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Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

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Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

>Are you sure that the >drivers you are using implement suspend and resume at all? I am using s3c6410 code.I2S driver has suspend/resume call(jassi brar maintained). Audio codec is also having generic code for suspend/resume but don't know if that is the problem or not.However how can i verify if my codec suspend/resume has got some problem?Most likely my codec has got some configuration problems but i am not getting any pointers to debug it.Any pointers in this direction would be very helpful.

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Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

Thanks Mark, >If your driver is not responding promptly to requests to terminate playback then an underrun doesn't seem surprising. Sometimes in my driver write calls doesn't return causing application to block forever and CPU pegged problem. Situation:Recording going on->go to sleep->come back from sleep->start recording again->try recording and playing back simultaneously(something like recording going on in background and you try to play short audio). This causes write call to get blocked in playback case causing application to hangup. This doesn't happen if we don't go to sleep. So i think somewhere there is some problem with suspend->resume case which is difficult to figure out. I checked driver logs after coming back from sleep.I found for some time interrupts are coming then stopped coming suddenly So now after waking up i am reopening the playback device before opening the record device.Which has fixed the problem but not the root cause.

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[alsa-devel] underrun problem when short audio play
by ANISH KUMAR 16 Mar '10

16 Mar '10

Hi all, I am facing a problem when i try to play short duration audio(problem is not observerd when i am playing tones which is also short duration audio). Somehow i am getting buffer overrun everytime i am playing short duration audio. It is happening because of avail >= stop_threshold. When i tried to increase the stop_threshold but then i got "same audio played again kind of echo" problem. I can't try the solution suggested by jaroslav i.e. checking for DRAINING state in lower level driver and stop queuing because of some design issue as in my code i don't call drain function as soon as all buffers are written. I wait for some time(in miliseconds) and then call drain function. Please suggest any pointers.Should i increase the buffer size or stop_threshold both of which i tried but to no use. I am always getting underrun problem that too always in the function update_hw_ptr_post function. How can i disable the stop_threshold check?? Thanks, anish

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[alsa-devel] [PATCH 2/4] mmp: support ssp in pxa168
by Haojian Zhuang 16 Mar '10

16 Mar '10

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Re: [alsa-devel] [PATCH 0/4] Some typo fixing
by Jiri Kosina 16 Mar '10

16 Mar '10

On Mon, 15 Mar 2010, Randy Dunlap wrote: > > I have fixed some typos. > > Acked-by: Randy Dunlap <rdunlap(a)xenotime.net> > > Jiri, can you merge these, please, unless someone objects (?). Yes, I will take it, thanks. A couple comments though: - [important!] Thomas, it's not necessary to CC zillions of people on such patches. Just submit them to trivial(a)kernel.org (and eventually CC lkml), and that's it. I believe many people might get annoyed by this. - I will not be applying the drivers/staging hunks. Staging patches are moving target, the code is changing quickly (including complete drops/rewrites) so we'll likely be getting conflicts there. I will reroute them to Greg. - I will fold all the patches into one. We don't need one commit per one specific typo. -- Jiri Kosina SUSE Labs, Novell Inc.

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[alsa-devel] [patch] sequencer: clean up remove bogus check
by Dan Carpenter 16 Mar '10

16 Mar '10

A few lines earlier bend is limited to 2399. So semitones is always less than 24 here. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index c798746..e85789e 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1631,8 +1631,6 @@ unsigned long compute_finetune(unsigned long base_freq, int bend, int range, } semitones = bend / 100; - if (semitones > 99) - semitones = 99; cents = bend % 100; amount = (int) (semitone_tuning[semitones] * multiplier * cent_tuning[cents]) / 10000;

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212
by Daniel Chen 16 Mar '10

16 Mar '10

Hi, Attached is a patch against current sound-2.6 master HEAD (applicable also to 2.6.32.10 and 2.6.33.1) that corrects the model quirk for the OR's machine and adds the LPIB quirk. Best, -Dan

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[alsa-devel] [PATCH 0/4] Some typo fixing
by Thomas Weber 15 Mar '10

15 Mar '10

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[alsa-devel] ALSA not working on HP Touchsmart 600-1100 with CentOS 5.4
by Robert Krakora 15 Mar '10

15 Mar '10

Hello, I have tried installing ALSA driver, lib and util for every version from 1.21 to the tip and I cannot get alsaconf to identify any audio device on my HP Touchsmart 600-1100 with CentOS 5.4 installed. Please see the attached lspci and lshw dumps. Thanks in advance. Best Regards, -- Rob Krakora Senior Software Engineer MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext. 206 (317)663-0808 Fax

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[Patch v2] aconnect -x: Do not update index after removal of connection. Signed-off-by: lieven moors <lievenmoors@gmail.com>
by lieven moors 15 Mar '10

15 Mar '10

--- seq/aconnect/aconnect.c | 43 ++++++++++++------------------------------- 1 files changed, 12 insertions(+), 31 deletions(-) diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 1a50666..6092748 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -192,52 +192,33 @@ static void remove_connection(snd_seq_t *seq, snd_seq_client_info_t *cinfo, snd_seq_port_info_t *pinfo, int count) { snd_seq_query_subscribe_t *query; - snd_seq_query_subscribe_alloca(&query); snd_seq_query_subscribe_set_root(query, snd_seq_port_info_get_addr(pinfo)); - snd_seq_query_subscribe_set_type(query, SND_SEQ_QUERY_SUBS_READ); snd_seq_query_subscribe_set_index(query, 0); - for (; snd_seq_query_port_subscribers(seq, query) >= 0; - snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1)) { + + while (snd_seq_query_port_subscribers(seq, query) >= 0) + { snd_seq_port_info_t *port; snd_seq_port_subscribe_t *subs; const snd_seq_addr_t *sender = snd_seq_query_subscribe_get_root(query); const snd_seq_addr_t *dest = snd_seq_query_subscribe_get_addr(query); snd_seq_port_info_alloca(&port); - if (snd_seq_get_any_port_info(seq, dest->client, dest->port, port) < 0) - continue; - if (!(snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_SUBS_WRITE)) - continue; - if (snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_NO_EXPORT) - continue; - snd_seq_port_subscribe_alloca(&subs); - snd_seq_port_subscribe_set_queue(subs, snd_seq_query_subscribe_get_queue(query)); - snd_seq_port_subscribe_set_sender(subs, sender); - snd_seq_port_subscribe_set_dest(subs, dest); - snd_seq_unsubscribe_port(seq, subs); - } - snd_seq_query_subscribe_set_type(query, SND_SEQ_QUERY_SUBS_WRITE); - snd_seq_query_subscribe_set_index(query, 0); - for (; snd_seq_query_port_subscribers(seq, query) >= 0; - snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1)) { - snd_seq_port_info_t *port; - snd_seq_port_subscribe_t *subs; - const snd_seq_addr_t *dest = snd_seq_query_subscribe_get_root(query); - const snd_seq_addr_t *sender = snd_seq_query_subscribe_get_addr(query); - snd_seq_port_info_alloca(&port); - if (snd_seq_get_any_port_info(seq, sender->client, sender->port, port) < 0) - continue; - if (!(snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_SUBS_READ)) - continue; - if (snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_NO_EXPORT) + if ((snd_seq_get_any_port_info(seq, dest->client, dest->port, port) < 0) || + !(snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_SUBS_WRITE) || + (snd_seq_port_info_get_capability(port) & SND_SEQ_PORT_CAP_NO_EXPORT)) + { + snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1); continue; + } snd_seq_port_subscribe_alloca(&subs); snd_seq_port_subscribe_set_queue(subs, snd_seq_query_subscribe_get_queue(query)); snd_seq_port_subscribe_set_sender(subs, sender); snd_seq_port_subscribe_set_dest(subs, dest); - snd_seq_unsubscribe_port(seq, subs); + if(snd_seq_unsubscribe_port(seq, subs) < 0){ + snd_seq_query_subscribe_set_index(query, snd_seq_query_subscribe_get_index(query) + 1); + } } } -- 1.6.3.3 --------------020302040200040109060709--

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[alsa-devel] Driver for a codec with built-in DSP for MP3 playback
by Dajun Chen 15 Mar '10

15 Mar '10

Hi all, I am wondering where to find the guidances of developing a device driver for a codec device which has a built-in DSP for MP3 (and tracks in other format) playback and recording? Regards David

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[alsa-devel] [PATCH] ALSA: hda - Add PCI quirks for MSI all-in-one computers NetOn AP1900 and Wind Top AE2220
by Anisse Astier 15 Mar '10

15 Mar '10

Signed-off-by: Anisse Astier <anisse(a)astier.eu> --- Hi, This should make the speakers and jack detection work on those computers. Regards, Anisse sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e8cbe21..970d0b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9191,6 +9191,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG), + SND_PCI_QUIRK(0x1462, 0x4570, "MSI Wind Top AE2220", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x6510, "MSI GX620", ALC883_TARGA_8ch_DIG), SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), @@ -9200,6 +9201,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7437, "MSI NetOn AP1900", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), -- 1.6.5.7

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Re: [alsa-devel] [2.6.34-rc1 REGRESSION] alsa: no mixer controls (and no sound) with 2.6.24-rc1+ on Realtek ALC260
by Takashi Iwai 15 Mar '10

15 Mar '10

At Mon, 15 Mar 2010 13:41:30 +0100, Alessandro Guido wrote: > > 2010/3/15 Takashi Iwai <tiwai(a)suse.de> > > > At Mon, 15 Mar 2010 01:06:17 +0100, > > Alessandro Guido wrote: > > > > > > With 2.6.34-rc1+ my laptop (Sony Vaio VGN-FS215S) can't reproduce any > > > sound. All controls are missing from alsamixer and this message is found > > > in dmesg: > > > > > > hda_codec: cannot build controlsfor #0 (error -22) > > > > > > Kernel config is attached, and the output from alsa-info.sh follows: > > > > Does the patch below fix? > > > > Yes, it does. Thanks for checking. The fix is merged in sound tree. Takashi

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[alsa-devel] [PATCH 2/2] Quick fix for the audio clocks on SAM9-L9261
by Benjamin Valentin 15 Mar '10

15 Mar '10

Signed-off-by: Dimitar Dimitrov <dimitar(a)olimex.com> From 10a85229b2357c89d5170efde5ac0e1cba4c9f9e Mon Sep 17 00:00:00 2001 From: Dimitar Dimitrov <dimitar(a)olimex.com> Date: Wed, 27 May 2009 08:11:37 +0300 Subject: [PATCH] Quick fix for the audio clocks on SAM9-L9261 This patch together with support from SAM9-L9261 hardware fix an issue with audio clocks. AT73C213 needs MCLK and BCLK with equal phase. The MCLK from AT91SAM9XXX clock output and BCLK from the SSC TK clock output are not phase-synchronized. Thus the SAM9-L9261 board provides external hardware divison by 8 from MCLK to BCLK, ensuring zero phase difference. See: http://www.at91.com/phpbb/viewtopic.php?t=2633 --- sound/spi/at73c213.c | 54 ++++++++++++++++++++++++++++++++++++++----------- 1 files changed, 42 insertions(+), 12 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4c7b051..325e534 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -29,6 +29,8 @@ #include <linux/atmel-ssc.h> +#include <asm/mach-types.h> + #include <linux/spi/spi.h> #include <linux/spi/at73c213.h> @@ -114,7 +116,11 @@ snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) static struct snd_pcm_hardware snd_at73c213_playback_hw = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, +#ifdef __BIG_ENDIAN .formats = SNDRV_PCM_FMTBIT_S16_BE, +#else + .formats = SNDRV_PCM_FMTBIT_S16_LE, +#endif .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, /* Replaced by chip->bitrate later. */ .rate_max = 50000, /* Replaced by chip->bitrate later. */ @@ -144,7 +150,8 @@ static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) * as practical to the desired target rate. * * The DAC master clock (MCLK) is programmable, and is either 256 - * or (not here) 384 times the I2S output clock (BCLK). + * or (not here) 384 times the I2S frame clock (LRC), that is + * 8 or 12 times the output bit clock (BCLK). */ /* SSC clock / (bitrate * stereo * 16-bit). */ @@ -747,17 +754,40 @@ cleanup: */ static int __devinit snd_at73c213_ssc_init(struct snd_at73c213 *chip) { - /* - * Continuous clock output. - * Starts on falling TF. - * Delay 1 cycle (1 bit). - * Periode is 16 bit (16 - 1). - */ - ssc_writel(chip->ssc->regs, TCMR, - SSC_BF(TCMR_CKO, 1) - | SSC_BF(TCMR_START, 4) - | SSC_BF(TCMR_STTDLY, 1) - | SSC_BF(TCMR_PERIOD, 16 - 1)); + if (machine_is_sam9_l9261()) { + /* + * dimitar(a)olimex.com : AT73C213 needs MCLK and BCLK with equal + * phase. The MCLK from AT91SAM9XXX clock output and BCLK from + * the SSC TK clock output are not phase-synchronized. Thus + * the SAM9-L9261 board provides external hardware divison by + * 8 from MCLK to BCLK, ensuring zero phase difference. + * See: http://www.at91.com/phpbb/viewtopic.php?t=2633 + * + * TK is clock input. + * Clock source is TK pin. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Period is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKS, 2) + | SSC_BF(TCMR_CKO, 0) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + } else { + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Period is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + } /* * Data length is 16 bit (16 - 1). * Transmit MSB first. -- 1.5.6.1

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[alsa-devel] 2.6.34-rc1: regression: ^Z no longer stops sound
by Pavel Machek 15 Mar '10

15 Mar '10

Hi! In 2.6.33, hitting ^Z stopped mpg123 in a way you'd expect (both on thinkpad and geode notebook). On 2.6.34-rc1, hit ^Z and sound starts looping. Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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Re: [alsa-devel] [2.6.34-rc1 REGRESSION] alsa: no mixer controls (and no sound) with 2.6.24-rc1+ on Realtek ALC260
by Takashi Iwai 15 Mar '10

15 Mar '10

At Mon, 15 Mar 2010 01:06:17 +0100, Alessandro Guido wrote: > > With 2.6.34-rc1+ my laptop (Sony Vaio VGN-FS215S) can't reproduce any > sound. All controls are missing from alsamixer and this message is found > in dmesg: > > hda_codec: cannot build controlsfor #0 (error -22) > > Kernel config is attached, and the output from alsa-info.sh follows: Does the patch below fix? Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a83719..ba45868 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6477,7 +6477,7 @@ static struct alc_config_preset alc260_presets[] = { .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_adc_nids, + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_capture_source,

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Re: [alsa-devel] [PATCH]Support MCP89 and GT21x hdmi audio
by Jaroslav Kysela 14 Mar '10

14 Mar '10

On Sat, 13 Mar 2010, VDR User wrote: > [pid 9629] 1268511516.215224 ioctl(18, 0x400c4150, 0xb43631a4) = 0 ^^^^ here xine feeds new samples to the ALSA driver (0x400c4150 == SNDRV_PCM_IOCTL_WRITEI_FRAMES) > [pid 9629] 1268511516.215275 ioctl(18, 0x80044121, 0xb43632cc) = 0 > [pid 9627] 1268511520.700109 ioctl(10, 0xc0104652, 0xb5443f00) = 0 > [pid 9629] 1268511520.700340 ioctl(18, 0x80044121, 0xb43632cc) = -1 > EPIPE (Broken pipe) ^^^^ next iteration (0x80044121 == SNDRV_PCM_IOCTL_DELAY) - see time difference - 4.485 seconds from last WRITEI It seems like a xine bug or the system scheduler bug or any kernel bug affecting the system scheduler. The ALSA subsystem is not reponsible for this behaviour - the ring buffer in your case is for about 0.372 seconds of samples. This limit must not be crossed for the click-free playback. Ask the xine developers for more help or try another player. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 13 Mar '10

13 Mar '10

The following changes since commit fad837c16cdd856c68ce2e1335ad0fe836ed8ecd: Mark Brown (1): Merge commit 'v2.6.34-rc1' into for-2.6.35 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Jassi Brar (3): ASoC: S3C64XX: I2S: Make BCLK independent of sample size ASoC: S3C: I2Sv2: Segregate hw_params callback ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags Kuninori Morimoto (1): ASoC: da7210: Add 11025/22050/44100/88200 rate support Miguel Aguilar (4): MFD: DaVinci Voice Codec ASoC: DaVinci: Voice Codec Interface ASoC: DaVinci: CQ93VC Voice Codec DaVinci: DM365: Voice Codec support for the DM365 EVM Peter Ujfalusi (4): ASoC: tlv320dac33: Start/stop sequence change ASoC: tlv320dac33: Add option for keeping the BCLK running OMAP3: McBSP: Add interface for FIFO caused delay query ASoC: OMAP3: Report delay caused by the internal FIFO arch/arm/mach-davinci/board-dm365-evm.c | 4 + arch/arm/plat-omap/include/plat/mcbsp.h | 6 + arch/arm/plat-omap/mcbsp.c | 55 ++++++ drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 +++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++ include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 298 +++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++ sound/soc/codecs/da7210.c | 90 +++++++++- sound/soc/codecs/tlv320dac33.c | 38 ++--- sound/soc/davinci/Kconfig | 27 +++- sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-evm.c | 45 +++++- sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 +++ sound/soc/omap/omap-mcbsp.c | 26 +++ sound/soc/s3c24xx/s3c-i2s-v2.c | 27 +--- sound/soc/s3c24xx/s3c2412-i2s.c | 37 ++++ sound/soc/soc-core.c | 12 ++ 23 files changed, 1270 insertions(+), 53 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h

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[alsa-devel] [PATCHv5 0/2] TWL6040 audio codec initial support
by Olaya, Margarita 13 Mar '10

13 Mar '10

Following patch series add initial support for TWL6040 codec driver. The patch set is based on the series sent from Misael Lopez in October 2009. Changes from v4: - Rename twl6030_codec as twl6040_codec - Rebase the 6 codec patches into a single patch - Rename twl6040_read as twl6040_read_reg_volatile - Use msleep instead of large mdelays - Add more trigger sources - Restructure set_bias_level function to avoid transitions like PREPARE->ON Regards, Margarita --- Misael Lopez Cruz (1): ASoC: TWL6040: Add twl6040 codec driver Margarita Olaya Cabrera (1): OMAP4: PMIC: Rename twl6030_codec as twl6040_codec drivers/mfd/twl-core.c | 4 +- include/linux/i2c/twl.h | 6 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6040.c | 1223 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 6 files changed, 1375 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

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[alsa-devel] [PATCH] ASoC: PCM_RATE: Check for KNOT and CONTINUOUS flags
by Jassi Brar 12 Mar '10

12 Mar '10

For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver should be given a chance to figure out if the dai, that set the flag, can accomodate a rate that it does not explicitly specify but is specified by the dai at the other end of the link. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/soc-core.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 06c38d1..eb73aab 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -404,6 +404,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->playback.formats & cpu_dai->playback.formats; runtime->hw.rates = codec_dai->playback.rates & cpu_dai->playback.rates; + if (codec_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->playback.rates; + if (cpu_dai->playback.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->playback.rates; } else { runtime->hw.rate_min = max(codec_dai->capture.rate_min, @@ -421,6 +427,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->capture.formats & cpu_dai->capture.formats; runtime->hw.rates = codec_dai->capture.rates & cpu_dai->capture.rates; + if (codec_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= cpu_dai->capture.rates; + if (cpu_dai->capture.rates + & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) + runtime->hw.rates |= codec_dai->capture.rates; } snd_pcm_limit_hw_rates(runtime); -- 1.6.2.5

4 4

[alsa-devel] [PATCH v2 1/5] MFD: DaVinci Voice Codec
by miguel.aguilar＠ridgerun.com 12 Mar '10

12 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This is the MFD driver for the DaVinci Voice codec, it has two clients: * Voice codec interface * Voice codec CQ93VC Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- drivers/mfd/Kconfig | 4 + drivers/mfd/Makefile | 1 + drivers/mfd/davinci_voicecodec.c | 189 ++++++++++++++++++++++++++++++++ include/linux/mfd/davinci_voicecodec.h | 126 +++++++++++++++++++++ 4 files changed, 320 insertions(+), 0 deletions(-) create mode 100644 drivers/mfd/davinci_voicecodec.c create mode 100644 include/linux/mfd/davinci_voicecodec.h diff --git a/drivers/mfd/Kconfig b/drivers/mfd/Kconfig index 8782978..6e93c0b 100644 --- a/drivers/mfd/Kconfig +++ b/drivers/mfd/Kconfig @@ -43,6 +43,10 @@ config MFD_SH_MOBILE_SDHI This driver supports the SDHI hardware block found in many SuperH Mobile SoCs. +config MFD_DAVINCI_VOICECODEC + tristate + select MFD_CORE + config MFD_DM355EVM_MSP bool "DaVinci DM355 EVM microcontroller" depends on I2C && MACH_DAVINCI_DM355_EVM diff --git a/drivers/mfd/Makefile b/drivers/mfd/Makefile index ca2f2c4..f5c617f 100644 --- a/drivers/mfd/Makefile +++ b/drivers/mfd/Makefile @@ -9,6 +9,7 @@ obj-$(CONFIG_MFD_SH_MOBILE_SDHI) += sh_mobile_sdhi.o obj-$(CONFIG_HTC_EGPIO) += htc-egpio.o obj-$(CONFIG_HTC_PASIC3) += htc-pasic3.o +obj-$(CONFIG_MFD_DAVINCI_VOICECODEC) += davinci_voicecodec.o obj-$(CONFIG_MFD_DM355EVM_MSP) += dm355evm_msp.o obj-$(CONFIG_MFD_T7L66XB) += t7l66xb.o diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c new file mode 100644 index 0000000..9886aa8 --- /dev/null +++ b/drivers/mfd/davinci_voicecodec.c @@ -0,0 +1,189 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/pcm.h> + +#include <linux/mfd/davinci_voicecodec.h> + +u32 davinci_vc_read(struct davinci_vc *davinci_vc, int reg) +{ + return __raw_readl(davinci_vc->base + reg); +} + +void davinci_vc_write(struct davinci_vc *davinci_vc, + int reg, u32 val) +{ + __raw_writel(val, davinci_vc->base + reg); +} + +static int __init davinci_vc_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc; + struct resource *res, *mem; + struct mfd_cell *cell = NULL; + int ret; + + davinci_vc = kzalloc(sizeof(struct davinci_vc), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + davinci_vc->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(davinci_vc->clk)) { + dev_dbg(&pdev->dev, + "could not get the clock for voice codec\n"); + ret = -ENODEV; + goto fail1; + } + clk_enable(davinci_vc->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no mem resource\n"); + ret = -ENODEV; + goto fail2; + } + + davinci_vc->pbase = res->start; + davinci_vc->base_size = resource_size(res); + + mem = request_mem_region(davinci_vc->pbase, davinci_vc->base_size, + pdev->name); + if (!mem) { + dev_err(&pdev->dev, "VCIF region already claimed\n"); + ret = -EBUSY; + goto fail2; + } + + davinci_vc->base = ioremap(davinci_vc->pbase, davinci_vc->base_size); + if (!davinci_vc->base) { + dev_err(&pdev->dev, "can't ioremap mem resource.\n"); + ret = -ENOMEM; + goto fail3; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_tx_channel = res->start; + davinci_vc->davinci_vcif.dma_tx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_WFIFO); + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "no DMA resource\n"); + return -ENXIO; + } + + davinci_vc->davinci_vcif.dma_rx_channel = res->start; + davinci_vc->davinci_vcif.dma_rx_addr = + (dma_addr_t)(io_v2p(davinci_vc->base) + DAVINCI_VC_RFIFO); + + davinci_vc->dev = &pdev->dev; + davinci_vc->pdev = pdev; + + /* Voice codec interface client */ + cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; + cell->name = "davinci_vcif"; + cell->driver_data = davinci_vc; + + /* Voice codec CQ93VC client */ + cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; + cell->name = "cq93vc"; + cell->driver_data = davinci_vc; + + ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, + DAVINCI_VC_CELLS, NULL, 0); + if (ret != 0) { + dev_err(&pdev->dev, "fail to register client devices\n"); + goto fail4; + } + + return 0; + +fail4: + iounmap(davinci_vc->base); +fail3: + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); +fail2: + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; +fail1: + kfree(davinci_vc); + + return ret; +} + +static int __devexit davinci_vc_remove(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + + mfd_remove_devices(&pdev->dev); + + iounmap(davinci_vc->base); + release_mem_region(davinci_vc->pbase, davinci_vc->base_size); + + clk_disable(davinci_vc->clk); + clk_put(davinci_vc->clk); + davinci_vc->clk = NULL; + + kfree(davinci_vc); + + return 0; +} + +static struct platform_driver davinci_vc_driver = { + .driver = { + .name = "davinci_voicecodec", + .owner = THIS_MODULE, + }, + .remove = __devexit_p(davinci_vc_remove), +}; + +static int __init davinci_vc_init(void) +{ + return platform_driver_probe(&davinci_vc_driver, davinci_vc_probe); +} +module_init(davinci_vc_init); + +static void __exit davinci_vc_exit(void) +{ + platform_driver_unregister(&davinci_vc_driver); +} +module_exit(davinci_vc_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci Voice Codec Core Interface"); +MODULE_LICENSE("GPL"); diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h new file mode 100644 index 0000000..0ab6132 --- /dev/null +++ b/include/linux/mfd/davinci_voicecodec.h @@ -0,0 +1,126 @@ +/* + * DaVinci Voice Codec Core Interface for TI platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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 + */ + +#ifndef __LINUX_MFD_DAVINCI_VOICECODEC_H_ +#define __LINUX_MFD_DAVINIC_VOICECODEC_H_ + +#include <linux/kernel.h> +#include <linux/platform_device.h> +#include <linux/mfd/core.h> + +#include <mach/edma.h> + +/* + * Register values. + */ +#define DAVINCI_VC_PID 0x00 +#define DAVINCI_VC_CTRL 0x04 +#define DAVINCI_VC_INTEN 0x08 +#define DAVINCI_VC_INTSTATUS 0x0c +#define DAVINCI_VC_INTCLR 0x10 +#define DAVINCI_VC_EMUL_CTRL 0x14 +#define DAVINCI_VC_RFIFO 0x20 +#define DAVINCI_VC_WFIFO 0x24 +#define DAVINCI_VC_FIFOSTAT 0x28 +#define DAVINCI_VC_TST_CTRL 0x2C +#define DAVINCI_VC_REG05 0x94 +#define DAVINCI_VC_REG09 0xA4 +#define DAVINCI_VC_REG12 0xB0 + +/* DAVINCI_VC_CTRL bit fields */ +#define DAVINCI_VC_CTRL_MASK 0x5500 +#define DAVINCI_VC_CTRL_RSTADC BIT(0) +#define DAVINCI_VC_CTRL_RSTDAC BIT(1) +#define DAVINCI_VC_CTRL_RD_BITS_8 BIT(4) +#define DAVINCI_VC_CTRL_RD_UNSIGNED BIT(5) +#define DAVINCI_VC_CTRL_WD_BITS_8 BIT(6) +#define DAVINCI_VC_CTRL_WD_UNSIGNED BIT(7) +#define DAVINCI_VC_CTRL_RFIFOEN BIT(8) +#define DAVINCI_VC_CTRL_RFIFOCL BIT(9) +#define DAVINCI_VC_CTRL_RFIFOMD_WORD_1 BIT(10) +#define DAVINCI_VC_CTRL_WFIFOEN BIT(12) +#define DAVINCI_VC_CTRL_WFIFOCL BIT(13) +#define DAVINCI_VC_CTRL_WFIFOMD_WORD_1 BIT(14) + +/* DAVINCI_VC_INT bit fields */ +#define DAVINCI_VC_INT_MASK 0x3F +#define DAVINCI_VC_INT_RDRDY_MASK BIT(0) +#define DAVINCI_VC_INT_RERROVF_MASK BIT(1) +#define DAVINCI_VC_INT_RERRUDR_MASK BIT(2) +#define DAVINCI_VC_INT_WDREQ_MASK BIT(3) +#define DAVINCI_VC_INT_WERROVF_MASKBIT BIT(4) +#define DAVINCI_VC_INT_WERRUDR_MASK BIT(5) + +/* DAVINCI_VC_REG05 bit fields */ +#define DAVINCI_VC_REG05_PGA_GAIN 0x07 + +/* DAVINCI_VC_REG09 bit fields */ +#define DAVINCI_VC_REG09_MUTE 0x40 +#define DAVINCI_VC_REG09_DIG_ATTEN 0x3F + +/* DAVINCI_VC_REG12 bit fields */ +#define DAVINCI_VC_REG12_POWER_ALL_ON 0xFD +#define DAVINCI_VC_REG12_POWER_ALL_OFF 0x00 + +#define DAVINCI_VC_CELLS 2 + +enum davinci_vc_cells { + DAVINCI_VC_VCIF_CELL, + DAVINCI_VC_CQ93VC_CELL, +}; + +struct davinci_vcif { + struct platform_device *pdev; + u32 dma_tx_channel; + u32 dma_rx_channel; + dma_addr_t dma_tx_addr; + dma_addr_t dma_rx_addr; +}; + +struct cq93vc { + struct platform_device *pdev; + struct snd_soc_codec *codec; + u32 sysclk; +}; + +struct davinci_vc; + +struct davinci_vc { + /* Device data */ + struct device *dev; + struct platform_device *pdev; + struct clk *clk; + + /* Memory resources */ + void __iomem *base; + resource_size_t pbase; + size_t base_size; + + /* MFD cells */ + struct mfd_cell cells[DAVINCI_VC_CELLS]; + + /* Client devices */ + struct davinci_vcif davinci_vcif; + struct cq93vc cq93vc; +}; + +#endif -- 1.6.0.4

3 3

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33 fixes
by Peter Ujfalusi 12 Mar '10

12 Mar '10

Hello, Revisiting the start/stop sequence used by the tlv320dac33 codec in order to avoid race conditions, which can occur, when the codec is in FIFO modes. It has been found that OMAP3 McBSP needs the BCLK continuously running on the serial interface when it is configured to be slave. DAC33 can be configured in a way, that it keeps the BCLK ticking, and only enable the FS when it is needed. Option for the platform data to select between continuous BCLK or non continuous BCLK. --- Peter Ujfalusi (2): ASoC: tlv320dac33: Start/stop sequence change ASoC: tlv320dac33: Add option for keeping the BCLK running include/sound/tlv320dac33-plat.h | 1 + sound/soc/codecs/tlv320dac33.c | 38 ++++++++++++++------------------------ 2 files changed, 15 insertions(+), 24 deletions(-)

4 8

[alsa-devel] [PATCHv2 0/5] FIFO caused playback delay (latency) handling in soc/omap
by Peter Ujfalusi 12 Mar '10

12 Mar '10

Hello, Changes since the RFC round on the alsa-devel list: - core is not limiting the query for the delay to the playback stream. It is no the dai and platform driver's responsibility handle that - delay interface added for platform driver - Capture path delay calculation is added to OMAP McBSP. - omap_mcbsp_get_tx_buffstat renamed as omap_mcbsp_get_tx_delay to reflect what it is actually doing. Tony: is it still possible to get this taken for 2.6.34? It would be nice to avoid cross tree problems during the 34 cycle... The initial RFC patch description: There has been discussion in alsa-devel in this subject several times, but no actual patches has been sent (it was not soc related, but anyway it was discussed). Based on the available information the latency caused by the HW buffer on some systems can be handled by updating the runtime->delay. It has been discussed, that the runtime->delay can also be updated dynamically to show more accurate delay. To further complicate things, in ASoC we could have more buffer in the chain. To handle this we need soc level support. This series tries to do that in soc by: - introducing a pcm_pointer wrapper - in this wrapper we call the original pcm_pointer functions to get the DMA pointer - introducing a new interface in dai_ops and for platfrom drivers to ask the delay - adding the cpu_dai, codec_dai and platform driver returned delay to form the actual delay - update the runtime->delay with this value. With this approach none of the existing drivers need change, but they can add support for specifying the FIFO caused delay. In this series on top of the core changes the omap(3) code is updated to take this delay reporting into use. I have not added the support to the tlv320dac33 codec driver, since it needs a bit more work, but along the same line it can be done, and if the tlv320dac33 is hooked to omap McBSP than applications can know the whole delay/latency on that path. The series is based on linux-omap master branch (with regcache and sidetone). However the first 3 patch for soc core applies on top of sound-2.6 topic/asoc cleanly as well. Obviously the last patch depends on the OMAP platform patch. --- Peter Ujfalusi (5): ASoC: core: fix tailing whitespace in soc_pcm_apply_symmetry ASoC: core: soc level wrapper for pcm_pointer callback ASoC: core: Add delay operation to snd_soc_dai_ops OMAP3: McBSP: Add interface for FIFO caused delay query ASoC: OMAP3: Report delay caused by the internal FIFO arch/arm/plat-omap/include/plat/mcbsp.h | 6 +++ arch/arm/plat-omap/mcbsp.c | 55 +++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 6 +++ include/sound/soc.h | 7 ++++ sound/soc/omap/omap-mcbsp.c | 26 ++++++++++++++ sound/soc/soc-core.c | 39 +++++++++++++++++++++- 6 files changed, 137 insertions(+), 2 deletions(-)

5 20

[alsa-devel] [PATCH v2 2/5] ASoC: DaVinci: Voice Codec Interface
by miguel.aguilar＠ridgerun.com 12 Mar '10

12 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the support for the interface needed by the DaVinci Voice Codec CQ93VC. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/davinci/Kconfig | 3 + sound/soc/davinci/Makefile | 2 + sound/soc/davinci/davinci-vcif.c | 272 ++++++++++++++++++++++++++++++++++++++ sound/soc/davinci/davinci-vcif.h | 28 ++++ 4 files changed, 305 insertions(+), 0 deletions(-) create mode 100644 sound/soc/davinci/davinci-vcif.c create mode 100644 sound/soc/davinci/davinci-vcif.h diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 047ee39..47e7cce 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -12,6 +12,9 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate +config SND_DAVINCI_SOC_VCIF + tristate + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a6939d7..a93679d 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -2,10 +2,12 @@ snd-soc-davinci-objs := davinci-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o +snd-soc-davinci-vcif-objs:= davinci-vcif.o obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o +obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o # DAVINCI Machine Support snd-soc-evm-objs := davinci-evm.o diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c new file mode 100644 index 0000000..03f3feb --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.c @@ -0,0 +1,272 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" +#include "davinci-i2s.h" +#include "davinci-vcif.h" + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +struct davinci_vcif_dev { + struct davinci_vc *davinci_vc; + struct davinci_pcm_dma_params dma_params[2]; +}; + +static void davinci_vcif_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static void davinci_vcif_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_vcif_dev *davinci_vcif_dev = + rtd->dai->cpu_dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + else + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); +} + +static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *davinci_vcif_dev = dai->private_data; + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; + struct davinci_pcm_dma_params *dma_params = + &davinci_vcif_dev->dma_params[substream->stream]; + u32 w; + + /* Restart the codec before setup */ + davinci_vcif_stop(substream); + davinci_vcif_start(substream); + + /* General line settings */ + writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); + + writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); + + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + dma_params->data_type = 0; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 1); + break; + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_WD_BITS_8, 1); + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | + DAVINCI_VC_CTRL_RD_UNSIGNED | + DAVINCI_VC_CTRL_WD_BITS_8 | + DAVINCI_VC_CTRL_WD_UNSIGNED, 0); + break; + default: + printk(KERN_WARNING "davinci-vcif: unsupported PCM format"); + return -EINVAL; + } + + dma_params->acnt = dma_params->data_type; + + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); + + return 0; +} + +static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_vcif_start(substream); + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_vcif_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +#define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 + +static struct snd_soc_dai_ops davinci_vcif_dai_ops = { + .trigger = davinci_vcif_trigger, + .hw_params = davinci_vcif_hw_params, +}; + +struct snd_soc_dai davinci_vcif_dai = { + .name = "davinci-vcif", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = DAVINCI_VCIF_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &davinci_vcif_dai_ops, + +}; +EXPORT_SYMBOL_GPL(davinci_vcif_dai); + +static int davinci_vcif_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct davinci_vcif_dev *davinci_vcif_dev; + int ret; + + davinci_vcif_dev = kzalloc(sizeof(struct davinci_vcif_dev), GFP_KERNEL); + if (!davinci_vc) { + dev_dbg(&pdev->dev, + "could not allocate memory for private data\n"); + return -ENOMEM; + } + + /* DMA tx params */ + davinci_vcif_dev->davinci_vc = davinci_vc; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = + davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + davinci_vc->davinci_vcif.dma_tx_addr; + + /* DMA rx params */ + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = + davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + davinci_vc->davinci_vcif.dma_rx_addr; + + davinci_vcif_dai.dev = &pdev->dev; + davinci_vcif_dai.dma_data = davinci_vcif_dev->dma_params; + davinci_vcif_dai.private_data = davinci_vcif_dev; + + ret = snd_soc_register_dai(&davinci_vcif_dai); + if (ret != 0) { + dev_err(&pdev->dev, "could not register dai\n"); + goto fail; + } + + return 0; + +fail: + kfree(davinci_vcif_dev); + + return ret; +} + +static int davinci_vcif_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&davinci_vcif_dai); + + return 0; +} + +static struct platform_driver davinci_vcif_driver = { + .probe = davinci_vcif_probe, + .remove = davinci_vcif_remove, + .driver = { + .name = "davinci_vcif", + .owner = THIS_MODULE, + }, +}; + +static int __init davinci_vcif_init(void) +{ + return platform_driver_probe(&davinci_vcif_driver, davinci_vcif_probe); +} +module_init(davinci_vcif_init); + +static void __exit davinci_vcif_exit(void) +{ + platform_driver_unregister(&davinci_vcif_driver); +} +module_exit(davinci_vcif_exit); + +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-vcif.h b/sound/soc/davinci/davinci-vcif.h new file mode 100644 index 0000000..571c994 --- /dev/null +++ b/sound/soc/davinci/davinci-vcif.h @@ -0,0 +1,28 @@ +/* + * ALSA SoC Voice Codec Interface for TI DAVINCI processor + * + * Copyright (C) 2010 Texas Instruments. + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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 + */ + +#ifndef _DAVINCI_VCIF_H +#define _DAVINCI_VCIF_H + +extern struct snd_soc_dai davinci_vcif_dai; + +#endif -- 1.6.0.4

2 1

[alsa-devel] ALSA: usb-mixer: Add support for UAC2 devices
by Daniel Mack 12 Mar '10

12 Mar '10

This patch series adds support for mixer interfaces in USB audio class v2 devices. Successfully tested on a XMOS L1 eval kit. They apply on top of the topic/usb branch. There's a number significant differences between v1 and v2 implementations, which the patches itself may explain best :) I moved most quirks out into a seperate file, and hope you agree that it's cleaner that way. Most descriptors are now parsed with structs which makes the code more readable. However, there are quite some descriptors with variable field lengths so they can't be mapped into structs. I introduces some macros to access these fields. Again, please test this with v1 devices, as I can't be entirely confident that I didn't break anything in the transition. Thanks, Daniel [PATCH 1/6] linux/usb/audio.h: split header [PATCH 2/6] ALSA: usb-mixer: use defines from audio.h [PATCH 3/6] ALSA: usb-mixer: factor out quirks [PATCH 4/6] ALSA: usb-mixer: rename usbmixer.[ch] -> mixer.[ch] [PATCH 5/6] ALSA: usb-mixer: parse descriptors with structs [PATCH 6/6] ALSA: usb-mixer: Add support for Audio Class v2.0

3 9

[alsa-devel] [PATCH] driver-core: Add device node pointer to struct device
by Grant Likely 11 Mar '10

11 Mar '10

Currently, platforms using CONFIG_OF add a 'struct device_node *of_node' to dev->archdata. However, with CONFIG_OF becoming generic for all architectures, it makes sense for commonality to move it out of archdata and into struct device proper. This patch adds a struct device_node *of_node member to struct device and updates all locations which currently write the device_node pointer into archdata to also update dev->of_node. Subsequent patches will modify callers to use the archdata location and ultimately remove the archdata member entirely. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> Acked-by: Greg Kroah-Hartman <gregkh(a)suse.de> --- This should have been the first patch in my 37 patch series "[PATCH 00/37] OF: move device node pointer into struct device.", but I accidentally missed it. Apply this patch before the rest of the series. I've also pushed out all the patches that I've just posted to the 'experimental-devicetree' branch of my git tree: git://git.secretlab.ca/git/linux-2.6 experimental-devicetree arch/microblaze/kernel/of_device.c | 3 ++- arch/powerpc/kernel/of_device.c | 3 ++- arch/powerpc/kernel/pci-common.c | 5 +++-- arch/powerpc/kernel/vio.c | 3 ++- arch/powerpc/platforms/ps3/system-bus.c | 3 ++- arch/sparc/kernel/pci.c | 3 ++- drivers/of/of_mdio.c | 3 ++- drivers/of/of_spi.c | 3 ++- include/linux/device.h | 4 ++++ 9 files changed, 21 insertions(+), 9 deletions(-) diff --git a/arch/microblaze/kernel/of_device.c b/arch/microblaze/kernel/of_device.c index 9a0f763..12371bd 100644 --- a/arch/microblaze/kernel/of_device.c +++ b/arch/microblaze/kernel/of_device.c @@ -53,7 +53,8 @@ struct of_device *of_device_alloc(struct device_node *np, dev->dev.dma_mask = &dev->dma_mask; dev->dev.parent = parent; dev->dev.release = of_release_dev; - dev->dev.archdata.of_node = np; + dev->dev.archdata.of_node = np; /* temporary; remove after merging */ + dev->dev.of_node = np; if (bus_id) dev_set_name(&dev->dev, bus_id); diff --git a/arch/powerpc/kernel/of_device.c b/arch/powerpc/kernel/of_device.c index a359cb0..efbc262 100644 --- a/arch/powerpc/kernel/of_device.c +++ b/arch/powerpc/kernel/of_device.c @@ -73,7 +73,8 @@ struct of_device *of_device_alloc(struct device_node *np, dev->dev.dma_mask = &dev->dma_mask; dev->dev.parent = parent; dev->dev.release = of_release_dev; - dev->dev.archdata.of_node = np; + dev->dev.archdata.of_node = np; /* temporary; remove after merging */ + dev->dev.of_node = np; if (bus_id) dev_set_name(&dev->dev, "%s", bus_id); diff --git a/arch/powerpc/kernel/pci-common.c b/arch/powerpc/kernel/pci-common.c index 2597f95..90501f2 100644 --- a/arch/powerpc/kernel/pci-common.c +++ b/arch/powerpc/kernel/pci-common.c @@ -1111,8 +1111,9 @@ void __devinit pcibios_setup_bus_devices(struct pci_bus *bus) if (dev->is_added) continue; - /* Setup OF node pointer in archdata */ - sd->of_node = pci_device_to_OF_node(dev); + /* Setup OF node pointer in the device */ + sd->of_node = pci_device_to_OF_node(dev); /* temporary */ + dev->dev.of_node = pci_device_to_OF_node(dev); /* Fixup NUMA node as it may not be setup yet by the generic * code and is needed by the DMA init diff --git a/arch/powerpc/kernel/vio.c b/arch/powerpc/kernel/vio.c index 77f6421..70b52c7 100644 --- a/arch/powerpc/kernel/vio.c +++ b/arch/powerpc/kernel/vio.c @@ -1229,7 +1229,8 @@ struct vio_dev *vio_register_device_node(struct device_node *of_node) if (unit_address != NULL) viodev->unit_address = *unit_address; } - viodev->dev.archdata.of_node = of_node_get(of_node); + viodev->dev.of_node = of_node_get(of_node); + viodev->dev.archdata.of_node = viodev->dev.of_node; /* temporary */ if (firmware_has_feature(FW_FEATURE_CMO)) vio_cmo_set_dma_ops(viodev); diff --git a/arch/powerpc/platforms/ps3/system-bus.c b/arch/powerpc/platforms/ps3/system-bus.c index e34b305..5f162c0 100644 --- a/arch/powerpc/platforms/ps3/system-bus.c +++ b/arch/powerpc/platforms/ps3/system-bus.c @@ -765,7 +765,8 @@ int ps3_system_bus_device_register(struct ps3_system_bus_device *dev) BUG(); }; - dev->core.archdata.of_node = NULL; + dev->core.of_node = NULL; + dev->core.archdata.of_node = NULL; /* temporary */ set_dev_node(&dev->core, 0); pr_debug("%s:%d add %s\n", __func__, __LINE__, dev_name(&dev->core)); diff --git a/arch/sparc/kernel/pci.c b/arch/sparc/kernel/pci.c index 5ac539a..cf73f43 100644 --- a/arch/sparc/kernel/pci.c +++ b/arch/sparc/kernel/pci.c @@ -261,7 +261,7 @@ static struct pci_dev *of_create_pci_dev(struct pci_pbm_info *pbm, sd->iommu = pbm->iommu; sd->stc = &pbm->stc; sd->host_controller = pbm; - sd->prom_node = node; + sd->prom_node = node; /* temporary */ sd->op = op = of_find_device_by_node(node); sd->numa_node = pbm->numa_node; @@ -285,6 +285,7 @@ static struct pci_dev *of_create_pci_dev(struct pci_pbm_info *pbm, dev->sysdata = node; dev->dev.parent = bus->bridge; dev->dev.bus = &pci_bus_type; + dev->dev.of_node = node; dev->devfn = devfn; dev->multifunction = 0; /* maybe a lie? */ set_pcie_port_type(dev); diff --git a/drivers/of/of_mdio.c b/drivers/of/of_mdio.c index 18ecae4..3bf20bf 100644 --- a/drivers/of/of_mdio.c +++ b/drivers/of/of_mdio.c @@ -79,7 +79,8 @@ int of_mdiobus_register(struct mii_bus *mdio, struct device_node *np) /* Associate the OF node with the device structure so it * can be looked up later */ of_node_get(child); - dev_archdata_set_node(&phy->dev.archdata, child); + dev_archdata_set_node(&phy->dev.archdata, child); /* temp */ + phy->dev.of_node = child; /* All data is now stored in the phy struct; register it */ rc = phy_device_register(phy); diff --git a/drivers/of/of_spi.c b/drivers/of/of_spi.c index f65f48b..433d76d 100644 --- a/drivers/of/of_spi.c +++ b/drivers/of/of_spi.c @@ -79,7 +79,8 @@ void of_register_spi_devices(struct spi_master *master, struct device_node *np) /* Store a pointer to the node in the device structure */ of_node_get(nc); - spi->dev.archdata.of_node = nc; + spi->dev.of_node = nc; + spi->dev.archdata.of_node = nc; /* temporary */ /* Register the new device */ request_module(spi->modalias); diff --git a/include/linux/device.h b/include/linux/device.h index 1821928..7a968bd 100644 --- a/include/linux/device.h +++ b/include/linux/device.h @@ -34,6 +34,7 @@ struct class; struct class_private; struct bus_type; struct bus_type_private; +struct device_node; struct bus_attribute { struct attribute attr; @@ -433,6 +434,9 @@ struct device { override */ /* arch specific additions */ struct dev_archdata archdata; +#ifdef CONFIG_OF + struct device_node *of_node; +#endif dev_t devt; /* dev_t, creates the sysfs "dev" */

1 0

[alsa-devel] [PATCH 00/27] Clean up of_platform drivers
by Grant Likely 11 Mar '10

11 Mar '10

struct of_platform_driver unnecessarily duplicates the .owner and .name fields from the encapsulated struct device_driver. This patch series goes through all of_platform drivers and makes it use the device_driver instance so that the duplicate field can be removed. Since every of_platform_driver is getting touched anyway, this series takes the opportunity to also move the of_device_id match table out of struct of_platform_driver and into struct device_driver proper. This makes it possible for all bus types to make use of OF-style binding when CONFIG_OF is set. I've compile tested this on sparc, powerpc and microblaze. It should all be fully bisectable, and only the last patch removes the old fields (ie. nothing should break without the last patch). g. --- Grant Likely (27): of: remove obsolete members from of_platform_driver arch/sparc/of: don't use deprecated field in of_platform_driver arch/powerpc/of: don't use deprecated field in of_platform_driver sound/of: don't use deprecated fields in of_platform_driver drivers/infiniband/of: don't use deprecated fields in of_platform_driver drivers/i2c/of: don't use deprecated field in of_platform_driver drivers/hwmon/of: don't use deprecated field in of_platform_driver drivers/mmc/of: don't use deprecated field in of_platform_driver drivers/pcmcia/of: don't use deprecated field in of_platform_driver drivers/parport/of: don't use deprecated field in of_platform_driver drivers/sbus/of: don't use deprecated field in of_platform_driver drivers/scsi/of: don't use deprecated field in of_platform_driver drivers/spi/of: don't use deprecated field in of_platform_driver drivers/serial/of: don't use deprecated field in of_platform_driver drivers/net/of: don't use deprecated field in of_platform_driver drivers/mtd/of: don't use deprecated field in of_platform_driver drivers/macintosh/of: don't use deprecated field in of_platform_driver drivers/input/of: don't use deprecated field in of_platform_driver drivers/atm/of: don't use deprecated field in of_platform_driver drivers/crypto/of: don't use deprecated field in of_platform_driver drivers/char/of: don't use deprecated field in of_platform_driver drivers/block/of: don't use deprecated field in of_platform_driver drivers/usb/of: don't use deprecated field in of_platform_driver drivers/video/of: don't use deprecated field in of_platform_driver drivers/watchdog/of: don't use deprecated .name and .owner in of_platform_driver drivers/gpio/of: don't use deprecated fields in of_platform_driver drivercore: Add of_match_table to the common device drivers arch/powerpc/kernel/ibmebus.c | 2 +- arch/powerpc/kernel/of_platform.c | 3 ++- arch/powerpc/platforms/52xx/mpc52xx_gpio.c | 14 ++++++++---- arch/powerpc/platforms/52xx/mpc52xx_gpt.c | 7 ++++-- arch/powerpc/platforms/52xx/mpc52xx_lpbfifo.c | 8 ++++--- arch/powerpc/platforms/82xx/ep8248e.c | 3 ++- arch/powerpc/platforms/83xx/suspend.c | 7 ++++-- arch/powerpc/platforms/cell/axon_msi.c | 7 +++--- arch/powerpc/platforms/pasemi/gpio_mdio.c | 7 +++--- arch/powerpc/sysdev/axonram.c | 8 +++---- arch/powerpc/sysdev/bestcomm/bestcomm.c | 10 +++----- arch/powerpc/sysdev/fsl_msi.c | 7 ++++-- arch/powerpc/sysdev/fsl_pmc.c | 7 ++++-- arch/powerpc/sysdev/fsl_rio.c | 7 ++++-- arch/powerpc/sysdev/pmi.c | 7 +++--- arch/powerpc/sysdev/qe_lib/qe.c | 7 ++++-- arch/sparc/include/asm/parport.h | 7 ++++-- arch/sparc/kernel/apc.c | 7 ++++-- arch/sparc/kernel/auxio_64.c | 7 +++--- arch/sparc/kernel/central.c | 14 +++++++----- arch/sparc/kernel/chmc.c | 7 ++++-- arch/sparc/kernel/pci_fire.c | 7 ++++-- arch/sparc/kernel/pci_psycho.c | 7 ++++-- arch/sparc/kernel/pci_sabre.c | 7 ++++-- arch/sparc/kernel/pci_schizo.c | 7 ++++-- arch/sparc/kernel/pci_sun4v.c | 7 ++++-- arch/sparc/kernel/pmc.c | 7 ++++-- arch/sparc/kernel/power.c | 7 +++--- arch/sparc/kernel/time_32.c | 7 +++--- arch/sparc/kernel/time_64.c | 21 ++++++++++-------- drivers/ata/pata_of_platform.c | 7 ++++-- drivers/ata/sata_fsl.c | 7 ++++-- drivers/atm/fore200e.c | 7 ++++-- drivers/block/xsysace.c | 5 ++-- drivers/char/hw_random/n2-drv.c | 7 ++++-- drivers/char/hw_random/pasemi-rng.c | 7 ++++-- drivers/char/ipmi/ipmi_si_intf.c | 7 ++++-- drivers/char/xilinx_hwicap/xilinx_hwicap.c | 5 ++-- drivers/crypto/amcc/crypto4xx_core.c | 7 ++++-- drivers/crypto/talitos.c | 7 ++++-- drivers/dma/fsldma.c | 13 +++++++---- drivers/dma/ppc4xx/adma.c | 2 +- drivers/edac/mpc85xx_edac.c | 30 ++++++++++--------------- drivers/edac/ppc4xx_edac.c | 10 ++++---- drivers/hwmon/ultra45_env.c | 7 ++++-- drivers/i2c/busses/i2c-cpm.c | 10 ++++---- drivers/i2c/busses/i2c-ibm_iic.c | 7 ++++-- drivers/i2c/busses/i2c-mpc.c | 8 +++---- drivers/infiniband/hw/ehca/ehca_main.c | 7 +++--- drivers/input/misc/sparcspkr.c | 14 ++++++++---- drivers/input/serio/i8042-sparcio.h | 7 ++++-- drivers/input/serio/xilinx_ps2.c | 7 ++++-- drivers/leds/leds-gpio.c | 2 +- drivers/macintosh/macio_asic.c | 5 ++-- drivers/macintosh/smu.c | 7 ++++-- drivers/macintosh/therm_pm72.c | 7 ++++-- drivers/macintosh/therm_windtunnel.c | 7 ++++-- drivers/mmc/host/sdhci-of-core.c | 7 ++++-- drivers/mtd/maps/physmap_of.c | 7 ++++-- drivers/mtd/maps/sun_uflash.c | 7 ++++-- drivers/mtd/nand/fsl_elbc_nand.c | 5 +++- drivers/mtd/nand/fsl_upm.c | 7 ++++-- drivers/mtd/nand/ndfc.c | 5 +++- drivers/mtd/nand/pasemi_nand.c | 7 ++++-- drivers/mtd/nand/socrates_nand.c | 7 ++++-- drivers/net/can/mscan/mpc5xxx_can.c | 8 ++++--- drivers/net/can/sja1000/sja1000_of_platform.c | 8 ++++--- drivers/net/ehea/ehea_main.c | 7 ++++-- drivers/net/fs_enet/fs_enet-main.c | 7 ++++-- drivers/net/fs_enet/mii-bitbang.c | 7 ++++-- drivers/net/fs_enet/mii-fec.c | 7 ++++-- drivers/net/fsl_pq_mdio.c | 7 ++++-- drivers/net/gianfar.c | 10 +++++--- drivers/net/ibm_newemac/core.c | 8 ++++--- drivers/net/ibm_newemac/mal.c | 8 ++++--- drivers/net/ibm_newemac/rgmii.c | 8 ++++--- drivers/net/ibm_newemac/tah.c | 8 ++++--- drivers/net/ibm_newemac/zmii.c | 8 ++++--- drivers/net/ll_temac_main.c | 2 +- drivers/net/myri_sbus.c | 7 ++++-- drivers/net/niu.c | 7 ++++-- drivers/net/phy/mdio-gpio.c | 7 ++++-- drivers/net/sunbmac.c | 7 ++++-- drivers/net/sunhme.c | 7 ++++-- drivers/net/sunlance.c | 7 ++++-- drivers/net/sunqe.c | 7 ++++-- drivers/net/ucc_geth.c | 7 ++++-- drivers/net/xilinx_emaclite.c | 7 ++++-- drivers/of/platform.c | 10 ++------ drivers/parport/parport_sunbpp.c | 7 ++++-- drivers/pcmcia/electra_cf.c | 7 ++++-- drivers/pcmcia/m8xx_pcmcia.c | 7 ++++-- drivers/sbus/char/bbc_i2c.c | 7 ++++-- drivers/sbus/char/display7seg.c | 7 ++++-- drivers/sbus/char/envctrl.c | 7 ++++-- drivers/sbus/char/flash.c | 7 ++++-- drivers/sbus/char/uctrl.c | 7 ++++-- drivers/scsi/qlogicpti.c | 7 ++++-- drivers/scsi/sun_esp.c | 7 ++++-- drivers/serial/apbuart.c | 8 +++---- drivers/serial/cpm_uart/cpm_uart_core.c | 7 ++++-- drivers/serial/mpc52xx_uart.c | 7 +++--- drivers/serial/of_serial.c | 8 ++++--- drivers/serial/sunhv.c | 7 ++++-- drivers/serial/sunsab.c | 7 ++++-- drivers/serial/sunsu.c | 7 ++++-- drivers/serial/sunzilog.c | 7 ++++-- drivers/serial/uartlite.c | 5 ++-- drivers/serial/ucc_uart.c | 8 ++++--- drivers/spi/mpc52xx_psc_spi.c | 4 +-- drivers/spi/mpc52xx_spi.c | 8 ++++--- drivers/spi/spi_mpc8xxx.c | 7 ++++-- drivers/spi/spi_ppc4xx.c | 2 +- drivers/spi/xilinx_spi_of.c | 2 +- drivers/usb/gadget/fsl_qe_udc.c | 7 ++++-- drivers/usb/host/ehci-ppc-of.c | 9 +++----- drivers/usb/host/ehci-xilinx-of.c | 9 +++----- drivers/usb/host/fhci-hcd.c | 7 ++++-- drivers/usb/host/isp1760-if.c | 7 ++++-- drivers/usb/host/ohci-ppc-of.c | 13 +++-------- drivers/video/bw2.c | 7 ++++-- drivers/video/cg14.c | 7 ++++-- drivers/video/cg3.c | 7 ++++-- drivers/video/cg6.c | 7 ++++-- drivers/video/ffb.c | 7 ++++-- drivers/video/fsl-diu-fb.c | 8 ++++--- drivers/video/leo.c | 7 ++++-- drivers/video/mb862xx/mb862xxfb.c | 8 ++++--- drivers/video/p9100.c | 7 ++++-- drivers/video/platinumfb.c | 7 ++++-- drivers/video/tcx.c | 7 ++++-- drivers/video/xilinxfb.c | 5 ++-- drivers/watchdog/cpwd.c | 7 ++++-- drivers/watchdog/gef_wdt.c | 8 ++++--- drivers/watchdog/mpc8xxx_wdt.c | 8 +++---- drivers/watchdog/riowd.c | 7 ++++-- include/linux/device.h | 4 +++ include/linux/of_platform.h | 4 --- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 8 ++++--- sound/sparc/amd7930.c | 7 ++++-- sound/sparc/cs4231.c | 7 ++++-- sound/sparc/dbri.c | 7 ++++-- 144 files changed, 665 insertions(+), 380 deletions(-) -- Signature

3 29

[alsa-devel] [PATCH 5/5] DaVinci: DM365: Voice Codec support for the DM365 EVM
by miguel.aguilar＠ridgerun.com 11 Mar '10

11 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> The DM365 EVM has two codecs: the Audio Codec (AIC3x) and the Voice Codec, the idea is to have both enabled in the same kernel simultaneously. However, the current soc-core doesn't support simultaneous codecs, once that support will have added, a patch will be posted to enable both codecs in the DM365 EVM. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/board-dm365-evm.c | 4 +++ sound/soc/davinci/Kconfig | 24 +++++++++++++++- sound/soc/davinci/davinci-evm.c | 45 +++++++++++++++++++++++++++++- 3 files changed, 69 insertions(+), 4 deletions(-) diff --git a/arch/arm/mach-davinci/board-dm365-evm.c b/arch/arm/mach-davinci/board-dm365-evm.c index 5c2636c..865d06a 100644 --- a/arch/arm/mach-davinci/board-dm365-evm.c +++ b/arch/arm/mach-davinci/board-dm365-evm.c @@ -516,7 +516,11 @@ static __init void dm365_evm_init(void) /* maybe setup mmc1/etc ... _after_ mmc0 */ evm_init_cpld(); +#ifdef CONFIG_SND_DM365_AIC3X_CODEC dm365_init_asp(&dm365_evm_snd_data); +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + dm365_init_vc(&dm365_evm_snd_data); +#endif dm365_init_rtc(); dm365_init_ks(&dm365evm_ks_data); } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 47e7cce..6bbf001 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -18,12 +18,32 @@ config SND_DAVINCI_SOC_VCIF config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DaVinci DM6446 or DM355 EVM platforms. + DaVinci DM6446, DM355 or DM365 EVM platforms. + +choice + prompt "DM365 codec select" + depends on SND_DAVINCI_SOC_EVM + depends on MACH_DAVINCI_DM365_EVM + default SND_DM365_EXTERNAL_CODEC + +config SND_DM365_AIC3X_CODEC + bool "Audio Codec - AIC3101" + help + Say Y if you want to add support for AIC3101 audio codec + +config SND_DM365_VOICE_CODEC + bool "Voice Codec - CQ93VC" + select MFD_DAVINCI_VOICECODEC + select SND_DAVINCI_SOC_VCIF + select SND_SOC_CQ0093VC + help + Say Y if you want to add support for SoC On-chip voice codec +endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..ef63096 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -28,10 +28,12 @@ #include <mach/mux.h> #include "../codecs/tlv320aic3x.h" +#include "../codecs/cq93vc.h" #include "../codecs/spdif_transciever.h" #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" +#include "davinci-vcif.h" #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) @@ -151,6 +153,22 @@ static struct snd_soc_dai_link evm_dai = { .ops = &evm_ops, }; +static struct snd_soc_dai_link dm365_evm_dai = { +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .name = "Voice Codec - CQ93VC", + .stream_name = "CQ93", + .cpu_dai = &davinci_vcif_dai, + .codec_dai = &cq93vc_dai, +#endif +}; + static struct snd_soc_dai_link dm6467_evm_dai[] = { { .name = "TLV320AIC3X", @@ -177,7 +195,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci dm6446, dm355 or dm365 evm audio machine driver */ +/* davinci dm6446, dm355 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -185,6 +203,15 @@ static struct snd_soc_card snd_soc_card_evm = { .num_links = 1, }; +/* davinci dm365 evm audio machine driver */ +static struct snd_soc_card dm365_snd_soc_card_evm = { + .name = "DaVinci DM365 EVM", + .platform = &davinci_soc_platform, + .dai_link = &dm365_evm_dai, + .num_links = 1, +}; + + /* davinci dm6467 evm audio machine driver */ static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", @@ -217,6 +244,17 @@ static struct snd_soc_device evm_snd_devdata = { }; /* evm audio subsystem */ +static struct snd_soc_device dm365_evm_snd_devdata = { + .card = &dm365_snd_soc_card_evm, +#ifdef CONFIG_SND_DM365_AIC3X_CODEC + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &aic3x_setup, +#elif defined(CONFIG_SND_DM365_VOICE_CODEC) + .codec_dev = &soc_codec_dev_cq93vc, +#endif +}; + +/* evm audio subsystem */ static struct snd_soc_device dm6467_evm_snd_devdata = { .card = &dm6467_snd_soc_card_evm, .codec_dev = &soc_codec_dev_aic3x, @@ -244,12 +282,15 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { + if (machine_is_davinci_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 1; + } else if (machine_is_davinci_dm365_evm()) { + evm_snd_dev_data = &dm365_evm_snd_devdata; + index = 0; } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; -- 1.6.0.4

