November 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] wm8974 and sam9g45 atmel
by Michele Da Rold 11 Jul '14

11 Jul '14

Hello everybody, I'm starting a new development with a Atmel SAM9G45 and a WM8974. I'm using kernel 2.6.33.2 I have connected the SSC1 pin TD,TF,TK and the RD pin, so I want to use WM8974 like a slave. I wrote a interface driver based on sam9263_wm8731.c The driver seem to load ok, I recognize the codec but when I start a mp3 playing (I use mpg123) the playback not start. Putting some dedug messagge the driver not call the function "set_bias_level", and call the mute function with mute=0. Testing the MCLK signal I not see any clock on pin because I never call clk_enable!! There is a way to identify the problem? Someone could help me? Thank you in advance Michele Da Rold I put the code of driver (don't care about printk I used it for debug purpose!!) /* * sam9g45_wm8974 -- SoC audio for AT91SAM9G20-based * ATMEL AT91SAM9G20ek board. * * Copyright (C) 2005 SAN People * Copyright (C) 2008 Atmel * * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> * * Based on sam9263_wm8731.c. * Frank Mandarino <fmandarino(a)endrelia.com> * Copyright 2006 Endrelia Technologies Inc. * Based on corgi.c by: * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that 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, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8974.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12000000 /* * As shipped the board does not have inputs. However, it is relatively * straightforward to modify the board to hook them up so support is left * in the driver. */ #undef ENABLE_MIC_INPUT #define ENABLE_MIC_INPUT 1 static struct clk *mclk; static int at91sam9g45ek_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->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_hw_params() called\n"); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; return 0; } static struct snd_soc_ops at91sam9g45ek_ops = { .hw_params = at91sam9g45ek_hw_params, }; static int at91sam9g45ek_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { static int mclk_on; int ret = 0; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_set_bias_level() called %d\n",level); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: if (!mclk_on) ret = clk_enable(mclk); if (ret == 0) mclk_on = 1; break; case SND_SOC_BIAS_OFF: case SND_SOC_BIAS_STANDBY: if (mclk_on) clk_disable(mclk); mclk_on = 0; break; } return ret; } #if 0 static const struct snd_soc_dapm_widget at91sam9g45ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* speaker connected to SPKOUTP */ {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Input PGA */ {"Input PGA", "Aux Switch", "Aux Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, }; /* * Logic for a wm974 as connected on a at91sam9g45ek board. */ static int at91sam9g45ek_wm8974_init(struct snd_soc_codec *codec) { struct snd_soc_dai *codec_dai = &codec->dai[0]; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45_wm8974_init() called\n"); /* ret = snd_soc_dai_set_sysclk(codec_dai, 0, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8974 SYSCLK: %d\n", ret); return ret; } */ /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g45ek_dapm_widgets, ARRAY_SIZE(at91sam9g45ek_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "MONOOUT"); snd_soc_dapm_nc_pin(codec, "AUX"); #ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); #else snd_soc_dapm_nc_pin(codec, "Int Mic"); #endif /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); return 0; } #endif static struct snd_soc_dai_link at91sam9g45ek_dai = { .name = "WM8974", .stream_name = "WM8974 PCM", .cpu_dai = &atmel_ssc_dai[1], .codec_dai = &wm8974_dai, #if 0 .init = at91sam9g45ek_wm8974_init, #endif .ops = &at91sam9g45ek_ops, }; static struct snd_soc_card snd_soc_at91sam9g45ek = { .name = "AT91SAMG45", .platform = &atmel_soc_platform, .dai_link = &at91sam9g45ek_dai, .num_links = 1, .set_bias_level = at91sam9g45ek_set_bias_level, }; static struct snd_soc_device at91sam9g45ek_snd_devdata = { .card = &snd_soc_at91sam9g45ek, .codec_dev = &soc_codec_dev_wm8974, }; static struct platform_device *at91sam9g45ek_snd_device; static int __init at91sam9g45ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; struct clk *main; int ret; at91_set_A_periph(AT91_PIN_PD27, 0); /* * Codec MCLK is supplied by PCK1 - set it up. */ mclk = clk_get(NULL, "pck1"); if (IS_ERR(mclk)) { printk(KERN_ERR "ASoC: Failed to get MCLK\n"); ret = PTR_ERR(mclk); goto err; } main = clk_get(NULL, "main"); if (IS_ERR(main)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk; } ret = clk_set_parent(mclk, main); clk_put(main); if (ret != 0) { printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); goto err_mclk; } clk_set_rate(mclk, MCLK_RATE); /* * Request SSC device */ ssc = ssc_request(1); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 1\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; at91sam9g45ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g45ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(at91sam9g45ek_snd_device, &at91sam9g45ek_snd_devdata); at91sam9g45ek_snd_devdata.dev = &at91sam9g45ek_snd_device->dev; ret = platform_device_add(at91sam9g45ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g45ek_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; err: return ret; } static void __exit at91sam9g45ek_exit(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(at91sam9g45ek_snd_device); at91sam9g45ek_snd_device = NULL; clk_put(mclk); mclk = NULL; } module_init(at91sam9g45ek_init); module_exit(at91sam9g45ek_exit); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("ALSA SoC AT91SAM9G45_WM8974"); MODULE_LICENSE("GPL");

