 right filter" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXL" }, { "Stereo DAC MIX", NULL, "Stereo DAC MIXR" }, -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH -next] ASoC: Intel: fix sparse non static symbol warnings
by weiyj_lk＠163.com 22 Dec '14

22 Dec '14

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warnings: sound/soc/intel/sst/sst_acpi.c:248:5: warning: symbol 'sst_acpi_probe' was not declared. Should it be static? sound/soc/intel/sst/sst_acpi.c:335:5: warning: symbol 'sst_acpi_remove' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/intel/sst/sst_acpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 31124aa..98a8f66 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -245,7 +245,7 @@ static struct sst_machines *sst_acpi_find_machine( return NULL; } -int sst_acpi_probe(struct platform_device *pdev) +static int sst_acpi_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; int ret = 0; @@ -332,7 +332,7 @@ do_sst_cleanup: * This function is called by OS when a device is unloaded * This frees the interrupt etc */ -int sst_acpi_remove(struct platform_device *pdev) +static int sst_acpi_remove(struct platform_device *pdev) { struct intel_sst_drv *ctx;

2 1

[alsa-devel] [PATCH] ASoC: pcm512x: Also support PCM514x devices
by Peter Rosin 22 Dec '14

22 Dec '14

From: Peter Rosin <peda(a)axentia.se> Signed-off-by: Peter Rosin <peda(a)axentia.se> --- Documentation/devicetree/bindings/sound/pcm512x.txt | 3 ++- sound/soc/codecs/pcm512x-i2c.c | 4 ++++ sound/soc/codecs/pcm512x-spi.c | 4 ++++ 3 files changed, 10 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt index faff75e64573..98e0d34915e8 100644 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -5,7 +5,8 @@ on the board). Required properties: - - compatible : One of "ti,pcm5121" or "ti,pcm5122" + - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141" or + "ti,pcm5142" - reg : the I2C address of the device for I2C, the chip select number for SPI. diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c index 4d62230bd378..08047ac952f4 100644 --- a/sound/soc/codecs/pcm512x-i2c.c +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -41,6 +41,8 @@ static int pcm512x_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id pcm512x_i2c_id[] = { { "pcm5121", }, { "pcm5122", }, + { "pcm5141", }, + { "pcm5142", }, { } }; MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); @@ -48,6 +50,8 @@ MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, + { .compatible = "ti,pcm5141", }, + { .compatible = "ti,pcm5142", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c index f297058c0038..7b64a9cef704 100644 --- a/sound/soc/codecs/pcm512x-spi.c +++ b/sound/soc/codecs/pcm512x-spi.c @@ -43,6 +43,8 @@ static int pcm512x_spi_remove(struct spi_device *spi) static const struct spi_device_id pcm512x_spi_id[] = { { "pcm5121", }, { "pcm5122", }, + { "pcm5141", }, + { "pcm5142", }, { }, }; MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); @@ -50,6 +52,8 @@ MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); static const struct of_device_id pcm512x_of_match[] = { { .compatible = "ti,pcm5121", }, { .compatible = "ti,pcm5122", }, + { .compatible = "ti,pcm5141", }, + { .compatible = "ti,pcm5142", }, { } }; MODULE_DEVICE_TABLE(of, pcm512x_of_match); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v3 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 22 Dec '14

22 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (5): ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Add documentation for I2S DT .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 293 +++++++++++++++----- 3 files changed, 258 insertions(+), 68 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

2 16

[alsa-devel] [PATCH 1/2] ASoC: Intel: Fix BYTCR firmware name
by Jarkko Nikula 22 Dec '14

22 Dec '14

BYTCR DSP firmware is in intel/ subdirectory. See linux-firmware.git commit d562a3b63632 ("linux-firmware: add sst audio firmware for baytrail platforms"). Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst/sst_acpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 3abc29e8a928..2ac72eb5e75d 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -343,7 +343,7 @@ int sst_acpi_remove(struct platform_device *pdev) } static struct sst_machines sst_acpi_bytcr[] = { - {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + {"10EC5640", "T100", "bytt100_rt5640", NULL, "intel/fw_sst_0f28.bin", &byt_rvp_platform_data }, {}, }; -- 2.1.3

2 2

[alsa-devel] X-Fi XtremeGamer front header ( headphones ) audio
by Marius Cirsta 22 Dec '14

22 Dec '14

Hi, I've got a X-Fi XtremeGamer ( SB0770 ) and I can't get sound output from the headphone plug in my case which is plugged into the front panel connector on the SB0770. This does work in Windows so I'm assuming that it's unimplemented in ALSO for the X-Fi. I can see no mention of headphones in the alsamixer either. Also looking at the code I can see nothing regarding the headphones part of this. If the implementation is indeed missing how difficult would it be to implement this ? I'd like try.