1 0

[alsa-devel] [PATCH 4/5] DaVinci: DM365: Voice codec support for the DM365 SoC
by miguel.aguilar＠ridgerun.com 11 Mar '10

11 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> This patch adds the generic Voice Codec support for the DM365 based platforms. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- arch/arm/mach-davinci/dm365.c | 37 +++++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/dm365.h | 5 ++++ arch/arm/mach-davinci/include/mach/mux.h | 2 + 3 files changed, 43 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/dm365.c b/arch/arm/mach-davinci/dm365.c index c63afc0..b5345d5 100644 --- a/arch/arm/mach-davinci/dm365.c +++ b/arch/arm/mach-davinci/dm365.c @@ -455,7 +455,7 @@ static struct clk_lookup dm365_clks[] = { CLK(NULL, "timer3", &timer3_clk), CLK(NULL, "usb", &usb_clk), CLK("davinci_emac.1", NULL, &emac_clk), - CLK("voice_codec", NULL, &voicecodec_clk), + CLK("davinci_voicecodec", NULL, &voicecodec_clk), CLK("davinci-asp.0", NULL, &asp0_clk), CLK(NULL, "rto", &rto_clk), CLK(NULL, "mjcp", &mjcp_clk), @@ -606,6 +606,8 @@ INT_CFG(DM365, INT_NSF_DISABLE, 25, 1, 0, false) EVT_CFG(DM365, EVT2_ASP_TX, 0, 1, 0, false) EVT_CFG(DM365, EVT3_ASP_RX, 1, 1, 0, false) +EVT_CFG(DM365, EVT2_VC_TX, 0, 1, 1, false) +EVT_CFG(DM365, EVT3_VC_RX, 1, 1, 1, false) #endif }; @@ -835,6 +837,31 @@ static struct platform_device dm365_asp_device = { .resource = dm365_asp_resources, }; +static struct resource dm365_vc_resources[] = { + { + .start = DAVINCI_DM365_VC_BASE, + .end = DAVINCI_DM365_VC_BASE + SZ_1K - 1, + .flags = IORESOURCE_MEM, + }, + { + .start = DAVINCI_DMA_VC_TX, + .end = DAVINCI_DMA_VC_TX, + .flags = IORESOURCE_DMA, + }, + { + .start = DAVINCI_DMA_VC_RX, + .end = DAVINCI_DMA_VC_RX, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device dm365_vc_device = { + .name = "davinci_voicecodec", + .id = -1, + .num_resources = ARRAY_SIZE(dm365_vc_resources), + .resource = dm365_vc_resources, +}; + static struct resource dm365_rtc_resources[] = { { .start = DM365_RTC_BASE, @@ -991,6 +1018,14 @@ void __init dm365_init_asp(struct snd_platform_data *pdata) platform_device_register(&dm365_asp_device); } +void __init dm365_init_vc(struct snd_platform_data *pdata) +{ + davinci_cfg_reg(DM365_EVT2_VC_TX); + davinci_cfg_reg(DM365_EVT3_VC_RX); + dm365_vc_device.dev.platform_data = pdata; + platform_device_register(&dm365_vc_device); +} + void __init dm365_init_ks(struct davinci_ks_platform_data *pdata) { dm365_ks_device.dev.platform_data = pdata; diff --git a/arch/arm/mach-davinci/include/mach/dm365.h b/arch/arm/mach-davinci/include/mach/dm365.h index 3c07a88..57e51eb 100644 --- a/arch/arm/mach-davinci/include/mach/dm365.h +++ b/arch/arm/mach-davinci/include/mach/dm365.h @@ -31,8 +31,13 @@ #define DM365_RTC_BASE (0x01C69000) +#define DAVINCI_DM365_VC_BASE (0x01D0C000) +#define DAVINCI_DMA_VC_TX 2 +#define DAVINCI_DMA_VC_RX 3 + void __init dm365_init(void); void __init dm365_init_asp(struct snd_platform_data *pdata); +void __init dm365_init_vc(struct snd_platform_data *pdata); void __init dm365_init_ks(struct davinci_ks_platform_data *pdata); void __init dm365_init_rtc(void); diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index 137bfba..2a68c1d 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -327,6 +327,8 @@ enum davinci_dm365_index { /* EDMA event muxing */ DM365_EVT2_ASP_TX, DM365_EVT3_ASP_RX, + DM365_EVT2_VC_TX, + DM365_EVT3_VC_RX, DM365_EVT26_MMC0_RX, }; -- 1.6.0.4

1 0

[alsa-devel] [PATCH v2 3/5] ASoC: DaVinci: CQ93VC Voice Codec
by miguel.aguilar＠ridgerun.com 11 Mar '10

11 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 298 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++++ 4 files changed, 333 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..0daca22 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..dff91fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..5132974 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,298 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_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; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + snd_soc_add_controls(codec, cq93vc_snd_controls, + ARRAY_SIZE(cq93vc_snd_controls)); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(davinci_vc->dev, "failed to register codec\n"); + goto fail1; + } + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail2; + } + return 0; + +fail2: + snd_soc_unregister_codec(codec); + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + snd_soc_unregister_codec(&codec); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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 + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif -- 1.6.0.4

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[alsa-devel] [PATCH 2/2 v2] ASoC: da7210: Add 11025/22050/44100/88200 rate support
by Kuninori Morimoto 11 Mar '10

11 Mar '10

This driver USE PLL for 11025/22050/44100/88200 rate. To enable switching to bypass mode, PLL is always turned on. Special thanks to Phil Signed-off-by: Phil Edworthy <Phil.Edworthy(a)renesas.com> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o Set symmetric_rates on da7210_dai o Add re-lock operation for DA7210_XXX_UNLOCK register sound/soc/codecs/da7210.c | 90 +++++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 86 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 3bd867d..35b8816 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -55,8 +55,14 @@ #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV1 0x29 +#define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_A_HID_UNLOCK 0x8A +#define DA7210_A_TEST_UNLOCK 0x8B +#define DA7210_A_PLL1 0x90 +#define DA7210_A_CP_MODE 0xA7 /* STARTUP1 bit fields */ #define DA7210_SC_MST_EN (1 << 0) @@ -124,13 +130,17 @@ /* PLL bit fields */ #define DA7210_PLL_FS_MASK (0xF << 0) #define DA7210_PLL_FS_8000 (0x1 << 0) +#define DA7210_PLL_FS_11025 (0x2 << 0) #define DA7210_PLL_FS_12000 (0x3 << 0) #define DA7210_PLL_FS_16000 (0x5 << 0) +#define DA7210_PLL_FS_22050 (0x6 << 0) #define DA7210_PLL_FS_24000 (0x7 << 0) #define DA7210_PLL_FS_32000 (0x9 << 0) +#define DA7210_PLL_FS_44100 (0xA << 0) #define DA7210_PLL_FS_48000 (0xB << 0) +#define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) - +#define DA7210_PLL_EN (0x1 << 7) #define DA7210_VERSION "0.0.1" @@ -249,7 +259,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u32 dai_cfg1; u32 hpf_reg, hpf_mask, hpf_value; - u32 fs; + u32 fs, bypass; /* set DAI source to Left and Right ADC */ da7210_write(codec, DA7210_DAI_SRC_SEL, @@ -281,38 +291,76 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, fs = DA7210_PLL_FS_8000; hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 11025: + fs = DA7210_PLL_FS_11025; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 22050: + fs = DA7210_PLL_FS_22050; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; hpf_mask = DA7210_VOICE_EN; hpf_value = 0; + bypass = DA7210_PLL_BYP; + break; + case 44100: + fs = DA7210_PLL_FS_44100; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; hpf_mask = DA7210_VOICE_EN; hpf_value = 0; + bypass = DA7210_PLL_BYP; + break; + case 88200: + fs = DA7210_PLL_FS_88200; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; hpf_mask = DA7210_VOICE_EN; hpf_value = 0; + bypass = DA7210_PLL_BYP; break; default: return -EINVAL; } + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, + DA7210_SC_MST_EN, DA7210_SC_MST_EN); return 0; } @@ -390,6 +438,7 @@ struct snd_soc_dai da7210_dai = { .formats = DA7210_FORMATS, }, .ops = &da7210_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(da7210_dai); @@ -444,7 +493,21 @@ static int da7210_init(struct da7210_priv *da7210) /* FIXME * * This driver use fixed value here + * And below settings expects MCLK = 12.288MHz + * + * When you select different MCLK, please check... + * DA7210_PLL_DIV1 val + * DA7210_PLL_DIV2 val + * DA7210_PLL_DIV3 val + * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx + */ + + /* + * make sure that DA7210 use bypass mode before start up */ + da7210_write(codec, DA7210_STARTUP1, 0); + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* * ADC settings @@ -482,9 +545,28 @@ static int da7210_init(struct da7210_priv *da7210) /* Diable PLL and bypass it */ da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* Bypass PLL and set MCLK freq rang to 10-20MHz */ - da7210_write(codec, DA7210_PLL_DIV3, + /* + * If 48kHz sound came, it use bypass mode, + * and when it is 44.1kHz, it use PLL. + * + * This time, this driver sets PLL always ON + * and controls bypass/PLL mode by switching + * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. + * see da7210_hw_params + */ + da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + da7210_write(codec, DA7210_PLL_DIV2, 0x99); + da7210_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + + /* As suggested by Dialog */ + da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ + da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + da7210_write(codec, DA7210_A_PLL1, 0x01); + da7210_write(codec, DA7210_A_CP_MODE, 0x7C); + da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ + da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); -- 1.6.3.3

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Re: [alsa-devel] underrun problem when short audio play
by ANISH KUMAR 11 Mar '10

11 Mar '10

Hi All, Just to add to my previous post. This problem of underrun is happening towards the end of the short audio. This causes unclear sound.This underrun is not happening with regular music (long duration audio). ALSA version:1.0.18a Regards, anish

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 10 Mar '10

10 Mar '10

The following changes since commit 57d54889cd00db2752994b389ba714138652e60c: Linus Torvalds (1): Linux 2.6.34-rc1 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Nothing terribly exciting in here yet, driver updates and minor things. The biggest block is the updats to the Samsung CPU drivers. Build testing has been limited by the fact that several of the machines I usually use fail to build with mainline at the minute (I have posted fixes) but the changes are fairly non-invasive. Barry Song (1): ASoC: bf5xx-sport: use common SPORT code for MMR info Chaithrika U S (1): ASoC: DaVinci: Add hw_param callback for S/PDIF DIT link Cliff Cai (1): ASoC: SSM2602: add SND control for mic boost2 and default it to off Jassi Brar (7): ASoC: Allow mulitple usage count of codec and cpu dai ASoC: S3C: I2Sv2: Add missing semicolon ASoC: S3C: I2Sv2: Unify clock source IDs ASoC: s3c-i2s-v2 remove unnecessary headers ASoC: s3c64xx-i2s remove unncessary headers ASoC: S3C64XX: I2S: Move RATE and FMT defines to header ASoC: S3C: I2Sv2: Reject immidiate register value Kuninori Morimoto (1): ASoC: da7210: Add 8/12/16/24/32/48/96 kHz rate support Mark Brown (9): ASoC: Wolfson Microelectronics 1133-EV1 audio support ASoC: Prettify wm8960 logging ASoC: Move WM8960 platform data into include/sound ASoC: Add support for WM8960 capless mode ASoC: Add delay information for Samsung IISv2 DAIs ASoC: Add 16/16 registers to soc-cache ASoC: Remove unused pmdown_time flag ASoC: Improve DAPM pop_wait delays Merge commit 'v2.6.34-rc1' into for-2.6.35 Peter Ujfalusi (3): ASoC: core: fix tailing whitespace in soc_pcm_apply_symmetry ASoC: core: soc level wrapper for pcm_pointer callback ASoC: core: Add delay operation to snd_soc_dai_ops jassi brar (3): ASoC: Pass dai_link as argument to platform suspend and resume ASoC: Change how suspend and resume obtain the PCM runtime ASoC: Remove runtime field from DAI include/sound/soc-dai.h | 7 +- include/sound/soc-dapm.h | 1 - include/sound/soc.h | 14 ++- include/sound/wm8960.h | 24 +++ sound/soc/atmel/atmel-pcm.c | 14 ++- sound/soc/blackfin/bf5xx-sport.h | 28 +---- sound/soc/codecs/da7210.c | 67 ++++++--- sound/soc/codecs/ssm2602.c | 4 +- sound/soc/codecs/wm8960.c | 209 +++++++++++++++++++++++---- sound/soc/codecs/wm8960.h | 10 -- sound/soc/davinci/davinci-evm.c | 16 ++- sound/soc/imx/Kconfig | 8 + sound/soc/imx/Makefile | 3 + sound/soc/imx/wm1133-ev1.c | 291 ++++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 93 +++++++------ sound/soc/s3c24xx/s3c-i2s-v2.h | 4 + sound/soc/s3c24xx/s3c2412-i2s.h | 4 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 12 -- sound/soc/s3c24xx/s3c64xx-i2s.h | 15 ++- sound/soc/soc-cache.c | 83 +++++++++++ sound/soc/soc-core.c | 76 ++++++++--- sound/soc/soc-dapm.c | 4 +- 22 files changed, 807 insertions(+), 180 deletions(-) create mode 100644 include/sound/wm8960.h create mode 100644 sound/soc/imx/wm1133-ev1.c

2 1

[alsa-devel] [PATCH v2 3/5] ASoC: DaVinci: CQ93VC Voice Codec
by miguel.aguilar＠ridgerun.com 10 Mar '10

10 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 298 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++++ 4 files changed, 333 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..0daca22 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..dff91fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..5132974 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,298 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_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; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + snd_soc_add_controls(codec, cq93vc_snd_controls, + ARRAY_SIZE(cq93vc_snd_controls)); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(davinci_vc->dev, "failed to register codec\n"); + goto fail1; + } + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail2; + } + return 0; + +fail2: + snd_soc_unregister_codec(codec); + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + snd_soc_unregister_codec(&codec); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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 + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif -- 1.6.0.4

2 1

[alsa-devel] [PATCH 4/4] mmp: enable sound in aspenite
by Haojian Zhuang 10 Mar '10

10 Mar '10

2 1

[alsa-devel] [PATCH 3/4] ASoC: enable ssp on aspenite
by Haojian Zhuang 10 Mar '10

10 Mar '10

2 1

[alsa-devel] [PATCH 1/4] pxa: move ssp into plat
by Haojian Zhuang 10 Mar '10

10 Mar '10

2 1

[alsa-devel] [PATCH 0/4] enable sound in pxa168
by Haojian Zhuang 10 Mar '10

10 Mar '10

These patches are used to enable sound in pxa168 SoC. The platform is aspenite. Since SSP clock is changed in PXA168, create a new pxa168-ssp.c file in sound/soc/pxa directory. 0001--ARM-pxa-move-ssp-into-plat.patch 0002--ARM-mmp-support-ssp-in-pxa168.patch 0003-ASoC-enable-ssp-on-aspenite.patch 0004--ARM-mmp-enable-sound-in-aspenite.patch Thanks Haojian

1 0

Re: [alsa-devel] [PATCH 1/4] From 75fe4b02234f6476e72827508f11f1035d7aaf68 Mon Sep 17 00:00:00 2001 From: Haojian Zhuang <haojian.zhuang@marvell.com> Date: Wed, 10 Mar 2010 06:32:43 -0500 Subject: [PATCH] [ARM] mmp: support ssp in pxa168 Support ssp in pxa168. Th
by Haojian Zhuang 10 Mar '10

10 Mar '10

Discard this patch. I'm sorry on it. Thanks Haojian On Wed, Mar 10, 2010 at 8:05 AM, Haojian Zhuang <haojian.zhuang(a)gmail.com> wrote: > From 75fe4b02234f6476e72827508f11f1035d7aaf68 Mon Sep 17 00:00:00 2001 > From: Haojian Zhuang <haojian.zhuang(a)marvell.com> > Date: Wed, 10 Mar 2010 06:32:43 -0500 > Subject: [PATCH] [ARM] mmp: support ssp in pxa168 > > Support ssp in pxa168. The basic function of SSP is same as pxa, but clock > source and IRQ is changed. > > Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> > --- > arch/arm/mach-mmp/include/mach/irqs.h | 10 +- > arch/arm/mach-mmp/include/mach/pxa168.h | 21 ++++ > arch/arm/mach-mmp/include/mach/regs-apbc.h | 10 +- > arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++++++ > arch/arm/mach-mmp/include/mach/regs-ssp.h | 141 ++++++++++++++++++++++++++++ > arch/arm/mach-mmp/pxa168.c | 15 +++ > arch/arm/plat-pxa/include/plat/ssp.h | 2 + > arch/arm/plat-pxa/ssp.c | 1 + > 8 files changed, 238 insertions(+), 10 deletions(-) > create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h > create mode 100644 arch/arm/mach-mmp/include/mach/regs-ssp.h > > diff --git a/arch/arm/mach-mmp/include/mach/irqs.h > b/arch/arm/mach-mmp/include/mach/irqs.h > index 0270119..b379cde 100644 > --- a/arch/arm/mach-mmp/include/mach/irqs.h > +++ b/arch/arm/mach-mmp/include/mach/irqs.h > @@ -5,10 +5,10 @@ > * Interrupt numbers for PXA168 > */ > #define IRQ_PXA168_NONE (-1) > -#define IRQ_PXA168_SSP3 0 > -#define IRQ_PXA168_SSP2 1 > -#define IRQ_PXA168_SSP1 2 > -#define IRQ_PXA168_SSP0 3 > +#define IRQ_PXA168_SSP4 0 > +#define IRQ_PXA168_SSP3 1 > +#define IRQ_PXA168_SSP2 2 > +#define IRQ_PXA168_SSP1 3 > #define IRQ_PXA168_PMIC_INT 4 > #define IRQ_PXA168_RTC_INT 5 > #define IRQ_PXA168_RTC_ALARM 6 > @@ -20,7 +20,7 @@ > #define IRQ_PXA168_TIMER2 14 > #define IRQ_PXA168_TIMER3 15 > #define IRQ_PXA168_CMU 16 > -#define IRQ_PXA168_SSP4 17 > +#define IRQ_PXA168_SSP5 17 > #define IRQ_PXA168_MSP_WAKEUP 19 > #define IRQ_PXA168_CF_WAKEUP 20 > #define IRQ_PXA168_XD_WAKEUP 21 > diff --git a/arch/arm/mach-mmp/include/mach/pxa168.h > b/arch/arm/mach-mmp/include/mach/pxa168.h > index 3ad612c..3b2bd5d 100644 > --- a/arch/arm/mach-mmp/include/mach/pxa168.h > +++ b/arch/arm/mach-mmp/include/mach/pxa168.h > @@ -14,6 +14,11 @@ extern struct pxa_device_desc pxa168_device_pwm1; > extern struct pxa_device_desc pxa168_device_pwm2; > extern struct pxa_device_desc pxa168_device_pwm3; > extern struct pxa_device_desc pxa168_device_pwm4; > +extern struct pxa_device_desc pxa168_device_ssp1; > +extern struct pxa_device_desc pxa168_device_ssp2; > +extern struct pxa_device_desc pxa168_device_ssp3; > +extern struct pxa_device_desc pxa168_device_ssp4; > +extern struct pxa_device_desc pxa168_device_ssp5; > extern struct pxa_device_desc pxa168_device_nand; > > static inline int pxa168_add_uart(int id) > @@ -67,6 +72,22 @@ static inline int pxa168_add_pwm(int id) > return pxa_register_device(d, NULL, 0); > } > > +static inline int pxa168_add_ssp(int id) > +{ > + struct pxa_device_desc *d = NULL; > + > + switch (id) { > + case 1: d = &pxa168_device_ssp1; break; > + case 2: d = &pxa168_device_ssp2; break; > + case 3: d = &pxa168_device_ssp3; break; > + case 4: d = &pxa168_device_ssp4; break; > + case 5: d = &pxa168_device_ssp5; break; > + default: > + return -EINVAL; > + } > + return pxa_register_device(d, NULL, 0); > +} > + > static inline int pxa168_add_nand(struct pxa3xx_nand_platform_data *info) > { > return pxa_register_device(&pxa168_device_nand, info, sizeof(*info)); > diff --git a/arch/arm/mach-mmp/include/mach/regs-apbc.h > b/arch/arm/mach-mmp/include/mach/regs-apbc.h > index 712af03..1a96585 100644 > --- a/arch/arm/mach-mmp/include/mach/regs-apbc.h > +++ b/arch/arm/mach-mmp/include/mach/regs-apbc.h > @@ -26,8 +26,6 @@ > #define APBC_PXA168_PWM2 APBC_REG(0x010) > #define APBC_PXA168_PWM3 APBC_REG(0x014) > #define APBC_PXA168_PWM4 APBC_REG(0x018) > -#define APBC_PXA168_SSP1 APBC_REG(0x01c) > -#define APBC_PXA168_SSP2 APBC_REG(0x020) > #define APBC_PXA168_RTC APBC_REG(0x028) > #define APBC_PXA168_TWSI0 APBC_REG(0x02c) > #define APBC_PXA168_KPC APBC_REG(0x030) > @@ -35,14 +33,16 @@ > #define APBC_PXA168_AIB APBC_REG(0x03c) > #define APBC_PXA168_SW_JTAG APBC_REG(0x040) > #define APBC_PXA168_ONEWIRE APBC_REG(0x048) > -#define APBC_PXA168_SSP3 APBC_REG(0x04c) > #define APBC_PXA168_ASFAR APBC_REG(0x050) > #define APBC_PXA168_ASSAR APBC_REG(0x054) > -#define APBC_PXA168_SSP4 APBC_REG(0x058) > -#define APBC_PXA168_SSP5 APBC_REG(0x05c) > #define APBC_PXA168_TWSI1 APBC_REG(0x06c) > #define APBC_PXA168_UART3 APBC_REG(0x070) > #define APBC_PXA168_AC97 APBC_REG(0x084) > +#define APBC_PXA168_SSP1 APBC_REG(0x81c) > +#define APBC_PXA168_SSP2 APBC_REG(0x820) > +#define APBC_PXA168_SSP3 APBC_REG(0x84c) > +#define APBC_PXA168_SSP4 APBC_REG(0x858) > +#define APBC_PXA168_SSP5 APBC_REG(0x85c) > > /* > * APB Clock register offsets for PXA910 > diff --git a/arch/arm/mach-mmp/include/mach/regs-mpmu.h > b/arch/arm/mach-mmp/include/mach/regs-mpmu.h > new file mode 100644 > index 0000000..0d57236 > --- /dev/null > +++ b/arch/arm/mach-mmp/include/mach/regs-mpmu.h > @@ -0,0 +1,48 @@ > +/* > + * linux/arch/arm/mach-mmp/include/mach/regs-mpmu.h > + * > + * Main Power Management Unit > + * > + * 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. > + */ > + > +#ifndef __ASM_MACH_REGS_MPMU_H > +#define __ASM_MACH_REGS_MPMU_H > + > +#include <mach/addr-map.h> > + > +#define MPMU_VIRT_BASE (APB_VIRT_BASE + 0x50000) > +#define MPMU_REG(off) (MPMU_VIRT_BASE + (off)) > + > +#define MPMU_CPCR MPMU_REG(0x0000) > +#define MPMU_FCCR MPMU_REG(0x0008) > +#define MPMU_POCR MPMU_REG(0x000c) > +#define MPMU_POSR MPMU_REG(0x0010) > +#define MPMU_SUCCR MPMU_REG(0x0014) > +#define MPMU_VRCR MPMU_REG(0x0018) > +#define MPMU_OHCR MPMU_REG(0x001c) > +#define MPMU_GPCR MPMU_REG(0x0030) > +#define MPMU_PLL2CR MPMU_REG(0x0034) > +#define MPMU_SCCR MPMU_REG(0x0038) > +#define MPMU_CWUCRM MPMU_REG(0x004c) > +#define MPMU_PLL1_REG1 MPMU_REG(0x0050) > +#define MPMU_PLL1_REG2 MPMU_REG(0x0054) > +#define MPMU_PLL1_SSC MPMU_REG(0x0058) > +#define MPMU_PLL2_REG1 MPMU_REG(0x0060) > +#define MPMU_PLL2_REG2 MPMU_REG(0x0064) > +#define MPMU_PLL2_SSC MPMU_REG(0x0068) > +#define MPMU_TS MPMU_REG(0x0080) > +#define MPMU_WDTPCR MPMU_REG(0x0200) > +#define MPMU_APCR MPMU_REG(0x1000) > +#define MPMU_APSR MPMU_REG(0x1004) > +#define MPMU_APRR MPMU_REG(0x1020) > +#define MPMU_ACGR MPMU_REG(0x1024) > +#define MPMU_ARSR MPMU_REG(0x1028) > +#define MPMU_AWUCRS MPMU_REG(0x1048) > +#define MPMU_AWUCRM MPMU_REG(0x104c) > +#define MPMU_ASYSDR MPMU_REG(0x1050) > +#define MPMU_ASSPDR MPMU_REG(0x1054) > + > +#endif /* __ASM_MACH_REGS_APMU_H */ > diff --git a/arch/arm/mach-mmp/include/mach/regs-ssp.h > b/arch/arm/mach-mmp/include/mach/regs-ssp.h > new file mode 100644 > index 0000000..7e16eeb > --- /dev/null > +++ b/arch/arm/mach-mmp/include/mach/regs-ssp.h > @@ -0,0 +1,141 @@ > +#ifndef __ASM_ARCH_REGS_SSP_H > +#define __ASM_ARCH_REGS_SSP_H > + > +/* > + * SSP Serial Port Registers > + */ > + > +#define SSCR0 (0x00) /* SSP Control Register 0 */ > +#define SSCR1 (0x04) /* SSP Control Register 1 */ > +#define SSSR (0x08) /* SSP Status Register */ > +#define SSITR (0x0C) /* SSP Interrupt Test Register */ > +#define SSDR (0x10) /* SSP Data Write/Data Read Register */ > + > +#define SSTO (0x28) /* SSP Time Out Register */ > +#define SSPSP (0x2C) /* SSP Programmable Serial Protocol */ > +#define SSTSA (0x30) /* SSP Tx Timeslot Active */ > +#define SSRSA (0x34) /* SSP Rx Timeslot Active */ > +#define SSTSS (0x38) /* SSP Timeslot Status */ > + > +#if defined(CONFIG_CPU_PXA910) > +#define SSACD (0x3C) /* SSP Audio Clock Divider */ > +#define SSACDD (0x40) /* SSP Audio Clock Dither Divider */ > +#endif > + > +/* Common bits first */ > +#define SSCR0_DSS_MASK (0x0000000f) /* Data Size Select (mask) */ > +#define SSCR0_DSS(x) ((x) - 1) /* Data Size Select [4..16] */ > +#define SSCR0_FRF (0x3 << 4) /* FRame Format (mask) */ > +#define SSCR0_Motorola (0x0 << 4) /* Motorola's Serial Peripheral > Interface (SPI) */ > +#define SSCR0_TI (0x1 << 4) /* Texas Instruments' Synchronous Serial > Protocol (SSP) */ > +#define SSCR0_National (0x2 << 4) /* National Microwire */ > +#define SSCR0_PSP (0x3 << 4) /* Programmable Serial Protocol (PSP) */ > +#define SSCR0_ECS (1 << 6) /* External clock select */ > +#define SSCR0_SSE (1 << 7) /* Synchronous Serial Port Enable */ > + > +#define SSCR0_SCR (0xfff << 8) /* Serial Clock Rate (mask) */ > +#define SSCR0_SerClkDiv(x) (((x) - 1) << 8) /* Divisor [1..4096] */ > + > +#define SSCR0_EDSS (1 << 20) /* Extended data size select */ > +#define SSCR0_NCS (1 << 21) /* Network clock select */ > +#define SSCR0_RIM (1 << 22) /* Receive FIFO overrrun interrupt mask */ > +#define SSCR0_TIM (1 << 23) /* Transmit FIFO underrun interrupt mask */ > +#define SSCR0_FRDC (0x7 << 24) /* Frame rate divider control (mask) */ > +#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24) /* Time slots per > frame [1..8] */ > +#define SSCR0_FPCKE (1 << 29) /* FIFO packing enable */ > +#define SSCR0_ACS (1 << 30) /* Audio clock select */ > +#define SSCR0_MOD (1 << 31) /* Mode (normal or network) */ > + > +#if defined(CONFIG_CPU_PXA168) || defined(CONFIG_CPU_PXA910) > +#define SSCR0_MODE_52M (1 << 27) /* 52Mbps Mode */ > +#endif > + > +#define SSCR1_RIE (1 << 0) /* Receive FIFO Interrupt Enable */ > +#define SSCR1_TIE (1 << 1) /* Transmit FIFO Interrupt Enable */ > +#define SSCR1_LBM (1 << 2) /* Loop-Back Mode */ > +#define SSCR1_SPO (1 << 3) /* Motorola SPI SSPSCLK polarity setting */ > +#define SSCR1_SPH (1 << 4) /* Motorola SPI SSPSCLK phase setting */ > +#define SSCR1_TFT_MASK (0xf << 6) /* Transmit FIFO Threshold (mask) */ > +#define SSCR1_TFT(x) (((x) - 1) << 6) /* level [1..16] */ > +#define SSCR1_RFT_MASK (0xf << 10) /* Receive FIFO Threshold (mask) */ > +#define SSCR1_RFT(x) (((x) - 1) << 10) /* level [1..16] */ > +#define SSCR1_EFWR (1 << 14) /* Enable FIFO Write/read */ > +#define SSCR1_STRF (1 << 15) /* Select FIFO for EFWR */ > +#define SSCR1_IFS (1 << 16) /* Invert Frame Signal */ > +#define SSCR1_PINTE (1 << 18) /* Peripheral Trailing Byte Interrupt Enable */ > +#define SSCR1_TINTE (1 << 19) /* Receiver Timeout Interrupt Enable */ > +#define SSCR1_RSRE (1 << 20) /* Receive Service Request Enable */ > +#define SSCR1_TSRE (1 << 21) /* Transmit Service Request Enable */ > +#define SSCR1_TRAIL (1 << 22) /* Trailing Byte */ > +#define SSCR1_RWOT (1 << 23) /* Receive Without Transmit */ > +#define SSCR1_SFRMDIR (1 << 24) /* SSP Frame (SSPSFRMx) Direction */ > +#define SSCR1_SCLKDIR (1 << 25) /* SSP Serial Bit Rate Clock > (SSPSCLKx) Direction */ > +#define SSCR1_ECRB (1 << 26) /* Enable Clock Request B */ > +#define SSCR1_ECRA (1 << 27) /* Enable Clock Request A */ > +#define SSCR1_SCFR (1 << 28) /* Slave Clock Free Running */ > +#define SSCR1_EBCEI (1 << 29) /* Enable Bit Count Error Interrupt */ > +#define SSCR1_TTE (1 << 30) /* TxD Three-state Enable */ > +#define SSCR1_TTELP (1 << 31) /* TxD Three-state Enable On Last Phase */ > + > +#if defined(CONFIG_CPU_PXA910) > +#define SSCR1_MWDS (1 << 5) /* Microwire Transmit Data Size */ > +#endif > + > +#define SSSR_TNF (1 << 2) /* Transmit FIFO Not Full */ > +#define SSSR_RNE (1 << 3) /* Receive FIFO Not Empty */ > +#define SSSR_BSY (1 << 4) /* SSP Busy */ > +#define SSSR_TFS (1 << 5) /* Transmit FIFO Service Request */ > +#define SSSR_RFS (1 << 6) /* Receive FIFO Service Request */ > +#define SSSR_ROR (1 << 7) /* Receive FIFO Overrun */ > +#define SSSR_TFL_MASK (0xf << 8) /* Transmit FIFO Level (mask) */ > +#define SSSR_TFL(x) ((x) << 8) > +#define SSSR_RFL_MASK (0xf << 12) /* Receive FIFO Level */ > +#define SSSR_RFL(x) ((x) << 12) > +#define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ > +#define SSSR_TINT (1 << 19) /* Receiver Timeout Interrupt */ > +#define SSSR_EOC (1 << 20) /* End of Chain */ > +#define SSSR_TUR (1 << 21) /* Transmit FIFO Underrun */ > +#define SSSR_CSS (1 << 22) /* Clock Synchronisation Status */ > +#define SSSR_BCE (1 << 23) /* Bit Count Error */ > +#define SSSR_TX_OSS (1 << 30) /* Tx FIFO Odd Sample Status */ > +#define SSSR_OSS (1 << 31) /* Odd Sample Status */ > + > +#define SSITR_TTFS (1 << 5) /* Test Tx FIFO Service Request */ > +#define SSITR_TRFS (1 << 6) /* Test Rx FIFO Service Request */ > +#define SSITR_TROR (1 << 7) /* Test Rx FIFO Overrun */ > + > +#define SSPSP_SCMODE_MASK (0x3) /* Serial Bit Rate Clock Mode */ > +#define SSPSP_SCMODE(x) ((x) << 0) > +#define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ > +#define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ > +#define SSPSP_STRTDLY_MASK (0x7 << 4) /* Start Delay */ > +#define SSPSP_STRTDLY(x) ((x) << 4) > +#define SSPSP_DMYSTRT_MASK (0x3 << 7) /* Dummy Start */ > +#define SSPSP_DMYSTRT(x) ((x) << 7) > +#define SSPSP_SFRMDLY_MASK (0x7f << 9) /* Serial Frame Delay */ > +#define SSPSP_SFRMDLY(x) ((x) << 9) > +#define SSPSP_SFRMWDTH_MASK (0x3f << 16) /* Serial Frame Width */ > +#define SSPSP_SFRMWDTH(x) ((x) << 16) > +#define SSPSP_DMYSTOP_MASK (0x3 << 23) /* Dummy Stop */ > +#define SSPSP_DMYSTOP(x) ((x) << 23) > +#define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ > +#define SSPSP_EDMYSTRT_MASK (0x3 << 26) /* Extended Dummy Start */ > +#define SSPSP_EDMYSTRT(x) ((x) << 26) > +#define SSPSP_EDMYSTOP_MASK (0x7 << 28) /* Extended Dummy Stop */ > +#define SSPSP_EDMYSTOP(x) ((x) << 28) > +#define SSPSP_TIMING_MASK (SSPSP_STRTDLY_MASK | SSPSP_DMYSTRT_MASK \ > + | SSPSP_SFRMDLY_MASK | SSPSP_SFRMWDTH_MASK \ > + | SSPSP_DMYSTOP_MASK | SSPSP_EDMYSTRT_MASK \ > + | SSPSP_EDMYSTOP_MASK | SSPSP_FSRT) > + > + > +#if defined(CONFIG_CPU_PXA910) > +#define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ > +#define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ > +#define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ > +#define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ > +#endif > + > + > +#endif /* __ASM_ARCH_REGS_SSP_H */ > + > diff --git a/arch/arm/mach-mmp/pxa168.c b/arch/arm/mach-mmp/pxa168.c > index 37dbdde..5f1a4dd 100644 > --- a/arch/arm/mach-mmp/pxa168.c > +++ b/arch/arm/mach-mmp/pxa168.c > @@ -72,6 +72,11 @@ static APBC_CLK(pwm1, PXA168_PWM1, 1, 13000000); > static APBC_CLK(pwm2, PXA168_PWM2, 1, 13000000); > static APBC_CLK(pwm3, PXA168_PWM3, 1, 13000000); > static APBC_CLK(pwm4, PXA168_PWM4, 1, 13000000); > +static APBC_CLK(ssp1, PXA168_SSP1, 4, 0); > +static APBC_CLK(ssp2, PXA168_SSP2, 4, 0); > +static APBC_CLK(ssp3, PXA168_SSP3, 4, 0); > +static APBC_CLK(ssp4, PXA168_SSP4, 4, 0); > +static APBC_CLK(ssp5, PXA168_SSP5, 4, 0); > > static APMU_CLK(nand, NAND, 0x01db, 208000000); > > @@ -85,6 +90,11 @@ static struct clk_lookup pxa168_clkregs[] = { > INIT_CLKREG(&clk_pwm2, "pxa168-pwm.1", NULL), > INIT_CLKREG(&clk_pwm3, "pxa168-pwm.2", NULL), > INIT_CLKREG(&clk_pwm4, "pxa168-pwm.3", NULL), > + INIT_CLKREG(&clk_ssp1, "pxa168-ssp.0", NULL), > + INIT_CLKREG(&clk_ssp2, "pxa168-ssp.1", NULL), > + INIT_CLKREG(&clk_ssp3, "pxa168-ssp.2", NULL), > + INIT_CLKREG(&clk_ssp4, "pxa168-ssp.3", NULL), > + INIT_CLKREG(&clk_ssp5, "pxa168-ssp.4", NULL), > INIT_CLKREG(&clk_nand, "pxa3xx-nand", NULL), > }; > > @@ -132,3 +142,8 @@ PXA168_DEVICE(pwm2, "pxa168-pwm", 1, NONE, > 0xd401a400, 0x10); > PXA168_DEVICE(pwm3, "pxa168-pwm", 2, NONE, 0xd401a800, 0x10); > PXA168_DEVICE(pwm4, "pxa168-pwm", 3, NONE, 0xd401ac00, 0x10); > PXA168_DEVICE(nand, "pxa3xx-nand", -1, NAND, 0xd4283000, 0x80, 97, 99); > +PXA168_DEVICE(ssp1, "pxa168-ssp", 0, SSP1, 0xd401b000, 0x40, 52, 53); > +PXA168_DEVICE(ssp2, "pxa168-ssp", 1, SSP2, 0xd401c000, 0x40, 54, 55); > +PXA168_DEVICE(ssp3, "pxa168-ssp", 2, SSP3, 0xd401f000, 0x40, 56, 57); > +PXA168_DEVICE(ssp4, "pxa168-ssp", 3, SSP4, 0xd4020000, 0x40, 58, 59); > +PXA168_DEVICE(ssp5, "pxa168-ssp", 4, SSP5, 0xd4021000, 0x40, 60, 61); > diff --git a/arch/arm/plat-pxa/include/plat/ssp.h > b/arch/arm/plat-pxa/include/plat/ssp.h > index fc83ba7..06cffae 100644 > --- a/arch/arm/plat-pxa/include/plat/ssp.h > +++ b/arch/arm/plat-pxa/include/plat/ssp.h > @@ -14,6 +14,7 @@ > * PXA26x SSP, NSSP, ASSP > * PXA27x SSP1, SSP2, SSP3 > * PXA3xx SSP1, SSP2, SSP3, SSP4 > + * PXA168 SSP1, SSP2, SSP3, SSP4, SSP5 > */ > > #ifndef __ASM_PLAT_SSP_H > @@ -27,6 +28,7 @@ enum pxa_ssp_type { > PXA25x_SSP, /* pxa 210, 250, 255, 26x */ > PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ > PXA27x_SSP, > + PXA168_SSP, > }; > > struct ssp_device { > diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c > index d4a7673..4dc2ee6 100644 > --- a/arch/arm/plat-pxa/ssp.c > +++ b/arch/arm/plat-pxa/ssp.c > @@ -465,6 +465,7 @@ static const struct platform_device_id ssp_id_table[] = { > { "pxa25x-ssp", PXA25x_SSP }, > { "pxa25x-nssp", PXA25x_NSSP }, > { "pxa27x-ssp", PXA27x_SSP }, > + { "pxa168-ssp", PXA168_SSP }, > { }, > }; > > -- > 1.5.6.5 >

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[alsa-devel] [PATCH 1/4] From 75fe4b02234f6476e72827508f11f1035d7aaf68 Mon Sep 17 00:00:00 2001 From: Haojian Zhuang <haojian.zhuang@marvell.com> Date: Wed, 10 Mar 2010 06:32:43 -0500 Subject: [PATCH] [ARM] mmp: support ssp in pxa168 Support ssp in pxa168. The ba
by Haojian Zhuang 10 Mar '10

10 Mar '10

1 0

[alsa-devel] [PATCH] hda-intel: Add position_fix quirk for ASUS M2V-MX SE.
by Paul Menzel 10 Mar '10

10 Mar '10

With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1]. Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE. The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker. $ lspci -vvnn | grep -A10 Audio 20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10) Subsystem: ASUSTeK Computer Inc. Device [1043:8290] 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 17 Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel [1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24… Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 051cf51..22dcdc2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2264,6 +2264,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), {} -- 1.6.6.1

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[alsa-devel] [PATCH] ASoC: bf5xx-sport: use common SPORT code for MMR info
by Mike Frysinger 10 Mar '10

10 Mar '10

From: Barry Song <barry.song(a)analog.com> No point in duplicating this structure layout in each driver. Signed-off-by: Barry Song <barry.song(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- this is fine for 2.6.34 or 2.6.35 sound/soc/blackfin/bf5xx-sport.h | 28 +--------------------------- 1 files changed, 1 insertions(+), 27 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h index 2e63dea..a86e8cc 100644 --- a/sound/soc/blackfin/bf5xx-sport.h +++ b/sound/soc/blackfin/bf5xx-sport.h @@ -34,33 +34,7 @@ #include <linux/wait.h> #include <linux/workqueue.h> #include <asm/dma.h> - -struct sport_register { - u16 tcr1; u16 reserved0; - u16 tcr2; u16 reserved1; - u16 tclkdiv; u16 reserved2; - u16 tfsdiv; u16 reserved3; - u32 tx; - u32 reserved_l0; - u32 rx; - u32 reserved_l1; - u16 rcr1; u16 reserved4; - u16 rcr2; u16 reserved5; - u16 rclkdiv; u16 reserved6; - u16 rfsdiv; u16 reserved7; - u16 stat; u16 reserved8; - u16 chnl; u16 reserved9; - u16 mcmc1; u16 reserved10; - u16 mcmc2; u16 reserved11; - u32 mtcs0; - u32 mtcs1; - u32 mtcs2; - u32 mtcs3; - u32 mrcs0; - u32 mrcs1; - u32 mrcs2; - u32 mrcs3; -}; +#include <asm/bfin_sport.h> #define DESC_ELEMENT_COUNT 9 -- 1.7.0.2

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[alsa-devel] [PATCH] ASoC: SSM2602: add SND control for mic boost2 and default it to off
by Mike Frysinger 10 Mar '10

10 Mar '10

From: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- this is fine for 2.6.34 or 2.6.35 sound/soc/codecs/ssm2602.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d2ff1cd..942f5dc 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -139,6 +139,7 @@ SOC_DOUBLE_R("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 31, 0), SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1), SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0), +SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 7, 1, 0), SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1), SOC_SINGLE("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1), @@ -604,8 +605,7 @@ static int ssm2602_init(struct snd_soc_device *socdev) reg = ssm2602_read_reg_cache(codec, SSM2602_ROUT1V); ssm2602_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH); /*select Line in as default input*/ - ssm2602_write(codec, SSM2602_APANA, - APANA_ENABLE_MIC_BOOST2 | APANA_SELECT_DAC | + ssm2602_write(codec, SSM2602_APANA, APANA_SELECT_DAC | APANA_ENABLE_MIC_BOOST); ssm2602_write(codec, SSM2602_PWR, 0); -- 1.7.0.2

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[alsa-devel] [PATCH] ASoC: DaVinci: Add hw_param callback for S/PDIF DIT link
by Chaithrika U S 10 Mar '10

10 Mar '10

On TI DM6467 EVM, S/PDIF DIT codec fails to open as it is unable to install hardware params. This dummy codec has no set_fmt and set_sysclk implementations and calls from the application to these functions cause errors. This patch adds a new hardware params callback function for S/PDIF transciever codec. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> Tested-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> --- This patch has been re-worked upon based on the review comments for http://mailman.alsa-project.org/pipermail/alsa-devel/2010-January/024621.ht… Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-evm.c | 16 +++++++++++++++- 1 files changed, 15 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7ccbe66..dba6651 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -81,10 +81,24 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return 0; } +static int evm_spdif_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 *cpu_dai = rtd->dai->cpu_dai; + + /* set cpu DAI configuration */ + return snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT); +} + static struct snd_soc_ops evm_ops = { .hw_params = evm_hw_params, }; +static struct snd_soc_ops evm_spdif_ops = { + .hw_params = evm_spdif_hw_params, +}; + /* davinci-evm machine dapm widgets */ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), @@ -165,7 +179,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .stream_name = "spdif", .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_DIT_DAI], .codec_dai = &dit_stub_dai, - .ops = &evm_ops, + .ops = &evm_spdif_ops, }, }; static struct snd_soc_dai_link da8xx_evm_dai = { -- 1.5.6

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[PATCH] [ARM] mmp: enable sound in aspenite
by Haojian Zhuang 10 Mar '10

10 Mar '10

Enable sound on aspenite. Hifi mode is supported. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/aspenite.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-mmp/aspenite.c b/arch/arm/mach-mmp/aspenite.c index a2d307e..244655d 100644 --- a/arch/arm/mach-mmp/aspenite.c +++ b/arch/arm/mach-mmp/aspenite.c @@ -59,6 +59,13 @@ static unsigned long common_pin_config[] __initdata = { /* UART1 */ GPIO107_UART1_RXD, GPIO108_UART1_TXD, + + /* SSP1 */ + GPIO113_I2S_MCLK, + GPIO114_I2S_FRM, + GPIO115_I2S_BCLK, + GPIO116_I2S_RXD, + GPIO117_I2S_TXD, }; static struct smc91x_platdata smc91x_info = { @@ -123,12 +130,18 @@ static struct pxa3xx_nand_platform_data aspenite_nand_info = { .nr_parts = ARRAY_SIZE(aspenite_nand_partitions), }; +static struct i2c_board_info aspenite_i2c_info[] __initdata = { + { I2C_BOARD_INFO("wm8753", 0x1b), }, +}; + static void __init common_init(void) { mfp_config(ARRAY_AND_SIZE(common_pin_config)); /* on-chip devices */ pxa168_add_uart(1); + pxa168_add_twsi(1, NULL, ARRAY_AND_SIZE(aspenite_i2c_info)); + pxa168_add_ssp(1); pxa168_add_nand(&aspenite_nand_info); /* off-chip devices */ -- 1.5.6.5

1 0

[PATCH] [ARM] mmp: support ssp in pxa168
by Haojian Zhuang 10 Mar '10

10 Mar '10

Support ssp in pxa168. The basic function of SSP is same as pxa, but clock source and IRQ is changed. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/include/mach/irqs.h | 10 +- arch/arm/mach-mmp/include/mach/pxa168.h | 21 ++++ arch/arm/mach-mmp/include/mach/regs-apbc.h | 10 +- arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++++++ arch/arm/mach-mmp/include/mach/regs-ssp.h | 141 ++++++++++++++++++++++++++++ arch/arm/mach-mmp/pxa168.c | 15 +++ arch/arm/plat-pxa/include/plat/ssp.h | 2 + arch/arm/plat-pxa/ssp.c | 1 + 8 files changed, 238 insertions(+), 10 deletions(-) create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h create mode 100644 arch/arm/mach-mmp/include/mach/regs-ssp.h diff --git a/arch/arm/mach-mmp/include/mach/irqs.h b/arch/arm/mach-mmp/include/mach/irqs.h index 0270119..b379cde 100644 --- a/arch/arm/mach-mmp/include/mach/irqs.h +++ b/arch/arm/mach-mmp/include/mach/irqs.h @@ -5,10 +5,10 @@ * Interrupt numbers for PXA168 */ #define IRQ_PXA168_NONE (-1) -#define IRQ_PXA168_SSP3 0 -#define IRQ_PXA168_SSP2 1 -#define IRQ_PXA168_SSP1 2 -#define IRQ_PXA168_SSP0 3 +#define IRQ_PXA168_SSP4 0 +#define IRQ_PXA168_SSP3 1 +#define IRQ_PXA168_SSP2 2 +#define IRQ_PXA168_SSP1 3 #define IRQ_PXA168_PMIC_INT 4 #define IRQ_PXA168_RTC_INT 5 #define IRQ_PXA168_RTC_ALARM 6 @@ -20,7 +20,7 @@ #define IRQ_PXA168_TIMER2 14 #define IRQ_PXA168_TIMER3 15 #define IRQ_PXA168_CMU 16 -#define IRQ_PXA168_SSP4 17 +#define IRQ_PXA168_SSP5 17 #define IRQ_PXA168_MSP_WAKEUP 19 #define IRQ_PXA168_CF_WAKEUP 20 #define IRQ_PXA168_XD_WAKEUP 21 diff --git a/arch/arm/mach-mmp/include/mach/pxa168.h b/arch/arm/mach-mmp/include/mach/pxa168.h index 3ad612c..3b2bd5d 100644 --- a/arch/arm/mach-mmp/include/mach/pxa168.h +++ b/arch/arm/mach-mmp/include/mach/pxa168.h @@ -14,6 +14,11 @@ extern struct pxa_device_desc pxa168_device_pwm1; extern struct pxa_device_desc pxa168_device_pwm2; extern struct pxa_device_desc pxa168_device_pwm3; extern struct pxa_device_desc pxa168_device_pwm4; +extern struct pxa_device_desc pxa168_device_ssp1; +extern struct pxa_device_desc pxa168_device_ssp2; +extern struct pxa_device_desc pxa168_device_ssp3; +extern struct pxa_device_desc pxa168_device_ssp4; +extern struct pxa_device_desc pxa168_device_ssp5; extern struct pxa_device_desc pxa168_device_nand; static inline int pxa168_add_uart(int id) @@ -67,6 +72,22 @@ static inline int pxa168_add_pwm(int id) return pxa_register_device(d, NULL, 0); } +static inline int pxa168_add_ssp(int id) +{ + struct pxa_device_desc *d = NULL; + + switch (id) { + case 1: d = &pxa168_device_ssp1; break; + case 2: d = &pxa168_device_ssp2; break; + case 3: d = &pxa168_device_ssp3; break; + case 4: d = &pxa168_device_ssp4; break; + case 5: d = &pxa168_device_ssp5; break; + default: + return -EINVAL; + } + return pxa_register_device(d, NULL, 0); +} + static inline int pxa168_add_nand(struct pxa3xx_nand_platform_data *info) { return pxa_register_device(&pxa168_device_nand, info, sizeof(*info)); diff --git a/arch/arm/mach-mmp/include/mach/regs-apbc.h b/arch/arm/mach-mmp/include/mach/regs-apbc.h index 712af03..1a96585 100644 --- a/arch/arm/mach-mmp/include/mach/regs-apbc.h +++ b/arch/arm/mach-mmp/include/mach/regs-apbc.h @@ -26,8 +26,6 @@ #define APBC_PXA168_PWM2 APBC_REG(0x010) #define APBC_PXA168_PWM3 APBC_REG(0x014) #define APBC_PXA168_PWM4 APBC_REG(0x018) -#define APBC_PXA168_SSP1 APBC_REG(0x01c) -#define APBC_PXA168_SSP2 APBC_REG(0x020) #define APBC_PXA168_RTC APBC_REG(0x028) #define APBC_PXA168_TWSI0 APBC_REG(0x02c) #define APBC_PXA168_KPC APBC_REG(0x030) @@ -35,14 +33,16 @@ #define APBC_PXA168_AIB APBC_REG(0x03c) #define APBC_PXA168_SW_JTAG APBC_REG(0x040) #define APBC_PXA168_ONEWIRE APBC_REG(0x048) -#define APBC_PXA168_SSP3 APBC_REG(0x04c) #define APBC_PXA168_ASFAR APBC_REG(0x050) #define APBC_PXA168_ASSAR APBC_REG(0x054) -#define APBC_PXA168_SSP4 APBC_REG(0x058) -#define APBC_PXA168_SSP5 APBC_REG(0x05c) #define APBC_PXA168_TWSI1 APBC_REG(0x06c) #define APBC_PXA168_UART3 APBC_REG(0x070) #define APBC_PXA168_AC97 APBC_REG(0x084) +#define APBC_PXA168_SSP1 APBC_REG(0x81c) +#define APBC_PXA168_SSP2 APBC_REG(0x820) +#define APBC_PXA168_SSP3 APBC_REG(0x84c) +#define APBC_PXA168_SSP4 APBC_REG(0x858) +#define APBC_PXA168_SSP5 APBC_REG(0x85c) /* * APB Clock register offsets for PXA910 diff --git a/arch/arm/mach-mmp/include/mach/regs-mpmu.h b/arch/arm/mach-mmp/include/mach/regs-mpmu.h new file mode 100644 index 0000000..0d57236 --- /dev/null +++ b/arch/arm/mach-mmp/include/mach/regs-mpmu.h @@ -0,0 +1,48 @@ +/* + * linux/arch/arm/mach-mmp/include/mach/regs-mpmu.h + * + * Main Power Management Unit + * + * 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. + */ + +#ifndef __ASM_MACH_REGS_MPMU_H +#define __ASM_MACH_REGS_MPMU_H + +#include <mach/addr-map.h> + +#define MPMU_VIRT_BASE (APB_VIRT_BASE + 0x50000) +#define MPMU_REG(off) (MPMU_VIRT_BASE + (off)) + +#define MPMU_CPCR MPMU_REG(0x0000) +#define MPMU_FCCR MPMU_REG(0x0008) +#define MPMU_POCR MPMU_REG(0x000c) +#define MPMU_POSR MPMU_REG(0x0010) +#define MPMU_SUCCR MPMU_REG(0x0014) +#define MPMU_VRCR MPMU_REG(0x0018) +#define MPMU_OHCR MPMU_REG(0x001c) +#define MPMU_GPCR MPMU_REG(0x0030) +#define MPMU_PLL2CR MPMU_REG(0x0034) +#define MPMU_SCCR MPMU_REG(0x0038) +#define MPMU_CWUCRM MPMU_REG(0x004c) +#define MPMU_PLL1_REG1 MPMU_REG(0x0050) +#define MPMU_PLL1_REG2 MPMU_REG(0x0054) +#define MPMU_PLL1_SSC MPMU_REG(0x0058) +#define MPMU_PLL2_REG1 MPMU_REG(0x0060) +#define MPMU_PLL2_REG2 MPMU_REG(0x0064) +#define MPMU_PLL2_SSC MPMU_REG(0x0068) +#define MPMU_TS MPMU_REG(0x0080) +#define MPMU_WDTPCR MPMU_REG(0x0200) +#define MPMU_APCR MPMU_REG(0x1000) +#define MPMU_APSR MPMU_REG(0x1004) +#define MPMU_APRR MPMU_REG(0x1020) +#define MPMU_ACGR MPMU_REG(0x1024) +#define MPMU_ARSR MPMU_REG(0x1028) +#define MPMU_AWUCRS MPMU_REG(0x1048) +#define MPMU_AWUCRM MPMU_REG(0x104c) +#define MPMU_ASYSDR MPMU_REG(0x1050) +#define MPMU_ASSPDR MPMU_REG(0x1054) + +#endif /* __ASM_MACH_REGS_APMU_H */ diff --git a/arch/arm/mach-mmp/include/mach/regs-ssp.h b/arch/arm/mach-mmp/include/mach/regs-ssp.h new file mode 100644 index 0000000..7e16eeb --- /dev/null +++ b/arch/arm/mach-mmp/include/mach/regs-ssp.h @@ -0,0 +1,141 @@ +#ifndef __ASM_ARCH_REGS_SSP_H +#define __ASM_ARCH_REGS_SSP_H + +/* + * SSP Serial Port Registers + */ + +#define SSCR0 (0x00) /* SSP Control Register 0 */ +#define SSCR1 (0x04) /* SSP Control Register 1 */ +#define SSSR (0x08) /* SSP Status Register */ +#define SSITR (0x0C) /* SSP Interrupt Test Register */ +#define SSDR (0x10) /* SSP Data Write/Data Read Register */ + +#define SSTO (0x28) /* SSP Time Out Register */ +#define SSPSP (0x2C) /* SSP Programmable Serial Protocol */ +#define SSTSA (0x30) /* SSP Tx Timeslot Active */ +#define SSRSA (0x34) /* SSP Rx Timeslot Active */ +#define SSTSS (0x38) /* SSP Timeslot Status */ + +#if defined(CONFIG_CPU_PXA910) +#define SSACD (0x3C) /* SSP Audio Clock Divider */ +#define SSACDD (0x40) /* SSP Audio Clock Dither Divider */ +#endif + +/* Common bits first */ +#define SSCR0_DSS_MASK (0x0000000f) /* Data Size Select (mask) */ +#define SSCR0_DSS(x) ((x) - 1) /* Data Size Select [4..16] */ +#define SSCR0_FRF (0x3 << 4) /* FRame Format (mask) */ +#define SSCR0_Motorola (0x0 << 4) /* Motorola's Serial Peripheral Interface (SPI) */ +#define SSCR0_TI (0x1 << 4) /* Texas Instruments' Synchronous Serial Protocol (SSP) */ +#define SSCR0_National (0x2 << 4) /* National Microwire */ +#define SSCR0_PSP (0x3 << 4) /* Programmable Serial Protocol (PSP) */ +#define SSCR0_ECS (1 << 6) /* External clock select */ +#define SSCR0_SSE (1 << 7) /* Synchronous Serial Port Enable */ + +#define SSCR0_SCR (0xfff << 8) /* Serial Clock Rate (mask) */ +#define SSCR0_SerClkDiv(x) (((x) - 1) << 8) /* Divisor [1..4096] */ + +#define SSCR0_EDSS (1 << 20) /* Extended data size select */ +#define SSCR0_NCS (1 << 21) /* Network clock select */ +#define SSCR0_RIM (1 << 22) /* Receive FIFO overrrun interrupt mask */ +#define SSCR0_TIM (1 << 23) /* Transmit FIFO underrun interrupt mask */ +#define SSCR0_FRDC (0x7 << 24) /* Frame rate divider control (mask) */ +#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24) /* Time slots per frame [1..8] */ +#define SSCR0_FPCKE (1 << 29) /* FIFO packing enable */ +#define SSCR0_ACS (1 << 30) /* Audio clock select */ +#define SSCR0_MOD (1 << 31) /* Mode (normal or network) */ + +#if defined(CONFIG_CPU_PXA168) || defined(CONFIG_CPU_PXA910) +#define SSCR0_MODE_52M (1 << 27) /* 52Mbps Mode */ +#endif + +#define SSCR1_RIE (1 << 0) /* Receive FIFO Interrupt Enable */ +#define SSCR1_TIE (1 << 1) /* Transmit FIFO Interrupt Enable */ +#define SSCR1_LBM (1 << 2) /* Loop-Back Mode */ +#define SSCR1_SPO (1 << 3) /* Motorola SPI SSPSCLK polarity setting */ +#define SSCR1_SPH (1 << 4) /* Motorola SPI SSPSCLK phase setting */ +#define SSCR1_TFT_MASK (0xf << 6) /* Transmit FIFO Threshold (mask) */ +#define SSCR1_TFT(x) (((x) - 1) << 6) /* level [1..16] */ +#define SSCR1_RFT_MASK (0xf << 10) /* Receive FIFO Threshold (mask) */ +#define SSCR1_RFT(x) (((x) - 1) << 10) /* level [1..16] */ +#define SSCR1_EFWR (1 << 14) /* Enable FIFO Write/read */ +#define SSCR1_STRF (1 << 15) /* Select FIFO for EFWR */ +#define SSCR1_IFS (1 << 16) /* Invert Frame Signal */ +#define SSCR1_PINTE (1 << 18) /* Peripheral Trailing Byte Interrupt Enable */ +#define SSCR1_TINTE (1 << 19) /* Receiver Timeout Interrupt Enable */ +#define SSCR1_RSRE (1 << 20) /* Receive Service Request Enable */ +#define SSCR1_TSRE (1 << 21) /* Transmit Service Request Enable */ +#define SSCR1_TRAIL (1 << 22) /* Trailing Byte */ +#define SSCR1_RWOT (1 << 23) /* Receive Without Transmit */ +#define SSCR1_SFRMDIR (1 << 24) /* SSP Frame (SSPSFRMx) Direction */ +#define SSCR1_SCLKDIR (1 << 25) /* SSP Serial Bit Rate Clock (SSPSCLKx) Direction */ +#define SSCR1_ECRB (1 << 26) /* Enable Clock Request B */ +#define SSCR1_ECRA (1 << 27) /* Enable Clock Request A */ +#define SSCR1_SCFR (1 << 28) /* Slave Clock Free Running */ +#define SSCR1_EBCEI (1 << 29) /* Enable Bit Count Error Interrupt */ +#define SSCR1_TTE (1 << 30) /* TxD Three-state Enable */ +#define SSCR1_TTELP (1 << 31) /* TxD Three-state Enable On Last Phase */ + +#if defined(CONFIG_CPU_PXA910) +#define SSCR1_MWDS (1 << 5) /* Microwire Transmit Data Size */ +#endif + +#define SSSR_TNF (1 << 2) /* Transmit FIFO Not Full */ +#define SSSR_RNE (1 << 3) /* Receive FIFO Not Empty */ +#define SSSR_BSY (1 << 4) /* SSP Busy */ +#define SSSR_TFS (1 << 5) /* Transmit FIFO Service Request */ +#define SSSR_RFS (1 << 6) /* Receive FIFO Service Request */ +#define SSSR_ROR (1 << 7) /* Receive FIFO Overrun */ +#define SSSR_TFL_MASK (0xf << 8) /* Transmit FIFO Level (mask) */ +#define SSSR_TFL(x) ((x) << 8) +#define SSSR_RFL_MASK (0xf << 12) /* Receive FIFO Level */ +#define SSSR_RFL(x) ((x) << 12) +#define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ +#define SSSR_TINT (1 << 19) /* Receiver Timeout Interrupt */ +#define SSSR_EOC (1 << 20) /* End of Chain */ +#define SSSR_TUR (1 << 21) /* Transmit FIFO Underrun */ +#define SSSR_CSS (1 << 22) /* Clock Synchronisation Status */ +#define SSSR_BCE (1 << 23) /* Bit Count Error */ +#define SSSR_TX_OSS (1 << 30) /* Tx FIFO Odd Sample Status */ +#define SSSR_OSS (1 << 31) /* Odd Sample Status */ + +#define SSITR_TTFS (1 << 5) /* Test Tx FIFO Service Request */ +#define SSITR_TRFS (1 << 6) /* Test Rx FIFO Service Request */ +#define SSITR_TROR (1 << 7) /* Test Rx FIFO Overrun */ + +#define SSPSP_SCMODE_MASK (0x3) /* Serial Bit Rate Clock Mode */ +#define SSPSP_SCMODE(x) ((x) << 0) +#define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ +#define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ +#define SSPSP_STRTDLY_MASK (0x7 << 4) /* Start Delay */ +#define SSPSP_STRTDLY(x) ((x) << 4) +#define SSPSP_DMYSTRT_MASK (0x3 << 7) /* Dummy Start */ +#define SSPSP_DMYSTRT(x) ((x) << 7) +#define SSPSP_SFRMDLY_MASK (0x7f << 9) /* Serial Frame Delay */ +#define SSPSP_SFRMDLY(x) ((x) << 9) +#define SSPSP_SFRMWDTH_MASK (0x3f << 16) /* Serial Frame Width */ +#define SSPSP_SFRMWDTH(x) ((x) << 16) +#define SSPSP_DMYSTOP_MASK (0x3 << 23) /* Dummy Stop */ +#define SSPSP_DMYSTOP(x) ((x) << 23) +#define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ +#define SSPSP_EDMYSTRT_MASK (0x3 << 26) /* Extended Dummy Start */ +#define SSPSP_EDMYSTRT(x) ((x) << 26) +#define SSPSP_EDMYSTOP_MASK (0x7 << 28) /* Extended Dummy Stop */ +#define SSPSP_EDMYSTOP(x) ((x) << 28) +#define SSPSP_TIMING_MASK (SSPSP_STRTDLY_MASK | SSPSP_DMYSTRT_MASK \ + | SSPSP_SFRMDLY_MASK | SSPSP_SFRMWDTH_MASK \ + | SSPSP_DMYSTOP_MASK | SSPSP_EDMYSTRT_MASK \ + | SSPSP_EDMYSTOP_MASK | SSPSP_FSRT) + + +#if defined(CONFIG_CPU_PXA910) +#define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ +#define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ +#define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ +#define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ +#endif + + +#endif /* __ASM_ARCH_REGS_SSP_H */ + diff --git a/arch/arm/mach-mmp/pxa168.c b/arch/arm/mach-mmp/pxa168.c index 37dbdde..5f1a4dd 100644 --- a/arch/arm/mach-mmp/pxa168.c +++ b/arch/arm/mach-mmp/pxa168.c @@ -72,6 +72,11 @@ static APBC_CLK(pwm1, PXA168_PWM1, 1, 13000000); static APBC_CLK(pwm2, PXA168_PWM2, 1, 13000000); static APBC_CLK(pwm3, PXA168_PWM3, 1, 13000000); static APBC_CLK(pwm4, PXA168_PWM4, 1, 13000000); +static APBC_CLK(ssp1, PXA168_SSP1, 4, 0); +static APBC_CLK(ssp2, PXA168_SSP2, 4, 0); +static APBC_CLK(ssp3, PXA168_SSP3, 4, 0); +static APBC_CLK(ssp4, PXA168_SSP4, 4, 0); +static APBC_CLK(ssp5, PXA168_SSP5, 4, 0); static APMU_CLK(nand, NAND, 0x01db, 208000000); @@ -85,6 +90,11 @@ static struct clk_lookup pxa168_clkregs[] = { INIT_CLKREG(&clk_pwm2, "pxa168-pwm.1", NULL), INIT_CLKREG(&clk_pwm3, "pxa168-pwm.2", NULL), INIT_CLKREG(&clk_pwm4, "pxa168-pwm.3", NULL), + INIT_CLKREG(&clk_ssp1, "pxa168-ssp.0", NULL), + INIT_CLKREG(&clk_ssp2, "pxa168-ssp.1", NULL), + INIT_CLKREG(&clk_ssp3, "pxa168-ssp.2", NULL), + INIT_CLKREG(&clk_ssp4, "pxa168-ssp.3", NULL), + INIT_CLKREG(&clk_ssp5, "pxa168-ssp.4", NULL), INIT_CLKREG(&clk_nand, "pxa3xx-nand", NULL), }; @@ -132,3 +142,8 @@ PXA168_DEVICE(pwm2, "pxa168-pwm", 1, NONE, 0xd401a400, 0x10); PXA168_DEVICE(pwm3, "pxa168-pwm", 2, NONE, 0xd401a800, 0x10); PXA168_DEVICE(pwm4, "pxa168-pwm", 3, NONE, 0xd401ac00, 0x10); PXA168_DEVICE(nand, "pxa3xx-nand", -1, NAND, 0xd4283000, 0x80, 97, 99); +PXA168_DEVICE(ssp1, "pxa168-ssp", 0, SSP1, 0xd401b000, 0x40, 52, 53); +PXA168_DEVICE(ssp2, "pxa168-ssp", 1, SSP2, 0xd401c000, 0x40, 54, 55); +PXA168_DEVICE(ssp3, "pxa168-ssp", 2, SSP3, 0xd401f000, 0x40, 56, 57); +PXA168_DEVICE(ssp4, "pxa168-ssp", 3, SSP4, 0xd4020000, 0x40, 58, 59); +PXA168_DEVICE(ssp5, "pxa168-ssp", 4, SSP5, 0xd4021000, 0x40, 60, 61); diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h index fc83ba7..06cffae 100644 --- a/arch/arm/plat-pxa/include/plat/ssp.h +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -14,6 +14,7 @@ * PXA26x SSP, NSSP, ASSP * PXA27x SSP1, SSP2, SSP3 * PXA3xx SSP1, SSP2, SSP3, SSP4 + * PXA168 SSP1, SSP2, SSP3, SSP4, SSP5 */ #ifndef __ASM_PLAT_SSP_H @@ -27,6 +28,7 @@ enum pxa_ssp_type { PXA25x_SSP, /* pxa 210, 250, 255, 26x */ PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ PXA27x_SSP, + PXA168_SSP, }; struct ssp_device { diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index d4a7673..4dc2ee6 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -465,6 +465,7 @@ static const struct platform_device_id ssp_id_table[] = { { "pxa25x-ssp", PXA25x_SSP }, { "pxa25x-nssp", PXA25x_NSSP }, { "pxa27x-ssp", PXA27x_SSP }, + { "pxa168-ssp", PXA168_SSP }, { }, }; -- 1.5.6.5