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[alsa-devel] Segfaults in snd_pcm_area_silence() when using softvol
by Tanu Kaskinen 18 May '14

18 May '14

Hi, A bug was reported to pulseaudio about a segfault in alsa-lib code when using softvol: https://bugs.freedesktop.org/show_bug.cgi?id=51060 . It has been observed by several people, and this mail was prompted by the last bug comment, which rightly recommends getting in touch with the alsa developers: > I've had this problem too and the problem only occurs with > -ftree-vectorize. > > It looks to me as if this is a gcc bug, as it produces a "movdqa" or > "vmovdqa" instruction which is used with an unaligned pointer. > > Now, I have no clue why gcc assumes the pointer would be an aligned > one, maybe it is even right and something else is wrong, but I haven't > spent too much time digging into this. > > (note that it still happens with gcc 4.7 and the current 4.8 > development one) > > Also, this bug should probably be brought up with the alsa guys as > well, as I don't think pulseaudio is at fault here. Here's the stack trace for your convenience (see the linked bug for more details) #0 snd_pcm_area_silence (dst_area=0x7fffe875bb00, dst_offset=<optimized out>, samples=2, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2549 #1 0x00007fffe87b4398 in snd_pcm_areas_silence (dst_areas=0x5555557fa100, dst_offset=891, channels=2, frames=439, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2662 #2 0x00007fffe87f9bdb in softvol_convert_stereo_vol (svol=0x555555813cf0, dst_areas=0x5555557fa0e0, dst_offset=891, src_areas=0x5555557fa0e0, src_offset=<optimized out>, channels=2, frames=439) at pcm_softvol.c:286 #3 0x00007fffe87fae58 in snd_pcm_softvol_write_areas (pcm=0x55555581db60, areas=0x5555557fa0e0, offset=891, size=439, slave_areas=0x5555557fa0e0, slave_offset=891, slave_sizep=0x7fffe875bcd0) at pcm_softvol.c:607 #4 0x00007fffe87c6fc7 in snd_pcm_plugin_mmap_commit (pcm=0x55555581db60, offset=<optimized out>, size=439) at pcm_plugin.c:424 #5 0x00007fffe87b9676 in snd_pcm_mmap_commit (pcm=<optimized out>, offset=<optimized out>, frames=<optimized out>) at pcm.c:6584 #6 0x00007fffe8a74876 in mmap_write (on_timeout=false, polled=false, sleep_usec=<synthetic pointer>, u=0x5555557e6d40) at modules/alsa/alsa-sink.c:636 #7 thread_func (userdata=0x5555557e6d40) at modules/alsa/alsa-sink.c:1694 #8 0x00007ffff722c039 in internal_thread_func (userdata=0x55555587ef80) at pulsecore/thread-posix.c:83 #9 0x00007ffff4a24e2c in ?? () from /lib64/libpthread.so.0 #10 0x00007ffff3ccc1dd in clone () from /lib64/libc.so.6 -- Tanu

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[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi: ATI/AMD multi-channel and HBR support
by Anssi Hannula 14 May '14

14 May '14

Hi all! Here's another revision of the ATI/AMD multichannel+HBR patchset. Changes: - added customizable callbacks instead of polluting generic code - fixed wrong codec id in rev3+ check (as previously noted) - cosmetics I've only tested this on NVIDIA/Intel, but I'd guess at least Peter should be able to test this on AMD in a few days or so. The earlier revisions were tested to fully work. Combined patch can be found at: http://onse.fi/files/atihdmi5.patch However, this patchset is on top recent sound-next commits so it does not directly apply on older kernels. Olivier, I hope you will be able to test this at some point as well (same things as last time), even though it will be a bit more work :) Anssi Hannula (5): ALSA: hda - hdmi: Allow HDA patches to customize more operations ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs sound/pci/hda/hda_eld.c | 151 +++++++++++ sound/pci/hda/hda_local.h | 4 + sound/pci/hda/patch_hdmi.c | 643 +++++++++++++++++++++++++++++++++++++-------- 3 files changed, 689 insertions(+), 109 deletions(-) -- Anssi Hannula

