April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC v2 0/13] dw_hdmi cleanups, audio preparation, helpers and ahb audio support
by Russell King - ARM Linux 28 May '15

28 May '15

I'm sending this series for comments, and to allow people to update their code bases (for those who are using my AHB audio driver on iMX6). This is really several series, and I'm not expecting this to be ready for the upcoming merge window. That said, if anyone wants to say that they're happy with some of the initial six patches, I'm happy to queue those for David, if he's willing to take them. This series applies on top of my previous series, with the three patches from the rockchip guys applied, iow: drm: rockchip/dw_hdmi-rockchip: improve for HDMI electrical test drm: bridge/dw_hdmi: separate VLEVCTRL settting into platform driver drm: bridge/dw_hdmi: fixed codec style drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: protect n/cts setting with a mutex drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() The first four patches are a few cleanups to the dw_hdmi driver. The next two patches introduce new interfaces to the dw_hdmi driver to support AHB audio, including the errata found on iMX6 which requires N to be programmed to zero when the audio FIFO is not full. I'm expecting some discussion with these last two as we try to work out how to deal with the two variants of audio support for this part (AHB audio vs I2S audio). The following three add various helpers to the DRM and ALSA subsystems for audio support which is non-specific to the AHB audio driver. For DRM, this is a helper macro to obtain the pointer into the short audio descriptors. For ALSA, it's a set of helpers to restrict an audio PCM device's capabilities according to the ELD, and to generate the IEC958 channel status data. If these helpers are acceptable, we can convert a number of drivers to them. The ELD helper isn't quite finished, but should be sufficient for initial testing. The last two patches add support for the dw_hdmi audio. This is an ALSA driver rather than an ASoC driver as it's a pretty simple affair. I couldn't find a suitable location in the sound/ subtree to place it, so I've placed it next to the bridge - yes, we probably need to find it a better home in the sound/ subtree, but it currently makes use of dw_hdmi.h in the directory it lives. drivers/gpu/drm/bridge/Kconfig | 11 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 566 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.h | 14 + drivers/gpu/drm/bridge/dw_hdmi.c | 321 +++++++++------- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + drivers/gpu/drm/imx/dw_hdmi-imx.c | 5 + include/drm/bridge/dw_hdmi.h | 7 + include/drm/drm_edid.h | 19 + include/sound/pcm_drm_eld.h | 6 + include/sound/pcm_iec958.h | 9 + sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/pcm_drm_eld.c | 92 +++++ sound/core/pcm_iec958.c | 95 +++++ 15 files changed, 1029 insertions(+), 128 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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Re: [alsa-devel] snd-bebob and PrismSound Orpheus fails with snd_pcm_hw_params_set_channels
by Mike Ood 24 May '15

24 May '15

Yet another experiment. Orpheus operation manual says there are six modes concerning Sync Source. -------------------------------------- Sync Source Master/Slave Description -------------------------------------- Local Master System is sync'ed to Orpheus' local clock Wordclock Master System is sync'ed to external wordclock applied to Orpheus DI Master System is sync'ed to external S/PDIF applied to Orpheus ADAT Master System is sync'ed to external ADAT (ADAT input must be enabled) PC DAW Slave Orpheus is slaved to system clock from host PC CSP Master System is sync'ed to Orpheus' 1394 clock (not recommended) -------------------------------------- With each of the five modes mentioned above (ADAT input not available), I tried to play music on Windows 7 using Audacity. The results are: ------------------- Local OK Wordclock NG DI NG PC DAW OK CSP OK ------------------- So it seems that if there is no external clock applied, when one is expected, Orpheus stops playing rather than trying to continue operation by free-running its clock. *** Takashi Sakamoto wrote: > Not funny, but headache. It's a customization by hardware vendor, I > mentioned. (in this case, by PrismSound Ltd.) > BridgeCo AG (currently ArchWave AG) produced DM1000/DM1100/DM1500 > chipset with BeBoB firmware. PrismSound Ltd. utilized the chipset and > firmware for streaming functionality of your Orpheus. > BeBoB firmware gives common ways to control itself, while the hadware > vendor added their own cusomization against it. I don't know exactly the > reason because there're few information about your device. I do not think I understand what you said here due to the lack of technical background, but what do you suggest at this moment to a casual applications programmer who wish to use snd-bebob with Orpheus? ------------ Answer 1 ------------ It should work. I received many successful reports for my final call for testing last year. It must be a hardware problem with your PC, or Firewire Cable, or Orpheus, or all of them. ------------ Answer 2 ------------ It should work. Check your software environment. Freshly installed Ubuntu 14.10 or later with the kernel 3.16 or later may not work. Always download the latest sources from GitHub, etc. ------------ Answer 3 ------------ Forget about snd-bebob, and fallback to libffado+jackd. The combination is loosing many packets, but if you say you do not notice any glitches, you should be completely happy with it. *** Regards, Mike Ood