1 0

[alsa-devel] [PATCH v3] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 22 Dec '14

22 Dec '14

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v2->v3: - drop .owner = THIS_MODULE from the struct platform_driver initialization; - rebase to 3.18-rc7. Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 675 +++++++++++++++++++++ 7 files changed, 706 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 180d560..6a09926 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10409,6 +10409,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 7d5d644..dcc79aa 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -55,6 +55,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index d88edfc..6c24e6a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -32,3 +32,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..1cfb19e --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,675 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

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[alsa-devel] [PATCH v3] ASoC: rt5670: set platform data by dmi
by Bard Liao 22 Dec '14

22 Dec '14

This patch set specific data according to dmi data. Signed-off-by: Jin, Yao <yao.jin(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Remove the __initconst in definition since the data of dmi_platform_intel_braswell[] could not be accessible by dmi_check_system(). --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8bf3a56..d6895aa 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include <linux/platform_device.h> #include <linux/acpi.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected
by Lars-Peter Clausen 22 Dec '14

22 Dec '14

The connected flag of a widget only affects widgets that are either a source or a sink. The MICBIAS widget is a simple pass-through widget though and hence its behavior is the same regardless of whether the connected flag is set or not. Hence there is not much point in trying to automatically mark MICBIAS widgets as non-connected, so just remove it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb..b8eba93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3880,7 +3880,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) switch (w->id) { case snd_soc_dapm_input: case snd_soc_dapm_output: - case snd_soc_dapm_micbias: dev_dbg(card->dev, "ASoC: Auto NC: Checking widget %s\n", w->name); if (!snd_soc_dapm_widget_in_card_paths(card, w)) { -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: intel: Remove superfluous backslash in Kconfig
by Takashi Iwai 22 Dec '14

22 Dec '14

The line continuation in Kconfig should be a '\' instead of '\\'. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e989ecf046c9..f06fcf1e21a5 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -46,7 +46,7 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \\ + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 @@ -76,7 +76,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \\ + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC && \ I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD -- 2.2.0

2 2

[alsa-devel] [PATCH] ASoC: Intel: Add I2C dependency to two new machines
by Jarkko Nikula 22 Dec '14

22 Dec '14

Fix following build error when CONFIG_I2C is not enabled: sound/soc/codecs/rt5640.c:2252:1: warning: data definition has no type or storage class module_i2c_driver(rt5640_i2c_driver); ^ sound/soc/codecs/rt5640.c:2252:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int] sound/soc/codecs/rt5640.c:2252:1: warning: parameter names (without types) in function declaration sound/soc/codecs/rt5640.c:2241:26: warning: ‘rt5640_i2c_driver’ defined but not used [-Wunused-variable] static struct i2c_driver rt5640_i2c_driver = { ^ cc1: some warnings being treated as errors Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/soc/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index e989ecf046c9..f86de1211b96 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -89,7 +89,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio DSP Support for MID BYT Platform" - depends on X86 + depends on X86 && I2C select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -101,7 +101,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS + depends on X86_INTEL_LPSS && I2C select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- 2.1.3

3 2

[alsa-devel] [PATCH 00/28] remove .owner for most platform_drivers: the missing bits
by Wolfram Sang 22 Dec '14

