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[alsa-devel] Alsa MMAP emulation
by Mathieu Monney 05 Mar '08

05 Mar '08

Hi, I'm trying to get working Alsa with the PortAudio library on an embedded device (Nokia N810). The problem is that the sound driver is not an hardware driver for Alsa but a plugin which deals with the internal DSP of the device. This plugin doesn't support mmap'ing at all. So I was wondering if it is possible to configure Alsa a way such PortAudio can use mmap'ing on my DSP plugin ? I saw there is a mmap_emulation flag but the problem is that it seems to be only available on HW driver and not on alsa plugins. Any clues ? Thank you in advance Mathieu

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[alsa-devel] HDMI v. ICH8 on Dell XPS M1330
by Armando Di Cianno 05 Mar '08

05 Mar '08

I figured this was more appropriate for devel, and not user. Sorry, if that's wrong. First, I'm using the 1.0.16 release tarballs. They work better than the in-kernel drivers, as they survive both suspend and hibernate, whereas the kernel drivers do not. I'm eager to see audio over HDMI working on my Dell XPS M1330. I'm pretty sure that this is /not supported/ at the moment -- if this isn't the case, please stop reading, and let me know the particulars. HDMI seems to be supported noted, but mainly for the ATI chips. I'm pretty sure the m1330 has a STAC9228 setup. None of the suggested model= module parameters succeeded in getting audio over HDMI, beyond what is autodetected without any parameter specified. model=5stack seems to show a digital SPDIF device added. If there is anyway I can help the core developers get HDMI audio working, I would be glad to help. I will post all the relevant information I can think of following; please let me know if other information would be helpful. I will gladly turn on any excessive debugging options needed. I am an ALSA n00b, but I will happily do any required work that I can do from this end. Thanks! -- Armando Di Cianno http://dicianno.org/blog armando(a)dicianno.org armando(a)goodship.net w/o model= parameter # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 w/ model=5stack # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 # lspci -nvvvv -d 8086:284b 00:1b.0 0403: 8086:284b (rev 02) Subsystem: 1028:0209 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 21 Region 0: Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot +,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 unlimited ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel

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[alsa-devel] [PATCH] model for alc883 to support M720R
by Jiang zhe 04 Mar '08

04 Mar '08

Hi, Please refer to [0003781] on the alsa mantis. There is no suitable model for M720R. This patch is to support HP jack-sensing and mixer. The reporter complained that there was a constant noise when he recorded from Mic, It existed even after he muted the input. I checked the codec info carefully, the input widget was muted indeed. It's strange, Is there anything special for this codec? Thanks! Signed-off-by: Jiang zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20080304/alsa-kernel/pci/hda/patch_realtek.c 2008-03-01 09:00:07.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2008-03-04 16:49:09.000000000 +0800 @@ -201,6 +201,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, + ALC883_CLEVO_M720R, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6645,6 +6646,33 @@ static struct snd_kcontrol_new alc883_mi { } /* end */ }; +static struct snd_kcontrol_new alc883_clevo_m720r_mixer[] = { + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -7178,6 +7206,20 @@ static struct hda_verb alc883_mitac_verb { } /* end */ }; +static struct hda_verb alc883_clevo_m720r_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Int speaker */ + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7361,6 +7403,25 @@ static void alc883_tagra_unsol_event(str alc883_tagra_automute(codec); } +/* toggle speaker-output according to the hp-jack state */ +static void alc883_clevo_m720r_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc883_clevo_m720r_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc883_clevo_m720r_automute(codec); +} + static void alc883_haier_w66_automute(struct hda_codec *codec) { unsigned int present; @@ -7598,6 +7659,7 @@ static const char *alc883_models[ALC883_ [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", + [ALC883_CLEVO_M720R] = "clevo-m720r", [ALC883_AUTO] = "auto", }; @@ -7638,6 +7700,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720R), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), @@ -7780,6 +7843,18 @@ static struct alc_config_preset alc883_p .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_CLEVO_M720R] = { + .mixers = { alc883_clevo_m720r_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m720r_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_clevo_m720r_unsol_event, + .init_hook = alc883_clevo_m720r_automute, + }, [ALC883_LENOVO_101E_2ch] = { .mixers = { alc883_lenovo_101e_2ch_mixer}, .init_verbs = { alc883_init_verbs, alc883_lenovo_101e_verbs},

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[alsa-devel] Driver recovers from underrun in plain ALSA mode, but not in OSS emulation
by Timur Tabi 04 Mar '08

04 Mar '08

I'm stress-testing my sound driver using mplayer to see how it handles underrun during jumps in CPU loads. That is, I start playing a movie, and then I cause the CPU to jump to 100%, and then I drop the CPU back to normal. During the times when the CPU is at 100%, I get underruns and other problems. That's normal. When I tell mplayer to use the ALSA API, my driver recovers nicely from an underrun situation when the CPU load drops back to normal. However, if I tell mplayer to use the OSS API (i.e. ALSA's OSS emulation), it does not recover from an underrun situation. The audio and video playback remains distorted. I need to tell mplayer to pause and resume to get it to recover. Is this normal for ALSA drivers? Before I try to debug this problem the hard way, I want to know if this could be the result of a common programming mistake in my driver or something like similar. (FYI, the source code for my driver is in sound/soc/fsl/) -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] mm functions vs. normal IO
by Reiner 04 Mar '08

04 Mar '08

Hello, I'am new for using ALSA API. I was astonished that a simple wav playback example does not exist. But after reading the pcm examples I have managed to make one (if I will finish it I can post it so it can be added to the examples) Now my question: What should be used: snd_pcm_mmap_commit <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___direct.html#g…> (I think snd_pcm_mmap_writei <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___direct.html#g…> is the same with an additional copy behind ?) or snd_pcm_writei <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gf13067c0e…> I think mmap will have better performance, but will it work with all soundcards? Or should it be in a config what to use? I'am also looking for a function to get the number of played frames. I have played around with "snd_pcm_status" but it seems not to have a counter for this. By the way is there a detailed tutorial (introduction) for ALSA available? The online help is a little bit too short for newbies. ;-) Reiner

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[alsa-devel] [PATCH] ALC288 - Add NEC S970 to the quirk table
by Pascal Terjan 04 Mar '08

04 Mar '08

Hi, NEC S970 has no sound in the internal speakers when autodetection is used. With targa-dig model, there is sound in the speakers and it gets correctly muted when pluging headphones.