1 0

[PATCH] [ARM] mmp: support ssp in pxa168
by Haojian Zhuang 10 Mar '10

10 Mar '10

Support ssp in pxa168. The basic function of SSP is same as pxa, but clock source and IRQ is changed. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/include/mach/irqs.h | 10 +- arch/arm/mach-mmp/include/mach/pxa168.h | 21 ++++ arch/arm/mach-mmp/include/mach/regs-apbc.h | 10 +- arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++++++ arch/arm/mach-mmp/include/mach/regs-ssp.h | 141 ++++++++++++++++++++++++++++ arch/arm/mach-mmp/pxa168.c | 15 +++ arch/arm/plat-pxa/include/plat/ssp.h | 2 + arch/arm/plat-pxa/ssp.c | 1 + 8 files changed, 238 insertions(+), 10 deletions(-) create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h create mode 100644 arch/arm/mach-mmp/include/mach/regs-ssp.h diff --git a/arch/arm/mach-mmp/include/mach/irqs.h b/arch/arm/mach-mmp/include/mach/irqs.h index 0270119..b379cde 100644 --- a/arch/arm/mach-mmp/include/mach/irqs.h +++ b/arch/arm/mach-mmp/include/mach/irqs.h @@ -5,10 +5,10 @@ * Interrupt numbers for PXA168 */ #define IRQ_PXA168_NONE (-1) -#define IRQ_PXA168_SSP3 0 -#define IRQ_PXA168_SSP2 1 -#define IRQ_PXA168_SSP1 2 -#define IRQ_PXA168_SSP0 3 +#define IRQ_PXA168_SSP4 0 +#define IRQ_PXA168_SSP3 1 +#define IRQ_PXA168_SSP2 2 +#define IRQ_PXA168_SSP1 3 #define IRQ_PXA168_PMIC_INT 4 #define IRQ_PXA168_RTC_INT 5 #define IRQ_PXA168_RTC_ALARM 6 @@ -20,7 +20,7 @@ #define IRQ_PXA168_TIMER2 14 #define IRQ_PXA168_TIMER3 15 #define IRQ_PXA168_CMU 16 -#define IRQ_PXA168_SSP4 17 +#define IRQ_PXA168_SSP5 17 #define IRQ_PXA168_MSP_WAKEUP 19 #define IRQ_PXA168_CF_WAKEUP 20 #define IRQ_PXA168_XD_WAKEUP 21 diff --git a/arch/arm/mach-mmp/include/mach/pxa168.h b/arch/arm/mach-mmp/include/mach/pxa168.h index 3ad612c..3b2bd5d 100644 --- a/arch/arm/mach-mmp/include/mach/pxa168.h +++ b/arch/arm/mach-mmp/include/mach/pxa168.h @@ -14,6 +14,11 @@ extern struct pxa_device_desc pxa168_device_pwm1; extern struct pxa_device_desc pxa168_device_pwm2; extern struct pxa_device_desc pxa168_device_pwm3; extern struct pxa_device_desc pxa168_device_pwm4; +extern struct pxa_device_desc pxa168_device_ssp1; +extern struct pxa_device_desc pxa168_device_ssp2; +extern struct pxa_device_desc pxa168_device_ssp3; +extern struct pxa_device_desc pxa168_device_ssp4; +extern struct pxa_device_desc pxa168_device_ssp5; extern struct pxa_device_desc pxa168_device_nand; static inline int pxa168_add_uart(int id) @@ -67,6 +72,22 @@ static inline int pxa168_add_pwm(int id) return pxa_register_device(d, NULL, 0); } +static inline int pxa168_add_ssp(int id) +{ + struct pxa_device_desc *d = NULL; + + switch (id) { + case 1: d = &pxa168_device_ssp1; break; + case 2: d = &pxa168_device_ssp2; break; + case 3: d = &pxa168_device_ssp3; break; + case 4: d = &pxa168_device_ssp4; break; + case 5: d = &pxa168_device_ssp5; break; + default: + return -EINVAL; + } + return pxa_register_device(d, NULL, 0); +} + static inline int pxa168_add_nand(struct pxa3xx_nand_platform_data *info) { return pxa_register_device(&pxa168_device_nand, info, sizeof(*info)); diff --git a/arch/arm/mach-mmp/include/mach/regs-apbc.h b/arch/arm/mach-mmp/include/mach/regs-apbc.h index 712af03..1a96585 100644 --- a/arch/arm/mach-mmp/include/mach/regs-apbc.h +++ b/arch/arm/mach-mmp/include/mach/regs-apbc.h @@ -26,8 +26,6 @@ #define APBC_PXA168_PWM2 APBC_REG(0x010) #define APBC_PXA168_PWM3 APBC_REG(0x014) #define APBC_PXA168_PWM4 APBC_REG(0x018) -#define APBC_PXA168_SSP1 APBC_REG(0x01c) -#define APBC_PXA168_SSP2 APBC_REG(0x020) #define APBC_PXA168_RTC APBC_REG(0x028) #define APBC_PXA168_TWSI0 APBC_REG(0x02c) #define APBC_PXA168_KPC APBC_REG(0x030) @@ -35,14 +33,16 @@ #define APBC_PXA168_AIB APBC_REG(0x03c) #define APBC_PXA168_SW_JTAG APBC_REG(0x040) #define APBC_PXA168_ONEWIRE APBC_REG(0x048) -#define APBC_PXA168_SSP3 APBC_REG(0x04c) #define APBC_PXA168_ASFAR APBC_REG(0x050) #define APBC_PXA168_ASSAR APBC_REG(0x054) -#define APBC_PXA168_SSP4 APBC_REG(0x058) -#define APBC_PXA168_SSP5 APBC_REG(0x05c) #define APBC_PXA168_TWSI1 APBC_REG(0x06c) #define APBC_PXA168_UART3 APBC_REG(0x070) #define APBC_PXA168_AC97 APBC_REG(0x084) +#define APBC_PXA168_SSP1 APBC_REG(0x81c) +#define APBC_PXA168_SSP2 APBC_REG(0x820) +#define APBC_PXA168_SSP3 APBC_REG(0x84c) +#define APBC_PXA168_SSP4 APBC_REG(0x858) +#define APBC_PXA168_SSP5 APBC_REG(0x85c) /* * APB Clock register offsets for PXA910 diff --git a/arch/arm/mach-mmp/include/mach/regs-mpmu.h b/arch/arm/mach-mmp/include/mach/regs-mpmu.h new file mode 100644 index 0000000..0d57236 --- /dev/null +++ b/arch/arm/mach-mmp/include/mach/regs-mpmu.h @@ -0,0 +1,48 @@ +/* + * linux/arch/arm/mach-mmp/include/mach/regs-mpmu.h + * + * Main Power Management Unit + * + * 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. + */ + +#ifndef __ASM_MACH_REGS_MPMU_H +#define __ASM_MACH_REGS_MPMU_H + +#include <mach/addr-map.h> + +#define MPMU_VIRT_BASE (APB_VIRT_BASE + 0x50000) +#define MPMU_REG(off) (MPMU_VIRT_BASE + (off)) + +#define MPMU_CPCR MPMU_REG(0x0000) +#define MPMU_FCCR MPMU_REG(0x0008) +#define MPMU_POCR MPMU_REG(0x000c) +#define MPMU_POSR MPMU_REG(0x0010) +#define MPMU_SUCCR MPMU_REG(0x0014) +#define MPMU_VRCR MPMU_REG(0x0018) +#define MPMU_OHCR MPMU_REG(0x001c) +#define MPMU_GPCR MPMU_REG(0x0030) +#define MPMU_PLL2CR MPMU_REG(0x0034) +#define MPMU_SCCR MPMU_REG(0x0038) +#define MPMU_CWUCRM MPMU_REG(0x004c) +#define MPMU_PLL1_REG1 MPMU_REG(0x0050) +#define MPMU_PLL1_REG2 MPMU_REG(0x0054) +#define MPMU_PLL1_SSC MPMU_REG(0x0058) +#define MPMU_PLL2_REG1 MPMU_REG(0x0060) +#define MPMU_PLL2_REG2 MPMU_REG(0x0064) +#define MPMU_PLL2_SSC MPMU_REG(0x0068) +#define MPMU_TS MPMU_REG(0x0080) +#define MPMU_WDTPCR MPMU_REG(0x0200) +#define MPMU_APCR MPMU_REG(0x1000) +#define MPMU_APSR MPMU_REG(0x1004) +#define MPMU_APRR MPMU_REG(0x1020) +#define MPMU_ACGR MPMU_REG(0x1024) +#define MPMU_ARSR MPMU_REG(0x1028) +#define MPMU_AWUCRS MPMU_REG(0x1048) +#define MPMU_AWUCRM MPMU_REG(0x104c) +#define MPMU_ASYSDR MPMU_REG(0x1050) +#define MPMU_ASSPDR MPMU_REG(0x1054) + +#endif /* __ASM_MACH_REGS_APMU_H */ diff --git a/arch/arm/mach-mmp/include/mach/regs-ssp.h b/arch/arm/mach-mmp/include/mach/regs-ssp.h new file mode 100644 index 0000000..7e16eeb --- /dev/null +++ b/arch/arm/mach-mmp/include/mach/regs-ssp.h @@ -0,0 +1,141 @@ +#ifndef __ASM_ARCH_REGS_SSP_H +#define __ASM_ARCH_REGS_SSP_H + +/* + * SSP Serial Port Registers + */ + +#define SSCR0 (0x00) /* SSP Control Register 0 */ +#define SSCR1 (0x04) /* SSP Control Register 1 */ +#define SSSR (0x08) /* SSP Status Register */ +#define SSITR (0x0C) /* SSP Interrupt Test Register */ +#define SSDR (0x10) /* SSP Data Write/Data Read Register */ + +#define SSTO (0x28) /* SSP Time Out Register */ +#define SSPSP (0x2C) /* SSP Programmable Serial Protocol */ +#define SSTSA (0x30) /* SSP Tx Timeslot Active */ +#define SSRSA (0x34) /* SSP Rx Timeslot Active */ +#define SSTSS (0x38) /* SSP Timeslot Status */ + +#if defined(CONFIG_CPU_PXA910) +#define SSACD (0x3C) /* SSP Audio Clock Divider */ +#define SSACDD (0x40) /* SSP Audio Clock Dither Divider */ +#endif + +/* Common bits first */ +#define SSCR0_DSS_MASK (0x0000000f) /* Data Size Select (mask) */ +#define SSCR0_DSS(x) ((x) - 1) /* Data Size Select [4..16] */ +#define SSCR0_FRF (0x3 << 4) /* FRame Format (mask) */ +#define SSCR0_Motorola (0x0 << 4) /* Motorola's Serial Peripheral Interface (SPI) */ +#define SSCR0_TI (0x1 << 4) /* Texas Instruments' Synchronous Serial Protocol (SSP) */ +#define SSCR0_National (0x2 << 4) /* National Microwire */ +#define SSCR0_PSP (0x3 << 4) /* Programmable Serial Protocol (PSP) */ +#define SSCR0_ECS (1 << 6) /* External clock select */ +#define SSCR0_SSE (1 << 7) /* Synchronous Serial Port Enable */ + +#define SSCR0_SCR (0xfff << 8) /* Serial Clock Rate (mask) */ +#define SSCR0_SerClkDiv(x) (((x) - 1) << 8) /* Divisor [1..4096] */ + +#define SSCR0_EDSS (1 << 20) /* Extended data size select */ +#define SSCR0_NCS (1 << 21) /* Network clock select */ +#define SSCR0_RIM (1 << 22) /* Receive FIFO overrrun interrupt mask */ +#define SSCR0_TIM (1 << 23) /* Transmit FIFO underrun interrupt mask */ +#define SSCR0_FRDC (0x7 << 24) /* Frame rate divider control (mask) */ +#define SSCR0_SlotsPerFrm(x) (((x) - 1) << 24) /* Time slots per frame [1..8] */ +#define SSCR0_FPCKE (1 << 29) /* FIFO packing enable */ +#define SSCR0_ACS (1 << 30) /* Audio clock select */ +#define SSCR0_MOD (1 << 31) /* Mode (normal or network) */ + +#if defined(CONFIG_CPU_PXA168) || defined(CONFIG_CPU_PXA910) +#define SSCR0_MODE_52M (1 << 27) /* 52Mbps Mode */ +#endif + +#define SSCR1_RIE (1 << 0) /* Receive FIFO Interrupt Enable */ +#define SSCR1_TIE (1 << 1) /* Transmit FIFO Interrupt Enable */ +#define SSCR1_LBM (1 << 2) /* Loop-Back Mode */ +#define SSCR1_SPO (1 << 3) /* Motorola SPI SSPSCLK polarity setting */ +#define SSCR1_SPH (1 << 4) /* Motorola SPI SSPSCLK phase setting */ +#define SSCR1_TFT_MASK (0xf << 6) /* Transmit FIFO Threshold (mask) */ +#define SSCR1_TFT(x) (((x) - 1) << 6) /* level [1..16] */ +#define SSCR1_RFT_MASK (0xf << 10) /* Receive FIFO Threshold (mask) */ +#define SSCR1_RFT(x) (((x) - 1) << 10) /* level [1..16] */ +#define SSCR1_EFWR (1 << 14) /* Enable FIFO Write/read */ +#define SSCR1_STRF (1 << 15) /* Select FIFO for EFWR */ +#define SSCR1_IFS (1 << 16) /* Invert Frame Signal */ +#define SSCR1_PINTE (1 << 18) /* Peripheral Trailing Byte Interrupt Enable */ +#define SSCR1_TINTE (1 << 19) /* Receiver Timeout Interrupt Enable */ +#define SSCR1_RSRE (1 << 20) /* Receive Service Request Enable */ +#define SSCR1_TSRE (1 << 21) /* Transmit Service Request Enable */ +#define SSCR1_TRAIL (1 << 22) /* Trailing Byte */ +#define SSCR1_RWOT (1 << 23) /* Receive Without Transmit */ +#define SSCR1_SFRMDIR (1 << 24) /* SSP Frame (SSPSFRMx) Direction */ +#define SSCR1_SCLKDIR (1 << 25) /* SSP Serial Bit Rate Clock (SSPSCLKx) Direction */ +#define SSCR1_ECRB (1 << 26) /* Enable Clock Request B */ +#define SSCR1_ECRA (1 << 27) /* Enable Clock Request A */ +#define SSCR1_SCFR (1 << 28) /* Slave Clock Free Running */ +#define SSCR1_EBCEI (1 << 29) /* Enable Bit Count Error Interrupt */ +#define SSCR1_TTE (1 << 30) /* TxD Three-state Enable */ +#define SSCR1_TTELP (1 << 31) /* TxD Three-state Enable On Last Phase */ + +#if defined(CONFIG_CPU_PXA910) +#define SSCR1_MWDS (1 << 5) /* Microwire Transmit Data Size */ +#endif + +#define SSSR_TNF (1 << 2) /* Transmit FIFO Not Full */ +#define SSSR_RNE (1 << 3) /* Receive FIFO Not Empty */ +#define SSSR_BSY (1 << 4) /* SSP Busy */ +#define SSSR_TFS (1 << 5) /* Transmit FIFO Service Request */ +#define SSSR_RFS (1 << 6) /* Receive FIFO Service Request */ +#define SSSR_ROR (1 << 7) /* Receive FIFO Overrun */ +#define SSSR_TFL_MASK (0xf << 8) /* Transmit FIFO Level (mask) */ +#define SSSR_TFL(x) ((x) << 8) +#define SSSR_RFL_MASK (0xf << 12) /* Receive FIFO Level */ +#define SSSR_RFL(x) ((x) << 12) +#define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ +#define SSSR_TINT (1 << 19) /* Receiver Timeout Interrupt */ +#define SSSR_EOC (1 << 20) /* End of Chain */ +#define SSSR_TUR (1 << 21) /* Transmit FIFO Underrun */ +#define SSSR_CSS (1 << 22) /* Clock Synchronisation Status */ +#define SSSR_BCE (1 << 23) /* Bit Count Error */ +#define SSSR_TX_OSS (1 << 30) /* Tx FIFO Odd Sample Status */ +#define SSSR_OSS (1 << 31) /* Odd Sample Status */ + +#define SSITR_TTFS (1 << 5) /* Test Tx FIFO Service Request */ +#define SSITR_TRFS (1 << 6) /* Test Rx FIFO Service Request */ +#define SSITR_TROR (1 << 7) /* Test Rx FIFO Overrun */ + +#define SSPSP_SCMODE_MASK (0x3) /* Serial Bit Rate Clock Mode */ +#define SSPSP_SCMODE(x) ((x) << 0) +#define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ +#define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ +#define SSPSP_STRTDLY_MASK (0x7 << 4) /* Start Delay */ +#define SSPSP_STRTDLY(x) ((x) << 4) +#define SSPSP_DMYSTRT_MASK (0x3 << 7) /* Dummy Start */ +#define SSPSP_DMYSTRT(x) ((x) << 7) +#define SSPSP_SFRMDLY_MASK (0x7f << 9) /* Serial Frame Delay */ +#define SSPSP_SFRMDLY(x) ((x) << 9) +#define SSPSP_SFRMWDTH_MASK (0x3f << 16) /* Serial Frame Width */ +#define SSPSP_SFRMWDTH(x) ((x) << 16) +#define SSPSP_DMYSTOP_MASK (0x3 << 23) /* Dummy Stop */ +#define SSPSP_DMYSTOP(x) ((x) << 23) +#define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ +#define SSPSP_EDMYSTRT_MASK (0x3 << 26) /* Extended Dummy Start */ +#define SSPSP_EDMYSTRT(x) ((x) << 26) +#define SSPSP_EDMYSTOP_MASK (0x7 << 28) /* Extended Dummy Stop */ +#define SSPSP_EDMYSTOP(x) ((x) << 28) +#define SSPSP_TIMING_MASK (SSPSP_STRTDLY_MASK | SSPSP_DMYSTRT_MASK \ + | SSPSP_SFRMDLY_MASK | SSPSP_SFRMWDTH_MASK \ + | SSPSP_DMYSTOP_MASK | SSPSP_EDMYSTRT_MASK \ + | SSPSP_EDMYSTOP_MASK | SSPSP_FSRT) + + +#if defined(CONFIG_CPU_PXA910) +#define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ +#define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ +#define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ +#define SSACD_SCDX8 (1 << 7) /* SYSCLK division ratio select */ +#endif + + +#endif /* __ASM_ARCH_REGS_SSP_H */ + diff --git a/arch/arm/mach-mmp/pxa168.c b/arch/arm/mach-mmp/pxa168.c index 37dbdde..5f1a4dd 100644 --- a/arch/arm/mach-mmp/pxa168.c +++ b/arch/arm/mach-mmp/pxa168.c @@ -72,6 +72,11 @@ static APBC_CLK(pwm1, PXA168_PWM1, 1, 13000000); static APBC_CLK(pwm2, PXA168_PWM2, 1, 13000000); static APBC_CLK(pwm3, PXA168_PWM3, 1, 13000000); static APBC_CLK(pwm4, PXA168_PWM4, 1, 13000000); +static APBC_CLK(ssp1, PXA168_SSP1, 4, 0); +static APBC_CLK(ssp2, PXA168_SSP2, 4, 0); +static APBC_CLK(ssp3, PXA168_SSP3, 4, 0); +static APBC_CLK(ssp4, PXA168_SSP4, 4, 0); +static APBC_CLK(ssp5, PXA168_SSP5, 4, 0); static APMU_CLK(nand, NAND, 0x01db, 208000000); @@ -85,6 +90,11 @@ static struct clk_lookup pxa168_clkregs[] = { INIT_CLKREG(&clk_pwm2, "pxa168-pwm.1", NULL), INIT_CLKREG(&clk_pwm3, "pxa168-pwm.2", NULL), INIT_CLKREG(&clk_pwm4, "pxa168-pwm.3", NULL), + INIT_CLKREG(&clk_ssp1, "pxa168-ssp.0", NULL), + INIT_CLKREG(&clk_ssp2, "pxa168-ssp.1", NULL), + INIT_CLKREG(&clk_ssp3, "pxa168-ssp.2", NULL), + INIT_CLKREG(&clk_ssp4, "pxa168-ssp.3", NULL), + INIT_CLKREG(&clk_ssp5, "pxa168-ssp.4", NULL), INIT_CLKREG(&clk_nand, "pxa3xx-nand", NULL), }; @@ -132,3 +142,8 @@ PXA168_DEVICE(pwm2, "pxa168-pwm", 1, NONE, 0xd401a400, 0x10); PXA168_DEVICE(pwm3, "pxa168-pwm", 2, NONE, 0xd401a800, 0x10); PXA168_DEVICE(pwm4, "pxa168-pwm", 3, NONE, 0xd401ac00, 0x10); PXA168_DEVICE(nand, "pxa3xx-nand", -1, NAND, 0xd4283000, 0x80, 97, 99); +PXA168_DEVICE(ssp1, "pxa168-ssp", 0, SSP1, 0xd401b000, 0x40, 52, 53); +PXA168_DEVICE(ssp2, "pxa168-ssp", 1, SSP2, 0xd401c000, 0x40, 54, 55); +PXA168_DEVICE(ssp3, "pxa168-ssp", 2, SSP3, 0xd401f000, 0x40, 56, 57); +PXA168_DEVICE(ssp4, "pxa168-ssp", 3, SSP4, 0xd4020000, 0x40, 58, 59); +PXA168_DEVICE(ssp5, "pxa168-ssp", 4, SSP5, 0xd4021000, 0x40, 60, 61); diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h index fc83ba7..06cffae 100644 --- a/arch/arm/plat-pxa/include/plat/ssp.h +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -14,6 +14,7 @@ * PXA26x SSP, NSSP, ASSP * PXA27x SSP1, SSP2, SSP3 * PXA3xx SSP1, SSP2, SSP3, SSP4 + * PXA168 SSP1, SSP2, SSP3, SSP4, SSP5 */ #ifndef __ASM_PLAT_SSP_H @@ -27,6 +28,7 @@ enum pxa_ssp_type { PXA25x_SSP, /* pxa 210, 250, 255, 26x */ PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ PXA27x_SSP, + PXA168_SSP, }; struct ssp_device { diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index d4a7673..4dc2ee6 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -465,6 +465,7 @@ static const struct platform_device_id ssp_id_table[] = { { "pxa25x-ssp", PXA25x_SSP }, { "pxa25x-nssp", PXA25x_NSSP }, { "pxa27x-ssp", PXA27x_SSP }, + { "pxa168-ssp", PXA168_SSP }, { }, }; -- 1.5.6.5

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[PATCH] [ARM] pxa: move ssp into plat
by Haojian Zhuang 10 Mar '10

10 Mar '10

Since ssp is shared between pxa and mmp, move ssp driver into plat directory. --- arch/arm/mach-mmp/Kconfig | 5 + arch/arm/mach-pxa/Kconfig | 10 - arch/arm/mach-pxa/Makefile | 3 - arch/arm/mach-pxa/corgi_ssp.c | 2 +- arch/arm/mach-pxa/include/mach/ssp.h | 109 -------- arch/arm/mach-pxa/littleton.c | 2 +- arch/arm/mach-pxa/poodle.c | 4 +- arch/arm/mach-pxa/pxa3xx.c | 2 +- arch/arm/mach-pxa/ssp.c | 510 ---------------------------------- arch/arm/plat-pxa/Kconfig | 10 + arch/arm/plat-pxa/Makefile | 1 + arch/arm/plat-pxa/include/plat/ssp.h | 109 ++++++++ arch/arm/plat-pxa/ssp.c | 510 ++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa-ssp.c | 2 +- 14 files changed, 641 insertions(+), 638 deletions(-) delete mode 100644 arch/arm/mach-pxa/include/mach/ssp.h delete mode 100644 arch/arm/mach-pxa/ssp.c create mode 100644 arch/arm/plat-pxa/include/plat/ssp.h create mode 100644 arch/arm/plat-pxa/ssp.c diff --git a/arch/arm/mach-mmp/Kconfig b/arch/arm/mach-mmp/Kconfig index 6ab843e..06728b1 100644 --- a/arch/arm/mach-mmp/Kconfig +++ b/arch/arm/mach-mmp/Kconfig @@ -5,6 +5,7 @@ menu "Marvell PXA168/910/MMP2 Implmentations" config MACH_ASPENITE bool "Marvell's PXA168 Aspenite Development Board" select CPU_PXA168 + select PXA_SSP help Say 'Y' here if you want to support the Marvell PXA168-based Aspenite Development Board. @@ -12,6 +13,7 @@ config MACH_ASPENITE config MACH_ZYLONITE2 bool "Marvell's PXA168 Zylonite2 Development Board" select CPU_PXA168 + select PXA_SSP help Say 'Y' here if you want to support the Marvell PXA168-based Zylonite2 Development Board. @@ -19,6 +21,7 @@ config MACH_ZYLONITE2 config MACH_AVENGERS_LITE bool "Marvell's PXA168 Avengers Lite Development Board" select CPU_PXA168 + select PXA_SSP help Say 'Y' here if you want to support the Marvell PXA168-based Avengers Lite Development Board. @@ -26,6 +29,7 @@ config MACH_AVENGERS_LITE config MACH_TAVOREVB bool "Marvell's PXA910 TavorEVB Development Board" select CPU_PXA910 + select PXA_SSP help Say 'Y' here if you want to support the Marvell PXA910-based TavorEVB Development Board. @@ -33,6 +37,7 @@ config MACH_TAVOREVB config MACH_TTC_DKB bool "Marvell's PXA910 TavorEVB Development Board" select CPU_PXA910 + select PXA_SSP help Say 'Y' here if you want to support the Marvell PXA910-based TTC_DKB Development Board. diff --git a/arch/arm/mach-pxa/Kconfig b/arch/arm/mach-pxa/Kconfig index 38fbd0a..ef8744a 100644 --- a/arch/arm/mach-pxa/Kconfig +++ b/arch/arm/mach-pxa/Kconfig @@ -654,16 +654,6 @@ config PXA_SHARP_Cxx00 help Enable common support for Sharp Cxx00 models -config PXA_SSP - tristate - help - Enable support for PXA2xx SSP ports - -config PXA_SSP_LEGACY - bool - help - Support of legacy SSP API - config TOSA_BT tristate "Control the state of built-in bluetooth chip on Sharp SL-6000" depends on MACH_TOSA diff --git a/arch/arm/mach-pxa/Makefile b/arch/arm/mach-pxa/Makefile index 86bc87b..299d4f5 100644 --- a/arch/arm/mach-pxa/Makefile +++ b/arch/arm/mach-pxa/Makefile @@ -13,9 +13,6 @@ obj-$(CONFIG_PXA27x) += cpufreq-pxa2xx.o obj-$(CONFIG_PXA3xx) += cpufreq-pxa3xx.o endif -# Generic drivers that other drivers may depend upon -obj-$(CONFIG_PXA_SSP) += ssp.o - # SoC-specific code obj-$(CONFIG_PXA25x) += mfp-pxa2xx.o pxa2xx.o pxa25x.o obj-$(CONFIG_PXA27x) += mfp-pxa2xx.o pxa2xx.o pxa27x.o diff --git a/arch/arm/mach-pxa/corgi_ssp.c b/arch/arm/mach-pxa/corgi_ssp.c index a5ee707..4ebc8e8 100644 --- a/arch/arm/mach-pxa/corgi_ssp.c +++ b/arch/arm/mach-pxa/corgi_ssp.c @@ -19,9 +19,9 @@ #include <mach/hardware.h> #include <asm/mach-types.h> -#include <mach/ssp.h> #include <mach/pxa2xx-gpio.h> #include <mach/regs-ssp.h> +#include <plat/ssp.h> #include "sharpsl.h" static DEFINE_SPINLOCK(corgi_ssp_lock); diff --git a/arch/arm/mach-pxa/include/mach/ssp.h b/arch/arm/mach-pxa/include/mach/ssp.h deleted file mode 100644 index be1be5b..0000000 --- a/arch/arm/mach-pxa/include/mach/ssp.h +++ /dev/null @@ -1,109 +0,0 @@ -/* - * ssp.h - * - * Copyright (C) 2003 Russell King, All Rights Reserved. - * - * 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. - * - * This driver supports the following PXA CPU/SSP ports:- - * - * PXA250 SSP - * PXA255 SSP, NSSP - * PXA26x SSP, NSSP, ASSP - * PXA27x SSP1, SSP2, SSP3 - * PXA3xx SSP1, SSP2, SSP3, SSP4 - */ - -#ifndef __ASM_ARCH_SSP_H -#define __ASM_ARCH_SSP_H - -#include <linux/list.h> -#include <linux/io.h> - -enum pxa_ssp_type { - SSP_UNDEFINED = 0, - PXA25x_SSP, /* pxa 210, 250, 255, 26x */ - PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ - PXA27x_SSP, -}; - -struct ssp_device { - struct platform_device *pdev; - struct list_head node; - - struct clk *clk; - void __iomem *mmio_base; - unsigned long phys_base; - - const char *label; - int port_id; - int type; - int use_count; - int irq; - int drcmr_rx; - int drcmr_tx; -}; - -#ifdef CONFIG_PXA_SSP_LEGACY -/* - * SSP initialisation flags - */ -#define SSP_NO_IRQ 0x1 /* don't register an irq handler in SSP driver */ - -struct ssp_state { - u32 cr0; - u32 cr1; - u32 to; - u32 psp; -}; - -struct ssp_dev { - struct ssp_device *ssp; - u32 port; - u32 mode; - u32 flags; - u32 psp_flags; - u32 speed; - int irq; -}; - -int ssp_write_word(struct ssp_dev *dev, u32 data); -int ssp_read_word(struct ssp_dev *dev, u32 *data); -int ssp_flush(struct ssp_dev *dev); -void ssp_enable(struct ssp_dev *dev); -void ssp_disable(struct ssp_dev *dev); -void ssp_save_state(struct ssp_dev *dev, struct ssp_state *ssp); -void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *ssp); -int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags); -int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed); -void ssp_exit(struct ssp_dev *dev); -#endif /* CONFIG_PXA_SSP_LEGACY */ - -/** - * ssp_write_reg - Write to a SSP register - * - * @dev: SSP device to access - * @reg: Register to write to - * @val: Value to be written. - */ -static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) -{ - __raw_writel(val, dev->mmio_base + reg); -} - -/** - * ssp_read_reg - Read from a SSP register - * - * @dev: SSP device to access - * @reg: Register to read from - */ -static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg) -{ - return __raw_readl(dev->mmio_base + reg); -} - -struct ssp_device *ssp_request(int port, const char *label); -void ssp_free(struct ssp_device *); -#endif /* __ASM_ARCH_SSP_H */ diff --git a/arch/arm/mach-pxa/littleton.c b/arch/arm/mach-pxa/littleton.c index fa527b2..b9d3c20 100644 --- a/arch/arm/mach-pxa/littleton.c +++ b/arch/arm/mach-pxa/littleton.c @@ -41,12 +41,12 @@ #include <mach/pxa300.h> #include <mach/pxafb.h> -#include <mach/ssp.h> #include <mach/mmc.h> #include <mach/pxa2xx_spi.h> #include <mach/pxa27x_keypad.h> #include <mach/littleton.h> #include <plat/i2c.h> +#include <plat/ssp.h> #include <plat/pxa3xx_nand.h> #include "generic.h" diff --git a/arch/arm/mach-pxa/poodle.c b/arch/arm/mach-pxa/poodle.c index d58a524..90343c3 100644 --- a/arch/arm/mach-pxa/poodle.c +++ b/arch/arm/mach-pxa/poodle.c @@ -40,13 +40,13 @@ #include <mach/pxa25x.h> #include <mach/mmc.h> #include <mach/udc.h> -#include <plat/i2c.h> #include <mach/irda.h> #include <mach/poodle.h> #include <mach/pxafb.h> #include <mach/sharpsl.h> -#include <mach/ssp.h> #include <mach/pxa2xx_spi.h> +#include <plat/i2c.h> +#include <plat/ssp.h> #include <asm/hardware/scoop.h> #include <asm/hardware/locomo.h> diff --git a/arch/arm/mach-pxa/pxa3xx.c b/arch/arm/mach-pxa/pxa3xx.c index fcb0721..bbb0b76 100644 --- a/arch/arm/mach-pxa/pxa3xx.c +++ b/arch/arm/mach-pxa/pxa3xx.c @@ -29,9 +29,9 @@ #include <mach/ohci.h> #include <mach/pm.h> #include <mach/dma.h> -#include <mach/ssp.h> #include <mach/regs-intc.h> #include <plat/i2c.h> +#include <plat/ssp.h> #include "generic.h" #include "devices.h" diff --git a/arch/arm/mach-pxa/ssp.c b/arch/arm/mach-pxa/ssp.c deleted file mode 100644 index a81d6db..0000000 --- a/arch/arm/mach-pxa/ssp.c +++ /dev/null @@ -1,510 +0,0 @@ -/* - * linux/arch/arm/mach-pxa/ssp.c - * - * based on linux/arch/arm/mach-sa1100/ssp.c by Russell King - * - * Copyright (C) 2003 Russell King. - * Copyright (C) 2003 Wolfson Microelectronics PLC - * - * 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. - * - * PXA2xx SSP driver. This provides the generic core for simple - * IO-based SSP applications and allows easy port setup for DMA access. - * - * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> - */ - -#include <linux/module.h> -#include <linux/kernel.h> -#include <linux/sched.h> -#include <linux/slab.h> -#include <linux/errno.h> -#include <linux/interrupt.h> -#include <linux/ioport.h> -#include <linux/init.h> -#include <linux/mutex.h> -#include <linux/clk.h> -#include <linux/err.h> -#include <linux/platform_device.h> -#include <linux/io.h> - -#include <asm/irq.h> -#include <mach/hardware.h> -#include <mach/ssp.h> -#include <mach/regs-ssp.h> - -#ifdef CONFIG_PXA_SSP_LEGACY - -#define TIMEOUT 100000 - -static irqreturn_t ssp_interrupt(int irq, void *dev_id) -{ - struct ssp_dev *dev = dev_id; - struct ssp_device *ssp = dev->ssp; - unsigned int status; - - status = __raw_readl(ssp->mmio_base + SSSR); - __raw_writel(status, ssp->mmio_base + SSSR); - - if (status & SSSR_ROR) - printk(KERN_WARNING "SSP(%d): receiver overrun\n", dev->port); - - if (status & SSSR_TUR) - printk(KERN_WARNING "SSP(%d): transmitter underrun\n", dev->port); - - if (status & SSSR_BCE) - printk(KERN_WARNING "SSP(%d): bit count error\n", dev->port); - - return IRQ_HANDLED; -} - -/** - * ssp_write_word - write a word to the SSP port - * @data: 32-bit, MSB justified data to write. - * - * Wait for a free entry in the SSP transmit FIFO, and write a data - * word to the SSP port. - * - * The caller is expected to perform the necessary locking. - * - * Returns: - * %-ETIMEDOUT timeout occurred - * 0 success - */ -int ssp_write_word(struct ssp_dev *dev, u32 data) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT; - - while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_TNF)) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - - __raw_writel(data, ssp->mmio_base + SSDR); - - return 0; -} - -/** - * ssp_read_word - read a word from the SSP port - * - * Wait for a data word in the SSP receive FIFO, and return the - * received data. Data is LSB justified. - * - * Note: Currently, if data is not expected to be received, this - * function will wait for ever. - * - * The caller is expected to perform the necessary locking. - * - * Returns: - * %-ETIMEDOUT timeout occurred - * 32-bit data success - */ -int ssp_read_word(struct ssp_dev *dev, u32 *data) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT; - - while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE)) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - - *data = __raw_readl(ssp->mmio_base + SSDR); - return 0; -} - -/** - * ssp_flush - flush the transmit and receive FIFOs - * - * Wait for the SSP to idle, and ensure that the receive FIFO - * is empty. - * - * The caller is expected to perform the necessary locking. - */ -int ssp_flush(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT * 2; - - /* ensure TX FIFO is empty instead of not full */ - if (cpu_is_pxa3xx()) { - while (__raw_readl(ssp->mmio_base + SSSR) & 0xf00) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - timeout = TIMEOUT * 2; - } - - do { - while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE) { - if (!--timeout) - return -ETIMEDOUT; - (void)__raw_readl(ssp->mmio_base + SSDR); - } - if (!--timeout) - return -ETIMEDOUT; - } while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_BSY); - - return 0; -} - -/** - * ssp_enable - enable the SSP port - * - * Turn on the SSP port. - */ -void ssp_enable(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - uint32_t sscr0; - - sscr0 = __raw_readl(ssp->mmio_base + SSCR0); - sscr0 |= SSCR0_SSE; - __raw_writel(sscr0, ssp->mmio_base + SSCR0); -} - -/** - * ssp_disable - shut down the SSP port - * - * Turn off the SSP port, optionally powering it down. - */ -void ssp_disable(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - uint32_t sscr0; - - sscr0 = __raw_readl(ssp->mmio_base + SSCR0); - sscr0 &= ~SSCR0_SSE; - __raw_writel(sscr0, ssp->mmio_base + SSCR0); -} - -/** - * ssp_save_state - save the SSP configuration - * @ssp: pointer to structure to save SSP configuration - * - * Save the configured SSP state for suspend. - */ -void ssp_save_state(struct ssp_dev *dev, struct ssp_state *state) -{ - struct ssp_device *ssp = dev->ssp; - - state->cr0 = __raw_readl(ssp->mmio_base + SSCR0); - state->cr1 = __raw_readl(ssp->mmio_base + SSCR1); - state->to = __raw_readl(ssp->mmio_base + SSTO); - state->psp = __raw_readl(ssp->mmio_base + SSPSP); - - ssp_disable(dev); -} - -/** - * ssp_restore_state - restore a previously saved SSP configuration - * @ssp: pointer to configuration saved by ssp_save_state - * - * Restore the SSP configuration saved previously by ssp_save_state. - */ -void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *state) -{ - struct ssp_device *ssp = dev->ssp; - uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; - - __raw_writel(sssr, ssp->mmio_base + SSSR); - - __raw_writel(state->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); - __raw_writel(state->cr1, ssp->mmio_base + SSCR1); - __raw_writel(state->to, ssp->mmio_base + SSTO); - __raw_writel(state->psp, ssp->mmio_base + SSPSP); - __raw_writel(state->cr0, ssp->mmio_base + SSCR0); -} - -/** - * ssp_config - configure SSP port settings - * @mode: port operating mode - * @flags: port config flags - * @psp_flags: port PSP config flags - * @speed: port speed - * - * Port MUST be disabled by ssp_disable before making any config changes. - */ -int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed) -{ - struct ssp_device *ssp = dev->ssp; - - dev->mode = mode; - dev->flags = flags; - dev->psp_flags = psp_flags; - dev->speed = speed; - - /* set up port type, speed, port settings */ - __raw_writel((dev->speed | dev->mode), ssp->mmio_base + SSCR0); - __raw_writel(dev->flags, ssp->mmio_base + SSCR1); - __raw_writel(dev->psp_flags, ssp->mmio_base + SSPSP); - - return 0; -} - -/** - * ssp_init - setup the SSP port - * - * initialise and claim resources for the SSP port. - * - * Returns: - * %-ENODEV if the SSP port is unavailable - * %-EBUSY if the resources are already in use - * %0 on success - */ -int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags) -{ - struct ssp_device *ssp; - int ret; - - ssp = ssp_request(port, "SSP"); - if (ssp == NULL) - return -ENODEV; - - dev->ssp = ssp; - dev->port = port; - - /* do we need to get irq */ - if (!(init_flags & SSP_NO_IRQ)) { - ret = request_irq(ssp->irq, ssp_interrupt, - 0, "SSP", dev); - if (ret) - goto out_region; - dev->irq = ssp->irq; - } else - dev->irq = NO_IRQ; - - /* turn on SSP port clock */ - clk_enable(ssp->clk); - return 0; - -out_region: - ssp_free(ssp); - return ret; -} - -/** - * ssp_exit - undo the effects of ssp_init - * - * release and free resources for the SSP port. - */ -void ssp_exit(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - - ssp_disable(dev); - if (dev->irq != NO_IRQ) - free_irq(dev->irq, dev); - clk_disable(ssp->clk); - ssp_free(ssp); -} -#endif /* CONFIG_PXA_SSP_LEGACY */ - -static DEFINE_MUTEX(ssp_lock); -static LIST_HEAD(ssp_list); - -struct ssp_device *ssp_request(int port, const char *label) -{ - struct ssp_device *ssp = NULL; - - mutex_lock(&ssp_lock); - - list_for_each_entry(ssp, &ssp_list, node) { - if (ssp->port_id == port && ssp->use_count == 0) { - ssp->use_count++; - ssp->label = label; - break; - } - } - - mutex_unlock(&ssp_lock); - - if (&ssp->node == &ssp_list) - return NULL; - - return ssp; -} -EXPORT_SYMBOL(ssp_request); - -void ssp_free(struct ssp_device *ssp) -{ - mutex_lock(&ssp_lock); - if (ssp->use_count) { - ssp->use_count--; - ssp->label = NULL; - } else - dev_err(&ssp->pdev->dev, "device already free\n"); - mutex_unlock(&ssp_lock); -} -EXPORT_SYMBOL(ssp_free); - -static int __devinit ssp_probe(struct platform_device *pdev) -{ - const struct platform_device_id *id = platform_get_device_id(pdev); - struct resource *res; - struct ssp_device *ssp; - int ret = 0; - - ssp = kzalloc(sizeof(struct ssp_device), GFP_KERNEL); - if (ssp == NULL) { - dev_err(&pdev->dev, "failed to allocate memory"); - return -ENOMEM; - } - ssp->pdev = pdev; - - ssp->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(ssp->clk)) { - ret = PTR_ERR(ssp->clk); - goto err_free; - } - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no memory resource defined\n"); - ret = -ENODEV; - goto err_free_clk; - } - - res = request_mem_region(res->start, res->end - res->start + 1, - pdev->name); - if (res == NULL) { - dev_err(&pdev->dev, "failed to request memory resource\n"); - ret = -EBUSY; - goto err_free_clk; - } - - ssp->phys_base = res->start; - - ssp->mmio_base = ioremap(res->start, res->end - res->start + 1); - if (ssp->mmio_base == NULL) { - dev_err(&pdev->dev, "failed to ioremap() registers\n"); - ret = -ENODEV; - goto err_free_mem; - } - - ssp->irq = platform_get_irq(pdev, 0); - if (ssp->irq < 0) { - dev_err(&pdev->dev, "no IRQ resource defined\n"); - ret = -ENODEV; - goto err_free_io; - } - - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (res == NULL) { - dev_err(&pdev->dev, "no SSP RX DRCMR defined\n"); - ret = -ENODEV; - goto err_free_io; - } - ssp->drcmr_rx = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (res == NULL) { - dev_err(&pdev->dev, "no SSP TX DRCMR defined\n"); - ret = -ENODEV; - goto err_free_io; - } - ssp->drcmr_tx = res->start; - - /* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id - * starts from 0, do a translation here - */ - ssp->port_id = pdev->id + 1; - ssp->use_count = 0; - ssp->type = (int)id->driver_data; - - mutex_lock(&ssp_lock); - list_add(&ssp->node, &ssp_list); - mutex_unlock(&ssp_lock); - - platform_set_drvdata(pdev, ssp); - return 0; - -err_free_io: - iounmap(ssp->mmio_base); -err_free_mem: - release_mem_region(res->start, res->end - res->start + 1); -err_free_clk: - clk_put(ssp->clk); -err_free: - kfree(ssp); - return ret; -} - -static int __devexit ssp_remove(struct platform_device *pdev) -{ - struct resource *res; - struct ssp_device *ssp; - - ssp = platform_get_drvdata(pdev); - if (ssp == NULL) - return -ENODEV; - - iounmap(ssp->mmio_base); - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(res->start, res->end - res->start + 1); - - clk_put(ssp->clk); - - mutex_lock(&ssp_lock); - list_del(&ssp->node); - mutex_unlock(&ssp_lock); - - kfree(ssp); - return 0; -} - -static const struct platform_device_id ssp_id_table[] = { - { "pxa25x-ssp", PXA25x_SSP }, - { "pxa25x-nssp", PXA25x_NSSP }, - { "pxa27x-ssp", PXA27x_SSP }, - { }, -}; - -static struct platform_driver ssp_driver = { - .probe = ssp_probe, - .remove = __devexit_p(ssp_remove), - .driver = { - .owner = THIS_MODULE, - .name = "pxa2xx-ssp", - }, - .id_table = ssp_id_table, -}; - -static int __init pxa_ssp_init(void) -{ - return platform_driver_register(&ssp_driver); -} - -static void __exit pxa_ssp_exit(void) -{ - platform_driver_unregister(&ssp_driver); -} - -arch_initcall(pxa_ssp_init); -module_exit(pxa_ssp_exit); - -#ifdef CONFIG_PXA_SSP_LEGACY -EXPORT_SYMBOL(ssp_write_word); -EXPORT_SYMBOL(ssp_read_word); -EXPORT_SYMBOL(ssp_flush); -EXPORT_SYMBOL(ssp_enable); -EXPORT_SYMBOL(ssp_disable); -EXPORT_SYMBOL(ssp_save_state); -EXPORT_SYMBOL(ssp_restore_state); -EXPORT_SYMBOL(ssp_init); -EXPORT_SYMBOL(ssp_exit); -EXPORT_SYMBOL(ssp_config); -#endif - -MODULE_DESCRIPTION("PXA SSP driver"); -MODULE_AUTHOR("Liam Girdwood"); -MODULE_LICENSE("GPL"); - diff --git a/arch/arm/plat-pxa/Kconfig b/arch/arm/plat-pxa/Kconfig index b158e98..994f4b8 100644 --- a/arch/arm/plat-pxa/Kconfig +++ b/arch/arm/plat-pxa/Kconfig @@ -1,3 +1,13 @@ if PLAT_PXA +config PXA_SSP + tristate + help + Enable support for PXA2xx SSP ports + +config PXA_SSP_LEGACY + bool + help + Support of legacy SSP API + endif diff --git a/arch/arm/plat-pxa/Makefile b/arch/arm/plat-pxa/Makefile index 0264bfb..4aacdd1 100644 --- a/arch/arm/plat-pxa/Makefile +++ b/arch/arm/plat-pxa/Makefile @@ -9,3 +9,4 @@ obj-$(CONFIG_PXA3xx) += mfp.o obj-$(CONFIG_ARCH_MMP) += mfp.o obj-$(CONFIG_HAVE_PWM) += pwm.o +obj-$(CONFIG_PXA_SSP) += ssp.o diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h new file mode 100644 index 0000000..fc83ba7 --- /dev/null +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -0,0 +1,109 @@ +/* + * ssp.h + * + * Copyright (C) 2003 Russell King, All Rights Reserved. + * + * 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. + * + * This driver supports the following PXA CPU/SSP ports:- + * + * PXA250 SSP + * PXA255 SSP, NSSP + * PXA26x SSP, NSSP, ASSP + * PXA27x SSP1, SSP2, SSP3 + * PXA3xx SSP1, SSP2, SSP3, SSP4 + */ + +#ifndef __ASM_PLAT_SSP_H +#define __ASM_PLAT_SSP_H + +#include <linux/list.h> +#include <linux/io.h> + +enum pxa_ssp_type { + SSP_UNDEFINED = 0, + PXA25x_SSP, /* pxa 210, 250, 255, 26x */ + PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ + PXA27x_SSP, +}; + +struct ssp_device { + struct platform_device *pdev; + struct list_head node; + + struct clk *clk; + void __iomem *mmio_base; + unsigned long phys_base; + + const char *label; + int port_id; + int type; + int use_count; + int irq; + int drcmr_rx; + int drcmr_tx; +}; + +#ifdef CONFIG_PXA_SSP_LEGACY +/* + * SSP initialisation flags + */ +#define SSP_NO_IRQ 0x1 /* don't register an irq handler in SSP driver */ + +struct ssp_state { + u32 cr0; + u32 cr1; + u32 to; + u32 psp; +}; + +struct ssp_dev { + struct ssp_device *ssp; + u32 port; + u32 mode; + u32 flags; + u32 psp_flags; + u32 speed; + int irq; +}; + +int ssp_write_word(struct ssp_dev *dev, u32 data); +int ssp_read_word(struct ssp_dev *dev, u32 *data); +int ssp_flush(struct ssp_dev *dev); +void ssp_enable(struct ssp_dev *dev); +void ssp_disable(struct ssp_dev *dev); +void ssp_save_state(struct ssp_dev *dev, struct ssp_state *ssp); +void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *ssp); +int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags); +int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed); +void ssp_exit(struct ssp_dev *dev); +#endif /* CONFIG_PXA_SSP_LEGACY */ + +/** + * ssp_write_reg - Write to a SSP register + * + * @dev: SSP device to access + * @reg: Register to write to + * @val: Value to be written. + */ +static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) +{ + __raw_writel(val, dev->mmio_base + reg); +} + +/** + * ssp_read_reg - Read from a SSP register + * + * @dev: SSP device to access + * @reg: Register to read from + */ +static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg) +{ + return __raw_readl(dev->mmio_base + reg); +} + +struct ssp_device *ssp_request(int port, const char *label); +void ssp_free(struct ssp_device *); +#endif /* __ASM_PLAT_SSP_H */ diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c new file mode 100644 index 0000000..d4a7673 --- /dev/null +++ b/arch/arm/plat-pxa/ssp.c @@ -0,0 +1,510 @@ +/* + * linux/arch/arm/mach-pxa/ssp.c + * + * based on linux/arch/arm/mach-sa1100/ssp.c by Russell King + * + * Copyright (C) 2003 Russell King. + * Copyright (C) 2003 Wolfson Microelectronics PLC + * + * 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. + * + * PXA2xx SSP driver. This provides the generic core for simple + * IO-based SSP applications and allows easy port setup for DMA access. + * + * Author: Liam Girdwood <liam.girdwood(a)wolfsonmicro.com> + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/sched.h> +#include <linux/slab.h> +#include <linux/errno.h> +#include <linux/interrupt.h> +#include <linux/ioport.h> +#include <linux/init.h> +#include <linux/mutex.h> +#include <linux/clk.h> +#include <linux/err.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <asm/irq.h> +#include <mach/hardware.h> +#include <mach/regs-ssp.h> +#include <plat/ssp.h> + +#ifdef CONFIG_PXA_SSP_LEGACY + +#define TIMEOUT 100000 + +static irqreturn_t ssp_interrupt(int irq, void *dev_id) +{ + struct ssp_dev *dev = dev_id; + struct ssp_device *ssp = dev->ssp; + unsigned int status; + + status = __raw_readl(ssp->mmio_base + SSSR); + __raw_writel(status, ssp->mmio_base + SSSR); + + if (status & SSSR_ROR) + printk(KERN_WARNING "SSP(%d): receiver overrun\n", dev->port); + + if (status & SSSR_TUR) + printk(KERN_WARNING "SSP(%d): transmitter underrun\n", dev->port); + + if (status & SSSR_BCE) + printk(KERN_WARNING "SSP(%d): bit count error\n", dev->port); + + return IRQ_HANDLED; +} + +/** + * ssp_write_word - write a word to the SSP port + * @data: 32-bit, MSB justified data to write. + * + * Wait for a free entry in the SSP transmit FIFO, and write a data + * word to the SSP port. + * + * The caller is expected to perform the necessary locking. + * + * Returns: + * %-ETIMEDOUT timeout occurred + * 0 success + */ +int ssp_write_word(struct ssp_dev *dev, u32 data) +{ + struct ssp_device *ssp = dev->ssp; + int timeout = TIMEOUT; + + while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_TNF)) { + if (!--timeout) + return -ETIMEDOUT; + cpu_relax(); + } + + __raw_writel(data, ssp->mmio_base + SSDR); + + return 0; +} + +/** + * ssp_read_word - read a word from the SSP port + * + * Wait for a data word in the SSP receive FIFO, and return the + * received data. Data is LSB justified. + * + * Note: Currently, if data is not expected to be received, this + * function will wait for ever. + * + * The caller is expected to perform the necessary locking. + * + * Returns: + * %-ETIMEDOUT timeout occurred + * 32-bit data success + */ +int ssp_read_word(struct ssp_dev *dev, u32 *data) +{ + struct ssp_device *ssp = dev->ssp; + int timeout = TIMEOUT; + + while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE)) { + if (!--timeout) + return -ETIMEDOUT; + cpu_relax(); + } + + *data = __raw_readl(ssp->mmio_base + SSDR); + return 0; +} + +/** + * ssp_flush - flush the transmit and receive FIFOs + * + * Wait for the SSP to idle, and ensure that the receive FIFO + * is empty. + * + * The caller is expected to perform the necessary locking. + */ +int ssp_flush(struct ssp_dev *dev) +{ + struct ssp_device *ssp = dev->ssp; + int timeout = TIMEOUT * 2; + + /* ensure TX FIFO is empty instead of not full */ + if (cpu_is_pxa3xx()) { + while (__raw_readl(ssp->mmio_base + SSSR) & 0xf00) { + if (!--timeout) + return -ETIMEDOUT; + cpu_relax(); + } + timeout = TIMEOUT * 2; + } + + do { + while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE) { + if (!--timeout) + return -ETIMEDOUT; + (void)__raw_readl(ssp->mmio_base + SSDR); + } + if (!--timeout) + return -ETIMEDOUT; + } while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_BSY); + + return 0; +} + +/** + * ssp_enable - enable the SSP port + * + * Turn on the SSP port. + */ +void ssp_enable(struct ssp_dev *dev) +{ + struct ssp_device *ssp = dev->ssp; + uint32_t sscr0; + + sscr0 = __raw_readl(ssp->mmio_base + SSCR0); + sscr0 |= SSCR0_SSE; + __raw_writel(sscr0, ssp->mmio_base + SSCR0); +} + +/** + * ssp_disable - shut down the SSP port + * + * Turn off the SSP port, optionally powering it down. + */ +void ssp_disable(struct ssp_dev *dev) +{ + struct ssp_device *ssp = dev->ssp; + uint32_t sscr0; + + sscr0 = __raw_readl(ssp->mmio_base + SSCR0); + sscr0 &= ~SSCR0_SSE; + __raw_writel(sscr0, ssp->mmio_base + SSCR0); +} + +/** + * ssp_save_state - save the SSP configuration + * @ssp: pointer to structure to save SSP configuration + * + * Save the configured SSP state for suspend. + */ +void ssp_save_state(struct ssp_dev *dev, struct ssp_state *state) +{ + struct ssp_device *ssp = dev->ssp; + + state->cr0 = __raw_readl(ssp->mmio_base + SSCR0); + state->cr1 = __raw_readl(ssp->mmio_base + SSCR1); + state->to = __raw_readl(ssp->mmio_base + SSTO); + state->psp = __raw_readl(ssp->mmio_base + SSPSP); + + ssp_disable(dev); +} + +/** + * ssp_restore_state - restore a previously saved SSP configuration + * @ssp: pointer to configuration saved by ssp_save_state + * + * Restore the SSP configuration saved previously by ssp_save_state. + */ +void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *state) +{ + struct ssp_device *ssp = dev->ssp; + uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; + + __raw_writel(sssr, ssp->mmio_base + SSSR); + + __raw_writel(state->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); + __raw_writel(state->cr1, ssp->mmio_base + SSCR1); + __raw_writel(state->to, ssp->mmio_base + SSTO); + __raw_writel(state->psp, ssp->mmio_base + SSPSP); + __raw_writel(state->cr0, ssp->mmio_base + SSCR0); +} + +/** + * ssp_config - configure SSP port settings + * @mode: port operating mode + * @flags: port config flags + * @psp_flags: port PSP config flags + * @speed: port speed + * + * Port MUST be disabled by ssp_disable before making any config changes. + */ +int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed) +{ + struct ssp_device *ssp = dev->ssp; + + dev->mode = mode; + dev->flags = flags; + dev->psp_flags = psp_flags; + dev->speed = speed; + + /* set up port type, speed, port settings */ + __raw_writel((dev->speed | dev->mode), ssp->mmio_base + SSCR0); + __raw_writel(dev->flags, ssp->mmio_base + SSCR1); + __raw_writel(dev->psp_flags, ssp->mmio_base + SSPSP); + + return 0; +} + +/** + * ssp_init - setup the SSP port + * + * initialise and claim resources for the SSP port. + * + * Returns: + * %-ENODEV if the SSP port is unavailable + * %-EBUSY if the resources are already in use + * %0 on success + */ +int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags) +{ + struct ssp_device *ssp; + int ret; + + ssp = ssp_request(port, "SSP"); + if (ssp == NULL) + return -ENODEV; + + dev->ssp = ssp; + dev->port = port; + + /* do we need to get irq */ + if (!(init_flags & SSP_NO_IRQ)) { + ret = request_irq(ssp->irq, ssp_interrupt, + 0, "SSP", dev); + if (ret) + goto out_region; + dev->irq = ssp->irq; + } else + dev->irq = NO_IRQ; + + /* turn on SSP port clock */ + clk_enable(ssp->clk); + return 0; + +out_region: + ssp_free(ssp); + return ret; +} + +/** + * ssp_exit - undo the effects of ssp_init + * + * release and free resources for the SSP port. + */ +void ssp_exit(struct ssp_dev *dev) +{ + struct ssp_device *ssp = dev->ssp; + + ssp_disable(dev); + if (dev->irq != NO_IRQ) + free_irq(dev->irq, dev); + clk_disable(ssp->clk); + ssp_free(ssp); +} +#endif /* CONFIG_PXA_SSP_LEGACY */ + +static DEFINE_MUTEX(ssp_lock); +static LIST_HEAD(ssp_list); + +struct ssp_device *ssp_request(int port, const char *label) +{ + struct ssp_device *ssp = NULL; + + mutex_lock(&ssp_lock); + + list_for_each_entry(ssp, &ssp_list, node) { + if (ssp->port_id == port && ssp->use_count == 0) { + ssp->use_count++; + ssp->label = label; + break; + } + } + + mutex_unlock(&ssp_lock); + + if (&ssp->node == &ssp_list) + return NULL; + + return ssp; +} +EXPORT_SYMBOL(ssp_request); + +void ssp_free(struct ssp_device *ssp) +{ + mutex_lock(&ssp_lock); + if (ssp->use_count) { + ssp->use_count--; + ssp->label = NULL; + } else + dev_err(&ssp->pdev->dev, "device already free\n"); + mutex_unlock(&ssp_lock); +} +EXPORT_SYMBOL(ssp_free); + +static int __devinit ssp_probe(struct platform_device *pdev) +{ + const struct platform_device_id *id = platform_get_device_id(pdev); + struct resource *res; + struct ssp_device *ssp; + int ret = 0; + + ssp = kzalloc(sizeof(struct ssp_device), GFP_KERNEL); + if (ssp == NULL) { + dev_err(&pdev->dev, "failed to allocate memory"); + return -ENOMEM; + } + ssp->pdev = pdev; + + ssp->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssp->clk)) { + ret = PTR_ERR(ssp->clk); + goto err_free; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no memory resource defined\n"); + ret = -ENODEV; + goto err_free_clk; + } + + res = request_mem_region(res->start, res->end - res->start + 1, + pdev->name); + if (res == NULL) { + dev_err(&pdev->dev, "failed to request memory resource\n"); + ret = -EBUSY; + goto err_free_clk; + } + + ssp->phys_base = res->start; + + ssp->mmio_base = ioremap(res->start, res->end - res->start + 1); + if (ssp->mmio_base == NULL) { + dev_err(&pdev->dev, "failed to ioremap() registers\n"); + ret = -ENODEV; + goto err_free_mem; + } + + ssp->irq = platform_get_irq(pdev, 0); + if (ssp->irq < 0) { + dev_err(&pdev->dev, "no IRQ resource defined\n"); + ret = -ENODEV; + goto err_free_io; + } + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no SSP RX DRCMR defined\n"); + ret = -ENODEV; + goto err_free_io; + } + ssp->drcmr_rx = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (res == NULL) { + dev_err(&pdev->dev, "no SSP TX DRCMR defined\n"); + ret = -ENODEV; + goto err_free_io; + } + ssp->drcmr_tx = res->start; + + /* PXA2xx/3xx SSP ports starts from 1 and the internal pdev->id + * starts from 0, do a translation here + */ + ssp->port_id = pdev->id + 1; + ssp->use_count = 0; + ssp->type = (int)id->driver_data; + + mutex_lock(&ssp_lock); + list_add(&ssp->node, &ssp_list); + mutex_unlock(&ssp_lock); + + platform_set_drvdata(pdev, ssp); + return 0; + +err_free_io: + iounmap(ssp->mmio_base); +err_free_mem: + release_mem_region(res->start, res->end - res->start + 1); +err_free_clk: + clk_put(ssp->clk); +err_free: + kfree(ssp); + return ret; +} + +static int __devexit ssp_remove(struct platform_device *pdev) +{ + struct resource *res; + struct ssp_device *ssp; + + ssp = platform_get_drvdata(pdev); + if (ssp == NULL) + return -ENODEV; + + iounmap(ssp->mmio_base); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, res->end - res->start + 1); + + clk_put(ssp->clk); + + mutex_lock(&ssp_lock); + list_del(&ssp->node); + mutex_unlock(&ssp_lock); + + kfree(ssp); + return 0; +} + +static const struct platform_device_id ssp_id_table[] = { + { "pxa25x-ssp", PXA25x_SSP }, + { "pxa25x-nssp", PXA25x_NSSP }, + { "pxa27x-ssp", PXA27x_SSP }, + { }, +}; + +static struct platform_driver ssp_driver = { + .probe = ssp_probe, + .remove = __devexit_p(ssp_remove), + .driver = { + .owner = THIS_MODULE, + .name = "pxa2xx-ssp", + }, + .id_table = ssp_id_table, +}; + +static int __init pxa_ssp_init(void) +{ + return platform_driver_register(&ssp_driver); +} + +static void __exit pxa_ssp_exit(void) +{ + platform_driver_unregister(&ssp_driver); +} + +arch_initcall(pxa_ssp_init); +module_exit(pxa_ssp_exit); + +#ifdef CONFIG_PXA_SSP_LEGACY +EXPORT_SYMBOL(ssp_write_word); +EXPORT_SYMBOL(ssp_read_word); +EXPORT_SYMBOL(ssp_flush); +EXPORT_SYMBOL(ssp_enable); +EXPORT_SYMBOL(ssp_disable); +EXPORT_SYMBOL(ssp_save_state); +EXPORT_SYMBOL(ssp_restore_state); +EXPORT_SYMBOL(ssp_init); +EXPORT_SYMBOL(ssp_exit); +EXPORT_SYMBOL(ssp_config); +#endif + +MODULE_DESCRIPTION("PXA SSP driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index cf00df9..ef75995 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -33,7 +33,7 @@ #include <mach/dma.h> #include <mach/regs-ssp.h> #include <mach/audio.h> -#include <mach/ssp.h> +#include <plat/ssp.h> #include "pxa2xx-pcm.h" #include "pxa-ssp.h" -- 1.5.6.5