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[alsa-devel] Ambiguous DAPM widget names and DPCM
by Daniel Mack 21 Mar '14

21 Mar '14

Hi Mark, Hi Liam, I spent some hours reverse-engineering the DPCM code, and I plan to prepare a patch to add some documentation once I've assembled a fully working setup. However, I have a general question regarding DAPM widget names. DPCM uses DAPM widgets and routes in order to determine the FE <-> BE connections at runtime. The problem with this is that widget names are very ambiguous, as the automatically created widgets for DAIs are named after their stream_name properties, which contain "Playback" or "Capture" for all components in my case. Hence, the code which walks the connections (snd_soc_dapm_dai_get_connected_widgets()) will most likely match the wrong one (which has empty sound/sink lists), which results in a failure like this: [ 16.606268] fe: ASoC: fe no valid playback route I hacked the names of all components so they are unique, but even then, the problem is that snd_soc_dapm_new_dai_widgets() is called twice for the CPU DAI, once from soc_probe_platform(), and later from soc_probe_link_dais(), and both calls would create DAPM widgets with identical names, resulting in the same problem as described above. I don't know how to handle this, but I believe that nobody was really hit by this issue yet, as DPCM doesn't seem to be used widely, at least not by any machine code in mainline. One idea would be to change the automatically generated names of DAI widgets in order to make them unique, or re-factor the lookup routines. Any opinion on that? Do I miss a general consideration here? With more hacks in place, the setup somehow succeeds, but when the stream is opened, we'll hit a BUG() in sound/core/pcm_memory.c, because the FE device's DMA type is undefined. But that's another topic I'll address later. Thanks, Daniel

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Re: [alsa-devel] Zoom R16
by Jason Mancine 11 Mar '14

11 Mar '14

Can't seem to compile 3.11 or 3.12 for the distro I am using (AVLInux)... tested it in Ubuntu Studio 13.10 and the quirk didn't seem to have any effect. The output of lsusb is the same, but the device doesn't show up as a sound card. On Sun, Nov 24, 2013 at 9:18 AM, Daniel Mack <daniel(a)zonque.org> wrote: > On Sun, Nov 24, 2013 at 09:03:47AM -0500, Jason Mancine wrote: > > Thanks, I will give it a try. By recent kernel do you mean latest > mainline > > or should my distro's 3.6.11 be new enough? > > Anything >= 3.11 should do. > > In further replies, please do not drop the alsa-devel mailing list. > > > Thanks, > Daniel >

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[alsa-devel] Update patch for codecs alias name for Dell
by Kailang 05 Mar '14

05 Mar '14

Hi Takashi, Dell assigned more codecs for alias name. I update in attach patch. Thanks. Kailang

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[alsa-devel] Unsupported Valley View 'LPE' adapter?
by Adam Williamson 27 Feb '14

27 Feb '14

Hi, folks. I recently bought a Dell Venue 8 Pro - a Bay Trail / Valley View-based Intel tablet - to try and get Fedora running on it. Among the many things I'm currently fighting :) , the sound doesn't work. snd-hda-intel is not loaded at all, and there are no audio devices present at any level. In Windows' Device Manager I see an 'Intel SST Audio Device' with PCI ID 0x86:0f28 . Indeed, the PCI ID database has this listed as "ValleyView LPE Audio Controller" - http://pci-ids.ucw.cz/read/PC/8086/0f28 . But hda-intel does not have 0f28 as a supported device. The commit for 'Bay Trail' audio support added the ID 0f04: https://lkml.org/lkml/2013/5/16/38 So...what is this 0f28? Is support for it going to be added? Is there any info I can provide to help? (I can boot Linux on the tablet and get info out of it _just barely_, but it's pretty painful - there's no wifi and an install to 'disk' (really a usb stick) doesn't work too well, so I'm stuck working in a live image with no network). -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | XMPP: adamw AT happyassassin . net http://www.happyassassin.net

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[alsa-devel] [PATCH RFC 0/9] Beaglebone-Black HDMI audio
by Jyri Sarha 24 Jan '14