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[alsa-devel] Question about snd_pcm_update_hw_ptr_post()
by Timur Tabi 04 Mar '08

04 Mar '08

Can someone explain to me this snippet from snd_pcm_update_hw_ptr_post(): if (avail >= runtime->stop_threshold) { if (substream->runtime->status->state == SNDRV_PCM_STATE_DRAINING) snd_pcm_drain_done(substream); else xrun(substream); return -EPIPE; } During playback, avail = snd_pcm_playback_avail(). Does this mean that avail is number of frames available in the playback DMA buffer for new data? If so, why is avail >= runtime->stop_threshold considered an underrun condition? Does ALSA try to keep the amount of available data to a minimum? If so, how can I adjust runtime->stop_threshold? I'd like to see what happens if I make stop_threshold equal to the size of my DMA buffer. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH 4/4] Convert non-SoC PXA2xx AC97 driver to clock API
by Mark Brown 04 Mar '08

04 Mar '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-ac97.c | 48 +++++++++++++++++++++++++++++++++++++++------- 1 files changed, 40 insertions(+), 8 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5d86e68..fd7bc86 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/wait.h> +#include <linux/clk.h> #include <linux/delay.h> #include <sound/core.h> @@ -35,6 +36,10 @@ static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; +static struct clk *ac97_clk; +#ifdef CONFIG_PXA27x +static struct clk *ac97conf_clk; +#endif /* * Beware PXA27x bugs: @@ -112,9 +117,9 @@ static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) gsr_bits = 0; #ifdef CONFIG_PXA27x /* PXA27x Developers Manual section 13.5.2.2.1 */ - pxa_set_cken(CKEN_AC97CONF, 1); + clk_enable(ac97conf_clk); udelay(5); - pxa_set_cken(CKEN_AC97CONF, 0); + clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); #else @@ -259,7 +264,7 @@ static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state) if (platform_ops && platform_ops->suspend) platform_ops->suspend(platform_ops->priv); GCR |= GCR_ACLINK_OFF; - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); return 0; } @@ -268,7 +273,7 @@ static int pxa2xx_ac97_do_resume(struct snd_card *card) { pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data; - pxa_set_cken(CKEN_AC97, 1); + clk_enable(ac97_clk); if (platform_ops && platform_ops->resume) platform_ops->resume(platform_ops->priv); snd_ac97_resume(pxa2xx_ac97_ac97); @@ -335,8 +340,21 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) #ifdef CONFIG_PXA27x /* Use GPIO 113 as AC97 Reset on Bulverde */ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); + if (IS_ERR(ac97conf_clk)) { + ret = PTR_ERR(ac97conf_clk); + ac97conf_clk = NULL; + goto err; + } #endif - pxa_set_cken(CKEN_AC97, 1); + + ac97_clk = clk_get(&dev->dev, "AC97CLK"); + if (IS_ERR(ac97_clk)) { + ret = PTR_ERR(ac97_clk); + ac97_clk = NULL; + goto err; + } + clk_enable(ac97_clk); ret = snd_ac97_bus(card, 0, &pxa2xx_ac97_ops, NULL, &ac97_bus); if (ret) @@ -361,11 +379,19 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev) err: if (card) snd_card_free(card); - if (CKEN & (1 << CKEN_AC97)) { + if (ac97_clk) { GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; + } +#ifdef CONFIG_PXA27x + if (ac97conf_clk) { + clk_put(ac97conf_clk); + ac97conf_clk = NULL; } +#endif return ret; } @@ -378,7 +404,13 @@ static int __devexit pxa2xx_ac97_remove(struct platform_device *dev) platform_set_drvdata(dev, NULL); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); - pxa_set_cken(CKEN_AC97, 0); + clk_disable(ac97_clk); + clk_put(ac97_clk); + ac97_clk = NULL; +#ifdef CONFIG_PXA27x + clk_put(ac97conf_clk); + ac97conf_clk = NULL; +#endif } return 0; -- 1.5.4.3

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[alsa-devel] [PATCH] Add Invert Switch for ROUT2
by Mark Brown 03 Mar '08

03 Mar '08

From: Graeme Gregory <graeme(a)openmoko.org> GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2 needs to be inverted. This patch adds a mixer control for this. Signed-off-by: Graeme Gregory <graeme(a)openmoko.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0288275..76a5c7b 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC", "Channel Swap"}; +static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"}; static const struct soc_enum wm8753_enum[] = { SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), @@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel), +SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase), }; @@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), SOC_ENUM("ADC Data Select", wm8753_enum[27]), +SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; /* add non dapm controls */ -- 1.5.4.3

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[alsa-devel] ALSA C++ API updated for 1.0.16
by Lasse Kärkkäinen 01 Mar '08

01 Mar '08

Here's an updated version of the C++ wrapper that I posted earlier in September 2007. I would like this to be included in the ALSA distribution after some peer review. This wrapper has been used in UltraStar NG (a game shipped by Gentoo and others) since September, with no known issues other than that it is incomplete (very PCM-centric). As a reminder, the API closely follows the C API, and also allows mixing direct C API calls with itself transparently.

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