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Re: [alsa-devel] State of the XFi driver on x86_64
by Toby 03 Nov '08

03 Nov '08

2008/11/3 Takashi Iwai <tiwai(a)suse.de>: > At Mon, 3 Nov 2008 10:42:55 +1100, > Toby wrote: >> >> 2008/11/2 WILMES Andreas <administrator(a)webspace.ms>: >> > Hello everbody, >> > >> > Can someone give the current state of the driver? What are his features, >> > what is working? >> >> I've been watching this thread for a while, and occasionally trying >> out new builds, as I have an original PCI X-Fi emu20k1. >> Everyone else seems to be reporting decent successes, but I have never >> managed it myself. >> >> Playing back sound files at any sample-rate results in a repeating >> burst of static. Adjusting the period/buffer parameters to aplay will >> cause the pitch/tone of the static burst to change, but never turn >> into anything remotely like the desired sound. >> >> I'm running this on a 64-bit system (Ubuntu 8.04 with a custom >> compiled kernel). Is anyone else having success on this platform? > > Maybe the difference of SB X-Fi model? > > IIRC, the boards people reported to work are either the original X-Fi, > SB055x or SB077x model. Is yours Vista-compatible one? > It appears in /proc/asound/cards as UAA, and can be identified via PCI > SSID. I believe I have the original model, as I bought this card shortly after the X-Fi was first announced. According to lspci: 05:02.0 Multimedia audio controller [0401]: Creative Labs SB X-Fi [1102:0005] Subsystem: Creative Labs X-Fi Platinum [1102:0021] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 32 (1000ns min, 1250ns max), Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 18 Region 0: I/O ports at d000 [size=32] Region 1: Memory at fb000000 (64-bit, non-prefetchable) [size=2M] Region 3: Memory at fb200000 (64-bit, non-prefetchable) [size=2M] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Some dmesg output: [ 245.518854] SBXFI: Allocate SRC 0 [ 245.537647] SBXFI: allocated TLB at 0 for 16 pages [ 245.537650] SBXFI: Setting TLB buffer page 0x7f9ea000 [ 245.538174] SBXFI: PLAYBACK PREPARE: rate=48000, period_size=4096, buffer_size=16384 [ 245.538177] SBXFI: Pitch [0:fa6] = 0x1000000 [ 245.538180] SBXFI: Pitch [1:fb6] = 0x1000000 [ 245.538183] SBXFI: Amp [00:0001] = 0x7b6 [ 245.538188] SBXFI: Amp [01:0011] = 0x7b6 [ 245.545546] SBXFI: PLAY TRIGGER START [ 245.545550] SBXFI: SETUP SRC: ch=0x0, start=0x0, loop=0x10000, cache=0x100, ctrl=0x1044c [ 245.545563] SBXFI: SETUP SRC: ch=0x1, start=0x0, loop=0x10000, cache=0x100, ctrl=0x4c [ 245.545578] SBXFI: PLAY UPDATE TIMER [ 246.996338] SBXFI: PLAY TRIGGER STOP [ 246.996352] SBXFI: PLAY UPDATE TIMER [ 246.996782] SBXFI: release TLB at 0 for 16 pages [ 246.996785] SBXFI: Disabling TLB buffer [ 246.997359] SBXFI: Release SRC 0 How could I help debug this? -- Turning and turning in the widening gyre The falcon cannot hear the falconer Things fall apart; the center cannot hold Mere anarchy is loosed upon the world

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[alsa-devel] [PATCH] Fix PIT lockup on some chipsets when using the PC-Speaker
by Zoltan Devai 03 Nov '08