1 0

[PATCH] ASoC: enable ssp on aspenite
by Haojian Zhuang 10 Mar '10

10 Mar '10

Append ssp driver for pxa168 since PLL could be generated by a new way. Append aspenite also that could support pxa168-ssp. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 21 ++ sound/soc/pxa/Makefile | 5 + sound/soc/pxa/aspenite.c | 206 ++++++++++++++ sound/soc/pxa/pxa168-ssp.c | 645 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/pxa168-ssp.h | 29 ++ 5 files changed, 906 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/aspenite.c create mode 100644 sound/soc/pxa/pxa168-ssp.c create mode 100644 sound/soc/pxa/pxa168-ssp.h diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 376e14a..71778be 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,3 +1,15 @@ +config SND_PXA168_SOC + tristate "SoC Audio for the Marvell chip" + depends on ARCH_MMP + select SND_PXA2XX_LIB + help + Say Y or M if you want to add support for codecs attached to + the PXA168 I2S or SSP interface. You will also need + to select the audio interfaces to support below. + +config SND_PXA168_SOC_SSP + tristate + config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA @@ -108,6 +120,15 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_ASPENITE + tristate "SoC Audio support for Marvell Aspenite" + depends on SND_PXA168_SOC && MACH_ASPENITE + select SND_PXA168_SOC_SSP + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on the + Marvell Aspenite reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index f3e08fd..46e9334 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,13 +3,17 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-pxa168-ssp-objs := pxa168-ssp.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o +obj-$(CONFIG_SND_PXA168_SOC) += snd-soc-pxa2xx.o +obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support +snd-soc-aspenite-objs := aspenite.o snd-soc-corgi-objs := corgi.o snd-soc-poodle-objs := poodle.o snd-soc-tosa-objs := tosa.o @@ -36,6 +40,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o +obj-$(CONFIG_SND_SOC_ASPENITE) += snd-soc-aspenite.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/aspenite.c b/sound/soc/pxa/aspenite.c new file mode 100644 index 0000000..12594fa --- /dev/null +++ b/sound/soc/pxa/aspenite.c @@ -0,0 +1,206 @@ +/* + * aspenite.c -- SoC audio for Aspenite + * + * Copyright (C) 2009-2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/ssp.h> + +#include "../codecs/wm8753.h" +#include "pxa2xx-pcm.h" +#include "pxa168-ssp.h" + +static struct snd_soc_card aspenite; + +/* aspenite machine dapm widgets */ +static const struct snd_soc_dapm_widget aspenite_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Headset Speaker", NULL), + SND_SOC_DAPM_LINE("Line LIN", NULL), + SND_SOC_DAPM_LINE("Line RIN", NULL), +}; + +/* aspenite machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Headphone connected to LOUT1/ROUT1 */ + {"Headphone", NULL, "LOUT1"}, + {"Headphone", NULL, "ROUT1"}, + + /* Speaker connected to LOUT2/OUT4 & OUT3/ROUT2 */ + {"Headset Speaker", NULL, "LOUT2"}, + {"Headset Speaker", NULL, "OUT4"}, + {"Headset Speaker", NULL, "OUT3"}, + {"Headset Speaker", NULL, "ROUT2"}, + + /* Line connected to LINE1/LINE2 */ + {"Line LIN", NULL, "LINE1"}, + {"Line RIN", NULL, "LINE2"}, + + /* Mic */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_aspenite_controls[] = { + SOC_DAPM_PIN_SWITCH("Headset Speaker"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Line LIN"), + SOC_DAPM_PIN_SWITCH("Line RIN"), +}; + +static int aspenite_hifi_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; + unsigned int rate, width, channel; + int index, mclk, ret; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + channel = params_channels(params); + ret = seek_mclk_conf(rate, width, channel); + if (ret < 0) + return ret; + index = ret; + mclk = get_mclk(ret); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_IF + | SND_SOC_DAIFMT_CBS_CFS); + 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_IF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA168_ASYSCLK_MASTER, index, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static int aspenite_hifi_hw_free(struct snd_pcm_substream *substream) +{ + return 0; +} + +static struct snd_soc_ops aspenite_hifi_ops = { + .hw_params = aspenite_hifi_hw_params, + .hw_free = aspenite_hifi_hw_free, +}; + +static int aspenite_wm8753_init(struct snd_soc_codec *codec) +{ + int ret; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "MONO1"); + snd_soc_dapm_nc_pin(codec, "MONO2"); + snd_soc_dapm_nc_pin(codec, "RXP"); + snd_soc_dapm_nc_pin(codec, "RXN"); + + snd_soc_dapm_new_controls(codec, aspenite_dapm_widgets, + ARRAY_SIZE(aspenite_dapm_widgets)); + ret = snd_soc_add_controls(codec, wm8753_aspenite_controls, + ARRAY_SIZE(wm8753_aspenite_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* Static setup for now */ + snd_soc_dapm_enable_pin(codec, "Headset Speaker"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headphone"); + + snd_soc_dapm_sync(codec); + return 0; +} + +static struct snd_soc_dai_link aspenite_dai[] = { + { + .name = "WM8753 HiFi", + .stream_name = "WM8753 HiFi", + .cpu_dai = &pxa168_ssp_dai[PXA168_DAI_SSP1], + .codec_dai = &wm8753_dai[0], + .init = aspenite_wm8753_init, + .ops = &aspenite_hifi_ops, + }, +}; + +static struct snd_soc_card aspenite = { + .name = "Aspenite", + .platform = &pxa2xx_soc_platform, + .dai_link = aspenite_dai, + .num_links = ARRAY_SIZE(aspenite_dai), +}; + +static struct snd_soc_device aspenite_snd_devdata = { + .card = &aspenite, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *aspenite_snd_device; + +static int __init aspenite_init(void) +{ + int ret; + + aspenite_snd_device = platform_device_alloc("soc-audio", -1); + if (!aspenite_snd_device) + return -ENOMEM; + + platform_set_drvdata(aspenite_snd_device, &aspenite_snd_devdata); + aspenite_snd_devdata.dev = &aspenite_snd_device->dev; + + ret = platform_device_add(aspenite_snd_device); + if (ret) + platform_device_put(aspenite_snd_device); + return ret; +} +module_init(aspenite_init); + +static void __exit aspenite_exit(void) +{ + platform_device_unregister(aspenite_snd_device); +} +module_exit(aspenite_exit); + +MODULE_DESCRIPTION("ALSA SoC WM8753 Aspenite"); +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa168-ssp.c b/sound/soc/pxa/pxa168-ssp.c new file mode 100644 index 0000000..2b20979 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.c @@ -0,0 +1,645 @@ +/* + * pxa168-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2009-2010 Marvell International 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. + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <asm/irq.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <mach/regs-ssp.h> +#include <mach/regs-apbc.h> +#include <mach/regs-apmu.h> +#include <mach/regs-mpmu.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa168-ssp.h" + +/* + * SSP audio private data + */ +struct ssp_priv { + struct ssp_device *ssp; + unsigned int sysclk; + int dai_fmt; +#ifdef CONFIG_PM + uint32_t cr0; + uint32_t cr1; + uint32_t to; + uint32_t psp; +#endif +}; + +struct pxa2xx_pcm_dma_data { + struct pxa2xx_pcm_dma_params params; + char name[20]; +}; + +struct ssp_mclk { + unsigned int rate; + unsigned int format; + unsigned int channel; + unsigned int mclk; + unsigned int mclk_denom; + unsigned int mclk_num; + unsigned int bclk; + unsigned int bclk_denom; + unsigned int bclk_num; +}; + +/* + * This table is used while CPU is clock master. + * MCLK = 312MHz * (ASYSCLK_DENOM + 1) / ASYSCLK_NUM + * BCLK = 2 * MCLK * (SSPSCLK_DENOM + 1) / SSPSCLK_NUM + */ +static const struct ssp_mclk mclk_conf[] = { + /* rate, fmt, chn, mclk, den, num, bclk, den, num */ + {96000, 16, 2, 12288000, 63, 1625, 3072000, 1, 2}, + {96000, 16, 1, 12288000, 63, 1625, 3072000, 1, 8}, + {88200, 16, 2, 11289600, 293, 8125, 2822400, 1, 2}, + {88200, 16, 1, 11289600, 293, 8125, 2822400, 1, 8}, + {48000, 16, 2, 12288000, 63, 1625, 1536000, 1, 4}, + {48000, 16, 1, 12288000, 63, 1625, 1536000, 1, 16}, + {44100, 16, 2, 11289600, 293, 8125, 1411200, 1, 4}, + {44100, 16, 1, 11289600, 293, 8125, 1411200, 1, 16}, + {32000, 16, 2, 12288000, 63, 1625, 1024000, 1, 6}, + {32000, 16, 1, 12288000, 63, 1625, 1024000, 1, 24}, + {22050, 16, 2, 11289600, 293, 8125, 705600, 1, 8}, + {22050, 16, 1, 11289600, 293, 8125, 705600, 1, 32}, + {16000, 16, 2, 12288000, 63, 1625, 512000, 1, 12}, + {16000, 16, 1, 12288000, 63, 1625, 512000, 1, 48}, + {11025, 16, 2, 11289600, 293, 8125, 352800, 1, 16}, + {11025, 16, 1, 11289600, 293, 8125, 352800, 1, 64}, + { 8000, 16, 2, 12288000, 63, 1625, 256000, 1, 24}, + { 8000, 16, 1, 12288000, 63, 1625, 256000, 1, 96}, +}; + +/* Seek the index of MCLK configuration table */ +int seek_mclk_conf(int rate, int format, int channel) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mclk_conf); i++) { + if ((mclk_conf[i].rate == rate) + && (mclk_conf[i].format == format) + && (mclk_conf[i].channel == channel)) + return i; + } + return -EINVAL; +} + +/* Get the MCLK frequency */ +int get_mclk(int i) +{ + if ((i < 0) || (i >= ARRAY_SIZE(mclk_conf))) + return -EINVAL; + return mclk_conf[i].mclk; +} + +/* Get the BCLK frequency */ +int get_bclk(int i) +{ + if ((i < 0) || (i >= ARRAY_SIZE(mclk_conf))) + return -EINVAL; + return mclk_conf[i].bclk; +} + +static void dump_registers(struct ssp_device *ssp) +{ + dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", + ssp_read_reg(ssp, SSCR0), ssp_read_reg(ssp, SSCR1), + ssp_read_reg(ssp, SSTO)); + + dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x\n", + ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR)); +} + +static void ssp_enable(struct ssp_device *ssp) +{ + uint32_t sscr0; + + sscr0 = __raw_readl(ssp->mmio_base + SSCR0) | SSCR0_SSE; + __raw_writel(sscr0, ssp->mmio_base + SSCR0); +} + +static void ssp_disable(struct ssp_device *ssp) +{ + uint32_t sscr0; + + sscr0 = __raw_readl(ssp->mmio_base + SSCR0) & ~SSCR0_SSE; + __raw_writel(sscr0, ssp->mmio_base + SSCR0); +} + +static struct pxa2xx_pcm_dma_params * +ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +{ + struct pxa2xx_pcm_dma_data *dma; + + dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL); + if (dma == NULL) + return NULL; + + snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id, + width4 ? "32-bit" : "16-bit", out ? "out" : "in"); + + dma->params.name = dma->name; + dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx); + dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) : + (DCMD_INCTRGADDR | DCMD_FLOWSRC)) | + (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16; + dma->params.dev_addr = ssp->phys_base + SSDR; + + return &dma->params; +} + +static int pxa168_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int ret = 0; + + if (!cpu_dai->active) { + clk_enable(ssp->clk); + ssp_disable(ssp); + } + return ret; +} + +static void pxa168_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + + if (!cpu_dai->active) { + ssp_disable(ssp); + clk_disable(ssp->clk); + } +} + +#ifdef CONFIG_PM +static int pxa168_ssp_suspend(struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + + if (!cpu_dai->active) + return 0; + + priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); + priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); + priv->to = __raw_readl(ssp->mmio_base + SSTO); + priv->psp = __raw_readl(ssp->mmio_base + SSPSP); + + ssp_disable(ssp); + clk_disable(ssp->clk); + return 0; +} + +static int pxa168_ssp_resume(struct snd_soc_dai *cpu_dai) +{ + struct ssp_priv *priv = cpu_dai->private_data; + + if (!cpu_dai->active) + return 0; + + __raw_writel(sssr, ssp->mmio_base + SSSR); + + __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); + __raw_writel(priv->cr1, ssp->mmio_base + SSCR1); + __raw_writel(priv->to, ssp->mmio_base + SSTO); + __raw_writel(priv->psp, ssp->mmio_base + SSPSP); + __raw_writel(priv->cr0 | SSCR0_SSE, ssp->mmio_base + SSCR0); + return 0; +} + +#else +#define pxa168_ssp_suspend NULL +#define pxa168_ssp_resume NULL +#endif + +/* + * Set the SSP ports SYSCLK only from Audio SYSCLK. + */ +static int pxa168_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + unsigned int sscr0, data, asysdr, asspdr; + + dev_dbg(&ssp->pdev->dev, "%s id: %d, clk_id %d, freq %u\n", + __func__, cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA168_ASYSCLK_MASTER: + case PXA168_ASYSCLK_SLAVE: + break; + default: + return -EINVAL; + } + + /* freq is the index of mclk_conf table */ + if ((freq < 0) || (freq >= ARRAY_SIZE(mclk_conf))) { + dev_warn(&ssp->pdev->dev, "Wrong frequency index:%d\n", freq); + return -EINVAL; + } + asysdr = (mclk_conf[freq].mclk_num << 16) + | mclk_conf[freq].mclk_denom; + asspdr = 0; + if (clk_id == PXA168_ASYSCLK_MASTER) + asspdr = (mclk_conf[freq].bclk_num << 16) + | mclk_conf[freq].bclk_denom; + + ssp_disable(ssp); + clk_disable(ssp->clk); /* SSP port internal clock */ + + /* clear ECS, NCS, MOD, ACS */ + sscr0 = ssp_read_reg(ssp, SSCR0); + data = sscr0 & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (sscr0 != data) + ssp_write_reg(ssp, SSCR0, data); + + /* update divider register in MPMU */ + __raw_writel(asysdr, MPMU_ASYSDR); + __raw_writel(asspdr, MPMU_ASSPDR); + + clk_enable(ssp->clk); /* SSP port internal clock */ + ssp_enable(ssp); + return 0; +} + +/* + * Tristate the SSP DAI lines + */ +static int pxa168_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, + int tristate) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 sscr1; + + sscr1 = ssp_read_reg(ssp, SSCR1); + if (tristate) + sscr1 &= ~SSCR1_TTE; + else + sscr1 |= SSCR1_TTE; + ssp_write_reg(ssp, SSCR1, sscr1); + + return 0; +} + +/* + * Set up the SSP DAI format. + * The SSP Port must be inactive before calling this function as the + * physical interface format is changed. + */ +static int pxa168_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 sscr0; + u32 sscr1; + u32 sspsp; + + /* check if we need to change anything at all */ + if (priv->dai_fmt == fmt) + return 0; + + ssp_disable(ssp); + + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr1 = SSCR1_RFT(8) | SSCR1_TFT(7); + sspsp = 0; + + /* we can only change the settings if the port is not in use */ + if (sscr0 & SSCR0_SSE) { + dev_err(&ssp->pdev->dev, + "can't change hardware dai format: stream is in use"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + sscr1 |= SSCR1_SCLKDIR; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspsp |= SSPSP_SFRMP; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + default: + return -EINVAL; + } + + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSCR1, sscr1); + ssp_write_reg(ssp, SSPSP, sspsp); + ssp_enable(ssp); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + priv->dai_fmt = fmt; + + dump_registers(ssp); + + return 0; +} + +static int pxa168_ssp_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int width = snd_pcm_format_physical_width(params_format(params)); + int channels = params_channels(params); + int dma_16b = 0, stream_out, data_size; + u32 sscr0, sspsp; + + /* generate correct DMA params */ + if (cpu_dai->dma_data) + kfree(cpu_dai->dma_data); + + if ((width == 16) && (params_channels(params) == 1)) + dma_16b = 1; + stream_out = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0; + cpu_dai->dma_data = ssp_get_dma_params(ssp, dma_16b, stream_out); + + /* clear selected SSP bits */ + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_DSS_MASK | SSCR0_EDSS); + + /* data_size should only be 16-bit or 32-bit because of DMA */ + data_size = width * channels; + switch (data_size) { + case 16: + sscr0 |= SSCR0_DSS(16); + break; + case 32: + sscr0 |= (SSCR0_EDSS | SSCR0_DSS(16)); + break; + } + + ssp_disable(ssp); + sspsp = ssp_read_reg(ssp, SSPSP); + sspsp &= ~SSPSP_TIMING_MASK; + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* The polarity of frame sync should be inverted at here. */ + sspsp |= SSPSP_SFRMWDTH(width); + if (channels == 1) { + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP((width - 1) & 0x3); + sspsp |= SSPSP_EDMYSTOP(((width - 1) >> 2) & 0x7); + } else if (channels == 2) { + if (width == 32) { + dev_err(&ssp->pdev->dev, "can't support %d-" + "data with %-channels in I2S mode\n", + width, channels); + return -EINVAL; + } + sspsp |= SSPSP_FSRT; + } + break; + case SND_SOC_DAIFMT_RIGHT_J: + /* Right Justified mode doesn't support 32-bit data */ + if (params_format(params) == SNDRV_PCM_FORMAT_S32_LE) + return -EINVAL; + break; + case SND_SOC_DAIFMT_LEFT_J: + sspsp |= SSPSP_SFRMWDTH(width); + break; + } + + /* update SSP register at the same time */ + ssp_write_reg(ssp, SSCR0, sscr0); + ssp_write_reg(ssp, SSPSP, sspsp); + ssp_enable(ssp); + + dump_registers(ssp); + + return 0; +} + +static int pxa168_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + ssp_enable(ssp); + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + val = ssp_read_reg(ssp, SSSR); + ssp_write_reg(ssp, SSSR, val); + break; + case SNDRV_PCM_TRIGGER_START: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= SSCR1_TSRE; + else + val |= SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + ssp_enable(ssp); + break; + case SNDRV_PCM_TRIGGER_STOP: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + ssp_disable(ssp); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + val = ssp_read_reg(ssp, SSCR1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val &= ~SSCR1_TSRE; + else + val &= ~SSCR1_RSRE; + ssp_write_reg(ssp, SSCR1, val); + break; + + default: + ret = -EINVAL; + } + + dump_registers(ssp); + + return ret; +} + +static int pxa168_ssp_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_priv *priv; + int ret; + + priv = kzalloc(sizeof(struct ssp_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { + ret = -ENODEV; + goto err_priv; + } + + priv->dai_fmt = (unsigned int) -1; + dai->private_data = priv; + + return 0; + +err_priv: + kfree(priv); + return ret; +} + +static void pxa168_ssp_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct ssp_priv *priv = dai->private_data; + ssp_free(priv->ssp); +} + +#define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops pxa168_ssp_dai_ops = { + .startup = pxa168_ssp_startup, + .shutdown = pxa168_ssp_shutdown, + .trigger = pxa168_ssp_trigger, + .hw_params = pxa168_ssp_hw_params, + .set_sysclk = pxa168_ssp_set_dai_sysclk, + .set_fmt = pxa168_ssp_set_dai_fmt, + .set_tristate = pxa168_ssp_set_dai_tristate, +}; + +#define PXA168_SSP_DAI(_id) \ +{ \ + .name = "pxa168-ssp" #_id, \ + .id = 0, \ + .probe = pxa168_ssp_probe, \ + .remove = pxa168_ssp_remove, \ + .suspend = pxa168_ssp_suspend, \ + .resume = pxa168_ssp_resume, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA_SSP_RATES, \ + .formats = PXA_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA_SSP_RATES, \ + .formats = PXA_SSP_FORMATS, \ + }, \ + .ops = &pxa168_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa168_ssp_dai[] = { + PXA168_SSP_DAI(0), + PXA168_SSP_DAI(1), + PXA168_SSP_DAI(2), + PXA168_SSP_DAI(3), + PXA168_SSP_DAI(4), +}; +EXPORT_SYMBOL_GPL(pxa168_ssp_dai); + +static int __init pxa168_ssp_init(void) +{ + return snd_soc_register_dais(pxa168_ssp_dai, ARRAY_SIZE(pxa168_ssp_dai)); +} +module_init(pxa168_ssp_init); + +static void __exit pxa168_ssp_exit(void) +{ + snd_soc_unregister_dais(pxa168_ssp_dai, ARRAY_SIZE(pxa168_ssp_dai)); +} +module_exit(pxa168_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("PXA168 SSP SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/pxa/pxa168-ssp.h b/sound/soc/pxa/pxa168-ssp.h new file mode 100644 index 0000000..10131fc --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.h @@ -0,0 +1,29 @@ +/* + * ASoC PXA168 SSP port support + * + * 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. + */ + +#ifndef _PXA168_SSP_H +#define _PXA168_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA168_DAI_SSP1 0 +#define PXA168_DAI_SSP2 1 +#define PXA168_DAI_SSP3 2 +#define PXA168_DAI_SSP4 3 +#define PXA168_DAI_SSP5 4 + +/* PXA168 SSP SYSCLK source */ +#define PXA168_ASYSCLK_MASTER 0 +#define PXA168_ASYSCLK_SLAVE 1 + +extern struct snd_soc_dai pxa168_ssp_dai[5]; + +extern int seek_mclk_conf(int rate, int format, int channel); +extern int get_mclk(int i); +extern int get_bclk(int i); + +#endif -- 1.5.6.5

1 0

[alsa-devel] [PATCH 2/2] ASoC: da7210: Add 11025/22050/44100/88200 rate support
by Kuninori Morimoto 10 Mar '10

10 Mar '10

This driver USE PLL for 11025/22050/44100/88200 rate. To enable switching to bypass mode, PLL is always turned on. Special thanks to Phil Signed-off-by: Phil Edworthy <Phil.Edworthy(a)renesas.com> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/codecs/da7210.c | 87 ++++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 83 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 3bd867d..4083c7e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -55,8 +55,14 @@ #define DA7210_DAI_SRC_SEL 0x25 #define DA7210_DAI_CFG1 0x26 #define DA7210_DAI_CFG3 0x28 +#define DA7210_PLL_DIV1 0x29 +#define DA7210_PLL_DIV2 0x2A #define DA7210_PLL_DIV3 0x2B #define DA7210_PLL 0x2C +#define DA7210_A_HID_UNLOCK 0x8A +#define DA7210_A_TEST_UNLOCK 0x8B +#define DA7210_A_PLL1 0x90 +#define DA7210_A_CP_MODE 0xA7 /* STARTUP1 bit fields */ #define DA7210_SC_MST_EN (1 << 0) @@ -124,13 +130,17 @@ /* PLL bit fields */ #define DA7210_PLL_FS_MASK (0xF << 0) #define DA7210_PLL_FS_8000 (0x1 << 0) +#define DA7210_PLL_FS_11025 (0x2 << 0) #define DA7210_PLL_FS_12000 (0x3 << 0) #define DA7210_PLL_FS_16000 (0x5 << 0) +#define DA7210_PLL_FS_22050 (0x6 << 0) #define DA7210_PLL_FS_24000 (0x7 << 0) #define DA7210_PLL_FS_32000 (0x9 << 0) +#define DA7210_PLL_FS_44100 (0xA << 0) #define DA7210_PLL_FS_48000 (0xB << 0) +#define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) - +#define DA7210_PLL_EN (0x1 << 7) #define DA7210_VERSION "0.0.1" @@ -249,7 +259,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u32 dai_cfg1; u32 hpf_reg, hpf_mask, hpf_value; - u32 fs; + u32 fs, bypass; /* set DAI source to Left and Right ADC */ da7210_write(codec, DA7210_DAI_SRC_SEL, @@ -281,38 +291,76 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, fs = DA7210_PLL_FS_8000; hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 11025: + fs = DA7210_PLL_FS_11025; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + bypass = DA7210_PLL_BYP; + break; + case 22050: + fs = DA7210_PLL_FS_22050; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; hpf_mask = DA7210_VOICE_EN; hpf_value = 0; + bypass = DA7210_PLL_BYP; + break; + case 44100: + fs = DA7210_PLL_FS_44100; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; hpf_mask = DA7210_VOICE_EN; hpf_value = 0; + bypass = DA7210_PLL_BYP; + break; + case 88200: + fs = DA7210_PLL_FS_88200; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; hpf_mask = DA7210_VOICE_EN; hpf_value = 0; + bypass = DA7210_PLL_BYP; break; default: return -EINVAL; } + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, + DA7210_SC_MST_EN, DA7210_SC_MST_EN); return 0; } @@ -444,9 +492,23 @@ static int da7210_init(struct da7210_priv *da7210) /* FIXME * * This driver use fixed value here + * And below settings expects MCLK = 12.288MHz + * + * When you select different MCLK, please check... + * DA7210_PLL_DIV1 val + * DA7210_PLL_DIV2 val + * DA7210_PLL_DIV3 val + * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx */ /* + * make sure that DA7210 use bypass mode before start up + */ + da7210_write(codec, DA7210_STARTUP1, 0); + da7210_write(codec, DA7210_PLL_DIV3, + DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + + /* * ADC settings */ @@ -482,9 +544,26 @@ static int da7210_init(struct da7210_priv *da7210) /* Diable PLL and bypass it */ da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* Bypass PLL and set MCLK freq rang to 10-20MHz */ - da7210_write(codec, DA7210_PLL_DIV3, + /* + * If 48kHz sound came, it use bypass mode, + * and when it is 44.1kHz, it use PLL. + * + * This time, this driver sets PLL always ON + * and controls bypass/PLL mode by switching + * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. + * see da7210_hw_params + */ + da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ + da7210_write(codec, DA7210_PLL_DIV2, 0x99); + da7210_write(codec, DA7210_PLL_DIV3, 0x0A | DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); + + /* As suggested by Dialog */ + da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); + da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); + da7210_write(codec, DA7210_A_PLL1, 0x01); + da7210_write(codec, DA7210_A_CP_MODE, 0x7C); /* Activate all enabled subsystem */ da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); -- 1.6.3.3

2 3

[alsa-devel] [PATCH] alsa: provide a more useful get_unmapped_area handler for pcm
by Mike Frysinger 10 Mar '10

10 Mar '10

From: Daniel Glöckner <dg(a)emlix.com> Shared memory mappings on nommu machines require a get_unmapped_area file operation that suggests an address for the mapping. The current implementation returns 0 and thus forces the driver to implement an mmap handler that fixes up the start and end address of the vma. This patch returns the address of the dma buffer, so it should work out of the box for all drivers that use the snd_pcm_runtime->dma_area pointer. Addresses for mapping the status and control pages are returned as well, but to make those work the conditional compilation of snd_pcm_mmap_{status,control} would need to be revised. URL: http://thread.gmane.org/gmane.linux.alsa.devel/61230 Signed-off-by: Daniel Glöckner <dg(a)emlix.com> Signed-off-by: Cliff Cai <cliff.cai(a)analog.com> Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- this has been posted before but didnt seem to get any feedback ... sound/core/pcm_native.c | 30 ++++++++++++++++++++++-------- 1 files changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8728876..9681518 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3434,14 +3434,28 @@ out: #endif /* CONFIG_SND_SUPPORT_OLD_API */ #ifndef CONFIG_MMU -unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr, - unsigned long len, unsigned long pgoff, - unsigned long flags) -{ - return 0; +static unsigned long snd_pcm_get_unmapped_area(struct file *file, + unsigned long addr, + unsigned long len, + unsigned long pgoff, + unsigned long flags) +{ + struct snd_pcm_file *pcm_file = file->private_data; + struct snd_pcm_substream *substream = pcm_file->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long offset = pgoff << PAGE_SHIFT; + + switch (offset) { + case SNDRV_PCM_MMAP_OFFSET_STATUS: + return (unsigned long)runtime->status; + case SNDRV_PCM_MMAP_OFFSET_CONTROL: + return (unsigned long)runtime->control; + default: + return (unsigned long)runtime->dma_area + offset; + } } #else -# define dummy_get_unmapped_area NULL +# define snd_pcm_get_unmapped_area NULL #endif /* @@ -3460,7 +3474,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, }, { .owner = THIS_MODULE, @@ -3473,6 +3487,6 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, } }; -- 1.7.0.2

2 1

[alsa-devel] [PATCH 20/20] ASoC: SMDK64XX: Switch to IISv4 CPU driver
by Jassi Brar 10 Mar '10

10 Mar '10

Switch the MACHINE driver to use IISv4 CPU dai. Remove BROKEN dependency now that we have proper CPU driver available. Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4 controller. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 7 +++---- sound/soc/s3c24xx/smdk64xx_wm8580.c | 6 ++---- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index c0c7edf..2ac1f33 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -64,12 +64,11 @@ config SND_S3C24XX_SOC_JIVE_WM8750 config SND_S3C64XX_SOC_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK64XX" - depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) - depends on BROKEN + depends on SND_S3C24XX_SOC && MACH_SMDK6410 select SND_SOC_WM8580 - select SND_S3C64XX_SOC_I2S + select SND_S3C64XX_SOC_I2S_V4 help - Sat Y if you want to add support for SoC audio on the SMDK64XX. + Say Y if you want to add support for SoC audio on the SMDK64XX. config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index efe4901..07e8e51 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -22,8 +22,6 @@ #include "s3c-dma.h" #include "s3c64xx-i2s.h" -#define S3C64XX_I2S_V4 2 - /* SMDK64XX has a 12MHZ crystal attached to WM8580 */ #define SMDK64XX_WM8580_FREQ 12000000 @@ -215,7 +213,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { { /* Primary Playback i/f */ .name = "WM8580 PAIF RX", .stream_name = "Playback", - .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .cpu_dai = &s3c64xx_i2s_v4_dai, .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], .init = smdk64xx_wm8580_init_paifrx, .ops = &smdk64xx_ops, @@ -223,7 +221,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { { /* Primary Capture i/f */ .name = "WM8580 PAIF TX", .stream_name = "Capture", - .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .cpu_dai = &s3c64xx_i2s_v4_dai, .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], .init = smdk64xx_wm8580_init_paiftx, .ops = &smdk64xx_ops, -- 1.6.2.5

1 0

[alsa-devel] [PATCH 19/20] ASoC: S3C64XX: IISv4: Add CPU driver
by Jassi Brar 10 Mar '10

10 Mar '10

Add the CPU driver for the IISv4 block found on S3C6410. For now, the driver is almost a copy of s3c64xx-i2s.c but it should diverge as more IISv4 specific stuff is added. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 5 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 206 ++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + 4 files changed, 214 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 15fe57e..c0c7edf 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,11 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + config SND_S3C_SOC_PCM tristate diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index df071a3..81d8dc5 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -4,6 +4,7 @@ snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c-ac97-objs := s3c-ac97.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o snd-soc-s3c-pcm-objs := s3c-pcm.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c new file mode 100644 index 0000000..d1fa837 --- /dev/null +++ b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c @@ -0,0 +1,206 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c + * + * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * 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/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include <plat/gpio-bank-c.h> +#include <plat/gpio-bank-h.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2Sv4 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2Sv4 PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; +static struct s3c_i2sv2_info s3c64xx_i2sv4; + +struct snd_soc_dai s3c64xx_i2s_v4_dai; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2sv4_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); + s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); + + return 0; +} + +static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod; + + dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_dai->dma_data = i2s->dma_playback; + else + cpu_dai->dma_data = i2s->dma_capture; + + iismod = readl(i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { + .hw_params = s3c_i2sv4_hw_params, +}; + +static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + i2s = &s3c64xx_i2sv4; + dai = &s3c64xx_i2s_v4_dai; + + if (dai->dev) { + dev_dbg(dai->dev, "%s: \ + I2Sv4 instance already registered!\n", __func__); + return -EBUSY; + } + + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s-v4"; + dai->id = 0; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2sv4_probe; + dai->ops = &s3c64xx_i2sv4_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; + i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; + + i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; + i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; + i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_i2sv4_driver = { + .probe = s3c64xx_i2sv4_dev_probe, + .remove = s3c64xx_i2sv4_dev_remove, + .driver = { + .name = "s3c64xx-iis-v4", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2sv4_init(void) +{ + return platform_driver_register(&s3c64xx_i2sv4_driver); +} +module_init(s3c64xx_i2sv4_init); + +static void __exit s3c64xx_i2sv4_exit(void) +{ + platform_driver_unregister(&s3c64xx_i2sv4_driver); +} +module_exit(s3c64xx_i2sv4_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index f27ed50..7a40f43 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -37,5 +37,6 @@ struct clk; SNDRV_PCM_FMTBIT_S24_LE) extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai; #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ -- 1.6.2.5

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[alsa-devel] [PATCH 18/20] ASoC: S3C: I2S: Move set_sysclk callback to common code
by Jassi Brar 10 Mar '10

10 Mar '10

Now that we can specify feature of a particular controller, we can avoid multiple copies of same code by defining the CDCLKCON bit feature in controller specific code and detecting that flag in the code common to all controllers. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 47 +++++++++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c2412-i2s.c | 27 ---------------------- sound/soc/s3c24xx/s3c64xx-i2s.c | 43 ++--------------------------------- 3 files changed, 50 insertions(+), 67 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index ecb3054..7d2edc0 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -325,6 +325,52 @@ static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, writel(iismod, i2s->regs + S3C2412_IISMOD); pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + pr_debug("Entered %s\n", __func__); + pr_debug("%s r: IISMOD: %x \n", __func__, iismod); + + switch (clk_id) { + case S3C_I2SV2_CLKSRC_PCLK: + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_AUDIOBUS: + iismod |= S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_CDCLK: + /* Error if controller doesn't have the CDCLKCON bit */ + if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) + return -EINVAL; + + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s w: IISMOD: %x \n", __func__, iismod); + return 0; } @@ -697,6 +743,7 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) ops->hw_params = s3c_i2sv2_hw_params; ops->set_fmt = s3c2412_i2s_set_fmt; ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + ops->set_sysclk = s3c_i2sv2_set_sysclk; /* Allow overriding by (for example) IISv4 */ if (!ops->delay) diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index fe90867..e351376 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -65,32 +65,6 @@ static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { static struct s3c_i2sv2_info s3c2412_i2s; -/* - * Set S3C2412 Clock source - */ -static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); - - pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, - freq, dir); - - switch (clk_id) { - case S3C2412_CLKSRC_PCLK: - iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; - break; - case S3C2412_CLKSRC_I2SCLK: - iismod |= S3C2412_IISMOD_IMS_SYSMUX; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); - return 0; -} - static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) { return cpu_dai->private_data; @@ -172,7 +146,6 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { - .set_sysclk = s3c2412_i2s_set_sysclk, .hw_params = s3c2412_i2s_hw_params, }; diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 1a29564..03cce8a 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -54,43 +54,6 @@ static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) return cpu_dai->private_data; } -static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - switch (clk_id) { - case S3C64XX_CLKSRC_PCLK: - iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C64XX_CLKSRC_MUX: - iismod |= S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C64XX_CLKSRC_CDCLK: - switch (dir) { - case SND_SOC_CLOCK_IN: - iismod |= S3C64XX_IISMOD_CDCLKCON; - break; - case SND_SOC_CLOCK_OUT: - iismod &= ~S3C64XX_IISMOD_CDCLKCON; - break; - default: - return -EINVAL; - } - break; - - default: - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - - return 0; -} - static int s3c64xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { @@ -115,9 +78,7 @@ static int s3c64xx_i2s_probe(struct platform_device *pdev, } -static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = { - .set_sysclk = s3c64xx_i2s_set_sysclk, -}; +static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) { @@ -147,6 +108,8 @@ static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) dai->probe = s3c64xx_i2s_probe; dai->ops = &s3c64xx_i2s_dai_ops; + i2s->feature |= S3C_FEATURE_CDCLKCON; + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; -- 1.6.2.5

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[alsa-devel] [PATCH 17/20] ASoC: S3C: I2Sv2: Define new field for controller features
by Jassi Brar 10 Mar '10

10 Mar '10

In order to make s3c-i2s-v2.c manage controllers with minor quirks and variation in features, we define a per-block flag that indicates the availability/lack of a particular feature to the s3c-i2s-v2.c While adding support for new SoCs' I2S, check for the blocks of older SoCs that have similar feature and set the flag for that feature. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.h | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index 39a6db6..766f43a 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -29,10 +29,16 @@ #define S3C_I2SV2_CLKSRC_AUDIOBUS 1 #define S3C_I2SV2_CLKSRC_CDCLK 2 +/* Set this flag for I2S controllers that have the bit IISMOD[12] + * bridge/break RCLK signal and external Xi2sCDCLK pin. + */ +#define S3C_FEATURE_CDCLKCON (1 << 0) + /** * struct s3c_i2sv2_info - S3C I2S-V2 information * @dev: The parent device passed to use from the probe. * @regs: The pointer to the device registe block. + * @feature: Set of bit-flags indicating features of the controller. * @master: True if the I2S core is the I2S bit clock master. * @dma_playback: DMA information for playback channel. * @dma_capture: DMA information for capture channel. @@ -47,6 +53,8 @@ struct s3c_i2sv2_info { struct device *dev; void __iomem *regs; + u32 feature; + struct clk *iis_pclk; struct clk *iis_cclk; -- 1.6.2.5

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[alsa-devel] [PATCH 16/20] ASoC: S3C64XX: I2S: Use s3c2412 defines
by Jassi Brar 10 Mar '10

10 Mar '10

Now that the fields are defined for s3c2412, use them and avoid having multiple copies of same defines. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/regs-i2s-v2.h | 3 --- sound/soc/s3c24xx/s3c64xx-i2s.c | 4 ++-- 2 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h index f612468..ad06937 100644 --- a/sound/soc/s3c24xx/regs-i2s-v2.h +++ b/sound/soc/s3c24xx/regs-i2s-v2.h @@ -73,9 +73,6 @@ #define S3C64XX_IISMOD_BLC_24BIT (2 << 13) #define S3C64XX_IISMOD_BLC_MASK (3 << 13) -#define S3C64XX_IISMOD_IMS_PCLK (0 << 10) -#define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) - #define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) #define S3C2412_IISMOD_SLAVE (1 << 11) #define S3C2412_IISMOD_MODE_TXONLY (0 << 8) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 9343349..1a29564 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -62,11 +62,11 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C64XX_CLKSRC_PCLK: - iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX; + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_MUX: - iismod |= S3C64XX_IISMOD_IMS_SYSMUX; + iismod |= S3C2412_IISMOD_IMS_SYSMUX; break; case S3C64XX_CLKSRC_CDCLK: -- 1.6.2.5

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[alsa-devel] [PATCH 15/20] ASoC: S3C: I2Sv2: Unify i2s_get_clock callback
by Jassi Brar 10 Mar '10

10 Mar '10

Now that we have two callbacks s3c2412_i2s_get_clock & s3c64xx_i2s_get_clock doing exactly the same thing, we can define one generic s3c_i2sv2_get_clock and discard other two copies. Also, switch the users to make calls to the newly defined and generic s3c_i2sv2_get_clock Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/jive_wm8750.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 12 ++++++++++++ sound/soc/s3c24xx/s3c-i2s-v2.h | 2 ++ sound/soc/s3c24xx/s3c2412-i2s.c | 13 ------------- sound/soc/s3c24xx/s3c2412-i2s.h | 2 -- sound/soc/s3c24xx/s3c64xx-i2s.c | 12 ------------ sound/soc/s3c24xx/s3c64xx-i2s.h | 2 -- 7 files changed, 15 insertions(+), 30 deletions(-) diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 59dc2c6..304240d 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -70,7 +70,7 @@ static int jive_hw_params(struct snd_pcm_substream *substream, } s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), - s3c2412_get_iisclk()); + s3c_i2sv2_get_clock(cpu_dai)); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 1b29b4b..ecb3054 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -499,6 +499,18 @@ static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, return delay; } +struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); + /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index ea56467..39a6db6 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -60,6 +60,8 @@ struct s3c_i2sv2_info { u32 suspend_iispsr; }; +extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); + struct s3c_i2sv2_rate_calc { unsigned int clk_div; /* for prescaler */ unsigned int fs_div; /* for root frame clock */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 4996843..fe90867 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -91,19 +91,6 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, return 0; } - -struct clk *s3c2412_get_iisclk(void) -{ - struct s3c_i2sv2_info *i2s = &s3c2412_i2s; - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISMOD_IMS_SYSMUX) - return i2s->iis_cclk; - else - return i2s->iis_pclk; -} -EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); - static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) { return cpu_dai->private_data; diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 60cac00..0b5686b 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -24,8 +24,6 @@ #define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK #define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS -extern struct clk *s3c2412_get_iisclk(void); - extern struct snd_soc_dai s3c2412_i2s_dai; #endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 9bcc99d..9343349 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -91,18 +91,6 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, return 0; } -struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) - return i2s->iis_cclk; - else - return i2s->iis_pclk; -} -EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); - static int s3c64xx_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index 53d2a0a..f27ed50 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -38,6 +38,4 @@ struct clk; extern struct snd_soc_dai s3c64xx_i2s_dai[]; -extern struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai); - #endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ -- 1.6.2.5

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[alsa-devel] [PATCH 14/20] ASoC: S3C: I2Sv2: Discard redundant field iis_clk
by Jassi Brar 10 Mar '10

10 Mar '10

No need to keep redundant field iis_clk in s3c_i2sv2_info. iis_cclk and iis_pclk is all we need. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c-i2s-v2.h | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index b094d3c..ea56467 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -49,7 +49,6 @@ struct s3c_i2sv2_info { struct clk *iis_pclk; struct clk *iis_cclk; - struct clk *iis_clk; unsigned char master; -- 1.6.2.5

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[alsa-devel] [PATCH 13/20] ASoC: S3C2412: I2S: Return correct source clock
by Jassi Brar 10 Mar '10

10 Mar '10

Until now, s3c2412_get_iisclk would return NULL since iis_clk was never initialized. Return appropriate pointer as per the selection made for source clock. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/s3c2412-i2s.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index e6871f7..4996843 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -94,7 +94,13 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, struct clk *s3c2412_get_iisclk(void) { - return s3c2412_i2s.iis_clk; + struct s3c_i2sv2_info *i2s = &s3c2412_i2s; + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; } EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); -- 1.6.2.5

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[alsa-devel] [PATCH 12/20] ASoC: S3C2412: I2S: Debug IMS field
by Jassi Brar 10 Mar '10

10 Mar '10

The IMS field of s3c2412/13 is essentially the same as that of s3c64xx. That is, the IISMOD[11] bit decides Master/Slave mode and IISMOD[10] bit selects source clock for signal generation. For that reason, remove improper defines for IISMOD[11:10] field mask and define two 1bit fields that can be set independent of each other. As a consequence, corresponding fields for PLAT_S3C64XX too get to use these new defines. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/regs-i2s-v2.h | 6 ++---- sound/soc/s3c24xx/s3c-i2s-v2.c | 25 ++----------------------- sound/soc/s3c24xx/s3c2412-i2s.c | 8 ++------ 3 files changed, 6 insertions(+), 33 deletions(-) diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h index dea199e..f612468 100644 --- a/sound/soc/s3c24xx/regs-i2s-v2.h +++ b/sound/soc/s3c24xx/regs-i2s-v2.h @@ -76,10 +76,8 @@ #define S3C64XX_IISMOD_IMS_PCLK (0 << 10) #define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) -#define S3C2412_IISMOD_MASTER_INTERNAL (0 << 10) -#define S3C2412_IISMOD_MASTER_EXTERNAL (1 << 10) -#define S3C2412_IISMOD_SLAVE (2 << 10) -#define S3C2412_IISMOD_MASTER_MASK (3 << 10) +#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) +#define S3C2412_IISMOD_SLAVE (1 << 11) #define S3C2412_IISMOD_MODE_TXONLY (0 << 8) #define S3C2412_IISMOD_MODE_RXONLY (1 << 8) #define S3C2412_IISMOD_MODE_TXRX (2 << 8) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index b690078..1b29b4b 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -251,35 +251,14 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("hw_params r: IISMOD: %x \n", iismod); -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) -#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK -#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE -#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL -#endif - -#if defined(CONFIG_PLAT_S3C64XX) -/* From Rev1.1 datasheet, we have two master and two slave modes: - * IMS[11:10]: - * 00 = master mode, fed from PCLK - * 01 = master mode, fed from CLKAUDIO - * 10 = slave mode, using PCLK - * 11 = slave mode, using I2SCLK - */ -#define IISMOD_MASTER_MASK (1 << 11) -#define IISMOD_SLAVE (1 << 11) -#define IISMOD_MASTER (0 << 11) -#endif - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: i2s->master = 0; - iismod &= ~IISMOD_MASTER_MASK; - iismod |= IISMOD_SLAVE; + iismod |= S3C2412_IISMOD_SLAVE; break; case SND_SOC_DAIFMT_CBS_CFS: i2s->master = 1; - iismod &= ~IISMOD_MASTER_MASK; - iismod |= IISMOD_MASTER; + iismod &= ~S3C2412_IISMOD_SLAVE; break; default: pr_err("unknwon master/slave format\n"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 42fb663..e6871f7 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -78,14 +78,10 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, switch (clk_id) { case S3C2412_CLKSRC_PCLK: - s3c2412_i2s.master = 1; - iismod &= ~S3C2412_IISMOD_MASTER_MASK; - iismod |= S3C2412_IISMOD_MASTER_INTERNAL; + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; break; case S3C2412_CLKSRC_I2SCLK: - s3c2412_i2s.master = 0; - iismod &= ~S3C2412_IISMOD_MASTER_MASK; - iismod |= S3C2412_IISMOD_MASTER_EXTERNAL; + iismod |= S3C2412_IISMOD_IMS_SYSMUX; break; default: return -EINVAL; -- 1.6.2.5

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[alsa-devel] [PATCH 11/20] ASoC: SAMSUNG: I2S: Add bit definitions
by Jassi Brar 10 Mar '10

10 Mar '10

Define more bit definitions in the order of mainline support for the SoC. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/regs-i2s-v2.h | 41 +++++++++++++++++++++++++++++++++++++++ 1 files changed, 41 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h index 8e1a585..dea199e 100644 --- a/sound/soc/s3c24xx/regs-i2s-v2.h +++ b/sound/soc/s3c24xx/regs-i2s-v2.h @@ -20,6 +20,24 @@ #define S3C2412_IISTXD (0x10) #define S3C2412_IISRXD (0x14) +#define S5PC1XX_IISFICS 0x18 +#define S5PC1XX_IISTXDS 0x1C + +#define S5PC1XX_IISCON_SW_RST (1 << 31) +#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) +#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) +#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) +#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) +#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) +#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) + +#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) +#define S3C64XX_IISCON_FTXURINTEN (1 << 16) +#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) +#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) +#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) +#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) + #define S3C2412_IISCON_LRINDEX (1 << 11) #define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) #define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) @@ -33,6 +51,23 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) +#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) +#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) +#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) +#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) +#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ +#define S5PC1XX_IISMOD_BLCS_MASK 0x3 +#define S5PC1XX_IISMOD_BLCS_SHIFT 26 +#define S5PC1XX_IISMOD_BLCP_MASK 0x3 +#define S5PC1XX_IISMOD_BLCP_SHIFT 24 + +#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ +#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ +#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) +#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) +#define S3C64XX_IISMOD_DC2_EN (1 << 17) +#define S3C64XX_IISMOD_DC1_EN (1 << 16) #define S3C64XX_IISMOD_BLC_16BIT (0 << 13) #define S3C64XX_IISMOD_BLC_8BIT (1 << 13) #define S3C64XX_IISMOD_BLC_24BIT (2 << 13) @@ -71,9 +106,15 @@ #define S3C2412_IISPSR_PSREN (1 << 15) +#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) +#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) + #define S3C2412_IISFIC_TXFLUSH (1 << 15) #define S3C2412_IISFIC_RXFLUSH (1 << 7) #define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) #define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) +#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) +#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) + #endif /* __REGS_IISV2_H */ -- 1.6.2.5

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Re: [alsa-devel] Bug#573243: Front Mic Boost mixer control missing on 2.6.33
by maximilian attems 10 Mar '10

10 Mar '10

On Tue, Mar 09, 2010 at 08:58:28PM -0500, Anthony DeRobertis wrote: > Package: linux-2.6 > Version: 2.6.33-1~experimental.2 > Severity: normal > > This is an Asus UL30A. > > On 2.6.32, I get two Mic Boosts: One called "Mic Boost" and one called > "Front Mic Boost". The "Front Mic Boost" is for the built-in microphone > mounted at the top of the display; the other is assumably the mic input > jack (have not tested). > > On 2.6.33, the Front Mic Boost goes away, making the built-in mic > unusable. > > Attached is the output of "amixer contents" on 2.6.33. Please let me > know if you need one from 2.6.32. > please sent output of alsa-info.sh of 2.6.32 and 2.6.33 thanks

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[alsa-devel] ALSA SDL VORBIS playback problem
by Christian Wolf 09 Mar '10

09 Mar '10

I'm using SDL on an ARM based board running Linux (Olimex-AT91SAM9261) . I successfully cross-compiled, installed and tested the libraries (SDL, SDL_mixer) on the system. The sound output is done thru ALSA (SDL_AUDIODRIVER=alsa). And the ogg file decoding thru libvorbis/libvorbisfile libraries. Now I'm trying to play ogg files. First I used the Mix_LoadWAV macro from the SDL_Mixer library for playback. This worked, but the macro decompresses first the complete audio file into memory and then starts the playback. Thus leading to a long delay before playing the music. Therefore I tried a different approach with the Mix_LoadMUS macro (SDL_mixer) to play the ogg file. This function streams the files directly from the hard-disk and avoids the long decoding time. Unfortunately ALSA shows several buffer under-runs leading to a disrupted playback. To fix this I tried different buffer sizes, but the problem is still existent. The code works fine on my host system, but not on the ARM target. And i'm not sure if these problems are caused by the ALSA lib? Any sort of help would be great! Thanks in advance Christian ___________________________________________________________ GRATIS für alle WEB.DE-Nutzer: Die maxdome Movie-FLAT! Jetzt freischalten unter http://movieflat.web.de

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[alsa-devel] [PATCH 1/2] ASoC: da7210: Add 8/12/16/24/32/48/96 kHz rate support
by Kuninori Morimoto 09 Mar '10

09 Mar '10

Signed-off-by: Phil Edworthy <Phil.Edworthy(a)renesas.com> Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/codecs/da7210.c | 67 ++++++++++++++++++++++++++++++++------------- 1 files changed, 48 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cf2975a..3bd867d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -74,15 +74,14 @@ /* INMIX_R bit fields */ #define DA7210_IN_R_EN (1 << 7) -/* ADC_HPF bit fields */ -#define DA7210_ADC_VOICE_EN (1 << 7) - /* ADC bit fields */ #define DA7210_ADC_L_EN (1 << 3) #define DA7210_ADC_R_EN (1 << 7) -/* DAC_HPF fields */ -#define DA7210_DAC_VOICE_EN (1 << 7) +/* DAC/ADC HPF fields */ +#define DA7210_VOICE_F0_MASK (0x7 << 4) +#define DA7210_VOICE_F0_25 (1 << 4) +#define DA7210_VOICE_EN (1 << 7) /* DAC_SEL bit fields */ #define DA7210_DAC_L_SRC_DAI_L (4 << 0) @@ -123,7 +122,15 @@ #define DA7210_PLL_BYP (1 << 6) /* PLL bit fields */ -#define DA7210_PLL_FS_48000 (11 << 0) +#define DA7210_PLL_FS_MASK (0xF << 0) +#define DA7210_PLL_FS_8000 (0x1 << 0) +#define DA7210_PLL_FS_12000 (0x3 << 0) +#define DA7210_PLL_FS_16000 (0x5 << 0) +#define DA7210_PLL_FS_24000 (0x7 << 0) +#define DA7210_PLL_FS_32000 (0x9 << 0) +#define DA7210_PLL_FS_48000 (0xB << 0) +#define DA7210_PLL_FS_96000 (0xF << 0) + #define DA7210_VERSION "0.0.1" @@ -241,7 +248,8 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; u32 dai_cfg1; - u32 reg, mask; + u32 hpf_reg, hpf_mask, hpf_value; + u32 fs; /* set DAI source to Left and Right ADC */ da7210_write(codec, DA7210_DAI_SRC_SEL, @@ -265,25 +273,46 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); - /* FIXME - * - * It support 48K only now - */ + hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? + DA7210_DAC_HPF : DA7210_ADC_HPF; + switch (params_rate(params)) { + case 8000: + fs = DA7210_PLL_FS_8000; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + break; + case 12000: + fs = DA7210_PLL_FS_12000; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + break; + case 16000: + fs = DA7210_PLL_FS_16000; + hpf_mask = DA7210_VOICE_F0_MASK | DA7210_VOICE_EN; + hpf_value = DA7210_VOICE_F0_25 | DA7210_VOICE_EN; + break; + case 32000: + fs = DA7210_PLL_FS_32000; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + break; case 48000: - if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { - reg = DA7210_DAC_HPF; - mask = DA7210_DAC_VOICE_EN; - } else { - reg = DA7210_ADC_HPF; - mask = DA7210_ADC_VOICE_EN; - } + fs = DA7210_PLL_FS_48000; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; + break; + case 96000: + fs = DA7210_PLL_FS_96000; + hpf_mask = DA7210_VOICE_EN; + hpf_value = 0; break; default: return -EINVAL; } - snd_soc_update_bits(codec, reg, mask, 0); + snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value); + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); return 0; } -- 1.6.3.3

2 1

[alsa-devel] semantics of SNDRV_PCM_INFO_BATCH (was: Re: safe support for rewind in ALSA
by Raymond Yau 09 Mar '10

09 Mar '10

>> Hi, >> On Sun, 21 Feb 2010, Lennart Poettering wrote: >>>> But you can query the SNDRV_PCM_INFO_BATCH flag (it's set by e.g. the USB >>>> driver) and adjust your hrtimer-based application's logic based on that [...] >>>>> This mean that the wake up time cannot be calculated using as number of >>>>> sample/rate since the fuction is not linear especially when using max buffer >>>>> size , min period --> max period size is much greater than the watermark >>>> >>>> Yep, that's what SNDRV_PCM_INFO_BATCH flag warns you about. >>> >>> Hmm, could you elaborate a little about SNDRV_PCM_INFO_BATCH? What >>> exactly does that mean in general and especially for the timer based >>> audio scheduling? >>> >>> I am currently not making use of this, but should I? >>I'm also looking at this topic from an application developer perspective, >>so I can only provide you my interpretation of reading current ALSA code, >>but here goes. >>I'll also add Eliot (who asked about the same thing, in >> http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025262.h… <http://mailman.alsa-project.org/pipermail/alsa-devel/2010-February/025262.h…> >> ) to the cc, and also Takashi, who git-annotate reveals has made most >> recent PCM_INFO_BATCH related changes to linux/sound/core/pcm_lib.c. ;) >> Anyway, SNDRV_PCM_INFO_BATCH is set by following drivers: >> - isa/msnd >> - pcii/bt87x >> - pci/korg1212 >> - pcmcia/pdaudiocf >> - soc/au1x >> - soc/fsl (mpc52xx/8610) >> - sparc/dbri >> - usbaudio >> Common to all these drivers is that audio samples are moved to/from the >> ALSA ringbuffer, in bursts, to an intermediary block of memory, and from >> there to/from the codec. And most importantly, pointer callback is based >> on last ack'ed burst. IOW, hw_ptr jumps in bursts with these drivers and >> granularity of hw_ptr is more coarse (affecting snd_pcm_avail(), >> snd_pcm_delay() and so forth). >> Now the existing documentation for the flag (and user-space >> snd_pcm_hw_params_is_batch() API) is rather brief as you've noted, and >> AFAIK very few apps use these, so I'm not sure, how much we as app >> developers should rely on these semantics (and to the fact that future >> drivers will use the BATCH flag in a similar way). >> But anyways, the fact remains that snd_pcm_hw_params_is_batch() exists, >> and based on a review of current (2.6.33-rc) drivers, majority/all of the >> drivers use the flag in the same way. >> So for apps like PA, snd_pcm_hw_params_is_batch() is a hint that app >> should not rewind closer to hw_ptr than period-size. It also hints that >> hw_ptr granularity is more coarse, which might be useful in setting sane >> defaults for buffer watermarks and such. if you look at PA src/modules/alsa/alsa-sink.c , PA try to use pa_smoother_translate() to solve the derviation of audio clock and system clock. this function assume two clocks montonic increase , I wonder that routine can really work if the value returned by pointer call back is a stepping function for usb driver i.e. PA expect the hw_ptr increase at a steady rate but usb can only return the number of urb sent if (u->use_tsched) { pa_usec_t cusec; if (u->since_start <= u->hwbuf_size) { /* USB devices on ALSA seem to hit a buffer * underrun during the first iterations much * quicker then we calculate here, probably due to * the transport latency. To accommodate for that * we artificially decrease the sleep time until * we have filled the buffer at least once * completely.*/ if (pa_log_ratelimit()) pa_log_debug("Cutting sleep time for the initial iterations by half."); sleep_usec /= 2; } /* OK, the playback buffer is now full, let's * calculate when to wake up next */ /* pa_log_debug("Waking up in %0.2fms (sound card clock).", (double) sleep_usec / PA_USEC_PER_MSEC); */ /* Convert from the sound card time domain to the * system time domain */ cusec = pa_smoother_translate(u->smoother, pa_rtclock_now(), sleep_usec);

