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Re: [alsa-devel] another ALSA question
by Jaroslav Kysela 28 May '07

28 May '07

I'm CCing alsa-devel because this discussion might be interesting for other developers, too. On Mon, 28 May 2007, Maarten Lankhorst wrote: > Jaroslav Kysela schreef: > > On Mon, 28 May 2007, Maarten Lankhorst wrote: > > > >> After playing around a little more with alsa I tried to do manipulation > >> of pointer using the instructions, I wrote some simple code either does > >> rewind or forward to get to the right point, taking in respect the > >> amount of delay and where to go. > >> > >> However it doesn't seem to work: Looking at the source code it seems > >> rewinding isn't implemented, using the #if 0'd code there gives problems > >> when doing snd_pcm_forward later: It then ends up at the pointer where > >> the sound is currently being played at. > >> > > > > Not all alsa-lib plugins are capable or have implemented this fine > > position control (check error value). > > > For a plugin like dmix, I would expect it to be implemented, especially > since most cards sold are integrated cards that don't have multiple > sound buffers. Go ahead and send us patches. We're doing open source :-) Anyway any rewind operation mens "undo" - it will eat extra CPU ticks for nothing. > >> As workaround I used the following in my own code: > >> typeof(snd_pcm_forward) snd_pcm_mmap_appl_forward; > >> #define snd_pcm_forward snd_pcm_mmap_appl_forward > >> > >> typeof(snd_pcm_rewind) snd_pcm_mmap_appl_backward; > >> #define snd_pcm_rewind snd_pcm_mmap_appl_backward > >> > >> This works, but I wonder if there is a cleaner way to do it. > >> > > > > It won't work with all plugins and mentioned functions are not part of > > public API. > > > As I work in wine on it's DirectSound implementation, I can tell it > relies heavily on the fact that it can lock a buffer at an arbitrary > position, and that it works with absolute pointers, rather then > relative. So if snd_pcm_rewind/snd_pcm_forward don't work as expected, > an alsa driver for it would become nearly useless, I don't expect it to > always work, but I would expect it to work for a common plugin like dmix > at least. > >> Also, I noticed some audio glitches when using this in combination with > >> alsa-lib's resampling, is that because of the way it is done or is > >> something else wrong? > >> > > > > You should not use mentioned functions. > > > > You can try to use the direct hardware access (hw:CARD_NAME) devices > > (you'll lose all stream conversions from the ALSA side) or use an > > intermediate buffer for fine position control and configure ALSA to use > > smallest ring buffer as possible (to minimalize latency). > > > > Jarosla > I wish that would work, directsound expects to have access to a fixed > size buffer, and if I go for direct hardware access, I lose the ability > to work properly with most cheap cards on the market, since people want > to be able to listen to music on background while gaming. For now it > seems I have to use the internal calls in case of dmix, or modify > already committed data, which is what winealsa's current driver does. I > don't know what is the least of those 2 evils, or if there is a third > option. I'm open for advice. I'm afraid, that you're trying to exactly copy DirectSound API. Playing with the buffer position in this way is always dangerous, because it's really hard to predict the final result in the multitasking environment. Think about these proposals: 1) what sample (ring buffer) rewritting is supposed to do - maybe you can change only not yet commited samples with satisfactory results 2) use a small ALSA ring buffer and use own "big" ring buffer for changes before you commit samples to ALSA's ring buffer - you'll increase a bit sample flow latency, but you can do anything you want (you just add extra buffering to avoid rewind / forward for ALSA API) 3) fix (or better implement) dmix plugin rewind function Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] [ALSA] ALSA Power Management, Drivers behaving unexpectedly after suspend/resume cycle
by Nobin Mathew 28 May '07

28 May '07

I am implementing power management in my ALSA sound drivers. ALSA drivers are not working properly after resume. All registers values are proper and I am able to read back the codec register contents. If I do a playback after resume then I am getting lots of "underrun". After a reboot playback is working fine. Is there any known issues with ALSA power management? Why Ubuntu and Redhat does ALSA modules removal before suspend and insertion after resume?