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 23 May '15

23 May '15

> > I have done a couple of things > First I left the laptop off with no power connected; > This made the laptop awake in the non functional state regarding the soundcard. > I then used this opportunity to update to the latest kernel. > > With the latest kernel, the sound is still not working. > Unmuting the S/PDIF in alsamixer turn on red light in the headphone jack. > > also, using 3stack-digout option doesn't bring back sound. > > This is the output from the alsa information tool > http://www.alsa-project.org/db/?f=df46a9ee01e04dde11b194e0278c9f22594540d0 > > after updating this is the output from various commands > > [ > Nothing plugged in: > ./hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > ./hda-verb /dev/snd/hwC0D0 0x1{4,5,8} SET_PIN_SENSE 0 nid > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > > nid = 0x18, verb = 0x709, param = 0x0 > value = 0x0 > > ./hda-verb /dev/snd/hwC0D0 0x1{4,5,8} GET_PIN_SENSE 0 nid > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x80001f41 > > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x80001f41 > > nid = 0x18, verb = 0xf09, param = 0x0 > value = 0x7fffffff > ] > > [ > With headphone plugged in > ./hdajacksensetest/hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > // and also ecact same output for hda-verb as with nothing plugged in > ] > > [ > with Mic plugged in, the following items changed > > ./hdajacksensetest/hdajacksensetest -a > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Internal Mic): present = Yes > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Not connected): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Internal SPDIF Out, Mobile-In): present = No > Pin 0x1f (Not connected): present = No > > ./hda-verb /dev/snd/hwC0D0 0x15 GET_PIN_SENSE 0 nid > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > ./hda-verb /dev/snd/hwC0D0 0x18 GET_PIN_SENSE 0 nid > nid = 0x18, verb = 0xf09, param = 0x0 > value = 0x80001f41 Seem impedance measurement is not correct, mic should have higher impedance than headphone https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… You can disable it by hint trigger_sense=0 - trigger_sense (bool): indicates that the jack detection needs the explicit call of AC_VERB_SET_PIN_SENSE verb Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Mic Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x0121411f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0c To get back headphone phantom jack if HP is really shared with speaker Disable hp mic by hint hp_mic_detect=0 - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true

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[alsa-devel] Assert in pcm_params.c
by Alan Horstmann 21 May '15

21 May '15

Hi, The following assert is being hit in pcm_params.c: "pcm_params.c:2249: snd1_pcm_hw_params_slave: Assertion `err >= 0' failed." This is from a developer using Portaudio in his own application; I have yet to ascertain full details, but it is trying successive sample rates to see which are valid. Does this point to a flaw in the configuration or possibly a bug in Alsa-lib or can it occur just as a result of misuse of the Alsa API? Regards Alan

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[alsa-devel] [RFC PATCH 00/14] ASoC: qcom: add support to apq8016 audio
by Srinivas Kandagatla 21 May '15