1 0

[alsa-devel] [PATCH 3/5] ASoC: DaVinci: CQ93VC Voice Codec
by miguel.aguilar＠ridgerun.com 08 Mar '10

08 Mar '10

From: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> Currently the DM365 is the only SoC that includes this Voice Codec. Signed-off-by: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cq93vc.c | 288 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cq93vc.h | 29 +++++ 4 files changed, 323 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cq93vc.c create mode 100644 sound/soc/codecs/cq93vc.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..0daca22 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -21,6 +21,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -108,6 +109,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_CQ0093VC + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..dff91fb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,7 @@ snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c new file mode 100644 index 0000000..37f6ba7 --- /dev/null +++ b/sound/soc/codecs/cq93vc.c @@ -0,0 +1,288 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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/init.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/platform_device.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/mfd/davinci_voicecodec.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include <mach/dm365.h> + +#include "cq93vc.h" + +static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + return readl(davinci_vc->base + reg); +} + +static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct davinci_vc *davinci_vc = codec->control_data; + + writel(value, davinci_vc->base + reg); + + return 0; +} + +static const struct snd_kcontrol_new cq93vc_snd_controls[] = { + SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), + SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), +}; + +static int cq93vc_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + + if (mute) + cq93vc_write(codec, DAVINCI_VC_REG09, + reg | DAVINCI_VC_REG09_MUTE); + else + cq93vc_write(codec, DAVINCI_VC_REG09, reg); + + return 0; +} + +static int cq93vc_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; + struct davinci_vc *davinci_vc = codec->control_data; + + switch (freq) { + case 22579200: + case 27000000: + case 33868800: + davinci_vc->cq93vc.sysclk = freq; + return 0; + } + + return -EINVAL; +} + +static int cq93vc_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + case SND_SOC_BIAS_OFF: + /* force all power off */ + cq93vc_write(codec, DAVINCI_VC_REG12, + DAVINCI_VC_REG12_POWER_ALL_OFF); + break; + } + codec->bias_level = level; + + return 0; +} + +#define CQ93VC_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) +#define CQ93VC_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) + +static struct snd_soc_dai_ops cq93vc_dai_ops = { + .digital_mute = cq93vc_mute, + .set_sysclk = cq93vc_set_dai_sysclk, +}; + +struct snd_soc_dai cq93vc_dai = { + .name = "CQ93VC", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CQ93VC_RATES, + .formats = CQ93VC_FORMATS,}, + .ops = &cq93vc_dai_ops, +}; +EXPORT_SYMBOL_GPL(cq93vc_dai); + +static int cq93vc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + cq93vc_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *cq93vc_codec; + +static int cq93vc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct device *dev = &pdev->dev; + struct snd_soc_codec *codec; + int ret; + + socdev->card->codec = cq93vc_codec; + codec = socdev->card->codec; + + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(dev, "%s: failed to create pcms\n", pdev->name); + return ret; + } + + /* Set controls */ + snd_soc_add_controls(codec, cq93vc_snd_controls, + ARRAY_SIZE(cq93vc_snd_controls)); + + /* Off, with power on */ + cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cq93vc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_cq93vc = { + .probe = cq93vc_probe, + .remove = cq93vc_remove, + .resume = cq93vc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_cq93vc); + +static __init int cq93vc_codec_probe(struct platform_device *pdev) +{ + struct davinci_vc *davinci_vc = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) { + dev_dbg(davinci_vc->dev, + "could not allocate memory for codec data\n"); + return -ENOMEM; + } + + davinci_vc->cq93vc.codec = codec; + + cq93vc_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "CQ93VC"; + codec->owner = THIS_MODULE; + codec->read = cq93vc_read; + codec->write = cq93vc_write; + codec->set_bias_level = cq93vc_set_bias_level; + codec->dai = &cq93vc_dai; + codec->num_dai = 1; + codec->control_data = davinci_vc; + + cq93vc_codec = codec; + + ret = snd_soc_register_dai(&cq93vc_dai); + if (ret != 0) { + dev_err(davinci_vc->dev, "could register dai\n"); + goto fail1; + } + return 0; + +fail1: + kfree(codec); + cq93vc_codec = NULL; + + return ret; +} + +static int __devexit cq93vc_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_unregister_dai(&cq93vc_dai); + + kfree(codec); + cq93vc_codec = NULL; + + return 0; +} + +static struct platform_driver cq93vc_codec_driver = { + .driver = { + .name = "cq93vc", + .owner = THIS_MODULE, + }, + .probe = cq93vc_codec_probe, + .remove = __devexit_p(cq93vc_codec_remove), +}; + +static __init int cq93vc_init(void) +{ + return platform_driver_probe(&cq93vc_codec_driver, cq93vc_codec_probe); +} +module_init(cq93vc_init); + +static __exit void cq93vc_exit(void) +{ + platform_driver_unregister(&cq93vc_codec_driver); +} +module_exit(cq93vc_exit); + +MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC CQ0093 Voice Codec Driver"); +MODULE_AUTHOR("Miguel Aguilar"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cq93vc.h b/sound/soc/codecs/cq93vc.h new file mode 100644 index 0000000..845b196 --- /dev/null +++ b/sound/soc/codecs/cq93vc.h @@ -0,0 +1,29 @@ +/* + * ALSA SoC CQ0093 Voice Codec Driver for DaVinci platforms + * + * Copyright (C) 2010 Texas Instruments, Inc + * + * Author: Miguel Aguilar <miguel.aguilar(a)ridgerun.com> + * + * 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 + */ + +#ifndef _CQ93VC_H +#define _CQ93VC_H + +extern struct snd_soc_dai cq93vc_dai; +extern struct snd_soc_codec_device soc_codec_dev_cq93vc; + +#endif -- 1.6.0.4

3 3

[alsa-devel] [PATCH] hdmi - merge common code for intelhdmi and nvhdmi
by Wu Fengguang 08 Mar '10

08 Mar '10

Create patch_hdmi.c to hold common code from intelhdmi and nvhdmi. For now the patch_hdmi.c file is simply included by patch_intelhdmi.c and patch_nvhdmi.c, and does not represent a real codec. There are no behavior changes to intelhdmi. However nvhdmi made several changes when copying code out of intelhdmi, which are all reverted in this patch. Wei Ni confirmed that the reverted code actually works fine. Tested-by: Wei Ni <wni(a)nvidia.com> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 845 ++++++++++++++++++++++++++++++ sound/pci/hda/patch_intelhdmi.c | 821 ----------------------------- sound/pci/hda/patch_nvhdmi.c | 829 ----------------------------- 3 files changed, 882 insertions(+), 1613 deletions(-) --- sound-2.6.orig/sound/pci/hda/patch_intelhdmi.c 2010-03-08 10:04:58.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_intelhdmi.c 2010-03-08 10:14:12.000000000 +0800 @@ -40,815 +40,20 @@ * * The HDA correspondence of pipes/ports are converter/pin nodes. */ -#define INTEL_HDMI_CVTS 2 -#define INTEL_HDMI_PINS 3 +#define MAX_HDMI_CVTS 2 +#define MAX_HDMI_PINS 3 -static char *intel_hdmi_pcm_names[INTEL_HDMI_CVTS] = { +#include "patch_hdmi.c" + +static char *intel_hdmi_pcm_names[MAX_HDMI_CVTS] = { "INTEL HDMI 0", "INTEL HDMI 1", }; -struct intel_hdmi_spec { - int num_cvts; - int num_pins; - hda_nid_t cvt[INTEL_HDMI_CVTS+1]; /* audio sources */ - hda_nid_t pin[INTEL_HDMI_PINS+1]; /* audio sinks */ - - /* - * source connection for each pin - */ - hda_nid_t pin_cvt[INTEL_HDMI_PINS+1]; - - /* - * HDMI sink attached to each pin - */ - struct hdmi_eld sink_eld[INTEL_HDMI_PINS]; - - /* - * export one pcm per pipe - */ - struct hda_pcm pcm_rec[INTEL_HDMI_CVTS]; -}; - -struct hdmi_audio_infoframe { - u8 type; /* 0x84 */ - u8 ver; /* 0x01 */ - u8 len; /* 0x0a */ - - u8 checksum; /* PB0 */ - u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ - u8 SS01_SF24; - u8 CXT04; - u8 CA; - u8 LFEPBL01_LSV36_DM_INH7; - u8 reserved[5]; /* PB6 - PB10 */ -}; - -/* - * CEA speaker placement: - * - * FLH FCH FRH - * FLW FL FLC FC FRC FR FRW - * - * LFE - * TC - * - * RL RLC RC RRC RR - * - * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to - * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. - */ -enum cea_speaker_placement { - FL = (1 << 0), /* Front Left */ - FC = (1 << 1), /* Front Center */ - FR = (1 << 2), /* Front Right */ - FLC = (1 << 3), /* Front Left Center */ - FRC = (1 << 4), /* Front Right Center */ - RL = (1 << 5), /* Rear Left */ - RC = (1 << 6), /* Rear Center */ - RR = (1 << 7), /* Rear Right */ - RLC = (1 << 8), /* Rear Left Center */ - RRC = (1 << 9), /* Rear Right Center */ - LFE = (1 << 10), /* Low Frequency Effect */ - FLW = (1 << 11), /* Front Left Wide */ - FRW = (1 << 12), /* Front Right Wide */ - FLH = (1 << 13), /* Front Left High */ - FCH = (1 << 14), /* Front Center High */ - FRH = (1 << 15), /* Front Right High */ - TC = (1 << 16), /* Top Center */ -}; - -/* - * ELD SA bits in the CEA Speaker Allocation data block - */ -static int eld_speaker_allocation_bits[] = { - [0] = FL | FR, - [1] = LFE, - [2] = FC, - [3] = RL | RR, - [4] = RC, - [5] = FLC | FRC, - [6] = RLC | RRC, - /* the following are not defined in ELD yet */ - [7] = FLW | FRW, - [8] = FLH | FRH, - [9] = TC, - [10] = FCH, -}; - -struct cea_channel_speaker_allocation { - int ca_index; - int speakers[8]; - - /* derived values, just for convenience */ - int channels; - int spk_mask; -}; - -/* - * ALSA sequence is: - * - * surround40 surround41 surround50 surround51 surround71 - * ch0 front left = = = = - * ch1 front right = = = = - * ch2 rear left = = = = - * ch3 rear right = = = = - * ch4 LFE center center center - * ch5 LFE LFE - * ch6 side left - * ch7 side right - * - * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} - */ -static int hdmi_channel_mapping[0x32][8] = { - /* stereo */ - [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* 2.1 */ - [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* Dolby Surround */ - [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* surround40 */ - [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, - /* 4ch */ - [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, - /* surround41 */ - [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround50 */ - [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround51 */ - [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, - /* 7.1 */ - [0x13] = { 0x00, 0x11, 0x26, 0x37, 0x43, 0x52, 0x64, 0x75 }, -}; - -/* - * This is an ordered list! - * - * The preceding ones have better chances to be selected by - * hdmi_setup_channel_allocation(). - */ -static struct cea_channel_speaker_allocation channel_allocations[] = { -/* channel: 7 6 5 4 3 2 1 0 */ -{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, - /* 2.1 */ -{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, - /* Dolby Surround */ -{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, - /* surround40 */ -{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, - /* surround41 */ -{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, - /* surround50 */ -{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, - /* surround51 */ -{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, -{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, -{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, -{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, -}; - -/* - * HDA/HDMI auto parsing - */ - -static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) -{ - int i; - - for (i = 0; nids[i]; i++) - if (nids[i] == nid) - return i; - - snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); - return -EINVAL; -} - -static int intel_hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct intel_hdmi_spec *spec = codec->spec; - hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; - int conn_len, curr; - int index; - - if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { - snd_printk(KERN_WARNING - "HDMI: pin %d wcaps %#x " - "does not support connection list\n", - pin_nid, get_wcaps(codec, pin_nid)); - return -EINVAL; - } - - conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, - HDA_MAX_CONNECTIONS); - if (conn_len > 1) - curr = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONNECT_SEL, 0); - else - curr = 0; - - index = hda_node_index(spec->pin, pin_nid); - if (index < 0) - return -EINVAL; - - spec->pin_cvt[index] = conn_list[curr]; - - return 0; -} - -static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - if (!snd_hdmi_get_eld(eld, codec, pin_nid)) - snd_hdmi_show_eld(eld); -} - -static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - int present = snd_hda_pin_sense(codec, pin_nid); - - eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - - if (present & AC_PINSENSE_ELDV) - hdmi_get_show_eld(codec, pin_nid, eld); -} - -static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct intel_hdmi_spec *spec = codec->spec; - - if (spec->num_pins >= INTEL_HDMI_PINS) { - snd_printk(KERN_WARNING - "HDMI: no space for pin %d \n", pin_nid); - return -EINVAL; - } - - hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); - - spec->pin[spec->num_pins] = pin_nid; - spec->num_pins++; - - /* - * It is assumed that converter nodes come first in the node list and - * hence have been registered and usable now. - */ - return intel_hdmi_read_pin_conn(codec, pin_nid); -} - -static int intel_hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) -{ - struct intel_hdmi_spec *spec = codec->spec; - - if (spec->num_cvts >= INTEL_HDMI_CVTS) { - snd_printk(KERN_WARNING - "HDMI: no space for converter %d \n", nid); - return -EINVAL; - } - - spec->cvt[spec->num_cvts] = nid; - spec->num_cvts++; - - return 0; -} - -static int intel_hdmi_parse_codec(struct hda_codec *codec) -{ - hda_nid_t nid; - int i, nodes; - - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); - if (!nid || nodes < 0) { - snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); - return -EINVAL; - } - - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; - - caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); - type = get_wcaps_type(caps); - - if (!(caps & AC_WCAP_DIGITAL)) - continue; - - switch (type) { - case AC_WID_AUD_OUT: - if (intel_hdmi_add_cvt(codec, nid) < 0) - return -EINVAL; - break; - case AC_WID_PIN: - caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) - continue; - if (intel_hdmi_add_pin(codec, nid) < 0) - return -EINVAL; - break; - } - } - - /* - * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event - * can be lost and presence sense verb will become inaccurate if the - * HDA link is powered off at hot plug or hw initialization time. - */ -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & - AC_PWRST_EPSS)) - codec->bus->power_keep_link_on = 1; -#endif - - return 0; -} - -/* - * HDMI routines - */ - -#ifdef BE_PARANOID -static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int *packet_index, int *byte_index) -{ - int val; - - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_INDEX, 0); - - *packet_index = val >> 5; - *byte_index = val & 0x1f; -} -#endif - -static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int packet_index, int byte_index) -{ - int val; - - val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); -} - -static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, - unsigned char val) -{ - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); -} - -static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) -{ - /* Unmute */ - if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out */ - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); -} - -/* - * Enable Audio InfoFrame Transmission - */ -static void hdmi_start_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); -} - -/* - * Disable Audio InfoFrame Transmission - */ -static void hdmi_stop_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_DISABLE); -} - -static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) -{ - return 1 + snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CVT_CHAN_COUNT, 0); -} - -static void hdmi_set_channel_count(struct hda_codec *codec, - hda_nid_t nid, int chs) -{ - if (chs != hdmi_get_channel_count(codec, nid)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); -} - -static void hdmi_debug_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int slot; - - for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); - printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0xf); - } -#endif -} - - -/* - * Audio InfoFrame routines - */ - -static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int size; - - size = snd_hdmi_get_eld_size(codec, pin_nid); - printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); - - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); - } -#endif -} - -static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef BE_PARANOID - int i, j; - int size; - int pi, bi; - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - if (size == 0) - continue; - - hdmi_set_dip_index(codec, pin_nid, i, 0x0); - for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, pin_nid, 0x0); - hdmi_get_dip_index(codec, pin_nid, &pi, &bi); - if (pi != i) - snd_printd(KERN_INFO "dip index %d: %d != %d\n", - bi, pi, i); - if (bi == 0) /* byte index wrapped around */ - break; - } - snd_printd(KERN_INFO - "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", - i, size, j); - } -#endif -} - -static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - u8 sum = 0; - int i; - - ai->checksum = 0; - - for (i = 0; i < sizeof(*ai); i++) - sum += bytes[i]; - - ai->checksum = - sum; -} - -static void hdmi_fill_audio_infoframe(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - int i; - - hdmi_debug_dip_size(codec, pin_nid); - hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - - hdmi_checksum_audio_infoframe(ai); - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) - hdmi_write_dip_byte(codec, pin_nid, bytes[i]); -} - -/* - * Compute derived values in channel_allocations[]. - */ -static void init_channel_allocations(void) -{ - int i, j; - struct cea_channel_speaker_allocation *p; - - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - p = channel_allocations + i; - p->channels = 0; - p->spk_mask = 0; - for (j = 0; j < ARRAY_SIZE(p->speakers); j++) - if (p->speakers[j]) { - p->channels++; - p->spk_mask |= p->speakers[j]; - } - } -} - -/* - * The transformation takes two steps: - * - * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask - * spk_mask => (channel_allocations[]) => ai->CA - * - * TODO: it could select the wrong CA from multiple candidates. -*/ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, - struct hdmi_audio_infoframe *ai) -{ - struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld; - int i; - int spk_mask = 0; - int channels = 1 + (ai->CC02_CT47 & 0x7); - char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; - - /* - * CA defaults to 0 for basic stereo audio - */ - if (channels <= 2) - return 0; - - i = hda_node_index(spec->pin_cvt, nid); - if (i < 0) - return 0; - eld = &spec->sink_eld[i]; - - /* - * HDMI sink's ELD info cannot always be retrieved for now, e.g. - * in console or for audio devices. Assume the highest speakers - * configuration, to _not_ prohibit multi-channel audio playback. - */ - if (!eld->spk_alloc) - eld->spk_alloc = 0xffff; - - /* - * expand ELD's speaker allocation mask - * - * ELD tells the speaker mask in a compact(paired) form, - * expand ELD's notions to match the ones used by Audio InfoFrame. - */ - for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { - if (eld->spk_alloc & (1 << i)) - spk_mask |= eld_speaker_allocation_bits[i]; - } - - /* search for the first working match in the CA table */ - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - if (channels == channel_allocations[i].channels && - (spk_mask & channel_allocations[i].spk_mask) == - channel_allocations[i].spk_mask) { - ai->CA = channel_allocations[i].ca_index; - break; - } - } - - snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); - snd_printdd(KERN_INFO - "HDMI: select CA 0x%x for %d-channel allocation: %s\n", - ai->CA, channels, buf); - - return ai->CA; -} - -static void hdmi_setup_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - int i; - int ca = ai->CA; - int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); - for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); - } - - for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); - if (err) { - snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); - break; - } - } - - hdmi_debug_channel_mapping(codec, pin_nid); -} - -static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - u8 val; - int i; - - if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) - != AC_DIPXMIT_BEST) - return false; - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) { - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_DATA, 0); - if (val != bytes[i]) - return false; - } - - return true; -} - -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, - struct snd_pcm_substream *substream) -{ - struct intel_hdmi_spec *spec = codec->spec; - hda_nid_t pin_nid; - int i; - struct hdmi_audio_infoframe ai = { - .type = 0x84, - .ver = 0x01, - .len = 0x0a, - .CC02_CT47 = substream->runtime->channels - 1, - }; - - hdmi_setup_channel_allocation(codec, nid, &ai); - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_cvt[i] != nid) - continue; - if (!spec->sink_eld[i].monitor_present) - continue; - - pin_nid = spec->pin[i]; - if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { - hdmi_setup_channel_mapping(codec, pin_nid, &ai); - hdmi_stop_infoframe_trans(codec, pin_nid); - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); - hdmi_start_infoframe_trans(codec, pin_nid); - } - } -} - - /* - * Unsolicited events + * HDMI callbacks */ -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - struct intel_hdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pind = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); - int index; - - printk(KERN_INFO - "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - tag, pind, eldv); - - index = hda_node_index(spec->pin, tag); - if (index < 0) - return; - - spec->sink_eld[index].monitor_present = pind; - spec->sink_eld[index].eld_valid = eldv; - - if (pind && eldv) { - hdmi_get_show_eld(codec, spec->pin[index], &spec->sink_eld[index]); - /* TODO: do real things about ELD */ - } -} - -static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); - int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); - - printk(KERN_INFO - "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", - tag, - subtag, - cp_state, - cp_ready); - - /* TODO */ - if (cp_state) - ; - if (cp_ready) - ; -} - - -static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct intel_hdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - - if (hda_node_index(spec->pin, tag) < 0) { - snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); - return; - } - - if (subtag == 0) - hdmi_intrinsic_event(codec, res); - else - hdmi_non_intrinsic_event(codec, res); -} - -/* - * Callbacks - */ - -static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, int format) -{ - int tag; - int fmt; - - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); -} - static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -882,7 +87,7 @@ static struct hda_pcm_stream intel_hdmi_ static int intel_hdmi_build_pcms(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; int i; @@ -908,7 +113,7 @@ static int intel_hdmi_build_pcms(struct static int intel_hdmi_build_controls(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int err; int i; @@ -923,7 +128,7 @@ static int intel_hdmi_build_controls(str static int intel_hdmi_init(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; for (i = 0; spec->pin[i]; i++) { @@ -937,7 +142,7 @@ static int intel_hdmi_init(struct hda_co static void intel_hdmi_free(struct hda_codec *codec) { - struct intel_hdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; for (i = 0; i < spec->num_pins; i++) @@ -951,12 +156,12 @@ static struct hda_codec_ops intel_hdmi_p .free = intel_hdmi_free, .build_pcms = intel_hdmi_build_pcms, .build_controls = intel_hdmi_build_controls, - .unsol_event = intel_hdmi_unsol_event, + .unsol_event = hdmi_unsol_event, }; static int patch_intel_hdmi(struct hda_codec *codec) { - struct intel_hdmi_spec *spec; + struct hdmi_spec *spec; int i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -964,7 +169,7 @@ static int patch_intel_hdmi(struct hda_c return -ENOMEM; codec->spec = spec; - if (intel_hdmi_parse_codec(codec) < 0) { + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ sound-2.6/sound/pci/hda/patch_hdmi.c 2010-03-08 10:11:00.000000000 +0800 @@ -0,0 +1,845 @@ +/* + * + * patch_hdmi.c - routines for HDMI/DisplayPort codecs + * + * Copyright(c) 2008-2010 Intel Corporation. All rights reserved. + * + * Authors: + * Wu Fengguang <wfg(a)linux.intel.com> + * + * Maintained by: + * Wu Fengguang <wfg(a)linux.intel.com> + * + * 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. + */ + + +struct hdmi_spec { + int num_cvts; + int num_pins; + hda_nid_t cvt[MAX_HDMI_CVTS+1]; /* audio sources */ + hda_nid_t pin[MAX_HDMI_PINS+1]; /* audio sinks */ + + /* + * source connection for each pin + */ + hda_nid_t pin_cvt[MAX_HDMI_PINS+1]; + + /* + * HDMI sink attached to each pin + */ + struct hdmi_eld sink_eld[MAX_HDMI_PINS]; + + /* + * export one pcm per pipe + */ + struct hda_pcm pcm_rec[MAX_HDMI_CVTS]; + + /* + * nvhdmi specific + */ + struct hda_multi_out multiout; + unsigned int codec_type; +}; + + +struct hdmi_audio_infoframe { + u8 type; /* 0x84 */ + u8 ver; /* 0x01 */ + u8 len; /* 0x0a */ + + u8 checksum; /* PB0 */ + u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ + u8 SS01_SF24; + u8 CXT04; + u8 CA; + u8 LFEPBL01_LSV36_DM_INH7; + u8 reserved[5]; /* PB6 - PB10 */ +}; + +/* + * CEA speaker placement: + * + * FLH FCH FRH + * FLW FL FLC FC FRC FR FRW + * + * LFE + * TC + * + * RL RLC RC RRC RR + * + * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to + * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. + */ +enum cea_speaker_placement { + FL = (1 << 0), /* Front Left */ + FC = (1 << 1), /* Front Center */ + FR = (1 << 2), /* Front Right */ + FLC = (1 << 3), /* Front Left Center */ + FRC = (1 << 4), /* Front Right Center */ + RL = (1 << 5), /* Rear Left */ + RC = (1 << 6), /* Rear Center */ + RR = (1 << 7), /* Rear Right */ + RLC = (1 << 8), /* Rear Left Center */ + RRC = (1 << 9), /* Rear Right Center */ + LFE = (1 << 10), /* Low Frequency Effect */ + FLW = (1 << 11), /* Front Left Wide */ + FRW = (1 << 12), /* Front Right Wide */ + FLH = (1 << 13), /* Front Left High */ + FCH = (1 << 14), /* Front Center High */ + FRH = (1 << 15), /* Front Right High */ + TC = (1 << 16), /* Top Center */ +}; + +/* + * ELD SA bits in the CEA Speaker Allocation data block + */ +static int eld_speaker_allocation_bits[] = { + [0] = FL | FR, + [1] = LFE, + [2] = FC, + [3] = RL | RR, + [4] = RC, + [5] = FLC | FRC, + [6] = RLC | RRC, + /* the following are not defined in ELD yet */ + [7] = FLW | FRW, + [8] = FLH | FRH, + [9] = TC, + [10] = FCH, +}; + +struct cea_channel_speaker_allocation { + int ca_index; + int speakers[8]; + + /* derived values, just for convenience */ + int channels; + int spk_mask; +}; + +/* + * ALSA sequence is: + * + * surround40 surround41 surround50 surround51 surround71 + * ch0 front left = = = = + * ch1 front right = = = = + * ch2 rear left = = = = + * ch3 rear right = = = = + * ch4 LFE center center center + * ch5 LFE LFE + * ch6 side left + * ch7 side right + * + * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} + */ +static int hdmi_channel_mapping[0x32][8] = { + /* stereo */ + [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* 2.1 */ + [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* Dolby Surround */ + [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, + /* surround40 */ + [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, + /* 4ch */ + [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, + /* surround41 */ + [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround50 */ + [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, + /* surround51 */ + [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, + /* 7.1 */ + [0x13] = { 0x00, 0x11, 0x26, 0x37, 0x43, 0x52, 0x64, 0x75 }, +}; + +/* + * This is an ordered list! + * + * The preceding ones have better chances to be selected by + * hdmi_setup_channel_allocation(). + */ +static struct cea_channel_speaker_allocation channel_allocations[] = { +/* channel: 7 6 5 4 3 2 1 0 */ +{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, + /* 2.1 */ +{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, + /* Dolby Surround */ +{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, + /* surround40 */ +{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, + /* surround41 */ +{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, + /* surround50 */ +{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, + /* surround51 */ +{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, + /* 6.1 */ +{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, + /* surround71 */ +{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, + +{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, +{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, +{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, +{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, +{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, +{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, +{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, +{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, +{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, +{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, +{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, +{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, +{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, +}; + + +/* + * HDMI routines + */ + +static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) +{ + int i; + + for (i = 0; nids[i]; i++) + if (nids[i] == nid) + return i; + + snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); + return -EINVAL; +} + +static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) + snd_hdmi_show_eld(eld); +} + +#ifdef BE_PARANOID +static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, + int *packet_index, int *byte_index) +{ + int val; + + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_INDEX, 0); + + *packet_index = val >> 5; + *byte_index = val & 0x1f; +} +#endif + +static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, + int packet_index, int byte_index) +{ + int val; + + val = (packet_index << 5) | (byte_index & 0x1f); + + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); +} + +static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, + unsigned char val) +{ + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); +} + +static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) +{ + /* Unmute */ + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + /* Enable pin out */ + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); +} + +static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) +{ + return 1 + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CVT_CHAN_COUNT, 0); +} + +static void hdmi_set_channel_count(struct hda_codec *codec, + hda_nid_t nid, int chs) +{ + if (chs != hdmi_get_channel_count(codec, nid)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); +} + + +/* + * Channel mapping routines + */ + +/* + * Compute derived values in channel_allocations[]. + */ +static void init_channel_allocations(void) +{ + int i, j; + struct cea_channel_speaker_allocation *p; + + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + p = channel_allocations + i; + p->channels = 0; + p->spk_mask = 0; + for (j = 0; j < ARRAY_SIZE(p->speakers); j++) + if (p->speakers[j]) { + p->channels++; + p->spk_mask |= p->speakers[j]; + } + } +} + +/* + * The transformation takes two steps: + * + * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask + * spk_mask => (channel_allocations[]) => ai->CA + * + * TODO: it could select the wrong CA from multiple candidates. +*/ +static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, + struct hdmi_audio_infoframe *ai) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_eld *eld; + int i; + int spk_mask = 0; + int channels = 1 + (ai->CC02_CT47 & 0x7); + char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; + + /* + * CA defaults to 0 for basic stereo audio + */ + if (channels <= 2) + return 0; + + i = hda_node_index(spec->pin_cvt, nid); + if (i < 0) + return 0; + eld = &spec->sink_eld[i]; + + /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!eld->spk_alloc) + eld->spk_alloc = 0xffff; + + /* + * expand ELD's speaker allocation mask + * + * ELD tells the speaker mask in a compact(paired) form, + * expand ELD's notions to match the ones used by Audio InfoFrame. + */ + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (eld->spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + /* search for the first working match in the CA table */ + for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { + if (channels == channel_allocations[i].channels && + (spk_mask & channel_allocations[i].spk_mask) == + channel_allocations[i].spk_mask) { + ai->CA = channel_allocations[i].ca_index; + break; + } + } + + snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); + snd_printdd(KERN_INFO + "HDMI: select CA 0x%x for %d-channel allocation: %s\n", + ai->CA, channels, buf); + + return ai->CA; +} + +static void hdmi_debug_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int slot; + + for (i = 0; i < 8; i++) { + slot = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_CHAN_SLOT, i); + printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", + slot >> 4, slot & 0xf); + } +#endif +} + + +static void hdmi_setup_channel_mapping(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + int i; + int ca = ai->CA; + int err; + + if (hdmi_channel_mapping[ca][1] == 0) { + for (i = 0; i < channel_allocations[ca].channels; i++) + hdmi_channel_mapping[ca][i] = i | (i << 4); + for (; i < 8; i++) + hdmi_channel_mapping[ca][i] = 0xf | (i << 4); + } + + for (i = 0; i < 8; i++) { + err = snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + hdmi_channel_mapping[ca][i]); + if (err) { + snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); + break; + } + } + + hdmi_debug_channel_mapping(codec, pin_nid); +} + + +/* + * Audio InfoFrame routines + */ + +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) +{ + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); +} + +/* + * Disable Audio InfoFrame Transmission + */ +static void hdmi_stop_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) +{ + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_DISABLE); +} + +static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) +{ +#ifdef CONFIG_SND_DEBUG_VERBOSE + int i; + int size; + + size = snd_hdmi_get_eld_size(codec, pin_nid); + printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); + + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); + } +#endif +} + +static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) +{ +#ifdef BE_PARANOID + int i, j; + int size; + int pi, bi; + for (i = 0; i < 8; i++) { + size = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_SIZE, i); + if (size == 0) + continue; + + hdmi_set_dip_index(codec, pin_nid, i, 0x0); + for (j = 1; j < 1000; j++) { + hdmi_write_dip_byte(codec, pin_nid, 0x0); + hdmi_get_dip_index(codec, pin_nid, &pi, &bi); + if (pi != i) + snd_printd(KERN_INFO "dip index %d: %d != %d\n", + bi, pi, i); + if (bi == 0) /* byte index wrapped around */ + break; + } + snd_printd(KERN_INFO + "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", + i, size, j); + } +#endif +} + +static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 sum = 0; + int i; + + ai->checksum = 0; + + for (i = 0; i < sizeof(*ai); i++) + sum += bytes[i]; + + ai->checksum = -sum; +} + +static void hdmi_fill_audio_infoframe(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + int i; + + hdmi_debug_dip_size(codec, pin_nid); + hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ + + hdmi_checksum_audio_infoframe(ai); + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) + hdmi_write_dip_byte(codec, pin_nid, bytes[i]); +} + +static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 val; + int i; + + if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) + != AC_DIPXMIT_BEST) + return false; + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) { + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_DATA, 0); + if (val != bytes[i]) + return false; + } + + return true; +} + +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t pin_nid; + int i; + struct hdmi_audio_infoframe ai = { + .type = 0x84, + .ver = 0x01, + .len = 0x0a, + .CC02_CT47 = substream->runtime->channels - 1, + }; + + hdmi_setup_channel_allocation(codec, nid, &ai); + + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (!spec->sink_eld[i].monitor_present) + continue; + + pin_nid = spec->pin[i]; + if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + hdmi_setup_channel_mapping(codec, pin_nid, &ai); + hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); + } + } +} + + +/* + * Unsolicited events + */ + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + struct hdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int pind = !!(res & AC_UNSOL_RES_PD); + int eldv = !!(res & AC_UNSOL_RES_ELDV); + int index; + + printk(KERN_INFO + "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + tag, pind, eldv); + + index = hda_node_index(spec->pin, tag); + if (index < 0) + return; + + spec->sink_eld[index].monitor_present = pind; + spec->sink_eld[index].eld_valid = eldv; + + if (pind && eldv) { + hdmi_get_show_eld(codec, spec->pin[index], + &spec->sink_eld[index]); + /* TODO: do real things about ELD */ + } +} + +static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); + int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); + + printk(KERN_INFO + "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + tag, + subtag, + cp_state, + cp_ready); + + /* TODO */ + if (cp_state) + ; + if (cp_ready) + ; +} + + +static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct hdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; + int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; + + if (hda_node_index(spec->pin, tag) < 0) { + snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); + return; + } + + if (subtag == 0) + hdmi_intrinsic_event(codec, res); + else + hdmi_non_intrinsic_event(codec, res); +} + +/* + * Callbacks + */ + +static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, int format) +{ + int tag; + int fmt; + + tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; + fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); + + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", + nid, + tag == stream_tag ? "" : "new-", + stream_tag, + fmt == format ? "" : "new-", + format); + + if (tag != stream_tag) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, + stream_tag << 4); + if (fmt != format) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); +} + +/* + * HDA/HDMI auto parsing + */ + +static int hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct hdmi_spec *spec = codec->spec; + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int conn_len, curr; + int index; + + if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { + snd_printk(KERN_WARNING + "HDMI: pin %d wcaps %#x " + "does not support connection list\n", + pin_nid, get_wcaps(codec, pin_nid)); + return -EINVAL; + } + + conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, + HDA_MAX_CONNECTIONS); + if (conn_len > 1) + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + else + curr = 0; + + index = hda_node_index(spec->pin, pin_nid); + if (index < 0) + return -EINVAL; + + spec->pin_cvt[index] = conn_list[curr]; + + return 0; +} + +static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + int present = snd_hda_pin_sense(codec, pin_nid); + + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->eld_valid = !!(present & AC_PINSENSE_ELDV); + + if (present & AC_PINSENSE_ELDV) + hdmi_get_show_eld(codec, pin_nid, eld); +} + +static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->num_pins >= MAX_HDMI_PINS) { + snd_printk(KERN_WARNING + "HDMI: no space for pin %d \n", pin_nid); + return -EINVAL; + } + + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); + + spec->pin[spec->num_pins] = pin_nid; + spec->num_pins++; + + /* + * It is assumed that converter nodes come first in the node list and + * hence have been registered and usable now. + */ + return hdmi_read_pin_conn(codec, pin_nid); +} + +static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->num_cvts >= MAX_HDMI_CVTS) { + snd_printk(KERN_WARNING + "HDMI: no space for converter %d \n", nid); + return -EINVAL; + } + + spec->cvt[spec->num_cvts] = nid; + spec->num_cvts++; + + return 0; +} + +static int hdmi_parse_codec(struct hda_codec *codec) +{ + hda_nid_t nid; + int i, nodes; + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (!nid || nodes < 0) { + snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); + return -EINVAL; + } + + for (i = 0; i < nodes; i++, nid++) { + unsigned int caps; + unsigned int type; + + caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + type = get_wcaps_type(caps); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + switch (type) { + case AC_WID_AUD_OUT: + if (hdmi_add_cvt(codec, nid) < 0) + return -EINVAL; + break; + case AC_WID_PIN: + caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) + continue; + if (hdmi_add_pin(codec, nid) < 0) + return -EINVAL; + break; + } + } + + /* + * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event + * can be lost and presence sense verb will become inaccurate if the + * HDA link is powered off at hot plug or hw initialization time. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & + AC_PWRST_EPSS)) + codec->bus->power_keep_link_on = 1; +#endif + + return 0; +} + --- sound-2.6.orig/sound/pci/hda/patch_nvhdmi.c 2010-03-08 10:04:58.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_nvhdmi.c 2010-03-08 10:12:47.000000000 +0800 @@ -29,6 +29,15 @@ #include "hda_codec.h" #include "hda_local.h" +#define MAX_HDMI_CVTS 1 +#define MAX_HDMI_PINS 1 + +#include "patch_hdmi.c" + +static char *nvhdmi_pcm_names[MAX_HDMI_CVTS] = { + "NVIDIA HDMI", +}; + /* define below to restrict the supported rates and formats */ /* #define LIMITED_RATE_FMT_SUPPORT */ @@ -83,802 +92,12 @@ static struct hda_verb nvhdmi_basic_init (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) #endif -#define NVIDIA_89_HDMI_CVTS 1 -#define NVIDIA_89_HDMI_PINS 1 - -static char *nvhdmi_pcm_names[NVIDIA_89_HDMI_CVTS] = { - "NVIDIA HDMI", -}; - -struct nvhdmi_spec { - int num_cvts; - int num_pins; - hda_nid_t cvt[NVIDIA_89_HDMI_CVTS+1]; /* audio sources */ - hda_nid_t pin[NVIDIA_89_HDMI_PINS+1]; /* audio sinks */ - hda_nid_t pin_cvt[NVIDIA_89_HDMI_PINS+1]; - struct hda_pcm pcm_rec[NVIDIA_89_HDMI_CVTS]; - struct hdmi_eld sink_eld[NVIDIA_89_HDMI_PINS]; - struct hda_multi_out multiout; - unsigned int codec_type; -}; - -struct hdmi_audio_infoframe { - u8 type; /* 0x84 */ - u8 ver; /* 0x01 */ - u8 len; /* 0x0a */ - - u8 checksum; /* PB0 */ - u8 CC02_CT47; /* CC in bits 0:2, CT in 4:7 */ - u8 SS01_SF24; - u8 CXT04; - u8 CA; - u8 LFEPBL01_LSV36_DM_INH7; -}; - -/* - * CEA speaker placement: - * - * FLH FCH FRH - * FLW FL FLC FC FRC FR FRW - * - * LFE - * TC - * - * RL RLC RC RRC RR - * - * The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M corresponds to - * CEA RL/RR; The SMPTE channel _assignment_ C/LFE is swapped to CEA LFE/FC. - */ -enum cea_speaker_placement { - FL = (1 << 0), /* Front Left */ - FC = (1 << 1), /* Front Center */ - FR = (1 << 2), /* Front Right */ - FLC = (1 << 3), /* Front Left Center */ - FRC = (1 << 4), /* Front Right Center */ - RL = (1 << 5), /* Rear Left */ - RC = (1 << 6), /* Rear Center */ - RR = (1 << 7), /* Rear Right */ - RLC = (1 << 8), /* Rear Left Center */ - RRC = (1 << 9), /* Rear Right Center */ - LFE = (1 << 10), /* Low Frequency Effect */ - FLW = (1 << 11), /* Front Left Wide */ - FRW = (1 << 12), /* Front Right Wide */ - FLH = (1 << 13), /* Front Left High */ - FCH = (1 << 14), /* Front Center High */ - FRH = (1 << 15), /* Front Right High */ - TC = (1 << 16), /* Top Center */ -}; - -/* - * ELD SA bits in the CEA Speaker Allocation data block - */ -static int eld_speaker_allocation_bits[] = { - [0] = FL | FR, - [1] = LFE, - [2] = FC, - [3] = RL | RR, - [4] = RC, - [5] = FLC | FRC, - [6] = RLC | RRC, - /* the following are not defined in ELD yet */ - [7] = FLW | FRW, - [8] = FLH | FRH, - [9] = TC, - [10] = FCH, -}; - -struct cea_channel_speaker_allocation { - int ca_index; - int speakers[8]; - - /* derived values, just for convenience */ - int channels; - int spk_mask; -}; - -/* - * ALSA sequence is: - * - * surround40 surround41 surround50 surround51 surround71 - * ch0 front left = = = = - * ch1 front right = = = = - * ch2 rear left = = = = - * ch3 rear right = = = = - * ch4 LFE center center center - * ch5 LFE LFE - * ch6 side left - * ch7 side right - * - * surround71 = {FL, FR, RLC, RRC, FC, LFE, RL, RR} - */ -static int hdmi_channel_mapping[0x32][8] = { - /* stereo */ - [0x00] = { 0x00, 0x11, 0xf2, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* 2.1 */ - [0x01] = { 0x00, 0x11, 0x22, 0xf3, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* Dolby Surround */ - [0x02] = { 0x00, 0x11, 0x23, 0xf2, 0xf4, 0xf5, 0xf6, 0xf7 }, - /* surround40 */ - [0x08] = { 0x00, 0x11, 0x24, 0x35, 0xf3, 0xf2, 0xf6, 0xf7 }, - /* 4ch */ - [0x03] = { 0x00, 0x11, 0x23, 0x32, 0x44, 0xf5, 0xf6, 0xf7 }, - /* surround41 */ - [0x09] = { 0x00, 0x11, 0x24, 0x34, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround50 */ - [0x0a] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0xf2, 0xf6, 0xf7 }, - /* surround51 */ - [0x0b] = { 0x00, 0x11, 0x24, 0x35, 0x43, 0x52, 0xf6, 0xf7 }, - /* 7.1 */ - [0x13] = { 0x00, 0x11, 0x32, 0x23, 0x64, 0x75, 0x46, 0x57 }, -}; - -/* - * This is an ordered list! - * - * The preceding ones have better chances to be selected by - * hdmi_setup_channel_allocation(). - */ -static struct cea_channel_speaker_allocation channel_allocations[] = { -/* channel: 7 6 5 4 3 2 1 0 */ -{ .ca_index = 0x00, .speakers = { 0, 0, 0, 0, 0, 0, FR, FL } }, - /* 2.1 */ -{ .ca_index = 0x01, .speakers = { 0, 0, 0, 0, 0, LFE, FR, FL } }, - /* Dolby Surround */ -{ .ca_index = 0x02, .speakers = { 0, 0, 0, 0, FC, 0, FR, FL } }, - /* surround40 */ -{ .ca_index = 0x08, .speakers = { 0, 0, RR, RL, 0, 0, FR, FL } }, - /* surround41 */ -{ .ca_index = 0x09, .speakers = { 0, 0, RR, RL, 0, LFE, FR, FL } }, - /* surround50 */ -{ .ca_index = 0x0a, .speakers = { 0, 0, RR, RL, FC, 0, FR, FL } }, - /* surround51 */ -{ .ca_index = 0x0b, .speakers = { 0, 0, RR, RL, FC, LFE, FR, FL } }, - /* 6.1 */ -{ .ca_index = 0x0f, .speakers = { 0, RC, RR, RL, FC, LFE, FR, FL } }, - /* surround71 */ -{ .ca_index = 0x13, .speakers = { RRC, RLC, RR, RL, FC, LFE, FR, FL } }, - -{ .ca_index = 0x03, .speakers = { 0, 0, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x04, .speakers = { 0, 0, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x05, .speakers = { 0, 0, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x06, .speakers = { 0, 0, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x07, .speakers = { 0, 0, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x0c, .speakers = { 0, RC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x0d, .speakers = { 0, RC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x0e, .speakers = { 0, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x10, .speakers = { RRC, RLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x11, .speakers = { RRC, RLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x12, .speakers = { RRC, RLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x14, .speakers = { FRC, FLC, 0, 0, 0, 0, FR, FL } }, -{ .ca_index = 0x15, .speakers = { FRC, FLC, 0, 0, 0, LFE, FR, FL } }, -{ .ca_index = 0x16, .speakers = { FRC, FLC, 0, 0, FC, 0, FR, FL } }, -{ .ca_index = 0x17, .speakers = { FRC, FLC, 0, 0, FC, LFE, FR, FL } }, -{ .ca_index = 0x18, .speakers = { FRC, FLC, 0, RC, 0, 0, FR, FL } }, -{ .ca_index = 0x19, .speakers = { FRC, FLC, 0, RC, 0, LFE, FR, FL } }, -{ .ca_index = 0x1a, .speakers = { FRC, FLC, 0, RC, FC, 0, FR, FL } }, -{ .ca_index = 0x1b, .speakers = { FRC, FLC, 0, RC, FC, LFE, FR, FL } }, -{ .ca_index = 0x1c, .speakers = { FRC, FLC, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x1d, .speakers = { FRC, FLC, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x1e, .speakers = { FRC, FLC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x1f, .speakers = { FRC, FLC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x20, .speakers = { 0, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x21, .speakers = { 0, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x22, .speakers = { TC, 0, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x23, .speakers = { TC, 0, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x24, .speakers = { FRH, FLH, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x25, .speakers = { FRH, FLH, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x26, .speakers = { FRW, FLW, RR, RL, 0, 0, FR, FL } }, -{ .ca_index = 0x27, .speakers = { FRW, FLW, RR, RL, 0, LFE, FR, FL } }, -{ .ca_index = 0x28, .speakers = { TC, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x29, .speakers = { TC, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2a, .speakers = { FCH, RC, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2b, .speakers = { FCH, RC, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2c, .speakers = { TC, FCH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2d, .speakers = { TC, FCH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x2e, .speakers = { FRH, FLH, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x2f, .speakers = { FRH, FLH, RR, RL, FC, LFE, FR, FL } }, -{ .ca_index = 0x30, .speakers = { FRW, FLW, RR, RL, FC, 0, FR, FL } }, -{ .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, -}; - -/* - * HDA/HDMI auto parsing - */ - -static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) -{ - int i; - - for (i = 0; nids[i]; i++) - if (nids[i] == nid) - return i; - - snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); - return -EINVAL; -} - -static int nvhdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct nvhdmi_spec *spec = codec->spec; - hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; - int conn_len, curr; - int index; - - if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { - snd_printk(KERN_WARNING - "HDMI: pin %d wcaps %#x " - "does not support connection list\n", - pin_nid, get_wcaps(codec, pin_nid)); - return -EINVAL; - } - - conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, - HDA_MAX_CONNECTIONS); - if (conn_len > 1) - curr = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONNECT_SEL, 0); - else - curr = 0; - - index = hda_node_index(spec->pin, pin_nid); - if (index < 0) - return -EINVAL; - - spec->pin_cvt[index] = conn_list[curr]; - - return 0; -} - -static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - if (!snd_hdmi_get_eld(eld, codec, pin_nid)) - snd_hdmi_show_eld(eld); -} - -static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_eld *eld) -{ - int present = snd_hda_pin_sense(codec, pin_nid); - - eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - eld->eld_valid = !!(present & AC_PINSENSE_ELDV); - - if (present & AC_PINSENSE_ELDV) - hdmi_get_show_eld(codec, pin_nid, eld); -} - -static int nvhdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) -{ - struct nvhdmi_spec *spec = codec->spec; - - if (spec->num_pins >= NVIDIA_89_HDMI_PINS) { - snd_printk(KERN_WARNING - "HDMI: no space for pin %d \n", pin_nid); - return -EINVAL; - } - - hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); - - spec->pin[spec->num_pins] = pin_nid; - spec->num_pins++; - - /* - * It is assumed that converter nodes come first in the node list and - * hence have been registered and usable now. - */ - return nvhdmi_read_pin_conn(codec, pin_nid); -} - -static int nvhdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) -{ - struct nvhdmi_spec *spec = codec->spec; - - if (spec->num_cvts >= NVIDIA_89_HDMI_CVTS) { - snd_printk(KERN_WARNING - "HDMI: no space for converter %d \n", nid); - return -EINVAL; - } - - spec->cvt[spec->num_cvts] = nid; - spec->num_cvts++; - - return 0; -} - - -static int nvhdmi_parse_codec(struct hda_codec *codec) -{ - hda_nid_t nid; - int i, nodes; - - nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); - if (!nid || nodes < 0) { - snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); - return -EINVAL; - } - - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; - - caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); - type = get_wcaps_type(caps); - - if (!(caps & AC_WCAP_DIGITAL)) - continue; - - switch (type) { - case AC_WID_AUD_OUT: - if (nvhdmi_add_cvt(codec, nid) < 0) - return -EINVAL; - break; - case AC_WID_PIN: - caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) - continue; - if (nvhdmi_add_pin(codec, nid) < 0) - return -EINVAL; - break; - } - } - - /* - * G45/IbexPeak don't support EPSS: the unsolicited pin hot plug event - * can be lost and presence sense verb will become inaccurate if the - * HDA link is powered off at hot plug or hw initialization time. - */ -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!(snd_hda_param_read(codec, codec->afg, AC_PAR_POWER_STATE) & - AC_PWRST_EPSS)) - codec->bus->power_keep_link_on = 1; -#endif - - return 0; -} - -/* - * HDMI routines - */ - -#ifdef BE_PARANOID -static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int *packet_index, int *byte_index) -{ - int val; - - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_INDEX, 0); - - *packet_index = val >> 5; - *byte_index = val & 0x1f; -} -#endif - -static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, - int packet_index, int byte_index) -{ - int val; - - val = (packet_index << 5) | (byte_index & 0x1f); - - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); -} - -static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, - unsigned char val) -{ - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); -} - -static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) -{ - /* Unmute */ - if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Enable pin out */ - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); -} - -/* - * Enable Audio InfoFrame Transmission - */ -static void hdmi_start_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); -} - -/* - * Disable Audio InfoFrame Transmission - */ -static void hdmi_stop_infoframe_trans(struct hda_codec *codec, - hda_nid_t pin_nid) -{ - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_DISABLE); -} - -static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) -{ - return 1 + snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CVT_CHAN_COUNT, 0); -} - -static void hdmi_set_channel_count(struct hda_codec *codec, - hda_nid_t nid, int chs) -{ - if (chs != hdmi_get_channel_count(codec, nid)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); -} - -static void hdmi_debug_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int slot; - - for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_CHAN_SLOT, i); - printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0xf); - } -#endif -} - - -/* - * Audio InfoFrame routines - */ - -static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef CONFIG_SND_DEBUG_VERBOSE - int i; - int size; - - size = snd_hdmi_get_eld_size(codec, pin_nid); - printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size); - - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size); - } -#endif -} - -static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) -{ -#ifdef BE_PARANOID - int i, j; - int size; - int pi, bi; - for (i = 0; i < 8; i++) { - size = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_SIZE, i); - if (size == 0) - continue; - - hdmi_set_dip_index(codec, pin_nid, i, 0x0); - for (j = 1; j < 1000; j++) { - hdmi_write_dip_byte(codec, pin_nid, 0x0); - hdmi_get_dip_index(codec, pin_nid, &pi, &bi); - if (pi != i) - snd_printd(KERN_INFO "dip index %d: %d != %d\n", - bi, pi, i); - if (bi == 0) /* byte index wrapped around */ - break; - } - snd_printd(KERN_INFO - "HDMI: DIP GP[%d] buf reported size=%d, written=%d\n", - i, size, j); - } -#endif -} - -static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) -{ - ai->checksum = 0; -} - -static void hdmi_fill_audio_infoframe(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - int i; - - hdmi_debug_dip_size(codec, pin_nid); - hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ - - hdmi_checksum_audio_infoframe(ai); - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) - hdmi_write_dip_byte(codec, pin_nid, bytes[i]); -} - -/* - * Compute derived values in channel_allocations[]. - */ -static void init_channel_allocations(void) -{ - int i, j; - struct cea_channel_speaker_allocation *p; - - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - p = channel_allocations + i; - p->channels = 0; - p->spk_mask = 0; - for (j = 0; j < ARRAY_SIZE(p->speakers); j++) - if (p->speakers[j]) { - p->channels++; - p->spk_mask |= p->speakers[j]; - } - } -} - -/* - * The transformation takes two steps: - * - * eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask - * spk_mask => (channel_allocations[]) => ai->CA - * - * TODO: it could select the wrong CA from multiple candidates. -*/ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, - struct hdmi_audio_infoframe *ai) -{ - struct nvhdmi_spec *spec = codec->spec; - struct hdmi_eld *eld; - int i; - int spk_mask = 0; - int channels = 1 + (ai->CC02_CT47 & 0x7); - char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; - - /* - * CA defaults to 0 for basic stereo audio - */ - if (channels <= 2) - return 0; - - i = hda_node_index(spec->pin_cvt, nid); - if (i < 0) - return 0; - eld = &spec->sink_eld[i]; - - /* - * HDMI sink's ELD info cannot always be retrieved for now, e.g. - * in console or for audio devices. Assume the highest speakers - * configuration, to _not_ prohibit multi-channel audio playback. - */ - if (!eld->spk_alloc) - eld->spk_alloc = 0xffff; - - /* - * expand ELD's speaker allocation mask - * - * ELD tells the speaker mask in a compact(paired) form, - * expand ELD's notions to match the ones used by Audio InfoFrame. - */ - for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { - if (eld->spk_alloc & (1 << i)) - spk_mask |= eld_speaker_allocation_bits[i]; - } - - /* search for the first working match in the CA table */ - for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) { - if (channels == channel_allocations[i].channels && - (spk_mask & channel_allocations[i].spk_mask) == - channel_allocations[i].spk_mask) { - ai->CA = channel_allocations[i].ca_index; - break; - } - } - - snd_print_channel_allocation(eld->spk_alloc, buf, sizeof(buf)); - snd_printdd(KERN_INFO - "HDMI: select CA 0x%x for %d-channel allocation: %s\n", - ai->CA, channels, buf); - - return ai->CA; -} - -static void hdmi_setup_channel_mapping(struct hda_codec *codec, - hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - int i; - int ca = ai->CA; - int err; - - if (hdmi_channel_mapping[ca][1] == 0) { - for (i = 0; i < channel_allocations[ca].channels; i++) - hdmi_channel_mapping[ca][i] = i | (i << 4); - for (; i < 8; i++) - hdmi_channel_mapping[ca][i] = 0xf | (i << 4); - } - - for (i = 0; i < 8; i++) { - err = snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_HDMI_CHAN_SLOT, - hdmi_channel_mapping[ca][i]); - if (err) { - snd_printdd(KERN_INFO "HDMI: channel mapping failed\n"); - break; - } - } - - hdmi_debug_channel_mapping(codec, pin_nid); -} - -static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, - struct hdmi_audio_infoframe *ai) -{ - u8 *bytes = (u8 *)ai; - u8 val; - int i; - - if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) - != AC_DIPXMIT_BEST) - return false; - - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(*ai); i++) { - val = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_HDMI_DIP_DATA, 0); - if (val != bytes[i]) - return false; - } - - return true; -} - -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, - struct snd_pcm_substream *substream) -{ - struct nvhdmi_spec *spec = codec->spec; - hda_nid_t pin_nid; - int i; - struct hdmi_audio_infoframe ai = { - .type = 0x84, - .ver = 0x01, - .len = 0x0a, - .CC02_CT47 = substream->runtime->channels - 1, - }; - - hdmi_setup_channel_allocation(codec, nid, &ai); - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_cvt[i] != nid) - continue; - if (!spec->sink_eld[i].monitor_present) - continue; - - pin_nid = spec->pin[i]; - if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { - hdmi_setup_channel_mapping(codec, pin_nid, &ai); - hdmi_stop_infoframe_trans(codec, pin_nid); - hdmi_fill_audio_infoframe(codec, pin_nid, &ai); - hdmi_start_infoframe_trans(codec, pin_nid); - } - } -} - -/* - * Unsolicited events - */ - -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - struct nvhdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pind = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); - int index; - - printk(KERN_INFO - "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - tag, pind, eldv); - - index = hda_node_index(spec->pin, tag); - if (index < 0) - return; - - spec->sink_eld[index].monitor_present = pind; - spec->sink_eld[index].eld_valid = eldv; - - if (eldv) { - spec->sink_eld[index].monitor_present = 1; - hdmi_get_show_eld(codec, spec->pin[index], - &spec->sink_eld[index]); - /* TODO: do real things about ELD */ - } -} - -static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) -{ - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); - int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); - - printk(KERN_INFO - "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", - tag, - subtag, - cp_state, - cp_ready); - - /* TODO */ - if (cp_state) - ; - if (cp_ready) - ; -} - -static void nvhdmi_unsol_event(struct hda_codec *codec, unsigned int res) -{ - struct nvhdmi_spec *spec = codec->spec; - int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - - if (hda_node_index(spec->pin, tag) < 0) { - snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); - return; - } - - if (subtag == 0) - hdmi_intrinsic_event(codec, res); - else - hdmi_non_intrinsic_event(codec, res); -} - -/* - * Callbacks - */ - -static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, int format) -{ - int tag; - int fmt; - - tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; - fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - - snd_printdd("hdmi_setup_stream: " - "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", - nid, - tag == stream_tag ? "" : "new-", - stream_tag, - fmt == format ? "" : "new-", - format); - - if (tag != stream_tag) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - stream_tag << 4); - if (fmt != format) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, format); -} - /* * Controls */ static int nvhdmi_build_controls(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int err; int i; @@ -902,7 +121,7 @@ static int nvhdmi_build_controls(struct static int nvhdmi_init(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) || (spec->codec_type == HDA_CODEC_NVIDIA_GT21X)) { @@ -920,7 +139,7 @@ static int nvhdmi_init(struct hda_codec static void nvhdmi_free(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; if ((spec->codec_type == HDA_CODEC_NVIDIA_MCP89) @@ -939,7 +158,7 @@ static int nvhdmi_dig_playback_pcm_open( struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_open(codec, &spec->multiout); } @@ -947,7 +166,7 @@ static int nvhdmi_dig_playback_pcm_close struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; int i; snd_hda_codec_write(codec, nvhdmi_master_con_nid_7x, @@ -968,7 +187,7 @@ static int nvhdmi_dig_playback_pcm_close struct hda_codec *codec, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -1121,7 +340,7 @@ static int nvhdmi_dig_playback_pcm_prepa unsigned int format, struct snd_pcm_substream *substream) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, format, substream); } @@ -1170,7 +389,7 @@ static struct hda_pcm_stream nvhdmi_pcm_ static int nvhdmi_build_pcms_8ch_89(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; int i; @@ -1196,7 +415,7 @@ static int nvhdmi_build_pcms_8ch_89(stru static int nvhdmi_build_pcms_8ch_7x(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; codec->num_pcms = 1; @@ -1212,7 +431,7 @@ static int nvhdmi_build_pcms_8ch_7x(stru static int nvhdmi_build_pcms_2ch(struct hda_codec *codec) { - struct nvhdmi_spec *spec = codec->spec; + struct hdmi_spec *spec = codec->spec; struct hda_pcm *info = spec->pcm_rec; codec->num_pcms = 1; @@ -1231,7 +450,7 @@ static struct hda_codec_ops nvhdmi_patch .build_pcms = nvhdmi_build_pcms_8ch_89, .init = nvhdmi_init, .free = nvhdmi_free, - .unsol_event = nvhdmi_unsol_event, + .unsol_event = hdmi_unsol_event, }; static struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = { @@ -1250,7 +469,7 @@ static struct hda_codec_ops nvhdmi_patch static int patch_nvhdmi_8ch_89(struct hda_codec *codec) { - struct nvhdmi_spec *spec; + struct hdmi_spec *spec; int i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -1260,7 +479,7 @@ static int patch_nvhdmi_8ch_89(struct hd codec->spec = spec; spec->codec_type = HDA_CODEC_NVIDIA_MCP89; - if (nvhdmi_parse_codec(codec) < 0) { + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; @@ -1277,7 +496,7 @@ static int patch_nvhdmi_8ch_89(struct hd static int patch_nvhdmi_8ch_7x(struct hda_codec *codec) { - struct nvhdmi_spec *spec; + struct hdmi_spec *spec; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1297,7 +516,7 @@ static int patch_nvhdmi_8ch_7x(struct hd static int patch_nvhdmi_2ch(struct hda_codec *codec) { - struct nvhdmi_spec *spec; + struct hdmi_spec *spec; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL)

2 2

Re: [alsa-devel] Bug#568806: linux-image-2.6.32-trunk-amd64: Internal mic of dell m1330 stopped working with 2.6.32 kernel
by maximilian attems 08 Mar '10

08 Mar '10

[ adding alsa dev on cc ] hda regression on 2.6.32, see full report on http://bugs.debian.org/568806 On Sun, Mar 07, 2010 at 10:29:49PM +0100, Sjoerd Hardeman wrote: > > > > you could try before unload hda snd module and pass all its different > > variants, see Documentation/sound/alsa/HD-Audio-Models.txt > That was a good idea. It took some time, but the option "model=dell-md6" > worked, as well as "model=dell-m23". SO I probably need to file a bug > that the auto-recognition stopped working? please provide alsa-info.sh output in working and nonworking case. so that alsa devs can see the diff. thanks

1 0

[alsa-devel] 2.6.30-rc8 skipping regression also affects cmipci
by Frédéric Brière 07 Mar '10

07 Mar '10

Hi everybody, I was only recently able to upgrade from 2.6.26 to 2.6.32 (long story). In so doing, I noticed infrequent occurrences of audio skipping, which I managed to bisect down to c87d973. It would appear that this commit also affects cmipci (CMI8738), albeit not as badly as it did for intel8x0. (I guess I'm the only person left on Earth still using a stand-alone PCI sound card, which is why this was never reported. <g>) Here's a sample log output after turning xrun debugging on: [ 171.555860] PCM: hw_ptr skipping! (pos=6341, delta=16383, period=1024, jdelta=0/85) [ 181.114680] PCM: hw_ptr skipping! (pos=12529, delta=16370, period=1024, jdelta=0/85) [ 181.114770] PCM: hw_ptr skipping! (pos=12534, delta=16375, period=1024, jdelta=0/85) [ 1713.774316] PCM: hw_ptr skipping! (pos=9456, delta=16369, period=1024, jdelta=0/85) [ 1713.774346] PCM: hw_ptr skipping! (pos=9458, delta=16371, period=1024, jdelta=0/85) [ 1713.774409] PCM: hw_ptr skipping! (pos=9461, delta=16374, period=1024, jdelta=0/85) [ 1713.774613] PCM: hw_ptr skipping! (pos=9470, delta=16383, period=1024, jdelta=0/85) [ 1831.092000] PCM: hw_ptr skipping! (pos=1712, delta=16373, period=1024, jdelta=0/85) [ 1953.347453] PCM: hw_ptr skipping! (pos=3723, delta=16316, period=1024, jdelta=0/84) [ 1953.347465] PCM: hw_ptr skipping! (pos=3724, delta=16317, period=1024, jdelta=0/84) [ 1953.347485] PCM: hw_ptr skipping! (pos=3725, delta=16318, period=1024, jdelta=0/84) You'll notice that this occurs much less frequently than it did for intel8x0. However, I left MPlayer on pause for the night, and at some point the kernel started spewing several syslog messages every couple of seconds: [25332.211286] PCM: hw_ptr skipping! [Q] (pos=4, delta=-1073740800, period=1024, jdelta=5/44744/0) [25332.232637] PCM: hw_ptr skipping! [Q] (pos=1029, delta=-1073739771, period=1024, jdelta=6/44749/0) [25332.253957] PCM: hw_ptr skipping! [Q] (pos=2053, delta=-1073738747, period=1024, jdelta=5/44755/0) [25332.275292] PCM: hw_ptr skipping! [Q] (pos=3077, delta=-1073737723, period=1024, jdelta=5/44760/0) [25332.296616] PCM: hw_ptr skipping! [Q] (pos=4101, delta=-1073736699, period=1024, jdelta=6/44765/0) [25332.317932] PCM: hw_ptr skipping! [Q] (pos=5125, delta=-1073735675, period=1024, jdelta=5/44771/0) [25332.339269] PCM: hw_ptr skipping! [Q] (pos=6149, delta=-1073734651, period=1024, jdelta=5/44776/0) [25332.360600] PCM: hw_ptr skipping! [Q] (pos=7173, delta=-1073733627, period=1024, jdelta=6/44781/0) [25332.381942] PCM: hw_ptr skipping! [Q] (pos=8197, delta=-1073732603, period=1024, jdelta=5/44787/0) [25332.403261] PCM: hw_ptr skipping! [Q] (pos=9221, delta=-1073731579, period=1024, jdelta=5/44792/0) [25337.244901] PCM: hw_ptr skipping! [Q] (pos=12293, delta=-1073728507, period=1024, jdelta=6/44808/0) [25337.266223] PCM: hw_ptr skipping! [Q] (pos=13317, delta=-1073727483, period=1024, jdelta=5/44813/0) This kept going on until I quit MPlayer (which, as I said, was not actually playing anything all this time). Not that skipped silence is much of an issue, but I thought this might add useful information. Let me know if I can be of any help in resolving this. -- If you want to travel around the world and be invited to speak at a lot of different places, just write a Unix operating system. -- Linus Torvalds

2 3

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for a Biostar Microtech board
by Daniel Chen 05 Mar '10

05 Mar '10

Hi, Attached is a patch against sound-2.6 master HEAD (applies with tweaking to 2.6.32.x) working around a hardware issue by using position_fix=1. Best, -Dan

2 1

[alsa-devel] usb-audio code refactoring, more v2 support [resend]
by Daniel Mack 05 Mar '10

05 Mar '10

This is a rebased version of the usb-audio cleanup patch set. On top of what's already been posted, there is one more fix to support sample rate configuration, and one bugfix. Commit 864c11080 was also ported over. And Clemens' latest patches are also included. With these patches applied, I can stream audio on a class v2 device. At least for awhile, then I get CRC errors in DATA0 packets. But that's a different issue. What's not yet done is all the feature unit parsing, mixers etc. Daniel [PATCH 1/8] ALSA: usb-audio: move ua101 driver [PATCH 2/8] ALSA: usb-audio: header file cleanups [PATCH 3/8] ALSA: usb-audio: refactor code [PATCH 4/8] ALSA: usb-audio: rename substream format field to altset_idx [PATCH 5/8] ALSA: usb-audio: use a format bitmask per alternate setting [PATCH 6/8] ALSA: usb-audio: support multiple formats with audio class v2 devices [PATCH 7/8] ALSA: usb-audio: add support for samplerate setting on v2 devices [PATCH 8/8] usb/audio.h: Fix field order

1 3

[alsa-devel] [PATCH 4/8] ALSA: usb-audio: rename substream format field to altset_idx
by Daniel Mack 05 Mar '10

05 Mar '10

From: Clemens Ladisch <clemens(a)ladisch.de> The snd_usb_substream::format field actually contains the index of the current alternate setting, so rename it to altset_idx to avoid confusion. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/card.h | 2 +- sound/usb/pcm.c | 8 ++++---- sound/usb/proc.c | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 71f03c1..856d71b 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -54,7 +54,7 @@ struct snd_usb_substream { struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ - unsigned int format; /* USB data format */ + unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int datapipe; /* the data i/o pipe */ unsigned int syncpipe; /* 1 - async out or adaptive in */ unsigned int datainterval; /* log_2 of data packet interval */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 87863cc..c3d5a97 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -202,11 +202,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) return -EIO; } subs->interface = -1; - subs->format = 0; + subs->altset_idx = 0; } /* set interface */ - if (subs->interface != fmt->iface || subs->format != fmt->altset_idx) { + if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", dev->devnum, fmt->iface, fmt->altsetting); @@ -214,7 +214,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); subs->interface = fmt->iface; - subs->format = fmt->altset_idx; + subs->altset_idx = fmt->altset_idx; } /* create a data pipe */ @@ -771,7 +771,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; subs->interface = -1; - subs->format = 0; + subs->altset_idx = 0; runtime->hw = snd_usb_hardware; runtime->private_data = subs; subs->pcm_substream = substream; diff --git a/sound/usb/proc.c b/sound/usb/proc.c index be3065e..78fc3ba 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -116,7 +116,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn unsigned int i; snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); - snd_iprintf(buffer, " Altset = %d\n", subs->format); + snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); for (i = 0; i < subs->nurbs; i++) snd_iprintf(buffer, "%d ", subs->dataurb[i].packets); -- 1.6.6.2

1 4

[alsa-devel] [PATCH] fix jazz16 compile (udelay)
by Meelis Roos 05 Mar '10

05 Mar '10

While trying to compile jazz16 isa sound driver on alpha (2.6.33+git), I found a compile failure in jazz16.c (udelay is unknown). Fix it by including delay.h. Signed-foo-by: Meelis Roos <mroos(a)linux.ee> diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8d21a3f..8ccbcdd 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -14,6 +14,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/io.h> +#include <linux/delay.h> #include <asm/dma.h> #include <linux/isa.h> #include <sound/core.h> -- Meelis Roos (mroos(a)linux.ee)

2 1

Re: [alsa-devel] [RFC 0/4] FIFO caused playback delay (latency) handling in soc
by Raymond Yau 05 Mar '10

05 Mar '10

>> Do soc support snd_pcm_rewind() and snd_pcm_forward ? >>No driver demand yet, though it'd be trivially implemented if anyone >>wanted to do it. if size of fifo is around 1x ~ 3x fames How accucate do those media player need to allow A/V lip sync , Video is about 24 ~ 30 frames per seconds and audio is 48000Hz ?