24 Jan '14

This set of patches implements PCM HDMI audio driver for Beaglebone-Black HDMI output. The patches have been rebased on top of: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound topic/davinci There is couple of earlier RFC versions of the clk patch floating around, but the one here is the latest version. I'll send the associated dts changes shortly as a separate patch. Best regards, Jyri Jyri Sarha (9): clk: add gpio controlled clock ASoC: davinci-evm: Add named clock reference to DT bindings ASoC: davinci-evm: HDMI audio support for TDA998x trough McASP I2S bus ASoC: hdmi-codec: Add devicetree binding with documentation ASoC: hdmi-codec: Add SNDRV_PCM_FMTBIT_32_LE playback format ASoC: davinci: HDMI audio build for AM33XX and TDA998x drm/tilcdc: Add I2C HDMI audio config for tda998x ARM: OMAP2+: omap2plus_defconfig: Enable tilcdc and TDA998X HDMI support ARM: OMAP2+: omap2plus_defconfig: Enable BeagleBone Black HDMI audio support .../devicetree/bindings/clock/gpio-clock.txt | 21 ++ .../bindings/sound/davinci-evm-audio.txt | 12 +- Documentation/devicetree/bindings/sound/hdmi.txt | 17 ++ arch/arm/configs/omap2plus_defconfig | 5 + drivers/clk/Makefile | 1 + drivers/clk/clk-gpio.c | 209 ++++++++++++++++++ drivers/gpu/drm/tilcdc/tilcdc_slave.c | 24 ++- include/linux/clk-provider.h | 25 +++ sound/soc/codecs/hdmi.c | 12 +- sound/soc/davinci/Kconfig | 12 ++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 227 +++++++++++++++++++- 12 files changed, 558 insertions(+), 8 deletions(-) create mode 100644 Documentation/devicetree/bindings/clock/gpio-clock.txt create mode 100644 Documentation/devicetree/bindings/sound/hdmi.txt create mode 100644 drivers/clk/clk-gpio.c -- 1.7.9.5

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[alsa-devel] HD-audio runtime PM
by Takashi Iwai 24 Jan '14

24 Jan '14

Hi, after my previous fix, the runtime PM seems working stably finally. However, there seem still some glitches: 1. The wakeup via jack or HDMI/DP detection doesn't seem to work on my test machines. WAKEEN is set properly. And its value can be read correctly at the point of runtime resume, too. 2. We don't change the device to D3 in runtime suspend. So I guess we save little power as of now. Actually, setting to D3 and resuming to D0 works fine, as far as I tested -- except for HDMI/DP ELD read out on Haswell. Although ELD read doesn't give any errors, the received bytes are corrupted. In my case, the read starts from offset 0x1c. The HDMI can be still played even at that state, but then it falls back to the stereo 2ch mode, of course. The tests were performed on both Haswell (DP and LynxPoint-LP) and IvyBridge (PantherPoint) machines. The first problem has been present, and it's also the behavior of some old chips where the codec can't go to sleep with unsol wakeup. So, in this regard, it's no regression, per se. But certainly it's no good thing. I thought Mengdong once made WAKEEN working. Mengdong, do you remember what was the condition? Could you check whether the latest kernel (at best sound git tree for-linus branch) still works? About the second point: we should do D3 if we do care power, and that's the very right reason for runtime PM. But the ELD issue is a bad bug, and it makes me wonder whether it's a graphics driver side issue or a codec side. As a quick workaround, we can implement D3 on controllers but for Haswell HDMI/DP. In anyway, this is no urgent issue, clearly targeted for 3.14. The whole patch series are found in my sound-unstable git tree test/hda branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git thanks, Takashi

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Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 08 Jan '14

08 Jan '14

At Thu, 6 Sep 2012 15:17:58 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 15:08 +0200, Takashi Iwai wrote: > > At Thu, 6 Sep 2012 11:43:48 +0200, > > Markus Trippelsdorf wrote: > > > > > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > > > At Thu, 06 Sep 2012 09:35:26 +0200, > > > > Takashi Iwai wrote: > > > > > > > > In short, a patch like below may fix the issue (note: completely > > > > untested!) > > > > > > No it doesn't, unfortunately... > > > > OK, I start tracking down the problem a bit more deeply now. > > > > The issue happens when the first two URBs are passed to > > retire_playback_urb(). These are URBs filled before start_endpoints() > > are set, so they contain actually zero size. Even though these are > > a sort of dummy packets, the driver still tries to check with the > > queued delay account, and gives bogus errors. > > > > So, essentially the messages are harmless and nothing to worry too > > much, but surely it doesn't look sexy. > > > > The patch below should fix the problem. Please give it a try. > > Yes, your patch finally fixes the problem. > Thank you Takashi-san. Thanks for your quick test! If Daniel has no objection with that patch, I'm going to merge it. Takashi

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