03 Nov '08

Hi, There've been reports [1] about the sysem bell causing a hard lockup. My machine was affected as well: Any speaker output hung the machine completely, nothing in the logs, no magic sysrq, etc., looked like a hardware problem. Had a closer look on the issue, and it turned out that the pcspkr module is responsible. The cause is the bad setup of Timer 2 in the i8253 controller, which probably hangs the whole PIT controller. Intel datasheets [2] state that the timer registers are in an undefined state after reset and they need to be programmed before enabling the timer. (And enabling without programming the frequency first doesn't make sense anyway). I don't know which chipsets are affected (if not all), it also depends on the BIOS whether it initializes the timer (e.g. to beep when you start the machine). The following patch solved the issue on my ICH6 notebook, couldn't test it with any others, but should be safe to apply. I've also sent a patch to the linux-input list for the pcspkr module. [1] https://bugs.launchpad.net/ubuntu/+source/linux/+bug/146151 https://bugs.launchpad.net/ubuntu/+bug/270790 http://ubuntuforums.org/showthread.php?t=227693 http://bugs.gentoo.org/show_bug.cgi?id=222583 https://bugzilla.redhat.com/show_bug.cgi?id=454225 [2] http://www.intel.com/assets/pdf/datasheet/252516.pdf http://www.intel.com/assets/pdf/datasheet/301473.pdf http://www.intel.com/design/chipsets/datashts/29065503.pdf ------------------ Fix PIT lockup on some chipsets when using the PC-Speaker. Signed-off-by: Zoltan Devai <zdevai(a)gmail.com> --- a/sound/drivers/pcsp/pcsp_input.c +++ b/sound/drivers/pcsp/pcsp_input.c @@ -24,13 +24,13 @@ spin_lock_irqsave(&i8253_lock, flags); if (count) { - /* enable counter 2 */ - outb_p(inb_p(0x61) | 3, 0x61); /* set command for counter 2, 2 byte write */ outb_p(0xB6, 0x43); /* select desired HZ */ outb_p(count & 0xff, 0x42); outb((count >> 8) & 0xff, 0x42); + /* enable counter 2 */ + outb_p(inb_p(0x61) | 3, 0x61); } else { /* disable counter 2 */ outb(inb_p(0x61) & 0xFC, 0x61);

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Re: [alsa-devel] State of the XFi driver on x86_64
by Toby 03 Nov '08

03 Nov '08

2008/11/2 WILMES Andreas <administrator(a)webspace.ms>: > Hello everbody, > > Can someone give the current state of the driver? What are his features, > what is working? I've been watching this thread for a while, and occasionally trying out new builds, as I have an original PCI X-Fi emu20k1. Everyone else seems to be reporting decent successes, but I have never managed it myself. Playing back sound files at any sample-rate results in a repeating burst of static. Adjusting the period/buffer parameters to aplay will cause the pitch/tone of the static burst to change, but never turn into anything remotely like the desired sound. I'm running this on a 64-bit system (Ubuntu 8.04 with a custom compiled kernel). Is anyone else having success on this platform? Thanks, Toby -- Turning and turning in the widening gyre The falcon cannot hear the falconer Things fall apart; the center cannot hold Mere anarchy is loosed upon the world

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[alsa-devel] M2N68-VM - No digital sound
by Ben Matthews 02 Nov '08

02 Nov '08

Hey Guys, I have the asus m2n68-vm motherboard currently have analog sound but no HDMI audio or digial audio. The subdevice 1043:8345 appears to be the problem, in windows HDMI sound works, and digial outputs are present. Chip VT1708B Chipset 630a see http://www.alsa-project.org/db/?f=37aec7ec77f7bd421ac7933702a5975e5764871f for more information. Any help would be wonderful happy to test / supply more information as required thanks, Ben.

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[alsa-devel] sound-unstable-snapshot doesn't compile
by Vedran Miletić 02 Nov '08

02 Nov '08

I get this error: CC [M] /home/vedran/bin/alsa-driver-unstable/soc/soc-core.o In file included from /home/vedran/bin/alsa-driver-unstable/soc/soc-core.c:6: /home/vedran/bin/alsa-driver-unstable/soc/../alsa-kernel/soc/soc-core.c: In function 'soc_init_debugfs': /home/vedran/bin/alsa-driver-unstable/soc/../alsa-kernel/soc/soc-core.c:1075: error: implicit declaration of function 'debugfs_remove_recursive' make[3]: *** [/home/vedran/bin/alsa-driver-unstable/soc/soc-core.o] Error 1 make[2]: *** [/home/vedran/bin/alsa-driver-unstable/soc] Error 2 make[1]: *** [_module_/home/vedran/bin/alsa-driver-unstable] Error 2 make[1]: Leaving directory `/usr/src/linux-headers-2.6.26-1-amd64' make: *** [compile] Error 2 Linux kalopsia 2.6.26-1-amd64 #1 SMP Thu Oct 9 14:16:53 UTC 2008 x86_64 GNU/Linux -- Vedran Miletić

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[alsa-devel] snd-ca0106
by Vedran Miletić 02 Nov '08