22 Dec '14

Generated with coccinelle. The big cleanup was pulled in this merge window. This series catches the bits fallen through. The patches shall go in via the subsystem trees. If possible for 3.19 to increase consistency I'd say, but you decide, of course. cocci-file used: @match1@ declarer name module_platform_driver; declarer name module_platform_driver_probe; declarer name for_each_node_by_type; identifier __driver; @@ ( module_platform_driver(__driver); | module_platform_driver_probe(__driver, ...); ) @fix1 depends on match1@ identifier match1.__driver; @@ static struct platform_driver __driver = { .driver = { - .owner = THIS_MODULE, } }; @match2@ identifier __driver; @@ ( platform_driver_register(&__driver) | platform_driver_probe(&__driver, ...) | platform_create_bundle(&__driver, ...) ) @fix2 depends on match2@ identifier match2.__driver; @@ static struct platform_driver __driver = { .driver = { - .owner = THIS_MODULE, } }; Thanks again to Julia Lawall for support. And hey, we fixed a coccinelle bug on the way :) Wolfram Sang (28): ARM: mach-exynos: drop owner assignment from platform_drivers mips: lantiq: xway: drop owner assignment from platform_drivers mips: pci: drop owner assignment from platform_drivers char: ipmi: drop owner assignment from platform_drivers cpufreq: drop owner assignment from platform_drivers dma: drop owner assignment from platform_drivers gpio: drop owner assignment from platform_drivers gpu: drm: rockchip: drop owner assignment from platform_drivers iommu: drop owner assignment from platform_drivers net: ethernet: stmicro: stmmac: drop owner assignment from platform_drivers net: wireless: ath: ath5k: drop owner assignment from platform_drivers of: drop owner assignment from platform_drivers pci: host: drop owner assignment from platform_drivers phy: drop owner assignment from platform_drivers pinctrl: intel: drop owner assignment from platform_drivers rtc: drop owner assignment from platform_drivers scsi: drop owner assignment from platform_drivers thermal: drop owner assignment from platform_drivers thermal: int340x_thermal: drop owner assignment from platform_drivers tty: serial: 8250: drop owner assignment from platform_drivers usb: gadget: udc: bdc: drop owner assignment from platform_drivers watchdog: drop owner assignment from platform_drivers ASoC: intel: drop owner assignment from platform_drivers ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers macintosh: drop owner assignment from platform_drivers arch/arm/mach-exynos/pmu.c | 1 - arch/mips/lantiq/xway/vmmc.c | 1 - arch/mips/pci/pci-ar2315.c | 1 - arch/mips/pci/pci-rt2880.c | 1 - drivers/char/ipmi/ipmi_powernv.c | 1 - drivers/cpufreq/ls1x-cpufreq.c | 1 - drivers/dma/at_xdmac.c | 1 - drivers/gpio/gpio-vf610.c | 1 - drivers/gpu/drm/rockchip/rockchip_drm_drv.c | 1 - drivers/iommu/rockchip-iommu.c | 1 - drivers/macintosh/windfarm_pm112.c | 1 - drivers/macintosh/windfarm_pm72.c | 1 - drivers/macintosh/windfarm_rm31.c | 1 - drivers/net/ethernet/stmicro/stmmac/stmmac_platform.c | 1 - drivers/net/wireless/ath/ath5k/ahb.c | 1 - drivers/of/unittest.c | 1 - drivers/pci/host/pci-layerscape.c | 1 - drivers/phy/phy-armada375-usb2.c | 1 - drivers/phy/phy-berlin-usb.c | 1 - drivers/phy/phy-miphy28lp.c | 1 - drivers/pinctrl/intel/pinctrl-cherryview.c | 1 - drivers/rtc/rtc-opal.c | 1 - drivers/scsi/atari_scsi.c | 1 - drivers/scsi/mac_scsi.c | 1 - drivers/scsi/sun3_scsi.c | 1 - drivers/thermal/int340x_thermal/int3400_thermal.c | 1 - drivers/thermal/int340x_thermal/int3402_thermal.c | 1 - drivers/thermal/rockchip_thermal.c | 1 - drivers/tty/serial/8250/8250_omap.c | 1 - drivers/usb/gadget/udc/bdc/bdc_core.c | 1 - drivers/watchdog/cadence_wdt.c | 1 - drivers/watchdog/meson_wdt.c | 1 - sound/soc/intel/bytcr_dpcm_rt5640.c | 1 - sound/soc/intel/cht_bsw_rt5672.c | 1 - sound/soc/intel/sst/sst_acpi.c | 1 - sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/pxa/spitz.c | 1 - sound/soc/samsung/arndale_rt5631.c | 1 - 38 files changed, 38 deletions(-) -- 2.1.3

2 10

[alsa-devel] [PATCH] ASoC: Intel: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 22 Dec '14

22 Dec '14

sound/soc/intel/cht_bsw_rt5672.c:273:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- cht_bsw_rt5672.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -270,7 +270,6 @@ static int snd_cht_mc_probe(struct platf static struct platform_driver snd_cht_mc_driver = { .driver = { - .owner = THIS_MODULE, .name = "cht-bsw-rt5672", .pm = &snd_soc_pm_ops, },

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[alsa-devel] [PATCH] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 20 Dec '14

20 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Unfortunately the driver is missing other work to be useful again on ARM, if anyone wants to actually use it, they need to add a DT binding, and the driver should really be converted to use the ASoC framework. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html --- sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..83975f8d5947 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() (0) #endif #include "ac97c.h" -- 2.1.0

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