1 0

[alsa-devel] snd-powermac @ G4 DA (PowerMac3, 4) - headphone detection doesn't work
by Douglas Mencken 04 Mar '10

04 Mar '10

I can hear sound with internal speakers only if I plug headphones jack, without plugged jack, there's no sound at all; to listen thru headphones, I need to half-insert that jack and this leads to distortions; with h/phones jack plugged fully, no sound in phones, but sound in internal speaker. With the same half-plug, in OS X I head sound both via h/phones and internal speaker. Driver: snd-powermac. Audio card: Tumbler @ i2ca @ KeyLargo i2c (I can provide the full dump of OpenFirmware device tree). As <gnubien> at #alsa @ irc.freenode.net suggested, I ran `alsamixer -c 0` and MM-ed 'Auto Mute'. After that, I can fully plug h/phones jack and enjoy full sounds w/o distortions but still can't get any sound when it's unplugged. The result of running ./alsa-info.sh: Your ALSA information is located at http://www.alsa-project.org/db/?f=daac9e06cbe772222b90d7fba14c1a49a65b980c

2 1

[alsa-devel] [RFC] Cleanup/refactor the usbaudio driver
by Daniel Mack 04 Mar '10

04 Mar '10

I've spent some time cleaning up the usbaudio driver. I felt that the existing implementation was quite difficult to follow because it contained code for all sorts of things mixed up in one big file. So I split the driver into pieces and tried to be as less intrusive as I can, meaning that I didn't touch much real code but rather copied the functions as they were and factored them out. Of course, non-static functions were properly prefixed to avoid name space pollution. I also moved almost all quirks to a seperate file and introduced two more generic hooks to handle them, so the excecption add less to the actual code of the driver. Unfortunately it wasn't easily possible to put that big change into smaller pieces, and so 3/3 became a huge chunk. After all, I feel that it is easier to read now, but I don't know whether you agree. So take this as RFC for now :) Daniel

3 11

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for Dell Latitude 131L
by Daniel Chen 04 Mar '10

04 Mar '10

Hi, Please apply the following patch against sound-2.6 master HEAD (applies with tweaking against 2.6.32.x) to work around hardware issues on Dell Latitude 131L. Best, -Dan

2 1

[alsa-devel] [PATCH] ASoC: Add delay information for Samsung IISv2 DAIs
by Mark Brown 04 Mar '10

04 Mar '10

Report the current FIFO depth when delay is queried. The FIFO is only 16 frames deep so the latency will be at most a couple of miliseconds (and we tend to end up reporting zero most of the time) but it may help some applications. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/s3c-i2s-v2.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d83..04a00fa 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -550,6 +550,21 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } +static long s3c2412_i2s_delay(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 reg = readl(i2s->regs + S3C2412_IISFIC); + int delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = S3C2412_IISFIC_TXCOUNT(reg); + else + delay = S3C2412_IISFIC_RXCOUNT(reg); + + return delay; +} + /* default table of all avaialable root fs divisors */ static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; @@ -736,6 +751,10 @@ int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) ops->set_fmt = s3c2412_i2s_set_fmt; ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + /* Allow overriding by (for example) IISv4 */ + if (!ops->delay) + ops->delay = s3c2412_i2s_delay, + dai->suspend = s3c2412_i2s_suspend; dai->resume = s3c2412_i2s_resume; -- 1.7.0

2 6

[alsa-devel] [RFC 0/4] FIFO caused playback delay (latency) handling in soc
by Peter Ujfalusi 04 Mar '10

04 Mar '10

Hello, There has been discussion in alsa-devel in this subject several times, but no actual patches has been sent. Based on the available information the latency caused by the HW buffer on some systems can be handled by updating the runtime->delay. It has been discussed, that the runtime->delay can also be updated dynamically to show more accurate delay. To further complicate things, in ASoC we could have more buffer in the chain. To handle this we need soc level support. This RFC series tries to do that in soc by: - introducing a pcm_pointer wrapper - in this wrapper we call the original pcm_pointer functions to get the DMA pointer - introducing a new interface in dai_ops to ask the delay from the dais - adding the cpu_dai and codec_dai returned delay to form the actual delay - update the runtime->delay with this value. With this approach none of the existing drivers need change, but they can add support for specifying the FIFO caused delay. In this series on top of the core changes the omap(3) code is updated to take this delay reporting into use. I have not added the support to the tlv320dac33 codec driver, since it needs a bit more work, but along the same line it can be done, and if the tlv320dac33 is hooked to omap McBSP than applications can know the whole delay/latency on that path. Since this is for RFC, I have based it on sound-2.6 topc/asoc branch, which does not have the McBSP regcache nor the sidetone patches, but I would like to get feedback on this method first than create a proper series. I have left out linux-omap from this RFC round, since the problem is ALSA related. Thank you, Péter Ujfalusi --- Peter Ujfalusi (4): ASoC: core: soc level wrapper for pcm_pointer callback ASoC: core: Add delay operation to snd_soc_dai_ops OMAP3: McBSP: Add interface for transmit FIFO state query ASoC: OMAP3: Report delay on playback caused by the internal FIFO arch/arm/plat-omap/include/plat/mcbsp.h | 4 +++ arch/arm/plat-omap/mcbsp.c | 27 ++++++++++++++++++++++++ include/sound/soc-dai.h | 6 +++++ sound/soc/omap/omap-mcbsp.c | 26 +++++++++++++++++++++++ sound/soc/soc-core.c | 34 ++++++++++++++++++++++++++++++- 5 files changed, 96 insertions(+), 1 deletions(-)

7 30

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 03 Mar '10

03 Mar '10

The following changes since commit a0b62329bb290c10d7278809af910ed115768991: Takashi Iwai (1): Merge branch 'for-2.6.34' of git://opensource.wolfsonmicro.com/linux-2.6-asoc into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (1): ASoC: fix ak4104 register array access Jassi Brar (1): ASoC: soc_pcm_open: Add missing bailout tag sound/soc/codecs/ak4104.c | 6 ++---- sound/soc/soc-core.c | 14 +++++++++----- 2 files changed, 11 insertions(+), 9 deletions(-)

2 1

[alsa-devel] [RESEND] opti92x: use PnP data to select Master Control port
by Krzysztof Helt 03 Mar '10

03 Mar '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The Master Control port (MC) is available as the last PnP resource (OPT005). Use this value instead fo guessing. Also, add some comments to the code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 120 +++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 43 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b865e45..5913717 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1558,7 +1558,7 @@ static int __devinit snd_card_miro_pnp(struct snd_miro *chip, err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); return err; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index a4af53b..becd90d 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -144,12 +144,8 @@ struct snd_opti9xx { spinlock_t lock; + long wss_base; int irq; - -#ifdef CONFIG_PNP - struct pnp_dev *dev; - struct pnp_dev *devmpu; -#endif /* CONFIG_PNP */ }; static int snd_opti9xx_pnp_is_probed; @@ -159,12 +155,17 @@ static int snd_opti9xx_pnp_is_probed; static struct pnp_card_device_id snd_opti9xx_pnpids[] = { #ifndef OPTi93X /* OPTi 82C924 */ - { .id = "OPT0924", .devs = { { "OPT0000" }, { "OPT0002" } }, .driver_data = 0x0924 }, + { .id = "OPT0924", + .devs = { { "OPT0000" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0924 }, /* OPTi 82C925 */ - { .id = "OPT0925", .devs = { { "OPT9250" }, { "OPT0002" } }, .driver_data = 0x0925 }, + { .id = "OPT0925", + .devs = { { "OPT9250" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0925 }, #else /* OPTi 82C931/3 */ - { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, .driver_data = 0x0931 }, + { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, + .driver_data = 0x0931 }, #endif /* OPTi93X */ { .id = "" } }; @@ -207,24 +208,34 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, chip->hardware = hardware; strcpy(chip->name, snd_opti9xx_names[hardware]); - chip->mc_base_size = opti9xx_mc_size[hardware]; - spin_lock_init(&chip->lock); chip->irq = -1; +#ifndef OPTi93X +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; +#endif /* CONFIG_PNP */ + else { + chip->mc_base = 0xf8c; + chip->mc_base_size = opti9xx_mc_size[hardware]; + } +#else + chip->mc_base_size = opti9xx_mc_size[hardware]; +#endif + switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = (hardware == OPTi9XX_HW_82C928) ? 0xe2 : 0xe3; chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: case OPTi9XX_HW_82C925: - chip->mc_base = 0xf8c; chip->password = 0xe5; chip->pwd_reg = 3; break; @@ -292,7 +303,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, spin_unlock_irqrestore(&chip->lock, flags); return retval; } - + static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, unsigned char value) { @@ -341,7 +352,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, - long wss_base, + long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) { @@ -354,16 +365,23 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, switch (chip->hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C924: + /* opti 929 mode (?), OPL3 clock output, audio enable */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); + /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); case OPTi9XX_HW_82C925: + /* enable WSS mode */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + /* OPL3 FM synthesis */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20); + /* disable Sound Blaster IRQ and DMA */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); #ifdef CS4231 + /* cs4231/4248 fix enabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); #else + /* cs4231/4248 fix disabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02); #endif /* CS4231 */ break; @@ -411,21 +429,26 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, return -EINVAL; } - switch (wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (port & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", port); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -487,7 +510,7 @@ __skip_base: #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, wss_base); + outb(irq_bits << 3 | dma_bits, chip->wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ @@ -729,15 +752,15 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, { struct pnp_dev *pdev; int err; + struct pnp_dev *devmpu; +#ifndef OPTi93X + struct pnp_dev *devmc; +#endif - chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - - pdev = chip->dev; - err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); @@ -750,9 +773,24 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else - if (pid->driver_data != 0x0924) - port = pnp_port_start(pdev, 1); + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); + return err; + } + + port = pnp_port_start(pdev, 1); fm_port = pnp_port_start(pdev, 2) + 8; + /* + * The MC(0) is never accessed and card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; #endif /* OPTi93X */ irq = pnp_irq(pdev, 0); dma1 = pnp_dma(pdev, 0); @@ -760,16 +798,16 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, dma2 = pnp_dma(pdev, 1); #endif /* CS4231 || OPTi93X */ - pdev = chip->devmpu; - if (pdev && mpu_port > 0) { - err = pnp_activate_dev(pdev); + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + + if (devmpu && mpu_port > 0) { + err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); mpu_port = -1; - chip->devmpu = NULL; } else { - mpu_port = pnp_port_start(pdev, 0); - mpu_irq = pnp_irq(pdev, 0); + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); } } return pid->driver_data; @@ -824,7 +862,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; - error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, + error = snd_wss_create(card, chip->wss_base + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -865,10 +903,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, port + 4, irq, dma1, xdma2); + card->shortname, pcm->name, + chip->wss_base + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, port + 4, irq, dma1); + card->shortname, pcm->name, chip->wss_base + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) @@ -1062,9 +1101,6 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, snd_card_free(card); return error; } - if (hw <= OPTi9XX_HW_82C930) - chip->mc_base -= 0x80; - error = snd_opti9xx_read_check(chip); if (error) { snd_printk(KERN_ERR "OPTI chip not found\n"); -- 1.6.4 ---------------------------------------------------------------------- Wygraj z armia wampirow! Kliknij >>> http://link.interia.pl/f23a0

2 1

[alsa-devel] [patch] riptide: clean up while loop
by Dan Carpenter 03 Mar '10

03 Mar '10

If getpaths() returned an odd number this would be a buffer under-run and an endless loop. It turns out that getpaths() can only return even numbers, but let's make it easy for people auditing code. With the new code you don't need to look at getpaths(). This silences a smatch warning. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e66ef2b..a5924af 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1974,9 +1974,9 @@ snd_riptide_proc_read(struct snd_info_entry *entry, } snd_iprintf(buffer, "Paths:\n"); i = getpaths(cif, p); - while (i--) { - snd_iprintf(buffer, "%x->%x ", p[i - 1], p[i]); - i--; + while (i >= 2) { + i -= 2; + snd_iprintf(buffer, "%x->%x ", p[i], p[i + 1]); } snd_iprintf(buffer, "\n"); }

2 1

[alsa-devel] [PATCH] ALSA: ASoC: fix ak4104 register array access
by Daniel Mack 03 Mar '10

03 Mar '10

Don't touch the variable 'reg' to construct the value for the actual SPI transport. This variable is again used to access the driver's register cache, and so random memory is overwritten. Compute the value in-place instead. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: stable(a)kernel.org --- sound/soc/codecs/ak4104.c | 6 ++---- 1 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b9ef7e4..b68d99f 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -90,12 +90,10 @@ static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, if (reg >= codec->reg_cache_size) return -EINVAL; - reg &= AK4104_REG_MASK; - reg |= AK4104_WRITE; - /* only write to the hardware if value has changed */ if (cache[reg] != value) { - u8 tmp[2] = { reg, value }; + u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value }; + if (spi_write(spi, tmp, sizeof(tmp))) { dev_err(&spi->dev, "SPI write failed\n"); return -EIO; -- 1.6.3.3

2 1

[alsa-devel] snd_pcm_drop causing lockup with DMix
by John Lindgren 03 Mar '10

03 Mar '10

Hi, A couple of weeks ago I reported a rather serious bug #4914 that can cause a multi-threaded using ALSA to lock up so that it can only be ended with SIGKILL. I maintain the ALSA output part of Audacious, and I need to be able to play audio in one thread of the program and then signal that thread to stop from another. The way to do this in ALSA, if I understand correctly, is to interrupt the thread that is sending audio with a call to snd_pcm_drop. But using DMix makes the first thread hang up in a snd_pcm_wait call with a timeout of -1. Is anyone working on this bug? Is there any more help I can provide? John Lindgren

2 9

[alsa-devel] ASoC: SAMSUNG: About I2S_v4 support
by jassi brar 03 Mar '10

03 Mar '10

Hello, Could we please reach some agreement about the support of I2Sv4 block of S3C64XX and newer SoCs? Kindly express your opinion. Otherwise, I am planning to re-factor(as and if needed) s3c-i2s-v2.c, s3c2412-i2s.c and s3c64xx-i2s.c and add a new s3c-i2s-v4.c to handle I2S_v4 controllers. Kindly ACK/NAK this idea if you are busy with other stuff and I am to start working on the issue. Thanks.

2 4

[alsa-devel] [patch] oxygen: change || to &&
by Dan Carpenter 03 Mar '10

03 Mar '10

In the original code the condition was always true (hopefully) because WM8776_HPLVOL is zero. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 7754db1..dbc4b89 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -68,7 +68,7 @@ static void wm8776_write(struct oxygen *chip, OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); if (reg < ARRAY_SIZE(data->wm8776_regs)) { - if (reg >= WM8776_HPLVOL || reg <= WM8776_DACMASTER) + if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER) value &= ~WM8776_UPDATE; data->wm8776_regs[reg] = value; }

2 1

[alsa-devel] alsa needs updating for stable kernel 2.6.33+, and compile error
by VDR User 02 Mar '10

02 Mar '10

First, linux/utsrelease.h was moved to generated/utsrelease.h so that needs to be fixed. Also, when will hdmi audio support for GT220 cards be included? And any ideas why this compile is failing?: test:/usrc/alsa/alsa-driver-1.0.22.1$ ./configure --with-cards=hda-intel --with-card-options=hda-hwdep,hda-reconfig,hda-codec-realtek,hda-codec-nvhdmi --with-isapnp=no --with-sequencer=no --with-oss=no --with-pcm-oss-plugins=no checking for gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for ranlib... ranlib checking for a BSD-compatible install... /usr/bin/install -c checking how to run the C preprocessor... gcc -E checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for ANSI C header files... yes checking for an ANSI C-conforming const... yes checking for inline... inline checking whether time.h and sys/time.h may both be included... yes checking whether gcc needs -traditional... no checking for current directory... /usr/local/src/alsa/alsa-driver-1.0.22.1 checking cross compile... checking for directory with ALSA kernel sources... ./configure: line 5107: cd: ../alsa-kmirror: No such file or directory ../alsa-kmirror checking for directory with kernel source... /lib/modules/2.6.33.amd64-x2.030210.1/source checking for directory with kernel build... /lib/modules/2.6.33.amd64-x2.030210.1/build checking for kernel linux/version.h... yes checking for kernel linux/autoconf.h... yes checking for kernel version... 2.6.33.amd64-x2.030210.1 checking for GCC version... ./configure: eval: line 5495: syntax error near unexpected token `)' ./configure: eval: line 5495: `my_compiler_version=4.4.2-9)' ./configure: line 5495: warning: syntax errors in . or eval will cause future versions of the shell to abort as Posix requires Kernel compiler: Used compiler: gcc (Debian 4.4.2-9) 4.4.3 20100108 (prerelease) *** NO PREDEFINED KERNEL COMPILER IS DETECTED *** Assuming the same compiler is used with the current system compiler. *** Please make sure that the same compiler version was used for building kernel. checking for built-in ALSA... no checking for existing ALSA module... yes checking for Red Hat kernel... auto checking for Red Hat kernel... no checking for SUSE kernel... auto checking for SUSE kernel... no checking for updating alsa-kernel version.h... yes checking for CONFIG_EXPERIMENTAL... yes checking for directory to store kernel modules... /lib/modules/2.6.33.amd64-x2.030210.1/kernel/sound checking for verbose procfs... on checking for verbose printk... on checking for debug level... none checking for ISA support in kernel... no checking for processor type... k8 checking for ISA DMA API... yes checking for kernel linux/config.h... no Creating <linux/config.h>... checking for deprecated linux/config.h... checking to modify of kernel linux/kmod.h... no checking for kernel linux/utsrelease.h... yes checking for kernel linux/compiler.h... yes checking for kernel linux/pm.h... yes checking for kernel linux/spinlock.h... yes checking for kernel linux/irq.h... yes checking for kernel linux/threads.h... yes checking for kernel linux/rwsem.h... yes checking for kernel linux/gameport.h... yes checking for kernel media/v4l2-dev.h... yes checking for kernel linux/devfs_fs_kernel.h... no Creating a dummy <linux/devfs_fs_kernel.h>... checking for kernel linux/highmem.h... yes checking for kernel linux/workqueue.h... yes checking for kernel linux/dma-mapping.h... yes checking for kernel asm/hw_irq.h... yes checking for kernel linux/device.h... yes checking for kernel linux/platform_device.h... yes checking for kernel linux/isa.h... yes checking for kernel linux/jiffies.h... yes checking for kernel linux/compat.h... yes checking for kernel linux/log2.h... yes checking for kernel linux/adb.h... yes checking for kernel linux/cuda.h... yes checking for kernel linux/pmu.h... yes checking for kernel linux/io.h... yes checking for kernel linux/moduleparam.h... yes checking for kernel linux/syscalls.h... yes checking for kernel linux/firmware.h... yes checking for kernel linux/err.h... yes checking for kernel linux/bitmap.h... yes checking for kernel linux/mutex.h... yes checking for kernel linux/pm_qos_params.h... yes checking for kernel linux/kthread.h... yes checking for kernel asm/irq_regs.h... yes checking for kernel linux/seq_file.h... yes checking for kernel linux/debugfs.h... yes checking for kernel linux/gpio.h... yes checking for kernel linux/bug.h... yes checking for kernel linux/math64.h... yes checking for kernel linux/regulator/consumer.h... yes checking for kernel linux/dmi.h... yes checking for kernel linux/bitrev.h... yes checking for kernel linux/hrtimer.h... yes checking for kernel linux/gcd.h... yes checking for kernel module symbol versions... no checking for has ioport support... yes checking for PCI support in kernel... yes checking for I2C driver in kernel... yes checking for I2C_POWERMAC in kernel... unknown checking for firmware loader... yes checking for input subsystem in kernel... yes checking for pc-speaker platform in kernel... yes checking for SMP... yes checking for Video device support in kernel... no checking for ISA PnP driver in kernel... yes checking for PnP driver in kernel... yes checking for Kernel ISA-PnP support... no checking for Kernel ISA-PnP module support... no checking for strlcpy... yes checking for snprintf... yes checking for vsnprintf... yes checking for scnprintf... yes checking for sscanf... yes checking for vmalloc_to_page... yes checking for old kmod... no checking for PDE... yes checking for pci_set_consistent_dma_mask... yes checking for pci_dev_present... yes checking for msleep... yes checking for msleep_interrupt... yes checking for msecs_to_jiffies... yes checking for tty->count is the atomic type... yes checking for io_remap_pfn_range... yes checking for kcalloc... yes checking for kstrdup... yes checking for kstrndup... yes checking for kzalloc... yes checking for create_workqueue with flags... no checking for pci_ioremap_bar... yes checking for saved_config_space in pci_dev... yes checking for new pci_save_state... yes checking for register_sound_special_device... yes checking for driver extra-version... checking for driver version... 1.0.22.1 checking for dynamic minor numbers... no checking for sequencer support... no checking for OSS/Free emulation... no checking for RTC support... yes checking for HPET support... no checking for Procfs support... yes checking for USB support... yes checking for class_simple... no checking for old driver suspend/resume callbacks... no checking for removal of page-reservation for nopage/mmap... no checking for nested class_device... no checking for new IRQ handler... yes checking for gfp_t... yes checking for GFP_DMA32... yes checking for page_to_pfn... yes checking for PnP suspend/resume... yes checking for device_create_drvdata... no checking for new unlocked/compat_ioctl... yes checking for builtin _Bool support... yes checking for x86-compatible PC... no checking for High-Res timers... yes checking for kernel PCMCIA checking for PCMCIA support... no checking for PCMCIA module support... no checking for PC9800 support in kernel... no checking for parallel port support... no checking for parallel port module support... no checking for power management... yes checking for CONFIG_HAS_DMA... yes checking for cards to compile driver for... hda-intel checking for additonal options to compile driver for... hda-hwdep hda-reconfig hda-codec-realtek hda-codec-nvhdmi configure: creating ./config.status config.status: creating version config.status: creating Makefile.conf config.status: WARNING: Makefile.conf.in seems to ignore the --datarootdir setting config.status: creating snddevices config.status: creating utils/alsa-driver.spec config.status: creating utils/buildrpm config.status: creating toplevel.config config.status: creating utils/alsasound config.status: creating utils/alsasound.posix config.status: creating include/pci_ids_compat.h config.status: creating include/i2c-id_compat.h config.status: creating include/config.h config.status: creating include/config1.h config.status: creating include/version.h config.status: include/version.h is unchanged config.status: creating include/autoconf-extra.h config.status: include/autoconf-extra.h is unchanged Hacking autoconf.h... test:/usrc/alsa/alsa-driver-1.0.22.1$ make make -C /lib/modules/2.6.33.amd64-x2.030210.1/source SUBDIRS=/usr/local/src/alsa/alsa-driver-1.0.22.1 CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/linux-2.6.33' CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/hwdep.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/memory_wrapper.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/memalloc.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/sgbuf.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/pcm.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/pcm_native.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/pcm_lib.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/pcm_timer.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/pcm_misc.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/pcm_memory.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/timer.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/wrappers.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/misc_driver.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/sound.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/init.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/memory.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/info.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/control.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/misc.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/device.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/isadma.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/vmaster.o LD [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/snd.o LD [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/snd-hwdep.o LD [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/snd-timer.o LD [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/snd-pcm.o LD [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/acore/snd-page-alloc.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.o In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:219:1: warning: "AC_VERB_GET_CVT_CHAN_COUNT" redefined In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:29, from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/hda_codec.h:93:1: warning: this is the location of the previous definition In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:220:1: warning: "AC_VERB_GET_HDMI_DIP_DATA" redefined In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:29, from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/hda_codec.h:97:1: warning: this is the location of the previous definition In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:221:1: warning: "AC_VERB_GET_HDMI_DIP_XMIT" redefined In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:29, from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/hda_codec.h:98:1: warning: this is the location of the previous definition In file included from /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c:3: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c: In function âhdmi_checksum_audio_infoframeâ: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:334: warning: unused variable âbytesâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c: In function âhdmi_setup_channel_allocationâ: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:397: warning: unused variable âspk_maskâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:396: warning: unused variable âiâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:395: warning: unused variable âeldâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:394: warning: unused variable âspecâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c: In function âhdmi_setup_audio_infoframeâ: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:545: warning: unused variable âiâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:543: warning: unused variable âspecâ /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_nvhdmi.c: At top level: /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:89: warning: âeld_speaker_allocation_bitsâ defined but not used /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:285: warning: âhdmi_enable_outputâ defined but not used /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/../../alsa-kernel/pci/hda/patch_nvhdmi.c:366: warning: âinit_channel_allocationsâ defined but not used CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/patch_realtek.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/hda_codec.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/hda_proc.o CC [M] /usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/hda_hwdep.o make[4]: *** No rule to make target `/usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/hda_intel.o', needed by `/usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda/snd-hda-codec.o'. Stop. make[3]: *** [/usr/local/src/alsa/alsa-driver-1.0.22.1/pci/hda] Error 2 make[2]: *** [/usr/local/src/alsa/alsa-driver-1.0.22.1/pci] Error 2 make[1]: *** [_module_/usr/local/src/alsa/alsa-driver-1.0.22.1] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.33' make: *** [compile] Error 2 test:/usrc/alsa/alsa-driver-1.0.22.1$

1 0

[alsa-devel] No CD mixer on Intel G35 (Asus P5E-V(M))
by Mario 'BitKoenig' Holbe 02 Mar '10

02 Mar '10

Hello, on my Asus P5E-V HDMI and my Asus P5E-VM HDMI (both Intel G35) alsamixer shows no CD mixer. Both boards have a CD audio connector and the CD-drive connected to it. holbe@darkside:/home/holbe% uname -a Linux darkside 2.6.32-3-686-bigmem #1 SMP Thu Feb 25 06:54:30 UTC 2010 i686 GNU/Linux holbe@darkside:/home/holbe% cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfe5f4000 irq 22 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xfdfff000, irq 16 alsamixer says: Card: HDA Intel Chip: Silicon Image SiI1392 HDMI Same with 2.6.33. On my P5Q-EM (Intel G45) the CD mixer appears in alsamixer. I have no idea how to dig deeper in this, if you need more information, please just tell me. Thanks for your work & regards Mario -- Do I contradict myself? Very well, then I contradict myself, I am large, I contain multitudes. -- Walt Whitman

2 2

[alsa-devel] [PATCH 22/66] sound/oss/v_midi.h: Checkpatch cleanup
by Andrea Gelmini 02 Mar '10

02 Mar '10

sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible sound/oss/v_midi.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- sound/oss/v_midi.h | 5 ++--- 1 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h index 1b86cb4..08e2185 100644 --- a/sound/oss/v_midi.h +++ b/sound/oss/v_midi.h @@ -2,9 +2,9 @@ typedef struct vmidi_devc { int dev; /* State variables */ - int opened; + int opened; spinlock_t lock; - + /* MIDI fields */ int my_mididev; int pair_mididev; @@ -12,4 +12,3 @@ typedef struct vmidi_devc { int intr_active; void (*midi_input_intr) (int dev, unsigned char data); } vmidi_devc; - -- 1.7.0.90.g251a4

2 2

Re: [alsa-devel] [Fwd: Re: [Regression] With the new 2.6.33 when I plug headphones, the speaker doen't get off anymore]
by Takashi Iwai 02 Mar '10

02 Mar '10

At Mon, 01 Mar 2010 14:46:10 +0100, Guillaume De Nayer wrote: > > Takashi Iwai a écrit : > > At Sun, 28 Feb 2010 15:29:25 +0100, > > giggz wrote: > >> giggz a écrit : > >>> giggzounet a écrit : > >>>> Hi, > >>>> > >>>> I have installed debian stable lenny + backports on an eeepc 1201n. With > >>>> the 2.6.30 or 2.6.32 from lenny-backport I don't have problem with sound > >>>> : when I plug the headphones, speakers get off. > >>>> > >>>> I have tested with 2.6.33 and when I plug headphones, speakers don't get > >>>> off anymore. so I have sound in headphones and speakers. > >>>> > >>>> 22:42 giggz@baal ~ % cat /proc/asound/card0/codec#* | grep Codec > >>>> Codec: Realtek ALC269 > >>>> Codec: Nvidia MCP7A HDMI > >>>> > >>>> I attach the output of alsa-info with the 2.6.32.9 and the 2.6.33 (I'm > >>>> in a Uni Campus and it's quite difficult to have internet on a external > >>>> laptop...so I attach them at this mail and I don't upload them on pastbin). > >>>> > >>>> I have opened a bug on the kernel bugzilla (bug 15399). Should I provide > >>>> anything more ? > >>>> > >>> Following the advice of Paul Menzel I did a "diff" of the sources of the > >>> 2.6.32.9 kernel and of the 2.6.33. In patch_realtek.c, we see that there > >>> is an new snd_hda_jack_detect funtion with 2 arguments. I have noticed > >>> that the second argument is "normaly" the second argument of > >>> snd_hda_codec_read. In the alc269_speaker_automute function there is > >>> this new snd_hda_jack_detect function, but the second argument is "nid". > >>> But in the old alc269_speaker_automute of the 2.6.32.9 the second > >>> argument of snd_hda_codec_read is 0x15. So I think there is perhaps a > >>> bug here...But I hesitate to modify the source...so I'm waiting the > >>> anwser of the dev. > >>> > >>> --- patch_realtek.c 2010-02-27 14:58:06.000000000 +0100 > >>> +++ patch_realtek_modif.c 2010-02-27 14:58:54.000000000 +0100 > >>> @@ -13381,7 +13381,7 @@ > >>> unsigned int present; > >>> unsigned char bits; > >>> > >>> - present = snd_hda_jack_detect(codec, nid); > >>> + present = snd_hda_jack_detect(codec, 0x15); > >>> bits = present ? AMP_IN_MUTE(0) : 0; > >>> snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, > >>> AMP_IN_MUTE(0), bits); > >>> > >> I tested it. And with this modif in the kernel, after recompilation and > >> reboot, my problem is headphones and speaker is gone. > > > > It implies that the setup of the headphone pin is wrong. This doesn't > > mean a bug of the driver, though. It might have uncovered the BIOS > > bug. > > > > ok. I don't have any knowledge in C or in bios...so you're problably > right :) > > but it seems strange that for all the new line with snd_hda_jack_detect > the second argument is always the second argument of snd_hda_codec_read > except for the alc269. > > > Could you provide alsa-info.sh without your patch to check the setup? > > > > > > In attachment of the first post on alsa-devel or on the kernel bugzilla > http://bugzilla.kernel.org/show_bug.cgi?id=15399, there are the output > of alsa-info.sh with the 2.6.32.9 (no problem) and the output of > alsa-info.sh with the 2.6.33 without any patch (problem occurs). OK, thanks. This looks actually like a bug :) But, the easiest (and probably better) fix is to use model=auto for your device. Just pass model=auto option. I guess everything should work with that. Takashi

2 1

[alsa-devel] dmix format S24_LE
by Bart De Vos 02 Mar '10

02 Mar '10

Hi, I've added support for the S24_LE format in the dmix plugin. This has been tested 'only' on i.MX31 (arm11) with sound DAC AK4367 over I2S, which supports this format natively. Patch is against alsa-lib-1.0.22. Please let me know your thoughts. diff -urN alsa-lib-1.0.22-orig/src/pcm/pcm_direct.c alsa-lib-1.0.22/src/pcm/pcm_direct.c --- alsa-lib-1.0.22-orig/src/pcm/pcm_direct.c 2009-12-16 16:18:51.000000000 +0100 +++ alsa-lib-1.0.22/src/pcm/pcm_direct.c 2010-03-02 10:01:00.000000000 +0100 @@ -888,6 +888,7 @@ SND_PCM_FORMAT_S32 ^ SND_PCM_FORMAT_S32_LE ^ SND_PCM_FORMAT_S32_BE, SND_PCM_FORMAT_S16, SND_PCM_FORMAT_S16 ^ SND_PCM_FORMAT_S16_LE ^ SND_PCM_FORMAT_S16_BE, + SND_PCM_FORMAT_S24_LE, SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_U8, }; diff -urN alsa-lib-1.0.22-orig/src/pcm/pcm_dmix.c alsa-lib-1.0.22/src/pcm/pcm_dmix.c --- alsa-lib-1.0.22-orig/src/pcm/pcm_dmix.c 2009-12-16 16:18:51.000000000 +0100 +++ alsa-lib-1.0.22/src/pcm/pcm_dmix.c 2010-03-02 10:02:39.000000000 +0100 @@ -174,6 +174,10 @@ sample_size = 4; do_mix_areas = (mix_areas_t *)dmix->u.dmix.mix_areas_32; break; + case SND_PCM_FORMAT_S24_LE: + sample_size = 4; + do_mix_areas = (mix_areas_t *)dmix->u.dmix.mix_areas_24; + break; case SND_PCM_FORMAT_S24_3LE: sample_size = 3; do_mix_areas = (mix_areas_t *)dmix->u.dmix.mix_areas_24; @@ -238,6 +242,10 @@ sample_size = 4; do_remix_areas = (mix_areas_t *)dmix->u.dmix.remix_areas_32; break; + case SND_PCM_FORMAT_S24_LE: + sample_size = 4; + do_remix_areas = (mix_areas_t *)dmix->u.dmix.remix_areas_24; + break; case SND_PCM_FORMAT_S24_3LE: sample_size = 3; do_remix_areas = (mix_areas_t *)dmix->u.dmix.remix_areas_24; diff -urN alsa-lib-1.0.22-orig/src/pcm/pcm_dmix_generic.c alsa-lib-1.0.22/src/pcm/pcm_dmix_generic.c --- alsa-lib-1.0.22-orig/src/pcm/pcm_dmix_generic.c 2009-12-16 16:18:51.000000000 +0100 +++ alsa-lib-1.0.22/src/pcm/pcm_dmix_generic.c 2010-03-02 10:37:16.000000000 +0100 @@ -122,7 +122,7 @@ #define generic_dmix_supported_format \ ((1ULL << SND_PCM_FORMAT_S16_LE) | (1ULL << SND_PCM_FORMAT_S32_LE) |\ (1ULL << SND_PCM_FORMAT_S16_BE) | (1ULL << SND_PCM_FORMAT_S32_BE) |\ - (1ULL << SND_PCM_FORMAT_S24_3LE) | \ + (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S24_3LE)|\ (1ULL << SND_PCM_FORMAT_U8)) #include <byteswap.h> -- -------------------------------------------------- Bart De Vos Development Engineer mailto:bart.devos@eia.be EIA Electronics Vluchtenburgstraat 3 B-2630 Aartselaar (Belgium) phone : +32(0)3 870 85 29 fax : +32(0)3 877 14 72 website : http://www.eia.be/ -------------------------------------------------- Basically, perfect development is impossible. Development can be fast, good and cheap. Pick two! — Larry Wall --------------------------------------------------

2 1

[alsa-devel] [PATCH 1/1] sound/pci/hda/hda_codec.c: various coding style fixes
by Norberto Lopes 02 Mar '10

02 Mar '10

Signed-off-by: Norberto Lopes <nlopes.ml(a)gmail.com> --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++-------------------- 1 files changed, 38 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f98b47c..444b989 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -955,8 +955,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * * Returns 0 if successful, or a negative error code. */ -int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp) +int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, + unsigned int codec_addr, + struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -1162,7 +1163,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); */ /* FIXME: more better hash key? */ -#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_KEY(nid, dir, idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) #define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) @@ -1332,7 +1333,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) if (!codec->no_trigger_sense) { pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); } return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); @@ -1348,8 +1350,8 @@ EXPORT_SYMBOL_HDA(snd_hda_pin_sense); */ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return !!(sense & AC_PINSENSE_PRESENCE); + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect); @@ -1891,7 +1893,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - + for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; int i = 0; @@ -2378,27 +2380,27 @@ static struct snd_kcontrol_new dig_mixes[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_default_get, .put = snd_hda_spdif_default_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), .info = snd_hda_spdif_out_switch_info, .get = snd_hda_spdif_out_switch_get, .put = snd_hda_spdif_out_switch_put, @@ -2549,7 +2551,7 @@ static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new dig_in_ctls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH), .info = snd_hda_spdif_in_switch_info, .get = snd_hda_spdif_in_switch_get, .put = snd_hda_spdif_in_switch_put, @@ -2557,7 +2559,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_in_status_get, }, @@ -2822,7 +2824,7 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) int err = snd_hda_codec_build_controls(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build controls" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -2918,8 +2920,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, val |= channels - 1; switch (snd_pcm_format_width(format)) { - case 8: val |= 0x00; break; - case 16: val |= 0x10; break; + case 8: + val |= 0x00; + break; + case 16: + val |= 0x10; + break; case 20: case 24: case 32: @@ -3235,7 +3241,8 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; - snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); + snd_printk(KERN_WARNING "Too many %s devices\n", + snd_hda_pcm_type_name[type]); return -EAGAIN; } @@ -3273,7 +3280,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) err = codec->patch_ops.build_pcms(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build PCMs" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -3403,8 +3410,8 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** * snd_hda_check_board_codec_sid_config - compare the current codec - subsystem ID with the - config table + subsystem ID with the + config table This is important for Gateway notebooks with SB450 HDA Audio where the vendor ID of the PCI device is: @@ -3542,7 +3549,7 @@ void snd_hda_update_power_acct(struct hda_codec *codec) * * Increment the power-up counter and power up the hardware really when * not turned on yet. - */ + */ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; @@ -3571,7 +3578,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); * * Decrement the power-up counter and schedules the power-off work if * the counter rearches to zero. - */ + */ void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; @@ -3597,7 +3604,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_down); * * This function is supposed to be set or called from the check_power_status * patch ops. - */ + */ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) @@ -3765,7 +3772,7 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, { /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) - set_dig_out_convert(codec, nid, + set_dig_out_convert(codec, nid, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff, -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -4024,13 +4031,13 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) /* * Sort an associated group of pins according to their sequence numbers. */ -static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, +static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, int num_pins) { int i, j; short seq; hda_nid_t nid; - + for (i = 0; i < num_pins; i++) { for (j = i + 1; j < num_pins; j++) { if (sequences[i] > sequences[j]) { @@ -4058,7 +4065,7 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. - * + * * The analog input pins are assigned to input_pins array. * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. @@ -4121,9 +4128,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (! assoc) + if (!assoc) continue; - if (! assoc_speaker) + if (!assoc_speaker) assoc_speaker = assoc; else if (assoc_speaker != assoc) continue; @@ -4221,7 +4228,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->speaker_outs); sort_pins_by_sequence(cfg->hp_pins, sequences_hp, cfg->hp_outs); - + /* if we have only one mic, make it AUTO_PIN_MIC */ if (!cfg->input_pins[AUTO_PIN_MIC] && cfg->input_pins[AUTO_PIN_FRONT_MIC]) { @@ -4371,7 +4378,7 @@ EXPORT_SYMBOL_HDA(snd_hda_resume); /** * snd_array_new - get a new element from the given array * @array: the array object - * + * * Get a new element from the given array. If it exceeds the * pre-allocated array size, re-allocate the array. * -- 1.6.4

2 1

Re: [alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 02 Mar '10

02 Mar '10

I forgot this attachment 1.patch and alsa-info.txt.

4 20

[alsa-devel] [PATCHv4 7/7] ASoC: TWL6030: Detect power-up sequence completion
by Olaya, Margarita 02 Mar '10

02 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. If NAUDINT irq is provided, CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. If AUDPWRON gpio line is provided but NAUDINT irq is not, then CODEC driver enables READYINT and polls on INTID register. If none of them are provided, then CODEC uses manual power sequences and disables all audio interrupts. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 61 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 1 + 2 files changed, 53 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index b8dd5ae..2847f1b 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -52,6 +52,7 @@ struct twl6030_data { int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; }; /* @@ -372,6 +373,7 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, static irqreturn_t twl6030_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; u8 intid; twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6030_REG_INTID); @@ -391,7 +393,7 @@ static irqreturn_t twl6030_naudint_handler(int irq, void *data) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + complete(&priv->ready); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -626,11 +628,45 @@ static int twl6030_add_widgets(struct snd_soc_codec *codec) return 0; } +static int twl6030_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6030_data *priv = codec->private_data; + int time_left; + u8 intid; + + if (naudint) { + /* wait for ready interrupt with 48 ms timeout */ + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + } else { + /* retry 3 times only */ + for (time_left = 3; time_left > 0; time_left--) { + mdelay(16); + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6030_REG_INTID); + if (intid & TWL6030_READYINT) + break; + } + } + + if (!time_left) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + + priv->codec_powered = 1; + + return 0; +} + static int twl6030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct twl6030_data *priv = codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -643,8 +679,10 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for power-up completion */ + ret = twl6030_power_up_completion(codec, naudint); + if (ret) + return ret; /* sync registers updated during power-up sequence */ twl6030_read(codec, TWL6030_REG_NCPCTL); @@ -653,12 +691,11 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -1067,6 +1104,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1089,10 +1127,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) if (ret) goto gpio2_err; } else { - dev_warn(codec->dev, - "no naudint irq, audio interrupts disabled\n"); - twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, - TWL6030_ALLINT_MSK); + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + ~TWL6030_READYMSK & TWL6030_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 90b8a44..d591be1 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6030_READYMSK 0x40 #define TWL6030_ALLINT_MSK 0x7B /* NCPCTL (0x05) fields */ -- 1.6.1.3

2 8

[alsa-devel] Audio4DJ (snd-usb-caiaq) noise when using DVB usb adap
by Pedro Ribeiro 02 Mar '10

02 Mar '10

Hi all, I'm encountering a problem with the Audio4DJ audio card with a DVB USB adapter (Win-TV NOVA-T USB2 STICK). The problem is strange because it only happens after I disconnect and reconnect the audio card. Procedure 1: - boot from fresh - plug the DVB adapter - plug the Audio4DJ Everything works correctly - meaning I can watch TV and my audio is fine. Procedure 2: - boot from fresh - plug the Audio4DJ - plug the DVB adapter The audio starts cracking every few seconds, and constantly if I use jackd. Procedure 3: - boot from fresh - plug the DVB adapter - plug the Audio4DJ (everything is correct, as above in Procedure 1) - unplug the Audio4DJ - plug it again Cracking audio. I'm insisting in "boot from fresh" because from now on (that is, from the moment that I unplug the Audio4DJ) I always get cracking audio. rmmod'ing the snd-usb-caiaq and the usb-dvb modules produces the same result. I have to boot from fresh and do Procedure 1 for it to work OK. Can anyone please give a hint on how to debug this? Regards, Pedro

1 0

[alsa-devel] Audio4DJ (snd-usb-caiaq) noise when using DVB usb adapter
by Pedro Ribeiro 02 Mar '10

02 Mar '10

Hi all, I'm encountering a problem with the Audio4DJ audio card with a DVB USB adapter (Win-TV NOVA-T USB2 STICK). The problem is strange because it only happens after I disconnect and reconnect the audio card. Procedure 1: - boot from fresh - plug the DVB adapter - plug the Audio4DJ Everything works correctly - meaning I can watch TV and my audio is fine. Procedure 2: - boot from fresh - plug the Audio4DJ - plug the DVB adapter The audio starts cracking every few seconds, and constantly if I use jackd. Procedure 3: - boot from fresh - plug the DVB adapter - plug the Audio4DJ (everything is correct, as above in Procedure 1) - unplug the Audio4DJ - plug it again Cracking audio. I'm insisting in "boot from fresh" because from now on (that is, from the moment that I unplug the Audio4DJ) I always get cracking audio. rmmod'ing the snd-usb-caiaq and the usb-dvb modules produces the same result. I have to boot from fresh and do Procedure 1 for it to work OK. Can anyone please give a hint on how to debug this? (Please CC me directly as I'm not in both lists). Regards, Pedro

1 0

[alsa-devel] [PATCH 2.6.34] ASoC: Fix S3C64xx IIS driver for Samsung header reorg
by Mark Brown 01 Mar '10

01 Mar '10

The reorgs of the Samsung headers have moved the GPIO bank definitions from plat/ to mach/ - the IIS driver needs to be updated to take care of this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Ideally this could also go in via the same path as the ARM/Samsung changes which do the moves. If those get merged without this fix then it may as well go via ASoC. sound/soc/s3c24xx/s3c64xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 93ed3aa..a72c251 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -22,8 +22,8 @@ #include <sound/soc.h> #include <plat/regs-s3c2412-iis.h> -#include <plat/gpio-bank-d.h> -#include <plat/gpio-bank-e.h> +#include <mach/gpio-bank-d.h> +#include <mach/gpio-bank-e.h> #include <plat/gpio-cfg.h> #include <mach/map.h> -- 1.7.0

1 0

[alsa-devel] [Regression] With the new 2.6.33 when I plug headphones, the speaker doen't get off anymore
by giggzounet 01 Mar '10

01 Mar '10

Hi, I have installed debian stable lenny + backports on an eeepc 1201n. With the 2.6.30 or 2.6.32 from lenny-backport I don't have problem with sound : when I plug the headphones, speakers get off. I have tested with 2.6.33 and when I plug headphones, speakers don't get off anymore. so I have sound in headphones and speakers. 22:42 giggz@baal ~ % cat /proc/asound/card0/codec#* | grep Codec Codec: Realtek ALC269 Codec: Nvidia MCP7A HDMI I attach the output of alsa-info with the 2.6.32.9 and the 2.6.33 (I'm in a Uni Campus and it's quite difficult to have internet on a external laptop...so I attach them at this mail and I don't upload them on pastbin). I have opened a bug on the kernel bugzilla (bug 15399). Should I provide anything more ? Thx a lot, Best regards Guillaume

4 14

[alsa-devel] [PATCH 3/3] ALSA: usb-audio: support multiple formats with audio class v2 devices
by Clemens Ladisch 01 Mar '10

01 Mar '10

Change the parser to correctly handle v2 descriptors with multiple format bits set. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- linux/sound/usb/format.c +++ linux/sound/usb/format.c @@ -37,19 +37,20 @@ * @format: the format tag (wFormatTag) * @fmt: the format type descriptor */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, +static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, int format, void *_fmt, int protocol) { - int pcm_format, i; int sample_width, sample_bytes; + u64 pcm_formats; switch (protocol) { case UAC_VERSION_1: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; + format = 1 << format; break; } @@ -57,24 +58,7 @@ static int parse_audio_format_i_type(str struct uac_format_type_i_ext_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubslotSize; - - /* - * FIXME - * USB audio class v2 devices specify a bitmap of possible - * audio formats rather than one fix value. For now, we just - * pick one of them and report that as the only possible - * value for this setting. - * The bit allocation map is in fact compatible to the - * wFormatTag of the v1 AS streaming descriptors, which is why - * we can simply map the matrix. - */ - - for (i = 0; i < 5; i++) - if (format & (1UL << i)) { - format = i + 1; - break; - } - + format <<= 1; break; } @@ -82,15 +66,15 @@ static int parse_audio_format_i_type(str return -EINVAL; } - /* FIXME: correct endianess and sign? */ - pcm_format = -1; + pcm_formats = 0; - switch (format) { - case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ + if (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED)) { + /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); - /* fall-through */ - case UAC_FORMAT_TYPE_I_PCM: + format = 1 << UAC_FORMAT_TYPE_I_PCM; + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -99,22 +83,22 @@ static int parse_audio_format_i_type(str /* check the format byte size */ switch (sample_bytes) { case 1: - pcm_format = SNDRV_PCM_FORMAT_S8; + pcm_formats |= SNDRV_PCM_FMTBIT_S8; break; case 2: if (snd_usb_is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; /* grrr, big endian!! */ + pcm_formats |= SNDRV_PCM_FMTBIT_S16_BE; /* grrr, big endian!! */ else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S16_LE; break; case 3: if (snd_usb_is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S24_3BE; /* grrr, big endian!! */ + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3BE; /* grrr, big endian!! */ else - pcm_format = SNDRV_PCM_FORMAT_S24_3LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3LE; break; case 4: - pcm_format = SNDRV_PCM_FORMAT_S32_LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S32_LE; break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", @@ -122,30 +106,29 @@ static int parse_audio_format_i_type(str sample_width, sample_bytes); break; } - break; - case UAC_FORMAT_TYPE_I_PCM8: - pcm_format = SNDRV_PCM_FORMAT_U8; - + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM8)) { /* Dallas DS4201 workaround: it advertises U8 format, but really supports S8. */ if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - pcm_format = SNDRV_PCM_FORMAT_S8; - break; - case UAC_FORMAT_TYPE_I_IEEE_FLOAT: - pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; - break; - case UAC_FORMAT_TYPE_I_ALAW: - pcm_format = SNDRV_PCM_FORMAT_A_LAW; - break; - case UAC_FORMAT_TYPE_I_MULAW: - pcm_format = SNDRV_PCM_FORMAT_MU_LAW; - break; - default: - snd_printk(KERN_INFO "%d:%u:%d : unsupported format type %d\n", + pcm_formats |= SNDRV_PCM_FMTBIT_S8; + else + pcm_formats |= SNDRV_PCM_FMTBIT_U8; + } + if (format & (1 << UAC_FORMAT_TYPE_I_IEEE_FLOAT)) { + pcm_formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; + } + if (format & (1 << UAC_FORMAT_TYPE_I_ALAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_A_LAW; + } + if (format & (1 << UAC_FORMAT_TYPE_I_MULAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_MU_LAW; + } + if (format & ~0x3f) { + snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n", chip->dev->devnum, fp->iface, fp->altsetting, format); - break; } - return pcm_format; + return pcm_formats; } @@ -313,14 +296,14 @@ static int parse_audio_format_i(struct s default: pcm_format = SNDRV_PCM_FORMAT_S16_LE; } + fp->formats = 1uLL << pcm_format; } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); - if (pcm_format < 0) + fp->formats = parse_audio_format_i_type(chip, fp, format, + fmt, protocol); + if (!fp->formats) return -1; } - fp->formats = 1uLL << pcm_format; - /* gather possible sample rates */ /* audio class v1 reports possible sample rates as part of the * proprietary class specific descriptor.