21 May '15

Hi All, This patchset adds apq8016 audio support into lpass driver. Existing Lpass driver can not be used as-it-is for apq8016 as it contains code specific to ipq806x. Also the driver only supports single i2s port, single dma channel and single bitclk control. APQ8016 has 4 MI2S( Primary, Secondary, Tertiary, Quaternary) which can be routed to internal wcd codec or external codecs. This routing is controlled by 2 mux registers. This patch series firstly re-organizes the lpass driver such that the SOC specific bits are moved away from the driver. And secondly the SOC specifics are now passed as lpass variant data which would include various register offsets, dma channel allocations and SOC specific clock handling. Finally the patchset add apq8016 lpass and machine driver. This patchset also has two trivial cleanup patches which are to do with redundant checks and removing unnecessary header files. All these patches are tested for HDMI audio via adv7533 bridge and Analog audio on APQ8016-SBC and msm8916-mtp boards. I dont have access to ipq806x boards to test these patches. This is very first version of the patches which was developed with very mimimal/no access to IP documentation. I would like to get your opinon on the over all approch. Kenneth/Patrick, Could you please try these patches on storm board? Thanks, srini Srinivas Kandagatla (14): ASoC: qcom: Remove redundant error check. ASoC: qcom: remove unnecessary header files ASoC: qcom: move ipq806x specific bits out of lpass driver. ASoC: qcom: remove hardcoded i2s port number ASoC: qcom: remove hardcoded dma channel ASoC: qcom: support bitclk and osrclk per i2s port ASoC: qcom: add no osr clk flag to lpass variant ASoC: qcom: add dma channel control offset to variant data ASoC: qcom: Add ability to handle interrupts per dma channel ASoC: qcom: add bit map to track static dma channel allocations ASoC: qcom: Add apq8016 lpass driver support ASoC: qcom: add apq8016 sound card support ASoC: qcom: Document apq8016 bindings. ASoC: qcom: document apq8016 machine driver bindings .../bindings/sound/qcom,apq8016-machine.txt | 61 ++++++ .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 13 +- sound/soc/qcom/Kconfig | 25 ++- sound/soc/qcom/Makefile | 5 + sound/soc/qcom/apq8016.c | 214 ++++++++++++++++++ sound/soc/qcom/lpass-apq8016.c | 243 +++++++++++++++++++++ sound/soc/qcom/lpass-cpu.c | 234 ++++++++++---------- sound/soc/qcom/lpass-ipq806x.c | 95 ++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 --------------- sound/soc/qcom/lpass-lpaif-reg.h | 167 ++++++++++++++ sound/soc/qcom/lpass-platform.c | 197 +++++++++++------ sound/soc/qcom/lpass.h | 54 ++++- 12 files changed, 1121 insertions(+), 359 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-machine.txt create mode 100644 sound/soc/qcom/apq8016.c create mode 100644 sound/soc/qcom/lpass-apq8016.c create mode 100644 sound/soc/qcom/lpass-ipq806x.c delete mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-lpaif-reg.h -- 1.9.1

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[alsa-devel] snd-hda-intel : issues with mic boost on ALC3236
by Simon Derr 21 May '15

21 May '15

Hello, I'm having issues with the internal microphone on an Asus R553 (or S551) laptop. With the boost set to zero the microphone works but the level of the recording is very low. If I try another boost setting I don't get any signal anymore. I have uploaded alsa-info data here : http://www.alsa-project.org/db/?f=55b7fb0b4998757d099addae0a411e6a7cebc079 This is on a Linux 4.0 kernel. Any help is welcome, Regards, Simon

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[alsa-devel] Intel HDA / ca0132: support for Alienware 15 Creative Sound Core3D-EX
by Gabriele Martino 19 May '15

19 May '15

Hi, the new Alienware 15 laptop (2015, not the M15x) has a Creative Sound Core3D-EX Audio which seems to work with ca0132. I'm using a 4.0.0 kernel with gentoo patchset, but with the default configuration: - the line out/headphone jack doesn't work, it only outputs some white noise - alsamixer shows only "Master" and "PCM" channel, no "Front", "Line Out" or similar - alsamixer has a toggle switch for the "HP/Speaker" but no headphone/line out switch - the jack detection is enabled on the "line in" jack instead of the "line out" (if I plug something on the "line in", the speakers are muted) - if I play too much with these switches, the speakers stop working until suspend/reboot To fix the first issue I run "hdajacksensetest" to get the headphone jack pin: hdajacksensetest -c 1 -a Pin 0x0b (Internal Speaker): present = Yes Pin 0x0c (Not connected): present = No Pin 0x0d (Not connected): present = No Pin 0x0e (Not connected): present = No Pin 0x0f (Not connected): present = Yes Pin 0x10 (Not connected): present = No Pin 0x11 (Black Line In, Left side): present = No Pin 0x12 (Internal Mic, Mobile-In): present = No Pin 0x13 (Not connected): present = No Pin 0x18 (Not connected): present = No and replaced the pin number: --- /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c.old 2015-04-13 00:12:50.000000000 +0200 +++ /usr/src/linux-4.0.0-gentoo/sound/pci/hda/patch_ca0132.c 2015-04-25 01:43:09.607531550 +0200 @@ -4625,7 +4625,7 @@ spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ - spec->out_pins[1] = 0x10; /* headphone out */ + spec->out_pins[1] = 0x0f; /* default: 0x10 - headphone out -0x0f on Alienware 15 2015 */ spec->shared_out_nid = 0x2; spec->num_inputs = 3; Now the "HP/Speaker Auto Detect" is ignored on both jacks, but I can switch between headphones and speakers manually with the "HP/Speaker" item. This is a dirty hack (and probably won't work on different cards), is there a better way to fix this and get at least the jack detection working? Regards, Gabriele

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[alsa-devel] Low volume since linux 3.19
by Jan Hinnerk Stosch 18 May '15