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[alsa-devel] Reasonable sound hardware parameters?
by Joachim Förster 28 May '07

28 May '07

Hi ALSA devs, as I already described in one of my last posts on the list, I'm going to write a ALSA driver for a sound controller which will also be "written" at the same time (VHDL for FPGA) - on top of an embedded system (Xilinx ML403 EvalBoard, PowerPC 405). We will do playback feature only. The point is, that we will have _no_ real DMA and we don't want to use the copy/silence or ack/workqueue (with intermediate buffer). So we decided to move the ring buffer, which usually exists in RAM into the sound controller and handle it as some kind of device IO memory, which should be mmap'able (kernel function io_remap_pfn_range()). So, the important goal is, to have mmap feature in the ALSA driver. First question: To implement mmap on such kind of device IO memory, I have to use snd_pcm_lib_mmap_iomem() as mmap() callback - like it is used in the rme32 driver, right? Second question: Do the following parameters make sense regarding the size of the HW ring buffer and period sizes: .buffer_bytes_max=16*1024; .period_bytes_max=8*1024; .period_bytes_min=?? resonable value ?? .periods_min=2; .periods_max=?? resonable value ?? Furthermore, the sound controller will fire an interrupt after one period is played. Are there any arguments against these values? Any comments? Thanks, Joachim

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[alsa-devel] alsa playback channel interleaving question
by Matthias Nyffenegger 27 May '07

27 May '07

Hi, Earlier this year I wrote to this list: > I am trying to write an alsa driver for my audiowerk8 soundcard. > The aw8 uses a Philips saa7146 chipset. There are 2 capture and > 8 playback channels mapped on the saa7146's DMA as follows: > > DMA1: 2xcapture > DMA2: 4xplayback > DMA3: 4xplayback > > Sample format is 16bit. > > Frame format is as follows: > (A number corresponds to a channel, a stereo channel-pair is embraced > by [] and 1 digit corresponds to one sample byte) > > capture channels [0,1] on DMA1: 00--11-- > playback channels [2,3],[4,5] on DMA2: 22443355 > playback channels [6,7],[8,9] on DMA3: 66887799 I tried to find a way to handle this in the card, but this special interleaving for the playback channels is by hardware restriction and can't be worked around in the soundcard itself. AFAIK alsa provides the samples interleaved like 22334455 (66778899). To avoid channel switching I guess some special processing in the alsa driver is required. Where is this done best? Do I have to move around the samples in the interrupt handler or does alsa provide some (possibly better) means to achive the correct (for the soundcard) interleaving? Regards, Matthias

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[alsa-devel] Suggestion for getting alsa to use Conexant
by Victor Schneider 26 May '07

26 May '07

How about being able to parameterize Alsa modules so that the intel 80xm module finds any hda modem that also acts as an Intel hda chip? I am referring to the Conexant combination modem and hda chip that gets identified as "Conexant Waikiki" by alsamixer, but that 80xm can't find as a modem. Giving the 80xm module the device:identifier value to override its detection of the Intel version of the Conexant hda modem might let slmodemd work.

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Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support for SMDK2440boards
by Zoltan Devai 25 May '07

25 May '07

2007/5/24, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > I have a SMDK2410 and I am trying to run kernel 2.6.20 on it. The kernel > is running Ok, but the sound is dead. I gonna try your patch and I would > like to know which files from the RK's patch have you used? The attached two patches should enable the sound support fully. One of them adds the l3-bus, the other one the necessary Kconfig and Makefile changes, along with an ASoC hack. Add this patch also: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/001105.html Everything should apply cleanly to 2.6.22-rc2, but probably also to 2.6.21. Zoltan

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[alsa-devel] Sound on MSI S420
by Baruch Even 25 May '07

25 May '07

Hello, I've succeeded by guessing to get my MSI S420 laptop sound to work. Attached is a one-liner patch to do that. The patch is against kernel 2.6.22-rc2. Since the patch is based on guesswork it might not be optimal but it does work for me. Baruch