2 2

[PATCH] ALSA: usb-audio: fix quirk formats
by Daniel Mack 01 Mar '10

01 Mar '10

'struct audioformat' takes a bit field of supported rates now rather than one fixed value. Adopt that change at two more places. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/quirks-table.h | 2 +- sound/usb/quirks.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index ff2df4e..2107d00 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2233,7 +2233,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3BE, + .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 1, .altsetting = 1, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 00e8a97..0119fa0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -89,7 +89,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua1000_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_S32_LE, .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, -- 1.6.6.2

1 0

[alsa-devel] [PATCH] opti92x: use PnP data to select Master Control port
by Krzysztof Helt 01 Mar '10

01 Mar '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The Master Control port (MC) is available as the last PnP resource (OPT005). Use this value instead fo guessing. Also, add some comments to the code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 120 +++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 43 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b865e45..5913717 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1558,7 +1558,7 @@ static int __devinit snd_card_miro_pnp(struct snd_miro *chip, err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); return err; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index a4af53b..becd90d 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -144,12 +144,8 @@ struct snd_opti9xx { spinlock_t lock; + long wss_base; int irq; - -#ifdef CONFIG_PNP - struct pnp_dev *dev; - struct pnp_dev *devmpu; -#endif /* CONFIG_PNP */ }; static int snd_opti9xx_pnp_is_probed; @@ -159,12 +155,17 @@ static int snd_opti9xx_pnp_is_probed; static struct pnp_card_device_id snd_opti9xx_pnpids[] = { #ifndef OPTi93X /* OPTi 82C924 */ - { .id = "OPT0924", .devs = { { "OPT0000" }, { "OPT0002" } }, .driver_data = 0x0924 }, + { .id = "OPT0924", + .devs = { { "OPT0000" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0924 }, /* OPTi 82C925 */ - { .id = "OPT0925", .devs = { { "OPT9250" }, { "OPT0002" } }, .driver_data = 0x0925 }, + { .id = "OPT0925", + .devs = { { "OPT9250" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0925 }, #else /* OPTi 82C931/3 */ - { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, .driver_data = 0x0931 }, + { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, + .driver_data = 0x0931 }, #endif /* OPTi93X */ { .id = "" } }; @@ -207,24 +208,34 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, chip->hardware = hardware; strcpy(chip->name, snd_opti9xx_names[hardware]); - chip->mc_base_size = opti9xx_mc_size[hardware]; - spin_lock_init(&chip->lock); chip->irq = -1; +#ifndef OPTi93X +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; +#endif /* CONFIG_PNP */ + else { + chip->mc_base = 0xf8c; + chip->mc_base_size = opti9xx_mc_size[hardware]; + } +#else + chip->mc_base_size = opti9xx_mc_size[hardware]; +#endif + switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = (hardware == OPTi9XX_HW_82C928) ? 0xe2 : 0xe3; chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: case OPTi9XX_HW_82C925: - chip->mc_base = 0xf8c; chip->password = 0xe5; chip->pwd_reg = 3; break; @@ -292,7 +303,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, spin_unlock_irqrestore(&chip->lock, flags); return retval; } - + static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, unsigned char value) { @@ -341,7 +352,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, - long wss_base, + long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) { @@ -354,16 +365,23 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, switch (chip->hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C924: + /* opti 929 mode (?), OPL3 clock output, audio enable */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); + /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); case OPTi9XX_HW_82C925: + /* enable WSS mode */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + /* OPL3 FM synthesis */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20); + /* disable Sound Blaster IRQ and DMA */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); #ifdef CS4231 + /* cs4231/4248 fix enabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); #else + /* cs4231/4248 fix disabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02); #endif /* CS4231 */ break; @@ -411,21 +429,26 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, return -EINVAL; } - switch (wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (port & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", port); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -487,7 +510,7 @@ __skip_base: #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, wss_base); + outb(irq_bits << 3 | dma_bits, chip->wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ @@ -729,15 +752,15 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, { struct pnp_dev *pdev; int err; + struct pnp_dev *devmpu; +#ifndef OPTi93X + struct pnp_dev *devmc; +#endif - chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - - pdev = chip->dev; - err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); @@ -750,9 +773,24 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else - if (pid->driver_data != 0x0924) - port = pnp_port_start(pdev, 1); + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); + return err; + } + + port = pnp_port_start(pdev, 1); fm_port = pnp_port_start(pdev, 2) + 8; + /* + * The MC(0) is never accessed and card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; #endif /* OPTi93X */ irq = pnp_irq(pdev, 0); dma1 = pnp_dma(pdev, 0); @@ -760,16 +798,16 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, dma2 = pnp_dma(pdev, 1); #endif /* CS4231 || OPTi93X */ - pdev = chip->devmpu; - if (pdev && mpu_port > 0) { - err = pnp_activate_dev(pdev); + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + + if (devmpu && mpu_port > 0) { + err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); mpu_port = -1; - chip->devmpu = NULL; } else { - mpu_port = pnp_port_start(pdev, 0); - mpu_irq = pnp_irq(pdev, 0); + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); } } return pid->driver_data; @@ -824,7 +862,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; - error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, + error = snd_wss_create(card, chip->wss_base + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -865,10 +903,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, port + 4, irq, dma1, xdma2); + card->shortname, pcm->name, + chip->wss_base + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, port + 4, irq, dma1); + card->shortname, pcm->name, chip->wss_base + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) @@ -1062,9 +1101,6 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, snd_card_free(card); return error; } - if (hw <= OPTi9XX_HW_82C930) - chip->mc_base -= 0x80; - error = snd_opti9xx_read_check(chip); if (error) { snd_printk(KERN_ERR "OPTI chip not found\n"); -- 1.6.4 ---------------------------------------------------------------------- Codziennie 11 tys. ofert pracy Sprawdz >>> http://link.interia.pl/f25ac

2 2

[alsa-devel] [PATCH 2/3] ALSA: usb-audio: use a format bitmask per alternate setting
by Clemens Ladisch 01 Mar '10

01 Mar '10

In preparation for USB audio 2.0 support, change the audioformat structure so that it uses a bitmask to specify possible formats. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- linux/sound/usb/card.h +++ linux/sound/usb/card.h @@ -9,7 +9,7 @@ struct audioformat { struct list_head list; - snd_pcm_format_t format; /* format type */ + u64 formats; /* ALSA format bits */ unsigned int channels; /* # channels */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int frame_size; /* samples per frame for non-audio */ --- linux/sound/usb/endpoint.c +++ linux/sound/usb/endpoint.c @@ -94,7 +94,7 @@ int snd_usb_add_audio_endpoint(struct sn if (subs->endpoint == fp->endpoint) { list_add_tail(&fp->list, &subs->fmt_list); subs->num_formats++; - subs->formats |= 1ULL << fp->format; + subs->formats |= fp->formats; return 0; } } @@ -268,7 +268,7 @@ int snd_usb_parse_audio_endpoints(struct */ if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && - fp->format == SNDRV_PCM_FORMAT_S16_LE && + fp->formats == SNDRV_PCM_FMTBIT_S16_LE && protocol == UAC_VERSION_1 && le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) --- linux/sound/usb/format.c +++ linux/sound/usb/format.c @@ -319,7 +319,7 @@ static int parse_audio_format_i(struct s return -1; } - fp->format = pcm_format; + fp->formats = 1uLL << pcm_format; /* gather possible sample rates */ /* audio class v1 reports possible sample rates as part of the @@ -361,16 +361,16 @@ static int parse_audio_format_ii(struct switch (format) { case UAC_FORMAT_TYPE_II_AC3: /* FIXME: there is no AC3 format defined yet */ - // fp->format = SNDRV_PCM_FORMAT_AC3; - fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ + // fp->formats = SNDRV_PCM_FMTBIT_AC3; + fp->formats = SNDRV_PCM_FMTBIT_U8; /* temporary hack to receive byte streams */ break; case UAC_FORMAT_TYPE_II_MPEG: - fp->format = SNDRV_PCM_FORMAT_MPEG; + fp->formats = SNDRV_PCM_FMTBIT_MPEG; break; default: snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", chip->dev->devnum, fp->iface, fp->altsetting, format); - fp->format = SNDRV_PCM_FORMAT_MPEG; + fp->formats = SNDRV_PCM_FMTBIT_MPEG; break; } --- linux/sound/usb/pcm.c +++ linux/sound/usb/pcm.c @@ -58,7 +58,9 @@ static struct audioformat *find_format(s list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (fp->format != format || fp->channels != channels) + if (!(fp->formats & (1uLL << format))) + continue; + if (fp->channels != channels) continue; if (rate < fp->rate_min || rate > fp->rate_max) continue; @@ -428,10 +430,15 @@ static int hw_check_valid_format(struct struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + struct snd_mask check_fmts; unsigned int ptime; /* check the format */ - if (!snd_mask_test(fmts, fp->format)) { + snd_mask_none(&check_fmts); + check_fmts.bits[0] = (u32)fp->formats; + check_fmts.bits[1] = (u32)(fp->formats >> 32); + snd_mask_intersect(&check_fmts, fmts); + if (snd_mask_empty(&check_fmts)) { hwc_debug(" > check: no supported format %d\n", fp->format); return 0; } @@ -584,7 +591,7 @@ static int hw_rule_format(struct snd_pcm fp = list_entry(p, struct audioformat, list); if (!hw_check_valid_format(subs, params, fp)) continue; - fbits |= (1ULL << fp->format); + fbits |= fp->formats; } oldbits[0] = fmt->bits[0]; --- linux/sound/usb/proc.c +++ linux/sound/usb/proc.c @@ -79,11 +79,16 @@ static void proc_dump_substream_formats( list_for_each(p, &subs->fmt_list) { struct audioformat *fp; + snd_pcm_format_t fmt; fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %s\n", - snd_pcm_format_name(fp->format)); + snd_iprintf(buffer, " Format:"); + for (fmt = 0; fmt <= SNDRV_PCM_FORMAT_LAST; ++fmt) + if (fp->formats & (1uLL << fmt)) + snd_iprintf(buffer, " %s", + snd_pcm_format_name(fmt)); + snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, --- linux/sound/usb/quirks-table.h +++ linux/sound/usb/quirks-table.h @@ -279,7 +279,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 4, .iface = 0, .altsetting = 1, @@ -296,7 +296,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -580,7 +580,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, @@ -597,7 +597,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -793,7 +793,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, @@ -810,7 +810,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 2, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 2, .altsetting = 1, --- linux/sound/usb/quirks.c +++ linux/sound/usb/quirks.c @@ -174,7 +174,7 @@ static int create_uaxx_quirk(struct snd_ const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua_format = { - .format = SNDRV_PCM_FORMAT_S24_3LE, + .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, --- linux/sound/usb/urb.c +++ linux/sound/usb/urb.c @@ -662,7 +662,7 @@ static int prepare_nodata_playback_urb(s urb->number_of_packets = ctx->packets; urb->transfer_buffer_length = offs * stride; memset(urb->transfer_buffer, - subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, + runtime->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, offs * stride); return 0; } @@ -924,7 +924,7 @@ void snd_usb_init_substream(struct snd_u snd_usb_set_pcm_ops(as->pcm, stream); list_add_tail(&fp->list, &subs->fmt_list); - subs->formats |= 1ULL << fp->format; + subs->formats |= fp->formats; subs->endpoint = fp->endpoint; subs->num_formats++; subs->fmt_type = fp->fmt_type;

1 0

[alsa-devel] [PATCH 1/3] ALSA: usb-audio: rename substream format field to altset_idx
by Clemens Ladisch 01 Mar '10

01 Mar '10

The snd_usb_substream::format field actually contains the index of the current alternate setting, so rename it to altset_idx to avoid confusion. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- linux/sound/usb/card.h +++ linux/sound/usb/card.h @@ -54,7 +54,7 @@ struct snd_usb_substream { struct audioformat *cur_audiofmt; /* current audioformat pointer (for hw_params callback) */ unsigned int cur_rate; /* current rate (for hw_params callback) */ unsigned int period_bytes; /* current period bytes (for hw_params callback) */ - unsigned int format; /* USB data format */ + unsigned int altset_idx; /* USB data format: index of alternate setting */ unsigned int datapipe; /* the data i/o pipe */ unsigned int syncpipe; /* 1 - async out or adaptive in */ unsigned int datainterval; /* log_2 of data packet interval */ --- linux/sound/usb/pcm.c +++ linux/sound/usb/pcm.c @@ -202,11 +202,11 @@ static int set_format(struct snd_usb_sub return -EIO; } subs->interface = -1; - subs->format = 0; + subs->altset_idx = 0; } /* set interface */ - if (subs->interface != fmt->iface || subs->format != fmt->altset_idx) { + if (subs->interface != fmt->iface || subs->altset_idx != fmt->altset_idx) { if (usb_set_interface(dev, fmt->iface, fmt->altsetting) < 0) { snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed\n", dev->devnum, fmt->iface, fmt->altsetting); @@ -214,7 +214,7 @@ static int set_format(struct snd_usb_sub } snd_printdd(KERN_INFO "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); subs->interface = fmt->iface; - subs->format = fmt->altset_idx; + subs->altset_idx = fmt->altset_idx; } /* create a data pipe */ @@ -771,7 +771,7 @@ static int snd_usb_pcm_open(struct snd_p struct snd_usb_substream *subs = &as->substream[direction]; subs->interface = -1; - subs->format = 0; + subs->altset_idx = 0; runtime->hw = snd_usb_hardware; runtime->private_data = subs; subs->pcm_substream = substream; --- linux/sound/usb/proc.c +++ linux/sound/usb/proc.c @@ -116,7 +116,7 @@ static void proc_dump_substream_status(s unsigned int i; snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); - snd_iprintf(buffer, " Altset = %d\n", subs->format); + snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); snd_iprintf(buffer, " URBs = %d [ ", subs->nurbs); for (i = 0; i < subs->nurbs; i++) snd_iprintf(buffer, "%d ", subs->dataurb[i].packets);

1 0

[alsa-devel] [PATCH] [patch 1/2]AC97: add AC97 support for AT91.
by Sedji Gaouaou 01 Mar '10

01 Mar '10

This patch add AC97 support for ATMEL AT91, using the AVR32 code. While AVR is using a DMA, the AT91 chips are using a Peripheral Data Controller. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/atmel/Kconfig | 2 +- sound/atmel/ac97c.c | 309 +++++++++++++++++++++++++++++++++++++-------------- 2 files changed, 226 insertions(+), 85 deletions(-) diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig index 6c228a9..94de43a 100644 --- a/sound/atmel/Kconfig +++ b/sound/atmel/Kconfig @@ -12,7 +12,7 @@ config SND_ATMEL_AC97C tristate "Atmel AC97 Controller (AC97C) driver" select SND_PCM select SND_AC97_CODEC - depends on DW_DMAC && AVR32 + depends on (DW_DMAC && AVR32) || ARCH_AT91 help ALSA sound driver for the Atmel AC97 controller. diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 0c0f877..e765d65 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -13,6 +13,7 @@ #include <linux/device.h> #include <linux/dmaengine.h> #include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/module.h> @@ -31,6 +32,10 @@ #include <linux/dw_dmac.h> +#include <mach/cpu.h> +#include <mach/hardware.h> +#include <mach/gpio.h> + #include "ac97c.h" enum { @@ -63,6 +68,7 @@ struct atmel_ac97c { u64 cur_format; unsigned int cur_rate; unsigned long flags; + int playback_period, capture_period; /* Serialize access to opened variable */ spinlock_t lock; void __iomem *regs; @@ -242,10 +248,12 @@ static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); - + if (cpu_is_at32ap7000()) { + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); chip->cur_rate = params_rate(hw_params); @@ -266,9 +274,14 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, if (retval < 0) return retval; /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) { + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -282,16 +295,20 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } return snd_pcm_lib_free_pages(substream); } static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } return snd_pcm_lib_free_pages(substream); } @@ -299,9 +316,11 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, OCA); int retval; + chip->playback_period = 0; word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); /* assign channels to AC97C channel A */ @@ -324,7 +343,8 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: - word |= AC97C_CMR_CEM_LITTLE; + if (cpu_is_at32ap7000()) + word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ word &= ~(AC97C_CMR_CEM_LITTLE); @@ -363,9 +383,18 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_TX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_TO_DEVICE); + if (cpu_is_at32ap7000()) { + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TNCR); + } return retval; } @@ -374,9 +403,11 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, ICA); int retval; + chip->capture_period = 0; word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); /* assign channels to AC97C channel A */ @@ -399,7 +430,8 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: - word |= AC97C_CMR_CEM_LITTLE; + if (cpu_is_at32ap7000()) + word |= AC97C_CMR_CEM_LITTLE; break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ word &= ~(AC97C_CMR_CEM_LITTLE); @@ -438,9 +470,18 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_RX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_FROM_DEVICE); + if (cpu_is_at32ap7000()) { + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RNCR); + } return retval; } @@ -458,15 +499,22 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.tx_chan); - if (retval) - goto out; + if (cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_TXTEN; + } camr |= AC97C_CMR_CENA; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) + dw_dma_cyclic_stop(chip->dma.tx_chan); + else + ptcr |= ATMEL_PDC_TXTDIS; if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -476,6 +524,8 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if (!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -484,24 +534,32 @@ static int atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, casr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); + ptcr = readl(chip->regs + ATMEL_PDC_PTSR); switch (cmd) { case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.rx_chan); - if (retval) - goto out; + if (cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_RXTEN; + } camr |= AC97C_CMR_CENA; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) + dw_dma_cyclic_stop(chip->dma.rx_chan); + else + ptcr |= (ATMEL_PDC_RXTDIS); if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -511,6 +569,8 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if (!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -523,7 +583,10 @@ atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + else + bytes = readl(chip->regs + ATMEL_PDC_TPR); bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -540,7 +603,10 @@ atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + if (cpu_is_at32ap7000()) + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + else + bytes = readl(chip->regs + ATMEL_PDC_RPR); bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -578,8 +644,11 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) u32 sr = ac97c_readl(chip, SR); u32 casr = ac97c_readl(chip, CASR); u32 cosr = ac97c_readl(chip, COSR); + u32 camr = ac97c_readl(chip, CAMR); if (sr & AC97C_SR_CAEVT) { + struct snd_pcm_runtime *runtime; + int offset, next_period, block_size; dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", @@ -587,6 +656,50 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", casr & AC97C_CSR_TXRDY ? " TXRDY" : "", !casr ? " NONE" : ""); + if (!cpu_is_at32ap7000()) { + if ((casr & camr) & AC97C_CSR_ENDTX) { + runtime = chip->playback_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->playback_period++; + + if (chip->playback_period == runtime->periods) + chip->playback_period = 0; + next_period = chip->playback_period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_TNCR); + + snd_pcm_period_elapsed( + chip->playback_substream); + } + if ((casr & camr) & AC97C_CSR_ENDRX) { + runtime = chip->capture_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->capture_period++; + + if (chip->capture_period == runtime->periods) + chip->capture_period = 0; + next_period = chip->capture_period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_RNCR); + snd_pcm_period_elapsed(chip->capture_substream); + } + } retval = IRQ_HANDLED; } @@ -617,6 +730,13 @@ static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + if (!cpu_is_at32ap7000()) { + err = snd_ac97_pcm_assign(chip->ac97_bus, + ARRAY_SIZE(at91_ac97_pcm_defs), + at91_ac97_pcm_defs); + if (err) + return err; + } retval = snd_pcm_new(chip->card, chip->card->shortname, chip->pdev->id, playback, capture, &pcm); if (retval) @@ -775,7 +895,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pclk = clk_get(&pdev->dev, "pclk"); + if (cpu_is_at32ap7000()) { + pclk = clk_get(&pdev->dev, "pclk"); + } else { + pclk = clk_get(&pdev->dev, "ac97_clk"); + } + if (IS_ERR(pclk)) { dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); @@ -844,44 +969,53 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) goto err_ac97_bus; } - if (pdata->rx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->rx_dws; - dma_cap_mask_t mask; + if (cpu_is_at32ap7000()) { + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; - dws->rx_reg = regs->start + AC97C_CARHR + 2; + dws->rx_reg = regs->start + AC97C_CARHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + chip->dma.rx_chan = dma_request_channel(mask, filter, + dws); - dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", dev_name(&chip->dma.rx_chan->dev->device)); - set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - } + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } - if (pdata->tx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->tx_dws; - dma_cap_mask_t mask; + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; - dws->tx_reg = regs->start + AC97C_CATHR + 2; + dws->tx_reg = regs->start + AC97C_CATHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + chip->dma.tx_chan = dma_request_channel(mask, filter, + dws); - dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", dev_name(&chip->dma.tx_chan->dev->device)); - set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - } + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } - if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && - !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { - dev_dbg(&pdev->dev, "DMA not available\n"); - retval = -ENODEV; - goto err_dma; - } + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + } else { + /* Just pretend that we have DMA channel(for at91 i is actually + * the PDC) */ + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } retval = atmel_ac97c_pcm_new(chip); if (retval) { @@ -897,20 +1031,22 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); - dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", - chip->regs); + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p, irq = %d\n", + chip->regs, irq); return 0; err_dma: - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } err_ac97_bus: snd_card_set_dev(card, NULL); @@ -934,10 +1070,12 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) struct snd_card *card = platform_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.tx_chan); + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + } clk_disable(chip->pclk); return 0; @@ -949,11 +1087,12 @@ static int atmel_ac97c_resume(struct platform_device *pdev) struct atmel_ac97c *chip = card->private_data; clk_enable(chip->pclk); - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.tx_chan); - + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + } return 0; } #else @@ -978,14 +1117,16 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) iounmap(chip->regs); free_irq(chip->irq, chip); - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if (cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } snd_card_set_dev(card, NULL); snd_card_free(card); -- 1.5.6.5

2 2

[alsa-devel] [PATCH 1/1] sound/pci/hda/hda_codec.c: various coding style fixes
by Wu Fengguang 01 Mar '10

01 Mar '10

Thanks. Could use shorter filename in the patch title :) Acked-by: Wu Fengguang <fengguang.wu(a)intel.com>

1 0

[alsa-devel] STAC9200 spdif output problem on Dell Precision M65
by Jonathan Thibault 01 Mar '10

01 Mar '10

Hello alsa devs :) I've tried to change the pin configuration for model=dell-m22 to match that of the windows driver as per this section of the relevant .INI file, but that hasn't really changed anything. I haven't really incorporated the windows gpio settings as I couldn't find figure out how to do so. Jonathan ---- relevant bits of windows .ini ---- [HKR\Settings\pin] CodecName = STAC9200-32 [HKR\Settings\pin\08] CfgDflt = dword: 0x40C003FA [HKR\Settings\pin\09] CfgDflt = dword: 0x01441340 [HKR\Settings\pin\10] CfgDflt = dword: 0x03A1102E AltCfg = hex: 2F,10,81,03 Bias = hex: FF,04,02 [HKR\Settings\pin\11] CfgDflt = dword: 0x401003FB [HKR\Settings\pin\12] CfgDflt = dword: 0x403003FC [HKR\Settings\pin\0d] CfgDflt = dword: 0x0321121F Action = hex: 80,00,81,0E,87,00 [HKR\Settings\pin\0e] CfgDflt = dword: 0x90170310 [HKR\Settings\pin\0F] CfgDflt = dword: 0x90A70320 ... [HKR\Settings\gpio] 02 = hex: 0x00, 0x01, 0x0E, 0x00, 0x00, 0x07, 0x00 01 = hex: 0x82 03 = hex: 0x82 ---------------------------------------- Here is the codec information. cat /proc/asound/card0/codec#0 Codec: SigmaTel STAC9200 Address: 0 Function Id: 0x1 Vendor Id: 0x83847690 Subsystem Id: 0x102801c8 Revision Id: 0x102201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=4, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x0a Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x08 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=5, channel=0 Digital: Digital category: 0x0 PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Delay: 3 samples Node 0x07 [Audio Selector] wcaps 0x300901: Stereo R/L Connection: 3 0x02* 0x08 0x0a Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x00010024: IN EAPD Detect EAPD 0x0: Pin Default 0x40c003fa: [N/A] SPDIF In at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0xa Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Delay: 3 samples Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x01441340: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x0a Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x08 0x08] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x19 0x19] Connection: 1 0x07 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 5 0x10 0x0f* 0x0e 0x0d 0x12 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x0321121f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x00: Unsolicited: tag=01, enabled=1 Connection: 1 0x0b Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x90170310: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x90a70320: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0381102f: [Jack] Line In at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=02, enabled=1 Connection: 1 0x0b Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x401003fb: [N/A] Speaker at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0xb Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 1 0x13 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x403003fc: [N/A] CD at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0xc Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x13 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] cat /proc/asound/card0/codec#1 Codec: Conexant ID 2bfa Address: 1 Function Id: 0x2 Vendor Id: 0x14f12bfa Subsystem Id: 0x14f100c3 Revision Id: 0x90000 Modem Function Group: 0x2

1 1

[PATCH 501/501] Fix a string format ambiguity
by Colin Guthrie 01 Mar '10

01 Mar '10

--- src/alisp/alisp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/alisp/alisp.c b/src/alisp/alisp.c index f3580f9..7575f55 100644 --- a/src/alisp/alisp.c +++ b/src/alisp/alisp.c @@ -1712,7 +1712,7 @@ static struct alisp_object * F_princ(struct alisp_instance *instance, struct ali delete_tree(instance, p1); p1 = eval(instance, car(p)); if (alisp_compare_type(p1, ALISP_OBJ_STRING)) - snd_output_printf(instance->out, p1->value.s); + snd_output_printf(instance->out, "%s", p1->value.s); else princ_object(instance->out, p1); n = cdr(p); -- 1.7.0 --------------090705070402050500050407--

1 0

[alsa-devel] [PATCH 48/66] sound/usb/caiaq/midi.h: Checkpatch cleanup
by Andrea Gelmini 28 Feb '10

28 Feb '10

sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar" Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- sound/usb/caiaq/midi.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/usb/caiaq/midi.h b/sound/usb/caiaq/midi.h index 9d16db0..380f984 100644 --- a/sound/usb/caiaq/midi.h +++ b/sound/usb/caiaq/midi.h @@ -3,6 +3,6 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *dev); void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len); -void snd_usb_caiaq_midi_output_done(struct urb* urb); +void snd_usb_caiaq_midi_output_done(struct urb *urb); #endif /* CAIAQ_MIDI_H */ -- 1.7.0.90.g251a4

2 1

[alsa-devel] [PATCH 38/66] sound/oss/coproc.h: Checkpatch cleanup
by Andrea Gelmini 27 Feb '10

27 Feb '10

sound/oss/coproc.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- sound/oss/coproc.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/oss/coproc.h b/sound/oss/coproc.h index 7306346..7bec21b 100644 --- a/sound/oss/coproc.h +++ b/sound/oss/coproc.h @@ -4,7 +4,7 @@ */ /* - * Coprocessor access types + * Coprocessor access types */ #define COPR_CUSTOM 0x0001 /* Custom applications */ #define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */ -- 1.7.0.90.g251a4

1 0

[alsa-devel] [PATCHv4 2/7] ASoC: TWL6030: Add twl6030 codec driver
by Olaya, Margarita 26 Feb '10

26 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6030 codec driver. The TWL6030 codec uses a propietary PDM-based digital audio interface. Audio paths supported are: - Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right - Output: Headset Left/Right, Handsfree Left/Right Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 823 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 94 +++++ 4 files changed, 923 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 52b005f..3b3d739 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_TWL6030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8350 if MFD_WM8350 @@ -155,6 +156,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6030 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index dbaecb1..e11a193 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -20,6 +20,7 @@ snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o +snd-soc-twl6030-objs := twl6030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -76,6 +77,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_TWL6030) += snd-soc-twl6030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c new file mode 100644 index 0000000..8b52aa1 --- /dev/null +++ b/sound/soc/codecs/twl6030.c @@ -0,0 +1,823 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/platform_device.h> +#include <linux/i2c/twl.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "twl6030.h" + +#define TWL6030_RATES (SNDRV_PCM_RATE_96000) +#define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6030_data { + struct snd_soc_codec codec; + int audpwron; + int codec_powered; +}; + +/* + * twl6030 register cache & default register settings + */ +static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6030_ASICID (ro) 0x01 */ + 0x00, /* TWL6030_ASICREV (ro) 0x02 */ + 0x00, /* TWL6030_INTID 0x03 */ + 0x7B, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_NCPCTRL 0x05 */ + 0x00, /* TWL6030_LDOCTL 0x06 */ + 0x00, /* TWL6030_HPPLLCTL 0x07 */ + 0x00, /* TWL6030_LPPLLCTL 0x08 */ + 0x00, /* TWL6030_LPPLLDIV 0x09 */ + 0x00, /* TWL6030_AMICBCTL 0x0A */ + 0x00, /* TWL6030_DMICBCTL 0x0B */ + 0x18, /* TWL6030_MICLCTL 0x0C */ + 0x18, /* TWL6030_MICRCTL 0x0D */ + 0x00, /* TWL6030_MICGAIN 0x0E */ + 0x1B, /* TWL6030_LINEGAIN 0x0F */ + 0x00, /* TWL6030_HSLCTL 0x10 */ + 0x00, /* TWL6030_HSRCTL 0x11 */ + 0x00, /* TWL6030_HSGAIN 0x12 */ + 0x06, /* TWL6030_EARCTL 0x13 */ + 0x00, /* TWL6030_HFLCTL 0x14 */ + 0x03, /* TWL6030_HFLGAIN 0x15 */ + 0x00, /* TWL6030_HFRCTL 0x16 */ + 0x03, /* TWL6030_HFRGAIN 0x17 */ + 0x00, /* TWL6030_VIBCTLL 0x18 */ + 0x00, /* TWL6030_VIBDATL 0x19 */ + 0x00, /* TWL6030_VIBCTLR 0x1A */ + 0x00, /* TWL6030_VIBDATR 0x1B */ + 0x00, /* TWL6030_HKCTL1 0x1C */ + 0x00, /* TWL6030_HKCTL2 0x1D */ + 0x00, /* TWL6030_GPOCTL 0x1E */ + 0x00, /* TWL6030_ALB 0x1F */ + 0x00, /* TWL6030_DLB 0x20 */ + 0x00, /* not used 0x21 */ + 0x00, /* not used 0x22 */ + 0x00, /* not used 0x23 */ + 0x00, /* not used 0x24 */ + 0x00, /* not used 0x25 */ + 0x00, /* not used 0x26 */ + 0x00, /* not used 0x27 */ + 0x00, /* TWL6030_TRIM1 0x28 */ + 0x00, /* TWL6030_TRIM2 0x29 */ + 0x00, /* TWL6030_TRIM3 0x2A */ + 0x00, /* TWL6030_HSOTRIM 0x2B */ + 0x00, /* TWL6030_HFOTRIM 0x2C */ + 0x09, /* TWL6030_ACCCTL 0x2D */ + 0x00, /* TWL6030_STATUS (ro) 0x2E */ +}; + +/* + * twl6030 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6030_vio_reg[TWL6030_VIOREGNUM] = { + TWL6030_REG_ASICID, + TWL6030_REG_ASICREV, + TWL6030_REG_INTID, + TWL6030_REG_INTMR, + TWL6030_REG_NCPCTL, + TWL6030_REG_LDOCTL, + TWL6030_REG_AMICBCTL, + TWL6030_REG_DMICBCTL, + TWL6030_REG_HKCTL1, + TWL6030_REG_HKCTL2, + TWL6030_REG_GPOCTL, + TWL6030_REG_TRIM1, + TWL6030_REG_TRIM2, + TWL6030_REG_TRIM3, + TWL6030_REG_HSOTRIM, + TWL6030_REG_HFOTRIM, + TWL6030_REG_ACCCTL, + TWL6030_REG_STATUS, +}; + +/* + * twl6030 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6030_vdd_reg[TWL6030_VDDREGNUM] = { + TWL6030_REG_HPPLLCTL, + TWL6030_REG_LPPLLCTL, + TWL6030_REG_LPPLLDIV, + TWL6030_REG_MICLCTL, + TWL6030_REG_MICRCTL, + TWL6030_REG_MICGAIN, + TWL6030_REG_LINEGAIN, + TWL6030_REG_HSLCTL, + TWL6030_REG_HSRCTL, + TWL6030_REG_HSGAIN, + TWL6030_REG_EARCTL, + TWL6030_REG_HFLCTL, + TWL6030_REG_HFLGAIN, + TWL6030_REG_HFRCTL, + TWL6030_REG_HFRGAIN, + TWL6030_REG_VIBCTLL, + TWL6030_REG_VIBDATL, + TWL6030_REG_VIBCTLR, + TWL6030_REG_VIBDATR, + TWL6030_REG_ALB, + TWL6030_REG_DLB, +}; + +/* + * read twl6030 register cache + */ +static inline unsigned int twl6030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6030 register cache + */ +static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6030 register space + */ +static int twl6030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6030_CACHEREGNUM) + return -EIO; + + twl6030_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6030_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6030_write(codec, TWL6030_REG_ACCCTL, cache[TWL6030_REG_ACCCTL]); + + for (i = 0; i < TWL6030_VIOREGNUM; i++) { + reg = twl6030_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + if ((reg == TWL6030_REG_ASICID) || + (reg == TWL6030_REG_ASICREV) || + (reg == TWL6030_REG_STATUS)) + continue; + twl6030_write(codec, reg, cache[reg]); + } +} + +static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6030_VDDREGNUM; i++) { + reg = twl6030_vdd_reg[i]; + twl6030_write(codec, reg, cache[reg]); + } +} + +/* + * MICATT volume control: + * from -6 to 0 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); + +/* + * MICGAIN volume control: + * from 6 to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); + +/* + * HSGAIN volume control: + * from -30 to 0 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); + +/* + * HFGAIN volume control: + * from -52 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); + +/* Left analog microphone selection */ +static const char *twl6030_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6030_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6030_enum[] = { + SOC_ENUM_SINGLE(TWL6030_REG_MICLCTL, 3, 3, twl6030_amicl_texts), + SOC_ENUM_SINGLE(TWL6030_REG_MICRCTL, 3, 3, twl6030_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6030_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6030_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6030_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6030_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6030_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6030_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6030_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6030_REG_HFLGAIN, TWL6030_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("AFML"), + SND_SOC_DAPM_INPUT("AFMR"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amicl_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amicr_control), + + /* Analog capture PGAs */ + SND_SOC_DAPM_PGA("MicAmpL", + TWL6030_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6030_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6030_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6030_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6030_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6030_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6030_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6030_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6030_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6030_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0), + SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0), + + /* Analog playback switches */ + SND_SOC_DAPM_SWITCH("HSDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hsdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HSDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hsdacr_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Left Playback", + SND_SOC_NOPM, 0, 0, &hfdacl_switch_controls), + SND_SOC_DAPM_SWITCH("HFDAC Right Playback", + SND_SOC_NOPM, 0, 0, &hfdacr_switch_controls), + + SND_SOC_DAPM_SWITCH("Headset Left Driver", + SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Headset Right Driver", + SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), + SND_SOC_DAPM_SWITCH("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6030_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6030_REG_HFRCTL, 1, 0, NULL, 0), + +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Capture path */ + {"Analog Left Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Left Capture Route", "Main Mic", "MAINMIC"}, + {"Analog Left Capture Route", "Aux/FM Left", "AFML"}, + + {"Analog Right Capture Route", "Headset Mic", "HSMIC"}, + {"Analog Right Capture Route", "Sub Mic", "SUBMIC"}, + {"Analog Right Capture Route", "Aux/FM Right", "AFMR"}, + + {"MicAmpL", NULL, "Analog Left Capture Route"}, + {"MicAmpR", NULL, "Analog Right Capture Route"}, + + {"ADC Left", NULL, "MicAmpL"}, + {"ADC Right", NULL, "MicAmpR"}, + + /* Headset playback path */ + {"HSDAC Left Playback", "Switch", "HSDAC Left"}, + {"HSDAC Right Playback", "Switch", "HSDAC Right"}, + + {"Headset Left Driver", "Switch", "HSDAC Left Playback"}, + {"Headset Right Driver", "Switch", "HSDAC Right Playback"}, + + {"HSOL", NULL, "Headset Left Driver"}, + {"HSOR", NULL, "Headset Right Driver"}, + + /* Handsfree playback path */ + {"HFDAC Left Playback", "Switch", "HFDAC Left"}, + {"HFDAC Right Playback", "Switch", "HFDAC Right"}, + + {"HFDAC Left PGA", NULL, "HFDAC Left Playback"}, + {"HFDAC Right PGA", NULL, "HFDAC Right Playback"}, + + {"Handsfree Left Driver", "Switch", "HFDAC Left PGA"}, + {"Handsfree Right Driver", "Switch", "HFDAC Right PGA"}, + + {"HFL", NULL, "Handsfree Left Driver"}, + {"HFR", NULL, "Handsfree Right Driver"}, +}; + +static int twl6030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6030_dapm_widgets, + ARRAY_SIZE(twl6030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6030_data *priv = codec->private_data; + int audpwron = priv->audpwron; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* power-up sequence latency */ + mdelay(16); + } + + /* sync registers updated during power-up sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + + /* initialize vdd/vss registers with reg_cache */ + twl6030_init_vdd_regs(codec); + + priv->codec_powered = 1; + break; + case SND_SOC_BIAS_OFF: + if (!priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 0); + + /* power-down sequence latency */ + udelay(500); + } + + /* sync registers updated during power-down sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + return 0; +} + +static int twl6030_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; + u8 hppllctl, lppllctl; + + hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); + hppllctl &= ~TWL6030_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_12000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6030_MCLK_19200KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_26000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6030_MCLK_38400KHZ | + TWL6030_HPLLSQRENA | + TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown sysclk freq %d\n", freq); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + + /* Disable LPPLL and select HPPLL */ + lppllctl = TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static struct snd_soc_dai_ops twl6030_dai_ops = { + .set_sysclk = twl6030_set_dai_sysclk, +}; + +struct snd_soc_dai twl6030_dai = { + .name = "twl6030", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6030_RATES, + .formats = TWL6030_FORMATS, + }, + .ops = &twl6030_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6030_dai); + +static int twl6030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6030_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6030_codec; + +static int twl6030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6030_codec); + + codec = twl6030_codec; + socdev->card->codec = codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + snd_soc_add_controls(codec, twl6030_snd_controls, + ARRAY_SIZE(twl6030_snd_controls)); + twl6030_add_widgets(codec); + + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int twl6030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl6030 = { + .probe = twl6030_probe, + .remove = twl6030_remove, + .suspend = twl6030_suspend, + .resume = twl6030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6030); + +static int __devinit twl6030_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6030_data *priv; + int audpwron; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) + audpwron = twl_codec->audpwron_gpio; + else + audpwron = -EINVAL; + + priv->audpwron = audpwron; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6030_dai.dev = &pdev->dev; + + codec->name = "twl6030"; + codec->owner = THIS_MODULE; + codec->read = twl6030_read_reg_cache; + codec->write = twl6030_write; + codec->set_bias_level = twl6030_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6030_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6030_reg); + codec->reg_cache = kmemdup(twl6030_reg, sizeof(twl6030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + if (gpio_is_valid(audpwron)) { + ret = gpio_request(audpwron, "audpwron"); + if (ret) + goto gpio1_err; + + ret = gpio_direction_output(audpwron, 0); + if (ret) + goto gpio2_err; + + priv->codec_powered = 0; + } else { + /* if no gpio is provided, then assume its always on */ + priv->codec_powered = 1; + } + + /* init vio registers */ + twl6030_init_vio_regs(codec); + + /* power on device */ + ret = twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto gpio2_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6030_codec = codec; + + ret = snd_soc_register_dai(&twl6030_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6030_codec = NULL; +reg_err: + twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); +gpio2_err: + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6030_codec_remove(struct platform_device *pdev) +{ + struct twl6030_data *priv = twl6030_codec->private_data; + int audpwron = priv->audpwron; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + snd_soc_unregister_dai(&twl6030_dai); + snd_soc_unregister_codec(twl6030_codec); + + kfree(twl6030_codec); + twl6030_codec = NULL; + + return 0; +} + +#ifdef CONFIG_PM +static int twl6030_codec_suspend(struct platform_device *pdev, + pm_message_t msg) +{ + return snd_soc_suspend_device(&pdev->dev); +} + +static int twl6030_codec_resume(struct platform_device *pdev) +{ + return snd_soc_resume_device(&pdev->dev); +} +#else +#define twl6030_codec_suspend NULL +#define twl6030_codec_resume NULL +#endif + +static struct platform_driver twl6030_codec_driver = { + .driver = { + .name = "twl6030_codec", + .owner = THIS_MODULE, + }, + .probe = twl6030_codec_probe, + .remove = __devexit_p(twl6030_codec_remove), + .suspend = twl6030_codec_suspend, + .resume = twl6030_codec_resume, +}; + +static int __init twl6030_codec_init(void) +{ + return platform_driver_register(&twl6030_codec_driver); +} +module_init(twl6030_codec_init); + +static void __exit twl6030_codec_exit(void) +{ + platform_driver_unregister(&twl6030_codec_driver); +} +module_exit(twl6030_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6030 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h new file mode 100644 index 0000000..7fba60e --- /dev/null +++ b/sound/soc/codecs/twl6030.h @@ -0,0 +1,94 @@ +/* + * ALSA SoC TWL6030 codec driver + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL6030_H__ +#define __TWL6030_H__ + +#define TWL6030_REG_ASICID 0x01 +#define TWL6030_REG_ASICREV 0x02 +#define TWL6030_REG_INTID 0x03 +#define TWL6030_REG_INTMR 0x04 +#define TWL6030_REG_NCPCTL 0x05 +#define TWL6030_REG_LDOCTL 0x06 +#define TWL6030_REG_HPPLLCTL 0x07 +#define TWL6030_REG_LPPLLCTL 0x08 +#define TWL6030_REG_LPPLLDIV 0x09 +#define TWL6030_REG_AMICBCTL 0x0A +#define TWL6030_REG_DMICBCTL 0x0B +#define TWL6030_REG_MICLCTL 0x0C +#define TWL6030_REG_MICRCTL 0x0D +#define TWL6030_REG_MICGAIN 0x0E +#define TWL6030_REG_LINEGAIN 0x0F +#define TWL6030_REG_HSLCTL 0x10 +#define TWL6030_REG_HSRCTL 0x11 +#define TWL6030_REG_HSGAIN 0x12 +#define TWL6030_REG_EARCTL 0x13 +#define TWL6030_REG_HFLCTL 0x14 +#define TWL6030_REG_HFLGAIN 0x15 +#define TWL6030_REG_HFRCTL 0x16 +#define TWL6030_REG_HFRGAIN 0x17 +#define TWL6030_REG_VIBCTLL 0x18 +#define TWL6030_REG_VIBDATL 0x19 +#define TWL6030_REG_VIBCTLR 0x1A +#define TWL6030_REG_VIBDATR 0x1B +#define TWL6030_REG_HKCTL1 0x1C +#define TWL6030_REG_HKCTL2 0x1D +#define TWL6030_REG_GPOCTL 0x1E +#define TWL6030_REG_ALB 0x1F +#define TWL6030_REG_DLB 0x20 +#define TWL6030_REG_TRIM1 0x28 +#define TWL6030_REG_TRIM2 0x29 +#define TWL6030_REG_TRIM3 0x2A +#define TWL6030_REG_HSOTRIM 0x2B +#define TWL6030_REG_HFOTRIM 0x2C +#define TWL6030_REG_ACCCTL 0x2D +#define TWL6030_REG_STATUS 0x2E + +#define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) + +#define TWL6030_VIOREGNUM 18 +#define TWL6030_VDDREGNUM 21 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6030_HPLLENA 0x01 +#define TWL6030_HPLLRST 0x02 +#define TWL6030_HPLLBP 0x04 +#define TWL6030_HPLLSQRENA 0x08 +#define TWL6030_HPLLSQRBP 0x10 +#define TWL6030_MCLK_12000KHZ (0 << 5) +#define TWL6030_MCLK_19200KHZ (1 << 5) +#define TWL6030_MCLK_26000KHZ (2 << 5) +#define TWL6030_MCLK_38400KHZ (3 << 5) +#define TWL6030_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6030_LPLLENA 0x01 +#define TWL6030_LPLLRST 0x02 +#define TWL6030_LPLLSEL 0x04 +#define TWL6030_LPLLFIN 0x08 +#define TWL6030_HPLLSEL 0x10 + +extern struct snd_soc_dai twl6030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6030; + +#endif /* End of __TWL6030_H__ */ -- 1.6.1.3

3 10

[alsa-devel] Listing virtual devices for a sound card (aplay -L)
by Nitin PAI 26 Feb '10

26 Feb '10

Hi, I have added proprietary sound cards which are statically compiled in the kernel. Card 0: Name=X, Card 1: Name=Y. I want to create virtual devices using these cards, like front, left,surround,iec958. I see that this is usually done at /usr/share/alsa/cards/.... Please let me know how this mapping works? I mean how does aplay -L list these devices by mapping the sound card to the ones in /usr/share/alsa/cards/... I tried creating X.conf and Y.conf there with front,left,surroud,iec958, but aplay -L doesnt seem to recognize them. Also, I want iec958 card to be one of the cards listed via aplay -L, and defining this in asound.conf gives error with aplay. Any help will be appreciated. Thanks, Nitin

1 1

[alsa-devel] [RFC] ASoC: Allow mulitple usage of codec and cpu dai
by Jassi Brar 26 Feb '10

26 Feb '10

We can have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two or more dai_links by avoiding shutting-down/powering-up if the dai is already in an appropriate state. For that, we change semantics of the snd_soc_dai.active flag from indicator to reference counter. This patch only attempts to block duplicate startup/shutdown and the mute callbacks, presuming currently no dai driver anticipates 'dai reuse'. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 47 ++++++++++++++++++++++++++++------------------- 2 files changed, 29 insertions(+), 20 deletions(-) The codec->active can also be deducted by codec_dai->capture/playback.active So, we can get rid of snd_soc_codec.active field as well to reduce number of variables we have to manage in future while making delicate changes in ASoC. diff --git a/include/sound/soc.h b/include/sound/soc.h index 27a2ad9..f792c18 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -375,7 +375,7 @@ struct snd_soc_pcm_stream { unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ - unsigned int active:1; /* stream is in use */ + unsigned int active; /* num of active users of the stream */ }; /* SoC audio ops */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2b66719..459f1f5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -352,7 +352,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops->startup) { + if (cpu_dai->active == 0 && cpu_dai->ops->startup) { ret = cpu_dai->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", @@ -369,7 +369,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->ops->startup) { + if (codec_dai->active == 0 && codec_dai->ops->startup) { ret = codec_dai->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", @@ -454,11 +454,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, runtime->hw.rate_max); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->playback.active = codec_dai->playback.active = 1; - else - cpu_dai->capture.active = codec_dai->capture.active = 1; - cpu_dai->active = codec_dai->active = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback.active++; + codec_dai->playback.active++; + } else { + cpu_dai->capture.active++; + codec_dai->capture.active++; + } + cpu_dai->active++; + codec_dai->active++; card->codec->active++; mutex_unlock(&pcm_mutex); return 0; @@ -534,27 +538,28 @@ static int soc_codec_close(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->playback.active = codec_dai->playback.active = 0; - else - cpu_dai->capture.active = codec_dai->capture.active = 0; - - if (codec_dai->playback.active == 0 && - codec_dai->capture.active == 0) { - cpu_dai->active = codec_dai->active = 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback.active--; + codec_dai->playback.active--; + } else { + cpu_dai->capture.active--; + codec_dai->capture.active--; } + + cpu_dai->active--; + codec_dai->active--; codec->active--; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (codec_dai->active == 0 && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops->shutdown) + if (cpu_dai->active == 0 && cpu_dai->ops->shutdown) cpu_dai->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops->shutdown) + if (codec_dai->active == 0 && codec_dai->ops->shutdown) codec_dai->ops->shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) @@ -563,6 +568,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (platform->pcm_ops->close) platform->pcm_ops->close(substream); + /* Skip DAPM if codec is still active via some other dai_link */ + if (codec_dai->active) + goto exit; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; @@ -574,7 +583,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) codec_dai->capture.stream_name, SND_SOC_DAPM_STREAM_STOP); } - +exit: mutex_unlock(&pcm_mutex); return 0; } -- 1.6.2.5

3 5

[alsa-devel] [PATCH] ASoC: Allow mulitple usage count of codec and cpu dai
by Jassi Brar 26 Feb '10

26 Feb '10

If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two or more dai_links we need to log the number of active users of the dai. For that, we change semantics of the snd_soc_dai.active flag from indicator to reference counter. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 31 ++++++++++++++++++------------- 2 files changed, 19 insertions(+), 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 27a2ad9..f792c18 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -375,7 +375,7 @@ struct snd_soc_pcm_stream { unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ - unsigned int active:1; /* stream is in use */ + unsigned int active; /* num of active users of the stream */ }; /* SoC audio ops */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2b66719..0211369 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -454,11 +454,15 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) pr_debug("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, runtime->hw.rate_max); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->playback.active = codec_dai->playback.active = 1; - else - cpu_dai->capture.active = codec_dai->capture.active = 1; - cpu_dai->active = codec_dai->active = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback.active++; + codec_dai->playback.active++; + } else { + cpu_dai->capture.active++; + codec_dai->capture.active++; + } + cpu_dai->active++; + codec_dai->active++; card->codec->active++; mutex_unlock(&pcm_mutex); return 0; @@ -534,15 +538,16 @@ static int soc_codec_close(struct snd_pcm_substream *substream) mutex_lock(&pcm_mutex); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->playback.active = codec_dai->playback.active = 0; - else - cpu_dai->capture.active = codec_dai->capture.active = 0; - - if (codec_dai->playback.active == 0 && - codec_dai->capture.active == 0) { - cpu_dai->active = codec_dai->active = 0; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback.active--; + codec_dai->playback.active--; + } else { + cpu_dai->capture.active--; + codec_dai->capture.active--; } + + cpu_dai->active--; + codec_dai->active--; codec->active--; /* Muting the DAC suppresses artifacts caused during digital -- 1.6.2.5

3 2

[alsa-devel] Procedure to add files in /usr/share/alsa/cards/
by Nitin PAI 26 Feb '10

26 Feb '10

Hi, What is the procedure to add a new conf file in /usr/share/alsa/cards ? What should be the name of the file and any other changes i need to make so that aplay -L displayes all the devices. Thanks, Nitin

1 0

[alsa-devel] [PATCH] [patch 2/2] AC97: add full duplex support for atmel AT91 and AVR.
by Sedji Gaouaou 26 Feb '10

26 Feb '10

This patch add full duplex support on AT91 and AVR. It was a bug: we needed to check first if there are some chips opened so we could enable both reception and sending of the data. Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/atmel/ac97c.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index d48a59a..6f5e22c 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -339,7 +339,11 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) ac97c_writel(chip, OCA, word); /* configure sample format and size */ - word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + word = ac97c_readl(chip, CAMR); + if (chip->opened <= 1) + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + else + word |= AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: @@ -426,7 +430,11 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) ac97c_writel(chip, ICA, word); /* configure sample format and size */ - word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + word = ac97c_readl(chip, CAMR); + if (chip->opened <= 1) + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + else + word |= AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: @@ -506,7 +514,7 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) } else { ptcr = ATMEL_PDC_TXTEN; } - camr |= AC97C_CMR_CENA; + camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ @@ -551,7 +559,7 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) } else { ptcr = ATMEL_PDC_RXTEN; } - camr |= AC97C_CMR_CENA; + camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ -- 1.5.6.5

3 2

[alsa-devel] Why use spin_lock or spin_lock_irq in trigger callback (audio capture)?
by Yiliang Bao 26 Feb '10

26 Feb '10

Hi, I looked at some PCI audio capture driver like BT878 (sound/pci/bt87x.c) or SAA7134 (drivers/media/video/saa7134-alsa.c), I found that spin_lock instead of spin_lock_irq (or spin_lock_irqsave) is used in trigger function. Is there any reason for that? Thanks, Yiliang

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.31-rc8-407-gfc9e4a3
by Takashi Iwai 26 Feb '10

26 Feb '10

At Tue, 6 Oct 2009 08:36:04 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > commit fc9e4a3811567f8d1d4c263a89e27b52c90d6098 > Author: Clemens Ladisch <clemens(a)ladisch.de> > AuthorDate: Tue Oct 6 08:21:04 2009 +0200 > Commit: Clemens Ladisch <clemens(a)ladisch.de> > CommitDate: Tue Oct 6 08:21:04 2009 +0200 > > sound: via82xx: move DXS volume controls to PCM interface > > The "VIA DXS" controls are actually volume controls that apply to the > four PCM substreams, so we better indicate this connection by moving the > controls to the PCM interface. > > Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the > restoring of these volumes by "amixer restore" that most distributions > use; the renaming in this patch cures that regression by preventing > amixer from applying the old, wrong volume levels to the new controls. > http://bugzilla.kernel.org/show_bug.cgi?id=14151 > http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613 > > Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> > Cc: <stable(a)kernel.org> Well, this control was once removed, and then revived due to complains by users who have used it in mixer. Moving this to PCM iface does almost the same thing, disabling an easy way to access it. I'm afraid that this patch might be regarded as another "regression". For such an old driver, the compatibility is the most important item. If any, we must provide a way to back to the old behavior. thanks, Takashi > > ----------------------------------------------------------------------- > > Diffstat: > sound/pci/via82xx.c | 27 ++++++++++++++++++--------- > 1 files changed, 18 insertions(+), 9 deletions(-) > > > Patch -------------- (max 1500 lines) > diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c > index acfa476..91683a3 100644 > --- a/sound/pci/via82xx.c > +++ b/sound/pci/via82xx.c > @@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > struct via82xx *chip = snd_kcontrol_chip(kcontrol); > - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); > + unsigned int idx = kcontrol->id.subdevice; > > ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0]; > ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1]; > @@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol, > struct snd_ctl_elem_value *ucontrol) > { > struct via82xx *chip = snd_kcontrol_chip(kcontrol); > - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); > + unsigned int idx = kcontrol->id.subdevice; > unsigned long port = chip->port + 0x10 * idx; > unsigned char val; > int i, change = 0; > @@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = > }; > > static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { > - .name = "VIA DXS Playback Volume", > - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .iface = SNDRV_CTL_ELEM_IFACE_PCM, > + .device = 0, > + /* .subdevice set later */ > + .name = "PCM Playback Volume", > .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | > SNDRV_CTL_ELEM_ACCESS_TLV_READ), > - .count = 4, > .info = snd_via8233_dxs_volume_info, > .get = snd_via8233_dxs_volume_get, > .put = snd_via8233_dxs_volume_put, > @@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) > } > else /* Using DXS when PCM emulation is enabled is really weird */ > { > - /* Standalone DXS controls */ > - err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip)); > - if (err < 0) > - return err; > + for (i = 0; i < 4; ++i) { > + struct snd_kcontrol *kctl; > + > + kctl = snd_ctl_new1( > + &snd_via8233_dxs_volume_control, chip); > + if (!kctl) > + return -ENOMEM; > + kctl->id.subdevice = i; > + err = snd_ctl_add(chip->card, kctl); > + if (err < 0) > + return err; > + } > } > } > /* select spdif data slot 10/11 */ > ------- End of patch > > hooks/update > --- > Git Source Code Management System > hooks/update refs/heads/master \ > 62f08ef9739c5cb5fff5b5bdec3bc02d522b9a12 \ > fc9e4a3811567f8d1d4c263a89e27b52c90d6098 > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry® Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9-12, 2009. Register now! > http://p.sf.net/sfu/devconf > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog >

6 22

[alsa-devel] issues porting codec to s3c2443
by Austin, Brian 26 Feb '10

26 Feb '10

Hello, I have forward ported the CS42L52 from 2.6.26.2 to the current asoc git tree. The codec and smdk2443-cs42l52 drivers load correctly, at least I think so, and would like to submit the CS42L52 driver but am having some issues with s3c2412-i2s. From my previous kernel version to the latest there has been significant changes to the S3C architecture and I pretty sure that I am missing something because of that. The question is, is it ok to submit the codec standalone and have someone look at it to make sure I made the correct changes. No compiler warnings occur. My port was based off of the WM8750 if that helps any Here is the output from my SMDK2443 Simtec Advanced Linux Sound Architecture Driver Version 1.0.18a. No device for DAI CS42L52 No device for DAI s3c2412-i2s SMDK2443 cs42l52 ALSA SoC driver ver1.1 10/2008 soc-audio soc-audio: RXEN: Invalid MODE in IISMOD CS42L52: Cirrus Logic ALSA SoC codec cs42l52 driver verison 1.1 10/2008 ALSA device list: No soundcards found. It seems that the S3C2412 I2S is not setup correctly for my platform. Thanks, Brian Austin

4 6

[alsa-devel] aplay: possible circular locking dependency detected
by Daniel Mack 25 Feb '10

25 Feb '10

I got this once while testing an USB audio device. Appeareantly not reproducable. Daniel Feb 21 14:26:29 jup kernel: [69661.385427] Feb 21 14:26:29 jup kernel: [69661.385430] ======================================================= Feb 21 14:26:29 jup kernel: [69661.385435] [ INFO: possible circular locking dependency detected ] Feb 21 14:26:29 jup kernel: [69661.385439] 2.6.33-rc8 #21 Feb 21 14:26:29 jup kernel: [69661.385441] ------------------------------------------------------- Feb 21 14:26:29 jup kernel: [69661.385444] aplay/11761 is trying to acquire lock: Feb 21 14:26:29 jup kernel: [69661.385446] (sysfs_mutex){+.+.+.}, at: [<c10f0668>] sysfs_get_dirent+0x15/0x4c Feb 21 14:26:29 jup kernel: [69661.385457] Feb 21 14:26:29 jup kernel: [69661.385457] but task is already holding lock: Feb 21 14:26:29 jup kernel: [69661.385460] (&pcm->open_mutex){+.+.+.}, at: [<fb68bc8e>] snd_pcm_release+0x55/0x9e [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385474] Feb 21 14:26:29 jup kernel: [69661.385474] which lock already depends on the new lock. Feb 21 14:26:29 jup kernel: [69661.385476] Feb 21 14:26:29 jup kernel: [69661.385478] Feb 21 14:26:29 jup kernel: [69661.385479] the existing dependency chain (in reverse order) is: Feb 21 14:26:29 jup kernel: [69661.385481] Feb 21 14:26:29 jup kernel: [69661.385482] -> #2 (&pcm->open_mutex){+.+.+.}: Feb 21 14:26:29 jup kernel: [69661.385487] [<c1059070>] __lock_acquire+0x10ff/0x1422 Feb 21 14:26:29 jup kernel: [69661.385493] [<c105943b>] lock_acquire+0xa8/0xbf Feb 21 14:26:29 jup kernel: [69661.385498] [<c13720d3>] mutex_lock_nested+0x47/0x2b2 Feb 21 14:26:29 jup kernel: [69661.385503] [<fb68bc8e>] snd_pcm_release+0x55/0x9e [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385513] [<c10aabe9>] __fput+0xeb/0x196 Feb 21 14:26:29 jup kernel: [69661.385517] [<c10aacad>] fput+0x19/0x1b Feb 21 14:26:29 jup kernel: [69661.385522] [<c109ae3b>] remove_vma+0x39/0x58 Feb 21 14:26:29 jup kernel: [69661.385528] [<c109bab3>] do_munmap+0x23c/0x259 Feb 21 14:26:29 jup kernel: [69661.385532] [<c109bb00>] sys_munmap+0x30/0x3f Feb 21 14:26:29 jup kernel: [69661.385536] [<c10028d0>] sysenter_do_call+0x12/0x36 Feb 21 14:26:29 jup kernel: [69661.385541] Feb 21 14:26:29 jup kernel: [69661.385542] -> #1 (&mm->mmap_sem){++++++}: Feb 21 14:26:29 jup kernel: [69661.385547] [<c1059070>] __lock_acquire+0x10ff/0x1422 Feb 21 14:26:29 jup kernel: [69661.385551] [<c105943b>] lock_acquire+0xa8/0xbf Feb 21 14:26:29 jup kernel: [69661.385556] [<c10968db>] might_fault+0x6b/0x88 Feb 21 14:26:29 jup kernel: [69661.385561] [<c11b3cf0>] copy_to_user+0x2c/0xfc Feb 21 14:26:29 jup kernel: [69661.385565] [<c10b5e73>] filldir64+0x97/0xcd Feb 21 14:26:29 jup kernel: [69661.385570] [<c10f0534>] sysfs_readdir+0x117/0x14b Feb 21 14:26:29 jup kernel: [69661.385574] [<c10b6085>] vfs_readdir+0x68/0x94 Feb 21 14:26:29 jup kernel: [69661.385578] [<c10b6114>] sys_getdents64+0x63/0xa0 Feb 21 14:26:29 jup kernel: [69661.385582] [<c10028d0>] sysenter_do_call+0x12/0x36 Feb 21 14:26:29 jup kernel: [69661.385587] Feb 21 14:26:29 jup kernel: [69661.385587] -> #0 (sysfs_mutex){+.+.+.}: Feb 21 14:26:29 jup kernel: [69661.385592] [<c1058dc6>] __lock_acquire+0xe55/0x1422 Feb 21 14:26:29 jup kernel: [69661.385597] [<c105943b>] lock_acquire+0xa8/0xbf Feb 21 14:26:29 jup kernel: [69661.385601] [<c13720d3>] mutex_lock_nested+0x47/0x2b2 Feb 21 14:26:29 jup kernel: [69661.385605] [<c10f0668>] sysfs_get_dirent+0x15/0x4c Feb 21 14:26:29 jup kernel: [69661.385609] [<c10f1b4a>] sysfs_remove_group+0x1e/0xa3 Feb 21 14:26:29 jup kernel: [69661.385614] [<c12359b8>] dpm_sysfs_remove+0x10/0x12 Feb 21 14:26:29 jup kernel: [69661.385619] [<c123114c>] device_del+0x33/0x154 Feb 21 14:26:29 jup kernel: [69661.385625] [<c1231278>] device_unregister+0xb/0x15 Feb 21 14:26:29 jup kernel: [69661.385629] [<c129a209>] usb_remove_ep_devs+0x15/0x1f Feb 21 14:26:29 jup kernel: [69661.385635] [<c12943ff>] remove_intf_ep_devs+0x21/0x32 Feb 21 14:26:29 jup kernel: [69661.385640] [<c12954ea>] usb_set_interface+0x18c/0x22c Feb 21 14:26:29 jup kernel: [69661.385644] [<fc703cd7>] snd_usb_playback_close+0x20/0x33 [snd_usb_audio] Feb 21 14:26:29 jup kernel: [69661.385654] [<fb68bc10>] snd_pcm_release_substream+0x3d/0x66 [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385664] [<fb68bc95>] snd_pcm_release+0x5c/0x9e [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385673] [<c10aabe9>] __fput+0xeb/0x196 Feb 21 14:26:29 jup kernel: [69661.385677] [<c10aacad>] fput+0x19/0x1b Feb 21 14:26:29 jup kernel: [69661.385681] [<c109ae3b>] remove_vma+0x39/0x58 Feb 21 14:26:29 jup kernel: [69661.385685] [<c109bab3>] do_munmap+0x23c/0x259 Feb 21 14:26:29 jup kernel: [69661.385689] [<c109bb00>] sys_munmap+0x30/0x3f Feb 21 14:26:29 jup kernel: [69661.385693] [<c10028d0>] sysenter_do_call+0x12/0x36 Feb 21 14:26:29 jup kernel: [69661.385698] Feb 21 14:26:29 jup kernel: [69661.385698] other info that might help us debug this: Feb 21 14:26:29 jup kernel: [69661.385700] Feb 21 14:26:29 jup kernel: [69661.385702] 2 locks held by aplay/11761: Feb 21 14:26:29 jup kernel: [69661.385705] #0: (&mm->mmap_sem){++++++}, at: [<c109baf3>] sys_munmap+0x23/0x3f Feb 21 14:26:29 jup kernel: [69661.385711] #1: (&pcm->open_mutex){+.+.+.}, at: [<fb68bc8e>] snd_pcm_release+0x55/0x9e [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385723] Feb 21 14:26:29 jup kernel: [69661.385724] stack backtrace: Feb 21 14:26:29 jup kernel: [69661.385727] Pid: 11761, comm: aplay Not tainted 2.6.33-rc8 #21 Feb 21 14:26:29 jup kernel: [69661.385730] Call Trace: Feb 21 14:26:29 jup kernel: [69661.385735] [<c137062c>] ? printk+0xf/0x11 Feb 21 14:26:29 jup kernel: [69661.385740] [<c1057b26>] print_circular_bug+0x90/0x9c Feb 21 14:26:29 jup kernel: [69661.385744] [<c1058dc6>] __lock_acquire+0xe55/0x1422 Feb 21 14:26:29 jup kernel: [69661.385748] [<c105917f>] ? __lock_acquire+0x120e/0x1422 Feb 21 14:26:29 jup kernel: [69661.385753] [<c103d4d3>] ? try_to_del_timer_sync+0x9c/0xa4 Feb 21 14:26:29 jup kernel: [69661.385758] [<c105943b>] lock_acquire+0xa8/0xbf Feb 21 14:26:29 jup kernel: [69661.385762] [<c10f0668>] ? sysfs_get_dirent+0x15/0x4c Feb 21 14:26:29 jup kernel: [69661.385766] [<c10f0668>] ? sysfs_get_dirent+0x15/0x4c Feb 21 14:26:29 jup kernel: [69661.385769] [<c13720d3>] mutex_lock_nested+0x47/0x2b2 Feb 21 14:26:29 jup kernel: [69661.385773] [<c10f0668>] ? sysfs_get_dirent+0x15/0x4c Feb 21 14:26:29 jup kernel: [69661.385777] [<c137187f>] ? __mutex_unlock_slowpath+0xe9/0x105 Feb 21 14:26:29 jup kernel: [69661.385781] [<c10572e5>] ? trace_hardirqs_on_caller+0xff/0x120 Feb 21 14:26:29 jup kernel: [69661.385785] [<c10f0668>] sysfs_get_dirent+0x15/0x4c Feb 21 14:26:29 jup kernel: [69661.385789] [<c10f1b4a>] sysfs_remove_group+0x1e/0xa3 Feb 21 14:26:29 jup kernel: [69661.385793] [<c12359b8>] dpm_sysfs_remove+0x10/0x12 Feb 21 14:26:29 jup kernel: [69661.385797] [<c123114c>] device_del+0x33/0x154 Feb 21 14:26:29 jup kernel: [69661.385801] [<c1231278>] device_unregister+0xb/0x15 Feb 21 14:26:29 jup kernel: [69661.385805] [<c129a209>] usb_remove_ep_devs+0x15/0x1f Feb 21 14:26:29 jup kernel: [69661.385809] [<c12943ff>] remove_intf_ep_devs+0x21/0x32 Feb 21 14:26:29 jup kernel: [69661.385813] [<c12954ea>] usb_set_interface+0x18c/0x22c Feb 21 14:26:29 jup kernel: [69661.385822] [<fc703cd7>] snd_usb_playback_close+0x20/0x33 [snd_usb_audio] Feb 21 14:26:29 jup kernel: [69661.385830] [<fb68bc10>] snd_pcm_release_substream+0x3d/0x66 [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385839] [<fb68bc95>] snd_pcm_release+0x5c/0x9e [snd_pcm] Feb 21 14:26:29 jup kernel: [69661.385843] [<c10aabe9>] __fput+0xeb/0x196 Feb 21 14:26:29 jup kernel: [69661.385847] [<c10aacad>] fput+0x19/0x1b Feb 21 14:26:29 jup kernel: [69661.385851] [<c109ae3b>] remove_vma+0x39/0x58 Feb 21 14:26:29 jup kernel: [69661.385854] [<c109bab3>] do_munmap+0x23c/0x259 Feb 21 14:26:29 jup kernel: [69661.385858] [<c109bb00>] sys_munmap+0x30/0x3f Feb 21 14:26:29 jup kernel: [69661.385862] [<c10028d0>] sysenter_do_call+0x12/0x36