18 May '15

Hi, I hope this is the right place to report my issue!? Since linux 3.19 I have very low volume on my Fujitsu E753 headphone even on highest volume level. As I don't have much experience with audio configuration and it worked before I did a kernel bisect and found out, that commit "03ad6a8c93b6df2d65c305b5b5f9474068b45bfb" is "responsible" for this. Is there something that I have to change in my configuration? Or do you need some further information? Here is some output from dmesg: [ 5.853009] sound hdaudioC0D0: ALC269VB: SKU not ready 0x909701f0 [ 5.853857] sound hdaudioC0D0: autoconfig: line_outs=1 (0x21/0x0/0x0/0x0/0x0) type:line [ 5.853860] sound hdaudioC0D0: speaker_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 5.853862] sound hdaudioC0D0: hp_outs=1 (0x1a/0x0/0x0/0x0/0x0) [ 5.853864] sound hdaudioC0D0: mono: mono_out=0x0 [ 5.853866] sound hdaudioC0D0: inputs: [ 5.853869] sound hdaudioC0D0: Mic=0x18 [ 5.853871] sound hdaudioC0D0: Dock Mic=0x1b [ 5.853873] sound hdaudioC0D0: Internal Mic=0x12 Thanks in advance, jhs

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Re: [alsa-devel] [pulseaudio-discuss] ThinkPad T-510 audio output mute LED non-workingness
by Takashi Iwai 16 May '15

16 May '15

[Switched Cc to alsa-devel ML, as it's irrelevant from PA but rather a driver issue.] At Mon, 27 Apr 2015 07:58:59 -0600, Glenn Golden wrote: > > Takashi Iwai <tiwai(a)suse.de> [2015-04-27 08:50:34 +0200]: > > > > Gah, I hate reports without alsa-info.sh output (not a link to > > somewhere else). > > > > Glenn, please attach alsa-info.sh outputs (run it with --no-upload > > option) and the whole dmesg output after boot. 3.19 should already > > contain the hook to thinkpad_acpi.c. If anything is missing, you > > should have a warning indicating it. > > > > Apologies for the omissions. Both are attached. Thanks! > Fwiw -- perhaps nothing at this point -- a good deal of info has also been > posted to the kernel bugtracker ticket: > > https://bugzilla.kernel.org/show_bug.cgi?id=96171 > > Re: "...you should have a warning indicating it": I'm an unskilled laboror > here w.r.t. kernel affairs, and not even sure what sort of warnings to be > looking for, so may have missed something important without realizing it. > Just let me know what add'l info you need, I'll be happy to post it. I do > appreciate your attention to the issue. I see no such warning in the log you attached, so it's not the issue, as it seems. Below is a debug patch, and now you should see messages like "XXX ..." in kernel log at loading the driver. If there is no such a message, it means that the thinkpad hook isn't built in at all. Please check the kernel log again with the patch. Takashi --- diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 0a4ad5feb82e..48ce61e46599 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -34,6 +34,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); + pr_info("XXX mute led updated %d\n", enabled); if (led_set_func) led_set_func(TPACPI_LED_MUTE, !enabled); } @@ -58,8 +59,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + codec_info(codec, "XXX hda_fixup_thinkpad_acpi init...\n"); if (!is_thinkpad(codec)) return; + codec_info(codec, "XXX ok, this is thinkpad\n"); if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -70,18 +73,24 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = true; if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + codec_info(codec, "XXX ok, mute LED is hooked\n"); old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; + } else { + codec_err(codec, "XXX mute LED NOT WORKING!\n"); } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { if (spec->num_adc_nids > 1) codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); else { + codec_info(codec, "XXX ok, mic LED is hooked\n"); spec->cap_sync_hook = update_tpacpi_micmute_led; removefunc = false; } + } else { + codec_err(codec, "XXX mic LED NOT WORKING!\n"); } }

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[alsa-devel] [PATCH v2 00/17] new locomo driver set
by Dmitry Eremin-Solenikov 13 May '15

13 May '15

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. With this patchset I tried to modernise and restructure the LoCoMo drivers. * Use platform bus and mfd core code * Use GPIO API instead of custom locomo GPIO code * Use irqdomains to manage IRQs * Split mixed locomolcd driver to lcd and backlight parts * Implement IRQ handling in GPIO driver * Implement SPI driver used as a host for mmc_spi device to control SD cards V1 was submitted about a half of a year ago. Major changes: * Use gpiod_* in most of places * Use regmap for accessing registers instead of manual RMW cycles * Split dac driver from core module to i2c bus and separate IIO driver * Replaced mdelay() calls with usleep_range or msleep where it was suitable * Rewrote SPI driver to use core transfer_one() interface

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