02 Nov '08

I played with CA0106 driver a bit. I wonder if we could make it set name for card and chip, in similar way to snd-emu10k1 and others. So... in emu10k1.c we have: 180 strcpy(card->driver, emu->card_capabilities->driver); 181 strcpy(card->shortname, emu->card_capabilities->name); While in ca0106_main.c there is: 1369 strcpy(card->driver, "CA0106"); 1370 strcpy(card->shortname, "CA0106"); Can it be changed? I want it do display Card in alsamixer as a board name from ca0106_card_capabailities list and Chip name to be CA0106 [chip->model]. Also, it would be great if we could name the four PCMs, like hw0,0 - card_name Front hw0,1 - card_name Rear hw0,2 - card_name Center/LFE hw0,3 - card_name Side (it's not unknown) Instead of just CA0106. Any ideas? -- Vedran Miletić

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[alsa-devel] Current state of the XFi driver?
by WILMES Andreas 02 Nov '08

02 Nov '08

Hello everbody, Can someone give the current state of the driver? What are his features, what is working? In what way is it different from the OSS driver? Regards, Andreas

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[alsa-devel] [PATCH 1/1] [ALSA]TLV320AIC23B: Support more sample rates
by Troy Kisky 02 Nov '08

02 Nov '08

Add support for more sample rates, different crystals and split playback/capture rates. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> Acked-by: Arun KS <arunks(a)mistralsolutions.com> diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308da..0096842 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -37,12 +37,6 @@ #define AIC23_VERSION "0.1" -struct tlv320aic23_srate_reg_info { - u32 sample_rate; - u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ - u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ -}; - /* * AIC23 register cache */ @@ -261,20 +255,160 @@ static const struct snd_soc_dapm_route intercon[] = { }; -/* tlv320aic23 related */ -static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { - {4000, 0x06, 1}, /* 4000 */ - {8000, 0x06, 0}, /* 8000 */ - {16000, 0x0C, 1}, /* 16000 */ - {22050, 0x11, 1}, /* 22050 */ - {24000, 0x00, 1}, /* 24000 */ - {32000, 0x0C, 0}, /* 32000 */ - {44100, 0x11, 0}, /* 44100 */ - {48000, 0x00, 0}, /* 48000 */ - {88200, 0x1F, 0}, /* 88200 */ - {96000, 0x0E, 0}, /* 96000 */ +/* AIC23 driver data */ +struct aic23 { + struct snd_soc_codec codec; + int mclk; + int requested_adc; + int requested_dac; +}; + +/* + * Common Crystals used + * 11.2896 Mhz /128 = *88.2k /192 = 58.8k + * 12.0000 Mhz /125 = *96k /136 = 88.235K + * 12.2880 Mhz /128 = *96k /192 = 64k + * 16.9344 Mhz /128 = 132.3k /192 = *88.2k + * 18.4320 Mhz /128 = 144k /192 = *96k + */ + +/* + * Normal BOSR 0-256/2 = 128, 1-384/2 = 192 + * USB BOSR 0-250/2 = 125, 1-272/2 = 136 + */ +static const int bosr_usb_divisor_table[] = { + 128, 125, 192, 136 +}; +#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) +#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) +static const unsigned short sr_valid_mask[] = { + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ + LOWER_GROUP, /* Usb, bosr - 0*/ + UPPER_GROUP, /* Usb, bosr - 1*/ +}; +/* + * Every divisor is a factor of 11*12 + */ +#define SR_MULT (11*12) +#define A(x) (x) ? (SR_MULT/x) : 0 +static const unsigned char sr_adc_mult_table[] = { + A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), + A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) +}; +static const unsigned char sr_dac_mult_table[] = { + A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), + A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) }; +unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, + int dac, int dac_l, int dac_h, int need_dac) +{ + if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { + unsigned diff_adc = need_adc - adc; + unsigned diff_dac = need_dac - dac; + unsigned score; + if (((int)diff_adc) < 0) + diff_adc = -diff_adc; + if (((int)diff_dac) < 0) + diff_dac = -diff_dac; + score = diff_adc + diff_dac; + return score; + } + return 0xffffffff; +} +int find_rate(int mclk, u32 need_adc, u32 need_dac) +{ + int i, j; + int best_i = -1; + int best_j = -1; + int best_div = 0; + unsigned best_score = 0xffffffff; + int adc_l, adc_h, dac_l, dac_h; + + need_adc *= SR_MULT; + need_dac *= SR_MULT; + /* + * rates given are +/- 1/32 + */ + adc_l = need_adc - (need_adc >> 5); + adc_h = need_adc + (need_adc >> 5); + dac_l = need_dac - (need_dac >> 5); + dac_h = need_dac + (need_dac >> 5); + for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { + int base = mclk / bosr_usb_divisor_table[i]; + int mask = sr_valid_mask[i]; + for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); + j++, mask >>= 1) { + int adc; + int dac; + int score; + if ((mask & 1) == 0) + continue; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score = get_score(adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); + if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div = 0; + } + score = get_score((adc >> 1), adc_l, adc_h, need_adc, + (dac >> 1), dac_l, dac_h, need_dac); + /* prefer to have a /2 */ + if ((score != 0xffffffff) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div = 1; + } + } + } + return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); +} + +#ifdef DEBUG +static void get_current_sample_rates(int sample_rate_control, int mclk, + u32 *sample_rate_adc, u32 *sample_rate_dac) +{ + int sr = (sample_rate_control >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[sample_rate_control & 3]); + int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; + int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; + if (sample_rate_control & TLV320AIC23_CLKIN_HALF) { + adc >>= 1; + dac >>= 1; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} +#endif + +static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, + u32 sample_rate_adc, u32 sample_rate_dac) +{ + /* Search for the right sample rate */ + int data = find_rate(mclk, sample_rate_adc, sample_rate_dac); + if (data < 0) { + printk(KERN_ERR "%s:Invalid rate %u,%u requested\n", + __func__, sample_rate_adc, sample_rate_dac); + return -EINVAL; + } + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); +#ifdef DEBUG + { + int adc, dac; + get_current_sample_rates(data, mclk, &adc, &dac); + printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", + adc, dac, data); + } +#endif + return 0; +} + static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, @@ -293,27 +427,30 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 iface_reg, data; - u8 count = 0; + u16 iface_reg; + int ret; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + u32 sample_rate_adc = aic23->requested_adc; + u32 sample_rate_dac = aic23->requested_dac; + u32 sample_rate = params_rate(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aic23->requested_dac = sample_rate_dac = sample_rate; + if (!sample_rate_adc) + sample_rate_adc = sample_rate; + } else { + aic23->requested_adc = sample_rate_adc = sample_rate; + if (!sample_rate_dac) + sample_rate_dac = sample_rate; + } + ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, + sample_rate_dac); + if (ret < 0) + return ret; iface_reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); - - /* Search for the right sample rate */ - /* Verify what happens if the rate is not supported - * now it goes to 96Khz */ - while ((srate_reg_info[count].sample_rate != params_rate(params)) && - (count < ARRAY_SIZE(srate_reg_info))) { - count++; - } - - data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | - (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | - TLV320AIC23_USB_CLK_ON; - - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -349,12 +486,17 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); /* deactivate */ if (!codec->active) { udelay(50); tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aic23->requested_dac = 0; + else + aic23->requested_adc = 0; } static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) @@ -422,12 +564,9 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - - switch (freq) { - case 12000000: - return 0; - } - return -EINVAL; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + aic23->mclk = freq; + return 0; } static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, @@ -659,14 +798,15 @@ static int tlv320aic23_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct aic23 *aic23; int ret = 0; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) + aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL); + if (aic23 == NULL) return -ENOMEM; - + codec = &aic23->codec; socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -687,6 +827,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); if (codec->control_data) tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -697,7 +838,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) i2c_del_driver(&tlv320aic23_i2c_driver); #endif kfree(codec->reg_cache); - kfree(codec); + kfree(aic23); return 0; } -- 1.5.4.3

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[alsa-devel] pulseaudio plugin configuration
by The Source 02 Nov '08

02 Nov '08

I use alsa-plugins-pulseaudio to make Doom3 use sound (alsa itself is used by pulseaudio and snd-sbxfi currently supports only one app at a time). But sound is 'delayed'. It plays 2 seconds after it actually should play. Should I change some configuration to make it work properly?

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[alsa-devel] ALSA bug-tracking
by A. C. Censi 01 Nov '08

01 Nov '08

On Thu, Oct 30, 2008 at 9:25 AM, Takashi Iwai <tiwai(a)suse.de> wrote, on the reply to this message Re: [alsa-devel] Automatic assignment/configuration of model-id?: > The best way to inform the model addition is to post to alsa-devel > ML, and/or post to me or other developers. > > A bad way is, unfortunately, to submit it on ALSA bug-tracking > system. It's handled right now very badly, and you post would be > often ignored. Why not shut down ALSA bug-tracking, or, at least, put this advice there (if not already, i've not checked)? -- A. C. Censi accensi [em] gmail [ponto] com accensi [em] montreal [ponto] com [ponto] br

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