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Re: [alsa-devel] [PATCH] Add support for Cyrix/NatSemi Geode CS5530 (VSA1)
by Ash Willis 25 May '07

25 May '07

> Thanks. Now committed to HG tree. > > One more possible fix is about the build of snd-sb16-dsp and dependent > modules. We don't need snd-sb16 itself. > > Could you try the patch below (both for alsa-driver and alsa-kernel)? > It introduces the intermediate configs for snd-sb16-dsp, snd-sb8-dsp > and snd-sb-common modules. > Yeah, that makes more sense...and it works fine. Thanks, Ash =

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[alsa-devel] Testing ALSA and OSS drivers
by Pharaoh . 24 May '07

24 May '07

Hi I am planning to write an ALSA driver for an OMAP based board, I already have an OSS driver written for my board, but it is a 2.4 kernel. Question: I am trying to bring up Linux for my board (the board is supported), how can I test whether the oss drivers are working fine or not? I just have to check whether operations like playback and recording etc work as expected. Any pointers will be helpful. Regards, Pharaoh.

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[alsa-devel] [RFC] [PATCH 2/2] ASoC sound support for SMDK2440 boards: board driver
by Zoltan Devai 24 May '07

24 May '07

--- /dev/null +++ /sound/soc/s3c24xx/smdk2440.c @@ -0,0 +1,162 @@ +/* + * smdk2440.c -- SMDK2440 ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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/module.h> +#include <linux/l3.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/arch/regs-iis.h> + +#include "../codecs/uda1341.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" + +static int smdk2440_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_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + unsigned int div = 0; + int ret = 0; + + switch (params_rate(params)) { + case 11025: + div = 16; + clk = 4233600; + break; + case 22050: + div = 8; + clk = 8467200; + break; + case 44100: + div = 4; + clk = 16934400; + break; + default: + return -EINVAL; + } + + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_384FS); + if (ret < 0) + return ret; + + ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div,div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static const char* smdk2440_l3_link_name = "s3c24xx-l3"; + +static struct snd_soc_ops smdk2440_ops = { + .hw_params = smdk2440_hw_params, +}; + +static struct snd_soc_dai_link smdk2440_dai_link = { + .name = "UDA1341", + .stream_name = "UDA1341", + .codec_dai = &uda1341_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &smdk2440_ops, +}; + +static struct snd_soc_machine snd_soc_machine_smdk2440 = { + .name = "SMDK2440", + .dai_link = &smdk2440_dai_link, + .num_links = 1, +}; + +static struct snd_soc_device smdk2440_snd_devdata = { + .machine = &snd_soc_machine_smdk2440, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_uda1341, + .codec_data = &smdk2440_l3_link_name, +}; + +static struct platform_device *smdk2440_snd_device; + +static int __init smdk2440_uda1341_init(void) +{ + int ret; + + printk(KERN_INFO "SMDK2440 ALSA SoC Audio driver\n"); + + if (!(machine_is_s3c2440())) + return -ENODEV; + + smdk2440_snd_devdata.codec_data = l3_get_adapter("s3c24xx-l3"); + if (smdk2440_snd_devdata.codec_data == NULL) { + printk(KERN_WARNING "SMDK2440 SoC Audio: \ + Unable to find L3 bus adapter\n"); + return -ENODEV; + } + + smdk2440_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk2440_snd_device) { + printk(KERN_ERR "SMDK2440 SoC Audio: Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(smdk2440_snd_device, &smdk2440_snd_devdata); + smdk2440_snd_devdata.dev = &smdk2440_snd_device->dev; + ret = platform_device_add(smdk2440_snd_device); + + if (ret) + platform_device_put(smdk2440_snd_device); + + return ret; +} + +static void __exit smdk2440_uda1341_exit(void) +{ + platform_device_unregister(smdk2440_snd_device); +} + +module_init(smdk2440_uda1341_init); +module_exit(smdk2440_uda1341_exit); + +MODULE_AUTHOR("Zoltan Devai"); +MODULE_DESCRIPTION("SMDK2440 ALSA SoC audio driver"); +MODULE_LICENSE("GPL");

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