2 2

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 25 Feb '10

25 Feb '10

The following changes since commit d62abe563fa4718e7f85f3e871655434db92366d: Misael Lopez Cruz (1): OMAP4: PMIC: Add support for twl6030 codec are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-2.6.34 These wind up being a bug fix, without fixing the reporting of the periods a lot of hardware is going to get lots of clock domain issues. Mark Brown (2): ASoC: Remove a unused variables from i.MX FIQ runtime data ASoC: Check progress when reporting periods from i.MX FIQ handler sound/soc/imx/imx-pcm-fiq.c | 40 ++++++++++++++++++++++++++++++---------- 1 files changed, 30 insertions(+), 10 deletions(-)

2 3

[alsa-devel] [PATCHv4 0/7] TWL6030 audio codec initial support
by Olaya, Margarita 25 Feb '10

25 Feb '10

Following patch series add initial support for TWL6030 codec driver. The patch set is based on the series sent from Misael Lopez in October 2009. Changes from v3: - Remove snd_soc_init_card function - Adapt to latest twl core changes based on Balaji's comment Regards, Margarita --- Misael Lopez Cruz (7): OMAP4: PMIC: Add support for twl6030 codec ASoC: TWL6030: Add twl6030 codec driver ASoC: TWL6030: Manual power-up/down sequences ASoC: TWL6030: Add support for low-power PLL ASoC: TWL6030: Add restrictions for low-power playback mode ASoC: TWL6030: Enable audio interrupt ASoC: TWL6030: Detect power-up sequence completion drivers/mfd/twl-core.c | 18 +- include/linux/i2c/twl.h | 4 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl6030.c | 1241 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6030.h | 141 +++++ 6 files changed, 1407 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/twl6030.c create mode 100644 sound/soc/codecs/twl6030.h

2 4

[alsa-devel] No sound on Dell Optiplex 380
by Dale Blount 25 Feb '10

25 Feb '10

Hello, I've been having issues with this new PC and sound since I got it last week. I'm not getting any sound out of the speakers. I've tried both rear speaker and front headphone jacks. It does "pop" when the snd-hda-intel module is loaded, though. I've tried all the model=X options listed for my Realtek chipset and made sure that everything is unmuted and non-zero volume. My alsa-info.sh report can be found here: http://www.alsa-project.org/db/?f=fa4aacb98b85d5a02e05df5e9bffa400409b0c21 Please let me know if I can help testing a possible fix or if any other information is needed. Thanks, Dale

1 0

[alsa-devel] [PATCH 1/2] ASoC: Remove a unused variables from i.MX FIQ runtime data
by Mark Brown 25 Feb '10

25 Feb '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/imx/imx-pcm-fiq.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 5532579..a1c4ce6 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -35,12 +35,8 @@ struct imx_pcm_runtime_data { int period; int periods; - unsigned long dma_addr; - int dma; unsigned long offset; unsigned long size; - unsigned long period_cnt; - void *buf; struct timer_list timer; int period_time; }; -- 1.7.0

3 3

[alsa-devel] [PATCH] ASoC: soc_pcm_open: Add missing bailout tag
by jassisinghbrar＠gmail.com 25 Feb '10

25 Feb '10

From: Jassi Brar <jassi.brar(a)samsung.com> The codec_dai needs to be shutdown should the machine startup fails. This patch adds another bailout tag for that case and rename the tag for configuration failures. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/soc-core.c | 14 +++++++++----- 1 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a03bac9..c81eb59 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -427,24 +427,24 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", codec_dai->name, cpu_dai->name); - goto machine_err; + goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", codec_dai->name, cpu_dai->name); - goto machine_err; + goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", codec_dai->name, cpu_dai->name); - goto machine_err; + goto config_err; } /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active || codec_dai->active) { ret = soc_pcm_apply_symmetry(substream); if (ret != 0) - goto machine_err; + goto config_err; } pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); @@ -464,10 +464,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&pcm_mutex); return 0; -machine_err: +config_err: if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); +machine_err: + if (codec_dai->ops->shutdown) + codec_dai->ops->shutdown(substream, codec_dai); + codec_dai_err: if (platform->pcm_ops->close) platform->pcm_ops->close(substream); -- 1.6.2.5

3 2

[alsa-devel] ATMEL AC97 status
by Sedji Gaouaou 25 Feb '10

25 Feb '10

Hi, I am writing this mail to know what is the status of the atmel at91 ac97 patch? There is a question in the arm mailing list: http://lists.arm.linux.org.uk/lurker/message/20100224.190605.0a85b6de.en.ht… The point is that I have done some work on the ac97c.c driver so it will now support full duplex mode. So I would like to know if I can just resubmit a new patch with all the changes? Regards, Sedji Gaouaou

2 1

[alsa-devel] [PATCH] hda - remove unnecessary msleep on power state transitions
by Wu Fengguang 25 Feb '10

25 Feb '10

From: "Zhang, Rui" <rui.zhang(a)intel.com> This will save ~15ms boot time. The first 10ms sleep was introduced in commit d2595d86e5 for (buggy) Cxt codecs, so better to limit the sleep to the problem hardware. For the second 10ms sleep, the HDA spec says: Power State[1:0]: 00: Node Power state (D0) is fully on. 01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog playback) which must remain fully on. 10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state. 11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software control. Note that any low power state set by software must retain sufficient operational capability to properly respond to subsequent software Power State command. So 10ms is actually the max wait time. It should be safe to remove/reduce it and rely on the loop of 1ms-sleeps. CC: Marc Boucher <marc(a)linuxant.com> CC: Arjan van de Ven <arjan(a)linux.intel.com> Signed-off-by: Zhang Rui <rui.zhang(a)intel.com> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- sound-2.6.orig/sound/pci/hda/hda_codec.c 2010-01-13 09:41:49.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_codec.c 2010-02-24 09:34:14.000000000 +0800 @@ -2767,7 +2767,8 @@ static void hda_set_power_state(struct h snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, power_state); /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0) + if (power_state == AC_PWRST_D0 && + codec->vendor_id & 0xffff0000 == 0x14f10000) msleep(10); nid = codec->start_nid; @@ -2801,7 +2802,6 @@ static void hda_set_power_state(struct h if (power_state == AC_PWRST_D0) { unsigned long end_time; int state; - msleep(10); /* wait until the codec reachs to D0 */ end_time = jiffies + msecs_to_jiffies(500); do {

2 2

[alsa-devel] [PATCHv4 1/7] OMAP4: PMIC: Add support for twl6030 codec
by Olaya, Margarita 25 Feb '10

25 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data may be passed: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected However, codec driver can operate if any or none of them are passed. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- drivers/mfd/twl-core.c | 18 +++++++++++++++--- include/linux/i2c/twl.h | 4 ++++ 2 files changed, 19 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 514e1e4..7e3b0d7 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -118,7 +118,8 @@ #define twl_has_watchdog() false #endif -#if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) +#if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ + defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -735,8 +736,19 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) return PTR_ERR(child); } - if (twl_has_codec() && pdata->codec) { - child = add_child(1, "twl4030_codec", + if (twl_has_codec() && pdata->codec && twl_class_is_4030()) { + sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; + child = add_child(sub_chip_id, "twl4030_codec", + pdata->codec, sizeof(*pdata->codec), + false, 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } + + /* Phoenix*/ + if (twl_has_codec() && pdata->codec && twl_class_is_6030()) { + sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid; + child = add_child(sub_chip_id, "twl6030_codec", pdata->codec, sizeof(*pdata->codec), false, 0, 0); if (IS_ERR(child)) diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index 9cf6cd7..d943c8b 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -556,6 +556,10 @@ struct twl4030_codec_data { unsigned int audio_mclk; struct twl4030_codec_audio_data *audio; struct twl4030_codec_vibra_data *vibra; + + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { -- 1.6.1.3

5 6

[alsa-devel] ASoC update for 2.6.34
by Mark Brown 25 Feb '10

25 Feb '10

The following changes since commit 47fc9a0a808f23b7b305f6c018e4882118b88d92: Kuninori Morimoto (1): ASoC: fsi: Modify over/under run error settlement are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Misael Lopez Cruz (1): OMAP4: PMIC: Add support for twl6030 codec drivers/mfd/twl-core.c | 18 +++++++++++++++--- include/linux/i2c/twl.h | 4 ++++ 2 files changed, 19 insertions(+), 3 deletions(-)

2 2

[alsa-devel] ASoC status to support multiple CODECs?
by Joonyoung Shim 25 Feb '10

25 Feb '10

Hi, Can i know the work progress status to support multiple CODECs at the ASoC? I just wonder. Thanks.

2 2

Re: [alsa-devel] Troubles using the external SDK
by Raymond Yau 25 Feb '10

25 Feb '10

>> Anyways. I want to know how you'd do this. >> The OSS driver for example doesn't even seem to define a delay() callback, >> which strikes me as odd.How can it know about the delay? http://git.alsa-project.org/?p=alsa-plugins.git;a=blob_plain;f=doc/README-p… This plugin converts the ALSA API over OSS API. With this plugin, ALSA native apps can run on OSS drivers. The function supported by this plugin is limited. For example, you cannot use dmix together with this plugin. Don't expect too much :) Please note that this plugin require a real OSS driver , not ALSA oss kernel emulation e.g. you can use this oss plugin to use the aureal binary driver http://aureal.sourceforge.net/ [^ <http://aureal.sourceforge.net/>] on Fedore Core 1 when alsa au88x0 driver was not written

1 0

[alsa-devel] [PATCHv4 3/7] ASoC: TWL6030: Manual power-up/down sequences
by Olaya, Margarita 25 Feb '10

25 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> TWL6030 codec device can be powered-up/down through a specific register writes sequence. These sequences can be used when no gpio line is provided for AUDPWRON. When the codec is powered-up in this way, automatic power-down sequence (triggered by thermal shutdown) is not possible. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 112 ++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/twl6030.h | 16 ++++++ 2 files changed, 115 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 8b52aa1..ec838b1 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -244,6 +244,88 @@ static void twl6030_init_vdd_regs(struct snd_soc_codec *codec) } } +/* twl6030 codec manual power-up sequence */ +static void twl6030_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6030_NCPENA | TWL6030_NCPOPEN; + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6030_RESETSPLIT; + twl6030_write(codec, TWL6030_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); +} + +/* twl6030 codec manual power-down sequence */ +static void twl6030_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6030_read_reg_cache(codec, TWL6030_REG_NCPCTL); + ldoctl = twl6030_read_reg_cache(codec, TWL6030_REG_LDOCTL); + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + accctl = twl6030_read_reg_cache(codec, TWL6030_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6030_LSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6030_NCPENA | TWL6030_NCPOPEN); + twl6030_write(codec, TWL6030_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6030_HSLDOENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6030_OSCENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6030_REFENA; + twl6030_write(codec, TWL6030_REG_LDOCTL, ldoctl); + mdelay(10); +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -480,12 +562,15 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-up sequence latency */ mdelay(16); - } - /* sync registers updated during power-up sequence */ - twl6030_read(codec, TWL6030_REG_NCPCTL); - twl6030_read(codec, TWL6030_REG_LDOCTL); - twl6030_read(codec, TWL6030_REG_LPPLLCTL); + /* sync registers updated during power-up sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6030_power_up(codec); + } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); @@ -502,12 +587,16 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* power-down sequence latency */ udelay(500); - } - /* sync registers updated during power-down sequence */ - twl6030_read(codec, TWL6030_REG_NCPCTL); - twl6030_read(codec, TWL6030_REG_LDOCTL); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + /* sync registers updated during power-down sequence */ + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6030_power_down(codec); + } priv->codec_powered = 0; break; @@ -722,9 +811,6 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) goto gpio2_err; priv->codec_powered = 0; - } else { - /* if no gpio is provided, then assume its always on */ - priv->codec_powered = 1; } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 7fba60e..81d25f6 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,18 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* NCPCTL (0x05) fields */ + +#define TWL6030_NCPENA 0x01 +#define TWL6030_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6030_LSLDOENA 0x01 +#define TWL6030_HSLDOENA 0x04 +#define TWL6030_REFENA 0x40 +#define TWL6030_OSCENA 0x80 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -88,6 +100,10 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* ACCCTL (0x2D) fields */ + +#define TWL6030_RESETSPLIT 0x04 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.6.1.3

2 2

[alsa-devel] Are there any quirks for getting mixers on USB audio devices working?
by Felix Homann 24 Feb '10

24 Feb '10

Hi, I'm still trying to get a M-Audio Fast Track Ultra 8R (FTU8R) working. Searching through the USB Device Class Definition for Audio devices and comparing with the findings and logs I've posted recently, it seems to me that the mixer on the FTU8R is controlled in a pretty much standard USB fashion. Unfortunately, it is not recognized as such. Are there any quirks to get such devices working? Kind regards, Felix

1 0

[alsa-devel] [PATCHv4 5/7] ASoC: TWL6030: Add restrictions for low-power playback mode
by Olaya, Margarita 24 Feb '10

24 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. Only in this mode the codec can support 44.1 and 48 kHz, low-power PLL should provide sysclk signal. Currently, handsfree DAC and driver are the only components that can prevent codec to enter to low-power playback mode. Other components like earphone driver, vibrator driver or loopback (which are not yet supported in the driver) can cause the same effect. In order to detect conflicting paths, CODEC driver supervises non-low-power widgets powered by DAPM mechanism, just at the point pcm trigger callback gets called. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 79 +++++++++++++++++++++++++++++++++++++++---- 1 files changed, 71 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 792407f..53aa837 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -48,6 +48,7 @@ struct twl6030_data { int audpwron; int codec_powered; int pll; + int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; }; @@ -352,6 +353,20 @@ static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) return 0; } +static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6030_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -485,10 +500,14 @@ static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { TWL6030_REG_HSLCTL, 0, 0), SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", TWL6030_REG_HSRCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Left", "Handsfree Playback", - TWL6030_REG_HFLCTL, 0, 0), - SND_SOC_DAPM_DAC("HFDAC Right", "Handsfree Playback", - TWL6030_REG_HFRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6030_REG_HFLCTL, 0, 0, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6030_REG_HFRCTL, 0, 0, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback switches */ SND_SOC_DAPM_SWITCH("HSDAC Left Playback", @@ -504,10 +523,14 @@ static const struct snd_soc_dapm_widget twl6030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0, &hsl_driver_switch_controls), SND_SOC_DAPM_SWITCH("Headset Right Driver", SND_SOC_NOPM, 0, 0, &hsr_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Handsfree Left Driver", - SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls), - SND_SOC_DAPM_SWITCH("Handsfree Right Driver", - SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls), + SND_SOC_DAPM_SWITCH_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Handsfree Right Driver", + SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, + twl6030_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HFDAC Left PGA", @@ -668,6 +691,17 @@ static int twl6030_startup(struct snd_pcm_substream *substream, return -EINVAL; } + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && substream->stream) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, priv->sysclk_constraints); @@ -712,6 +746,34 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, return 0; } +static int twl6030_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* + * low-power playback mode is restricted + * for headset path only + */ + if ((priv->sysclk == 17640000) && priv->non_lp) { + dev_err(codec->dev, + "some enabled paths aren't supported at %dHz\n", + priv->sysclk); + return -EPERM; + } + break; + default: + break; + } + + return 0; +} + static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -822,6 +884,7 @@ static int twl6030_set_dai_sysclk(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .startup = twl6030_startup, .hw_params = twl6030_hw_params, + .trigger = twl6030_trigger, .set_sysclk = twl6030_set_dai_sysclk, }; -- 1.6.1.3

2 1

[alsa-devel] alsa-compile.sh - easy testing of latest ALSA code
by Jaroslav Kysela 24 Feb '10

24 Feb '10

Hi all, I'm instroducing a new bash script - alsa-compile.sh - to allow easy testing of new ALSA code by all users using standard Linux desktop distributions. For example to get latest driver snapshot from www.alsa-project.org, compile it and insert new kernel modules, just type: wget -O alsa-compile.sh http://www.alsa-project.org/alsa-compile.sh chmod 755 alsa-compile.sh ./alsa-compile.sh --driver --kmodules More information are available using --help option. More examples are available using --examples option. Actually, Fedora and openSUSE distributions are supported. Please, send patches extending functionality for other distributions to this list or directly to me. Takashi, you might want to add support (option) for your driver tarballs. It would be probably a good idea to create a link to latest tarball in your web directory. Please, send ideas, bugs, patches and any other issues to this list. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] [PATCH v4 2/3] ASoC: OMAP4: Add support for McPDM
by Candelaria Villareal, Jorge 24 Feb '10

24 Feb '10

McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> --- Resending patch to fix an issue reset_playback function sound/soc/omap/mcpdm.c | 484 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++++ 2 files changed, 635 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 0000000..6ee9067 --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,484 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Copyright (C) 2009 - Texas Instruments, Inc. + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/wait.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/irq.h> + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static inline void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +static void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +static void omap_mcpdm_reset_capture(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +static void omap_mcpdm_reset_playback(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + ret = -EBUSY; + goto err; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + goto err; + } + + return 0; + +err: + clk_disable(mcpdm->clk); + return ret; +} + +void omap_mcpdm_free(void) +{ + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + clk_disable(mcpdm->clk); + + free_irq(mcpdm->irq, (void *)mcpdm); +} + +/* Enable/disable DC offset cancelation for the analog + * headset path (PDM channels 1 and 2). + */ +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_resource: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + + kfree(mcpdm_ptr); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static int __init omap_mcpdm_init(void) +{ + return platform_driver_register(&omap_mcpdm_driver); +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 0000000..41e50cc --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,151 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +extern void omap_mcpdm_start(int stream); +extern void omap_mcpdm_stop(int stream); +extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_request(void); +extern void omap_mcpdm_free(void); +extern int omap_mcpdm_set_offset(int offset1, int offset2); -- 1.6.0.4

2 3

[alsa-devel] Built-in speaker was not auto-muted on EeePC 1000H
by Bin Gao 24 Feb '10

24 Feb '10

The built-in speaker was not muted when a headphone was plugged. I found this issue on EeePC 1000H and 1005HA, both of them have ALC269 codec. The kernel tested is 2.6.33-rc8. Is this a known alsa driver regression? I remember 2.6.32 kernel worked fine for me. Thanks, Bin

1 0

[alsa-devel] [PATCHv4 6/7] ASoC: TWL6030: Enable audio interrupt
by Olaya, Margarita 24 Feb '10

24 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 70 ++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl6030.h | 14 +++++++++ 2 files changed, 79 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 53aa837..b8dd5ae 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -46,6 +46,7 @@ struct twl6030_data { struct snd_soc_codec codec; int audpwron; + int naudint; int codec_powered; int pll; int non_lp; @@ -61,7 +62,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x7B, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -367,6 +368,39 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, return 0; } +/* audio interrupt handler */ +static irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + dev_alert(codec->dev, "codec is ready\n"); + break; + default: + dev_err(codec->dev, "unknown audio interrupt %d\n", intid); + break; + } + + return IRQ_HANDLED; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -992,19 +1026,23 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; struct snd_soc_codec *codec; struct twl6030_data *priv; - int audpwron; + int audpwron, naudint; int ret = 0; priv = kzalloc(sizeof(struct twl6030_data), GFP_KERNEL); if (priv == NULL) return -ENOMEM; - if (twl_codec) + if (twl_codec) { audpwron = twl_codec->audpwron_gpio; - else + naudint = twl_codec->naudint_irq; + } else { audpwron = -EINVAL; + naudint = 0; + } priv->audpwron = audpwron; + priv->naudint = naudint; codec = &priv->codec; codec->dev = &pdev->dev; @@ -1042,13 +1080,28 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) priv->codec_powered = 0; } + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6030_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6030_codec", codec); + if (ret) + goto gpio2_err; + } else { + dev_warn(codec->dev, + "no naudint irq, audio interrupts disabled\n"); + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } + /* init vio registers */ twl6030_init_vio_regs(codec); /* power on device */ ret = twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret) - goto gpio2_err; + goto irq_err; ret = snd_soc_register_codec(codec); if (ret) @@ -1067,6 +1120,9 @@ dai_err: twl6030_codec = NULL; reg_err: twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); +irq_err: + if (naudint) + free_irq(naudint, codec); gpio2_err: if (gpio_is_valid(audpwron)) gpio_free(audpwron); @@ -1081,10 +1137,14 @@ static int __devexit twl6030_codec_remove(struct platform_device *pdev) { struct twl6030_data *priv = twl6030_codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; if (gpio_is_valid(audpwron)) gpio_free(audpwron); + if (naudint) + free_irq(naudint, twl6030_codec); + snd_soc_unregister_dai(&twl6030_dai); snd_soc_unregister_codec(twl6030_codec); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index c321347..90b8a44 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -67,6 +67,20 @@ #define TWL6030_VIOREGNUM 18 #define TWL6030_VDDREGNUM 21 +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6030_ALLINT_MSK 0x7B + /* NCPCTL (0x05) fields */ #define TWL6030_NCPENA 0x01 -- 1.6.1.3

1 0

[alsa-devel] [PATCHv4 4/7] ASoC: TWL6030: Add support for low-power PLL
by Olaya, Margarita 24 Feb '10

24 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> TWL6030 codec sysclk can be provided by: low-power or high performance PLL. The low-power PLL takes a low-frequency input at 32,768 Hz and generates an approximate of 17.64 or 19.2 MHz. The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. Headset DAC and output driver should be configured according to the selected mode. 17.64 MHz sysclk is recommended only for headset low-power playback mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 232 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 16 +++ 2 files changed, 215 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index ec838b1..792407f 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -39,7 +39,7 @@ #include "twl6030.h" -#define TWL6030_RATES (SNDRV_PCM_RATE_96000) +#define TWL6030_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) /* codec private data */ @@ -47,6 +47,9 @@ struct twl6030_data { struct snd_soc_codec codec; int audpwron; int codec_powered; + int pll; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; }; /* @@ -326,6 +329,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) mdelay(10); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -607,55 +633,195 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, return 0; } +/* set of rates for each pll: low-power and high-performance */ + +static unsigned int lp_rates[] = { + 88200, + 96000, +}; + +static struct snd_pcm_hw_constraint_list lp_constraints = { + .count = ARRAY_SIZE(lp_rates), + .list = lp_rates, +}; + +static unsigned int hp_rates[] = { + 96000, +}; + +static struct snd_pcm_hw_constraint_list hp_constraints = { + .count = ARRAY_SIZE(hp_rates), + .list = hp_rates, +}; + +static int twl6030_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + priv->sysclk_constraints); + + return 0; +} + +static int twl6030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6030_HPPLL_ID) + return 0; + + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6030_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6030_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + return 0; +} + static int twl6030_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; + struct twl6030_data *priv = codec->private_data; u8 hppllctl, lppllctl; hppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppllctl &= ~TWL6030_MCLK_MSK; - - switch (freq) { - case 12000000: - /* mclk input, pll enabled */ - hppllctl |= TWL6030_MCLK_12000KHZ | - TWL6030_HPLLSQRBP | - TWL6030_HPLLENA; - break; - case 19200000: - /* mclk input, pll disabled */ - hppllctl |= TWL6030_MCLK_19200KHZ | - TWL6030_HPLLSQRBP | - TWL6030_HPLLBP; - break; - case 26000000: - /* mclk input, pll enabled */ - hppllctl |= TWL6030_MCLK_26000KHZ | - TWL6030_HPLLSQRBP | - TWL6030_HPLLENA; + lppllctl = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + switch (freq) { + case 32768: + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + /* clk32k input requires low-power pll */ + lppllctl |= TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* lppll divider */ + switch (priv->sysclk) { + case 17640000: + lppllctl |= TWL6030_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6030_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6030_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6030_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; break; - case 38400000: - /* clk slicer, pll disabled */ - hppllctl |= TWL6030_MCLK_38400KHZ | - TWL6030_HPLLSQRENA | - TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6030_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_12000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6030_MCLK_19200KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6030_MCLK_26000KHZ | + TWL6030_HPLLSQRBP | + TWL6030_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6030_MCLK_38400KHZ | + TWL6030_HPLLSQRENA | + TWL6030_HPLLBP; + break; + default: + dev_err(codec->dev, "unknown mclk freq %d\n", freq); + return -EINVAL; + } + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6030_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; break; default: - dev_err(codec->dev, "unknown sysclk freq %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppllctl); - - /* Disable LPPLL and select HPPLL */ - lppllctl = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppllctl); - return 0; } static struct snd_soc_dai_ops twl6030_dai_ops = { + .startup = twl6030_startup, + .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, }; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 81d25f6..c321347 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -100,10 +100,26 @@ #define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + /* ACCCTL (0x2D) fields */ #define TWL6030_RESETSPLIT 0x04 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 + +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.6.1.3

1 0

[alsa-devel] usbaudio: Support for USB audio v2.0 devices
by Daniel Mack 24 Feb '10

24 Feb '10

Here's a new version of the patchset to start support for USB audio v2.0 devices. I moved all the definitions to linux/usb/audio.h. As I did that, I realized that there's actually many definitions in sound/usb/usbaudio.h which don't need to be there at all. Most of them are already defined in audio.h and ch9.h. Hence, I added a 5th patch to consolidate things and ease the chaos a bit. Hope I didn't break anything :) Daniel

5 19

[alsa-devel] 2.6.31.x vs. rme vs. alsa 1.0.22?
by Fernando Lopez-Lezcano 23 Feb '10

23 Feb '10

Hey, has anyone been seeing strange behavior from this combination? kernel 2.6.31.x rt20 + alsa 1.0.22 userland RME card (pcmcia card + multiface) hdspmixer is not doing the right thing (does not initialize the card in a way in which playback works), it does not see the hwdep interface (or something like that) and disables metering, alsamixer even segfaults when I reach the end of the controls listed. Plain weird. Smells like something changed deep in the kernel that makes alsa-lib very unhappy. Alsa-tools rebuilt from source does not make a difference. Weirdness goes away when I boot into 2.6.29.6 rt23... Is there anything in alsa-* that depends on which _kernel_ is available at compile time? -- Fernando

2 6

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 23 Feb '10

23 Feb '10

The following changes since commit 6c5f1fed49f96a0600aa9a97ac3faf972c33a341: Mark Brown (1): ASoC: Make pmdown_time a long are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Candelaria Villareal, Jorge (1): ASoC: OMAP4: Add support for McPDM Kuninori Morimoto (1): ASoC: fsi: Modify over/under run error settlement Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver Peter Ujfalusi (1): ASoC: core: On resume also check the soc device state sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 484 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 ++++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 251 ++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- sound/soc/sh/fsi.c | 46 +++-- sound/soc/soc-core.c | 6 + 11 files changed, 964 insertions(+), 29 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h

2 1

[alsa-devel] [PATCH v4 0/3] McPDM driver patch series
by Candelaria Villareal, Jorge 23 Feb '10

23 Feb '10

McPDM module is the interface between the OMAP and TWL6030 audio codec. The interface supports the following: * Up to 5 downlink channels and 3 uplink channels * 88.2 and 96 KHz * 32 bit sample size Currently, driver is set to support up to 4 downlink channels. If 2 channels are selected, output will go to the headset, and if 4 channels are selected, output will go to the headset and speakers. Functionality to select the output at runtime will be added afterwards. Command and status mode is not supported in current patches. Changes from v3: - Implemented changes based on comments from Liam Girdwood. -- Jorge Eduardo Candelaria (1): ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 486 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 251 ++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- 9 files changed, 935 insertions(+), 8 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h

4 3

[alsa-devel] [PATCH] ALSA: hda - Refer to macmini3 channel mode functions
by Luke Yelavich 23 Feb '10

23 Feb '10

From: Luke Yelavich <luke.yelavich(a)canonical.com> Refer to alc885_macmini3_{2,6}ch_mode_init functions instead of those for the mb5 model quirk. While they are the functino contents are the same now, things may change later. This also gets rid of compiler warnings about initialized, but unused functions. Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f8fb260..9e5ea9e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7033,8 +7033,8 @@ static struct hda_verb alc885_macmini3_ch6_init[] = { }; static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { - { 2, alc885_mb5_ch2_init }, - { 6, alc885_mb5_ch6_init }, + { 2, alc885_macmini3_ch2_init }, + { 6, alc885_macmini3_ch6_init }, }; -- 1.7.0

3 2

[alsa-devel] [PATCH] ASoC: fsi: Modify over/under run error settlement
by Kuninori Morimoto 23 Feb '10

23 Feb '10

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- sound/soc/sh/fsi.c | 46 +++++++++++++++++++++++++--------------------- 1 files changed, 25 insertions(+), 21 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c36d24..993abb7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -388,7 +388,7 @@ static void fsi_soft_all_reset(struct fsi_master *master) } /* playback interrupt */ -static int fsi_data_push(struct fsi_priv *fsi) +static int fsi_data_push(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -397,7 +397,7 @@ static int fsi_data_push(struct fsi_priv *fsi) int fifo_free; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -453,24 +453,26 @@ static int fsi_data_push(struct fsi_priv *fsi) fsi->byte_offset += send * width; - ret = 0; status = fsi_reg_read(fsi, DOFF_ST); - if (status & ERR_OVER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "over run error\n"); - fsi_reg_write(fsi, DOFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DOFF_ST, 0); fsi_irq_enable(fsi, 1); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } -static int fsi_data_pop(struct fsi_priv *fsi) +static int fsi_data_pop(struct fsi_priv *fsi, int startup) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; @@ -479,7 +481,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) int fifo_fill; int width; u8 *start; - int i, ret, over_period; + int i, over_period; if (!fsi || !fsi->substream || @@ -534,21 +536,23 @@ static int fsi_data_pop(struct fsi_priv *fsi) fsi->byte_offset += fifo_fill * width; - ret = 0; status = fsi_reg_read(fsi, DIFF_ST); - if (status & ERR_UNDER) { + if (!startup) { struct snd_soc_dai *dai = fsi_get_dai(substream); - dev_err(dai->dev, "under run error\n"); - fsi_reg_write(fsi, DIFF_ST, status & ~ST_ERR); - ret = -EIO; + + if (status & ERR_OVER) + dev_err(dai->dev, "over run\n"); + if (status & ERR_UNDER) + dev_err(dai->dev, "under run\n"); } + fsi_reg_write(fsi, DIFF_ST, 0); fsi_irq_enable(fsi, 0); if (over_period) snd_pcm_period_elapsed(substream); - return ret; + return 0; } static irqreturn_t fsi_interrupt(int irq, void *data) @@ -562,13 +566,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_master_write(master, SOFT_RST, status | 0x00000010); if (int_st & INT_A_OUT) - fsi_data_push(&master->fsia); + fsi_data_push(&master->fsia, 0); if (int_st & INT_B_OUT) - fsi_data_push(&master->fsib); + fsi_data_push(&master->fsib, 0); if (int_st & INT_A_IN) - fsi_data_pop(&master->fsia); + fsi_data_pop(&master->fsia, 0); if (int_st & INT_B_IN) - fsi_data_pop(&master->fsib); + fsi_data_pop(&master->fsib, 0); fsi_master_write(master, INT_ST, 0x0000000); @@ -726,7 +730,7 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); + ret = is_play ? fsi_data_push(fsi, 1) : fsi_data_pop(fsi, 1); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); -- 1.6.3.3

2 3

[alsa-devel] Separate dma driver for cpu_dais
by jassi brar 23 Feb '10

23 Feb '10

Hi, The query concerns ASOC. Currently we have one platform per card. What if my card has two dai_links with each cpu_dai fed data with different mechanism(one with system dma controller and the other with a dma dedicated to the Audio block). The cpu_dai's are tightly coupled and I don't wanna have separate socdev's for them. Presently, I define a dma_id flag in snd_soc_dai, make the snd_soc_card point to a 'wrapper/muxed' platform driver that checks the dma_id flag and calls the relevant platform's callbacks. Of course, each cpu_dai initializes the dma_id to a unique value that corresponds to it's real dma mechanism. What approach/changes would you suggest I take, so that someday I could submit the implementation for mainline inclusion? (For those curious to have a reference - I am talking about Samsung's SoCs <http://dev.odroid.com/wiki/odroid/pds/HardwareInformation/S5PC100_UM_REV101… refer to figures on page-1635 & 1637> that have I2S controllers with h/w mixing capability and where 'secondary fifo' can be fed with I2S internal DMA controller with dedicated h/w ring buffer) Thanks.

3 23

[alsa-devel] [PATCH] ALSA: add support for Macbook Air 2, 1 internal speaker.
by Reimundo Heluani 23 Feb '10

23 Feb '10

From: Reimundo Heluani <rheluani(a)gmail.com> Add support for Macbook Air 2,1 (late 2008) internal speaker and headphones. Create a "mba21" model for snd-hda-intel. Signed-off-by: Reimundo Heluani <rheluani(a)gmail.com> --- Still missing internal microphone support --- linux-2.6/sound/pci/hda/patch_realtek.c.orig 2010-02-23 00:36:16.000000000 -0800 +++ linux-2.6/sound/pci/hda/patch_realtek.c 2010-02-23 00:44:35.000000000 -0800 @@ -207,6 +207,7 @@ enum { ALC882_ASUS_A7J, ALC882_ASUS_A7M, ALC885_MACPRO, + ALC885_MBA21, ALC885_MBP3, ALC885_MB5, ALC885_IMAC24, @@ -6852,6 +6853,13 @@ static struct hda_channel_mode alc882_si { 8, alc882_sixstack_ch8_init }, }; + +/* Macbook Air 2,1 */ + +static struct hda_channel_mode alc885_mba21_ch_modes[1] = { + { 2, NULL }, +}; + /* * macbook pro ALC885 can switch LineIn to LineOut without losing Mic */ @@ -7123,6 +7131,15 @@ static struct snd_kcontrol_new alc882_ba { } /* end */ }; +/* Macbook Air 2,1 same control for HP and internal Speaker */ + +static struct snd_kcontrol_new alc885_mba21_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), + { } +}; + + static struct snd_kcontrol_new alc885_mbp3_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7562,6 +7579,29 @@ static struct hda_verb alc885_mb5_init_v { } }; + +static struct hda_verb alc885_mba21_init_verbs[] = { + /*Internal and HP Speaker Mixer*/ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /*Internal Speaker Pin (0x0c)*/ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP Pin: output 0 (0x0e) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)}, + /* Line in (is hp when jack connected)*/ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + { } + }; + + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7724,6 +7764,17 @@ static void alc885_imac24_setup(struct h spec->autocfg.speaker_pins[1] = 0x1a; } +/* Macbook Air 2,1 */ +static void alc885_mba21_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; +} + + + static void alc885_mbp3_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -8938,6 +8989,7 @@ static const char *alc882_models[ALC882_ [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MBA21] = "mba21", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9172,6 +9224,18 @@ static struct alc_config_preset alc882_p .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, }, + [ALC885_MBA21] = { + .mixers = { alc885_mba21_mixer }, + .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .num_dacs = 2, + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_mba21_ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_mba21_setup, + .init_hook = alc_automute_amp, + }, [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs,

2 1

[alsa-devel] Conexant CX20561 alsa driver not muting headphone jack on Thinkpads when mute button pressed .. any ideas?
by Jerone Young 23 Feb '10

23 Feb '10

There is an interesting problem with Lenovo Thinkpads X200, X200s, X301, & .. looks like W500 using the Conexant CX20561 codec under Linux with alsa. But if you have your headphone plugged in. You get audio .. but if you press the hardware mute button .. it will not mute the headphones. By pulling out the headphones you will notice though that the internal speakers are muted. At first I blamed the bios. But then I did a test under FreeDOS where I used the beep command for sound. Sure enough I press the mute button and the headphone jack is muted. I've been looking for any documentation I can find on the CX20561 but there is non out there I can find. I could be wrong, but it seems some bit isn't being flipped for this codec to work correctly with this functionality with the alsa driver. Any ideas? I've created a bug here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4928 Thanks, Jerone

1 1

[alsa-devel] [PATCHES_v1] ASoC: Towards dropping runtime field from snd_soc_dai
by jassi brar 23 Feb '10

23 Feb '10

Hi, <Due to some problem I am unable to send inline patches, please accept the attached patches for this time> The following three patches are aimed at discarding runtime field from struct snd_soc_dai, which comes in the way of working towards making it possible to share cpu_dai/codec_dai among two or more dai_links. [PATCH 1/3] ASoC: PLATFORM: Pass dai_link as argument to suspend, resume Also a good idea to provide pointer to relevant dai_link to platform suspend/resume. It also makes easy traversing the ASOC links to reach the substream(when snd_soc_dai.runtime is dropped). [PATCH 2/3] ASOC: ATMEL: Getting cpu_dai method change [PATCH 3/3] ASOC: SND_SOC_DAI: Get rid of 'runtime' field Regards.

2 3

[alsa-devel] Macbook Air 2,1 - no mic nor main speakers
by Reimundo Heluani 23 Feb '10

23 Feb '10

Hello list, this came up a year ago and Takashi's answer was "a broken bios". I was hoping someone here could point me to documentation to read in order to diagnose the problem if indeed is a broken bios. I don't want to start randombly reading the driver's code. Macbook Air 2,1 2.6.32-gentoo-r3 SMP x86_64 Alsa Driver 1.0.21 libs 1.0.22 nVidia Corporation MCP79 High Definition Audio (rev b1) HDA NVidia at 0x93380000 irq 19 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb79 Codec: Realtek ALC889A Booting with snd_hda_intel.model=mbp3 get sound on headphones, but not on mainspeakers. The mic never worked.

2 7

[alsa-devel] [PATCH v4 3/3] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 22 Feb '10

22 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap-mcpdm.c | 251 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 4 files changed, 285 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..b96e9d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..41ae838 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := omap-mcpdm.o mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..25f19e4 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,251 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/control.h> +#include <plat/dma.h> +#include <plat/mcbsp.h> +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio playback", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio capture", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_capture_open(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int err; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + err = omap_mcpdm_playback_close(&mcpdm_links[stream]); + else + err = omap_mcpdm_capture_close(&mcpdm_links[stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- 1.6.0.4

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[alsa-devel] [PATCH v4 2/3] ASoC: OMAP4: Add support for McPDM
by Candelaria Villareal, Jorge 22 Feb '10

22 Feb '10

McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> --- sound/soc/omap/mcpdm.c | 486 ++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++++ 2 files changed, 637 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 0000000..33f3ed6 --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,486 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Copyright (C) 2009 - Texas Instruments, Inc. + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/wait.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/irq.h> + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static inline void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +static void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +static void omap_mcpdm_reset_capture(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +static void omap_mcpdm_reset_playback(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (links & MCPDM_DOWNLINK) { + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + } + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + ret = -EBUSY; + goto err; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + goto err; + } + + return 0; + +err: + clk_disable(mcpdm->clk); + return ret; +} + +void omap_mcpdm_free(void) +{ + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + clk_disable(mcpdm->clk); + + free_irq(mcpdm->irq, (void *)mcpdm); +} + +/* Enable/disable DC offset cancelation for the analog + * headset path (PDM channels 1 and 2). + */ +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_resource: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + + kfree(mcpdm_ptr); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static int __init omap_mcpdm_init(void) +{ + return platform_driver_register(&omap_mcpdm_driver); +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 0000000..41e50cc --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,151 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +extern void omap_mcpdm_start(int stream); +extern void omap_mcpdm_stop(int stream); +extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_request(void); +extern void omap_mcpdm_free(void); +extern int omap_mcpdm_set_offset(int offset1, int offset2); -- 1.6.0.4

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[alsa-devel] [PATCH v4 1/3] ASoC: OMAP: data_type and sync_mode configurable in audio dma
by Candelaria Villareal, Jorge 22 Feb '10

22 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 ++ sound/soc/omap/omap-pcm.c | 15 ++++++++------- sound/soc/omap/omap-pcm.h | 4 +++- 3 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2a..d297256 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9db2770..825db38 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, @@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_runtime_data *prtd = runtime->private_data; struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + int bytes; /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - /* - * Note: Regardless of interface data formats supported by OMAP McBSP - * or EAC blocks, internal representation is always fixed 16-bit/sample - */ - dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; dma_params.dst_port = OMAP_DMA_PORT_MPUI; + dma_params.dst_fi = dma_data->packet_size; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; @@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; dma_params.src_port = OMAP_DMA_PORT_MPUI; + dma_params.src_fi = dma_data->packet_size; } /* * Set DMA transfer frame size equal to ALSA period size and frame @@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * we can transfer the whole ALSA buffer with single DMA transfer but * still can get an interrupt at each period bounary */ - dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + bytes = snd_pcm_lib_period_bytes(substream); + dma_params.elem_count = bytes >> dma_data->data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 38a821d..b19975d 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,8 +29,10 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - int sync_mode; /* DMA sync mode */ void (*set_threshold)(struct snd_pcm_substream *substream); + int data_type; /* data type 8,16,32 */ + int sync_mode; /* DMA sync mode */ + int packet_size; /* packet size only in PACKET mode */ }; extern struct snd_soc_platform omap_soc_platform; -- 1.6.0.4

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[alsa-devel] [PATCHv5 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 22 Feb '10

22 Feb '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. These patches apply on top of Janusz Krzysztofik's McBSP register cache patches in linux-omap-2.6.git Since the version 4, ASoC part has been modified to deal with McBSP ids consistently (zero based indexing). Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 60 +++++ arch/arm/plat-omap/mcbsp.c | 402 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 138 +++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 603 insertions(+), 1 deletions(-)

6 7

[alsa-devel] [PATCH] ASoC: core: On resume also check the soc device state
by Peter Ujfalusi 22 Feb '10

22 Feb '10

Check the card->codec on soc_resume to detect if the soc device is properly initialized. If the card->codec is NULL, than do not continue the resume operation, since the device is not initialized properly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The soc_susbend code is checking, if the soc device is properly initialized, but on soc_resume the check is missing, and it is unlikely that the soc device is actually initialized between these two calls. This fixes a crash, when doing suspend on a system where a soc device is not properly initialized. The crash happens on resume path, since the suspend has been protected against such a scenario. sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1c0336..a03bac9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -963,6 +963,12 @@ static int soc_resume(struct device *dev) struct snd_soc_card *card = socdev->card; struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; + /* If the initialization of this soc device failed, there is no codec + * associated with it. Just bail out in this case. + */ + if (!card->codec) + return 0; + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume -- 1.7.0

3 2

[alsa-devel] [PATCHv4 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 22 Feb '10

22 Feb '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. These patches apply on top of Janusz Krzysztofik's McBSP register cache patches in linux-omap-2.6.git Since the version 3, the buggy channel gain control has been fixed. ASoC part have been simplified resulting in removing a couple unnecessary functions. Added a comment to ASoC part explaining McBSP port number vs. index values used in OMAP McBSP implementation. Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 60 +++++ arch/arm/plat-omap/mcbsp.c | 402 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 138 +++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 603 insertions(+), 1 deletions(-)

7 15

[alsa-devel] M-Audio Fast Track Ultra 8R - USB Sniffs and first layman's analysis (Next try)
by Felix Homann 22 Feb '10

22 Feb '10

[Note, somehow my mails won't get through to the list. Maybe this time...] Hi, as posted on the alsa-user list, I could get the Fast Track Ultra 8R (USB Id 0763:2881) partially working with Clemes Ladisch's patch for the Fast Track Ultra (w/o 8R) [1] by just changing the USB Id. What's not working is setting the sample rate (at least I don't know how to set it up for 44100 Hz ). And there is no access to the DSP mixer and effects section. The nice thing on the DSP mixer is that it remembers it's settings and operates even without a computer attached. In other words, once set up, the device can operate as a simple stand alone mixer. In order to help developing a driver, I've gathered some information on how the M-Audio Fast Track Ultra 8R interface is set up in Windows (using Virtualbox and usbmon). I'm not familiar with USB programming nor with programming audio devices or Alsa. But I hope, I can give the relevant parts of the usbmon output nonetheless. I've grepped the usbmon logs for "Co", the output below contains only the lines with "Co" in them. Most of them carry a "Co:1:003:0" while some of them have "Co:1:006:0". I guess that's because of my replugging the device as at the time of this writing all output has the 006. I hope somebody can take this information and put in in a driver. You can find further information on this device on http://www.m-audio.com/products/en_us/FastTrackUltra8R.html. Some parts of my description might make a little more sense looking at a screenshot of the Windows settings dialog or the flow schematics, both of which you can find in the manual, linked to on the site above. See below my findings. And please, don't hesitate to aks for more information. Kind regards, Felix [1] http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg25385.html 1. Settings on Win** -------------------- There are 4 regions in the settings tab. (A nice overview on the Control Panel can be found in the manual, which you can download at www.m-audio.com.) 1.1 Effect Program ------------------ 1.1.1 The Effect Program ------------------------ There are 8 effect programs to choose from a drop down menu: Room 1-3, Hall 1-2, Plate, Delay and Echo. Choosing an effect programm yields these relevant parts of the usbmon output: ffff8800a00d0d80 595057804 S Co:1:003:0 s 21 01 0100 0600 0002 2 = [VAL] ffff8800a00d0d80 595058325 C Co:1:003:0 0 2 > ffff8800a00d0d80 595079774 S Co:1:003:0 s 21 01 0200 0600 0001 1 = 40 ffff8800a00d0d80 595080194 C Co:1:003:0 0 1 > ffff8800a00d0d80 595084732 S Co:1:003:0 s 21 01 0300 0600 0002 2 = 0040 ffff8800a00d0d80 595085070 C Co:1:003:0 0 2 > ffff8800a00d0d80 595089766 S Co:1:003:0 s 21 01 0400 0600 0001 1 = 40 ffff8800a00d0d80 595090192 C Co:1:003:0 0 1 > where [VAL] is one of the following values: 0000 for "Room 1" 0100 for "Room 2" 0200 for "Room 3" 0300 for "Hall 1" 0400 for "Hall 2" 0500 for "Plate" 0600 for "Delay" 0700 for "Echo". 1.1.2 Duration -------------- There's a control "Duration". Changing it yields these relevant parts of the usbmon output: ffff8800608fed80 771218819 S Co:1:003:0 s 21 01 0300 0600 0002 2 = [VAL] ffff8800608fed80 771219174 C Co:1:003:0 0 2 > where [VAL] seems to range from 0000 (control knob left) to 007f (control knob right). 1.1.3 Feedback -------------- There's a control "Feedback", which is only adjustable in the effect programs "Delay" and "Echo". Changing it yields these relevant parts of the usbmon output: ffff8800662a6600 3120473733 S Co:1:006:0 s 21 01 0400 0600 0001 1 = [VAL] ffff8800662a6600 3120474160 C Co:1:006:0 0 1 > where [VAL] seems to range from 00 (control knob left) to 7f (control knob right). 1.1.4 Volume ------------ There's a control "Volume". Changing it yields these relevant parts of the usbmon output: ffff8800ac1ed900 2882138576 S Co:1:006:0 s 21 01 0200 0600 0001 1 = [VAL] ffff8800ac1ed900 2882138990 C Co:1:006:0 0 1 > where [VAL] seems to range from 00 (control knob left) to 7f (control knob right). 1.2 Sample rate --------------- 1.2.1 Sync Source ----------------- There's a drop down menu for selecting the sync source. The choices are "Internal" and "External" (<- SPDIF). When set to "External" there are lots of "Zi" lines in the log like these: ffff88006638d000 3325892013 C Zi:1:006:1 0:1:904:0 8 0:0:168 0:312:144 0:624:120 0:936:144 0:1248:168 1152 = a7e4fff5 d7ff6fe3 ffb0e0ff a9e9ff65 dcff0000 00000000 cce4fffa d7ff8ee3 ffff88004e611800 3325893010 C Zi:1:006:1 0:1:912:0 8 0:0:168 0:312:144 0:624:120 0:936:168 0:1248:144 1176 = d7e5ff0a d9ff6ce4 ff0be2ff 17eaff66 ddff0000 00000000 eee5ff32 d9ff59e4 Don't know if they are relevant. 1.2.1.1 From "Internal" to "External" ------------------------------------- Here's what happens when switching the "Sync Source" from "Internal" to "External": ffff8800662a60c0 3325542111 S Co:1:006:0 s 01 0b 0000 0001 0000 0 ffff8800662a60c0 3325543386 C Co:1:006:0 0 0 ffff88009159e740 3325594872 S Co:1:006:0 s 01 0b 0000 0002 0000 0 ffff88009159e740 3325596263 C Co:1:006:0 0 0 ffff8800a00d0900 3325633405 S Co:1:006:0 s 01 0b 0001 0001 0000 0 ffff8800a00d0900 3325633637 C Co:1:006:0 0 0 ffff88009159e440 3325644251 S Co:1:006:0 s 01 0b 0002 0002 0000 0 ffff88009159e440 3325644631 C Co:1:006:0 0 0 ffff8800a00d0480 3326127863 S Co:1:006:0 s 22 01 0100 0001 0003 3 = 80bb00 ffff8800a00d0480 3326128270 C Co:1:006:0 0 3 > ffff8800a00d0480 3326132492 S Co:1:006:0 s 22 01 0100 0081 0003 3 = 80bb00 ffff8800a00d0480 3326132895 C Co:1:006:0 0 3 > 1.2.1.2 From "External" to "Internal" ------------------------------------- Here's what happens when switching the "Sync Source" from "External" to "Internal": ffff8800a00d0480 3349133460 S Co:1:006:0 s 01 0b 0000 0001 0000 0 ffff8800a00d0480 3349134696 C Co:1:006:0 0 0 ffff88009159e740 3349186489 S Co:1:006:0 s 01 0b 0000 0002 0000 0 ffff88009159e740 3349187696 C Co:1:006:0 0 0 ffff88009159e440 3349239544 S Co:1:006:0 s 01 0b 0001 0001 0000 0 ffff88009159e440 3349239823 C Co:1:006:0 0 0 ffff8800a00d0900 3349255385 S Co:1:006:0 s 01 0b 0001 0002 0000 0 ffff8800a00d0900 3349256567 C Co:1:006:0 0 0 1.2.2 Sample Rate ----------------- When "Sync Source" is set to "Internal" you can choose a sample rate 44.1 kHz, 48 kHz, 88.2 kHz an 96 kHz. ffff88007a826a40 4164558574 S Co:1:003:0 s 22 01 0100 0001 0003 3 = 007701 ffff88007a826a40 4164559977 C Co:1:003:0 0 3 > ffff88007a826a40 4164563251 S Co:1:003:0 s 22 01 0100 0081 0003 3 = 007701 ffff88007a826a40 4164563597 C Co:1:003:0 0 3 > where [Val] is one of the following values: 44ac00 for 44.1 kHz 80bb00 for 48 kHz 885801 for 88.2 kHz 007701 for 96 kHz 1.3 Other settings ------------------ The other settings do not seem to change anything on the device but are only related to the driver or the GUI. (Well, maybe the "High Performance Mode" button changes something on the device, when the driver is loaded next time.) 2. The mixer (on Win**) ----------------------- The Fast Track Ultra 8R features 4 monitor mixers. Monitor 1/2, 3/4, 5/6 and 7/8. They are mixing to the corresponding outputs 1-8. Each mixer is represented by controls for "Analog In 1-8" and "Software Return 1-8" and "Master Out 1/2". For each of those 16 channels ("Analog In"s and "Software Return"s) there is a level slider, a pan knob, a mute button and a solo button. "Master Out" 1+2 are represented by a level slider. For "Monitor 1/2" and "Monitor 3/4" in addition to the controls mentioned above there's an "Effects Send" control knob for each "Analog In" and "Software Return", a control knob "Effects Return" for each of the "Master Out"s and a "Effects Bypass" button. 2.1 Effects Bypass ------------------ 2.1.1 Monitor 1/2 Bypass ------------------------ Setting the Monitor 1/2 to bypass the effect yields this: ffff8800ac1edcc0 734991273 S Co:1:006:0 s 21 01 0201 0700 0002 2 = 0080 ffff8800ac1edcc0 734991583 C Co:1:006:0 0 2 > ffff8800ac1edcc0 734999254 S Co:1:006:0 s 21 01 0202 0700 0002 2 = 0080 ffff8800ac1edcc0 734999581 C Co:1:006:0 0 2 > Reenabling the effects on Monitor 1/2 yields this: ffff8800662a63c0 1084925217 S Co:1:006:0 s 21 01 0201 0700 0002 2 = e7cd ffff8800662a63c0 1084925584 C Co:1:006:0 0 2 > ffff8800662a63c0 1084934745 S Co:1:006:0 s 21 01 0202 0700 0002 2 = e7cd ffff8800662a63c0 1084935078 C Co:1:006:0 0 2 > 2.1.2 Monitor 3/4 Bypass ------------------------ Setting the Monitor 3/4 to bypass the effect yields this: ffff8800ac1ed0c0 3024153559 S Co:1:006:0 s 21 01 0203 0700 0002 2 = 0080 ffff8800ac1ed0c0 3024153901 C Co:1:006:0 0 2 > ffff8800ac1ed0c0 3024158859 S Co:1:006:0 s 21 01 0204 0700 0002 2 = 0080 ffff8800ac1ed0c0 3024159277 C Co:1:006:0 0 2 > Reenabling the effects on Monitor 3/4 yields this: ffff8800662a6840 3034928368 S Co:1:006:0 s 21 01 0203 0700 0002 2 = e7cd ffff8800662a6840 3034928788 C Co:1:006:0 0 2 > ffff8800662a6840 3034936142 S Co:1:006:0 s 21 01 0204 0700 0002 2 = e7cd ffff8800662a6840 3034936540 C Co:1:006:0 0 2 > 2.2 Levels ---------- Changing a level for a channel yields this: ffff880093800e00 3649051254 S Co:1:003:0 s 21 01 [OutIn] 0500 0002 2 = [VAL] ffff880093800e00 3649051527 C Co:1:003:0 0 2 > where [VAL] ranges from 0080 to ffff and [OutIn] is a number composed of the Out-channel (01-08) and In ranges from 01 to 10 (Hex, Software returns starting at 09). Example: [OutIn] = 0209 for the Level of "Software Return 1" on Out 2. 2.2.1 Panning ------------- Panning is done by changing the respective Levels for 2 channels. When panning is in the middle position, the max value for each level seems to be "fdfc". Example: ffff8800ac1edd80 1908059496 S Co:1:003:0 s 21 01 0402 0500 0002 2 = fdfc ffff8800ac1edd80 1908059852 C Co:1:003:0 0 2 > ffff8800ac1edd80 1908064495 S Co:1:003:0 s 21 01 0302 0500 0002 2 = fdfc ffff8800ac1edd80 1908064850 C Co:1:003:0 0 2 > This sets the level of "Analog In 2" in "Monitor 3/4" to the maximum value while the panning control is in the middle position. 2.3 Solo and Mute ----------------- Solo and Mute are operated by changing the respective levels to minimum value. In other words: Not a seperate action from changing levels. 2.4 Effects Send ---------------- Changing the Effects send value for a Channel ffff8800ac1ed840 958480452 S Co:1:[MON]:0 s 21 01 09[CH] 0500 0002 2 = [VAL] ffff8800ac1ed840 958480800 C Co:1:[MON]:0 0 2 > where [VAL] ranges from e7cd (knob left, minimum) to daff (knob right, maximum), [MON] = 003 for Monitor 1/2 and 006 for Monitor 3/4 and [CH] is the channel number (0-10, Hex, Software returns starting at 09) 2.5 Effects Return ------------------ 2.5.1 Monitor 1/2 ----------------- Setting the "Effects Return 1" ('left') Value in Monitor 1/2 yields ffff8800608fe9c0 1220780329 S Co:1:003:0 s 21 01 0201 0700 0002 2 = [VAL] ffff8800608fe9c0 1220780648 C Co:1:003:0 0 2 > with [VAL] from e7cd to daff. For "Effect Return 2" ('right') it's ffff880093800080 1281633920 S Co:1:003:0 s 21 01 0202 0700 0002 2 = [VAL] ffff880093800080 1281634358 C Co:1:003:0 0 2 > 2.5.2 Monitor 3/4 ----------------- Setting the "Effects Return 1" ('left') Value in Monitor 3/4 yields ffff8800ac1ed240 45150924 S Co:1:006:0 s 21 01 0203 0700 0002 2 = [VAL] ffff8800ac1ed240 45151245 C Co:1:006:0 0 2 > with [VAL] from e7cd to daff. For "Effect Return 2" ('right') it's ffff8800a00d0480 31181802 S Co:1:006:0 s 21 01 0204 0700 0002 2 = [VAL] ffff8800a00d0480 31182181 C Co:1:006:0 0 2 > 2.6 Master Out -------------- 2.6.1 Monitor 1/2 ----------------- Changing the "Master Out 1" ('left') level in Monitor 1/2 gives this ffff880093800440 1385027420 S Co:1:003:0 s 21 01 0109 0500 0002 2 = [VAL] ffff880093800440 1385027770 C Co:1:003:0 0 2 > changing the "Master Out 2" ('right') level in Monitor 1/2 gives this ffff8800ac1ede40 1410769402 S Co:1:003:0 s 21 01 020a 0500 0002 2 = [VAL] ffff8800ac1ede40 1410769744 C Co:1:003:0 0 2 > with [VAL] ranging from e7cd to 0000 (loudest). 2.6.2 Monitor 3/4 ----------------- Changing the "Master Out 1" ('left') level in Monitor 3/4 gives this ffff8800662a6540 4152434840 S Co:1:006:0 s 21 01 030b 0500 0002 2 = [VAL] ffff8800662a6540 4152435226 C Co:1:006:0 0 2 > changing the "Master Out 2" ('right') level in Monitor 3/4 gives this ffff88008280ca40 14381909 S Co:1:006:0 s 21 01 040c 0500 0002 2 = [VAL] ffff88008280ca40 14382274 C Co:1:006:0 0 2 > with [VAL] ranging from e7cd to 0000 (loudest). 3. Startup ---------- I've attached a log file showing what happens when the device is attached to Windows. Please note, that this is done under VirtualBox, so there might be some noise at the beginning of the file. [Edit: My mail doesn't seem to get through with the log file attached. If somone's interested, please send a mail an I will send you the log. But maybe the log file will get through sometime... ] I hope, there's some information in the log on how the Windows control panel gathers the current mixer values from the device.

1 0

[alsa-devel] M-Audio Fast Track Ultra 8R - USB Sniffs and first layman's analysis
by Felix Homann 22 Feb '10

22 Feb '10

Hi, as posted on the alsa-user list, I could get the Fast Track Ultra 8R (USB Id 0763:2881) partially working with Clemes Ladisch's patch for the Fast Track Ultra (w/o 8R) [1] by just changing the USB Id. What's not working is setting the sample rate (at least I don't know how to set it up for 44100 Hz ). And there is no access to the DSP mixer and effects section. The nice thing on the DSP mixer is that it remembers it's settings and operates even without a computer attached. In other words, once set up, the device can operate as a simple stand alone mixer. In order to help developing a driver, I've gathered some information on how the M-Audio Fast Track Ultra 8R interface is set up in Windows (using Virtualbox and usbmon). I'm not familiar with USB programming nor with programming audio devices or Alsa. But I hope, I can give the relevant parts of the usbmon output nonetheless. I've grepped the usbmon logs for "Co", the output below contains only the lines with "Co" in them. Most of them carry a "Co:1:003:0" while some of them have "Co:1:006:0". I guess that's because of my replugging the device as at the time of this writing all output has the 006. I hope somebody can take this information and put in in a driver. You can find further information on this device on http://www.m-audio.com/products/en_us/FastTrackUltra8R.html. Some parts of my description might make a little more sense looking at a screenshot of the Windows settings dialog or the flow schematics, both of which you can find in the manual, linked to on the site above. See below my findings. And please, don't hesitate to aks for more information. Kind regards, Felix [1] http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg25385.html 1. Settings on Win** -------------------- There are 4 regions in the settings tab. (A nice overview on the Control Panel can be found in the manual, which you can download at www.m-audio.com.) 1.1 Effect Program ------------------ 1.1.1 The Effect Program ------------------------ There are 8 effect programs to choose from a drop down menu: Room 1-3, Hall 1-2, Plate, Delay and Echo. Choosing an effect programm yields these relevant parts of the usbmon output: ffff8800a00d0d80 595057804 S Co:1:003:0 s 21 01 0100 0600 0002 2 = [VAL] ffff8800a00d0d80 595058325 C Co:1:003:0 0 2 > ffff8800a00d0d80 595079774 S Co:1:003:0 s 21 01 0200 0600 0001 1 = 40 ffff8800a00d0d80 595080194 C Co:1:003:0 0 1 > ffff8800a00d0d80 595084732 S Co:1:003:0 s 21 01 0300 0600 0002 2 = 0040 ffff8800a00d0d80 595085070 C Co:1:003:0 0 2 > ffff8800a00d0d80 595089766 S Co:1:003:0 s 21 01 0400 0600 0001 1 = 40 ffff8800a00d0d80 595090192 C Co:1:003:0 0 1 > where [VAL] is one of the following values: 0000 for "Room 1" 0100 for "Room 2" 0200 for "Room 3" 0300 for "Hall 1" 0400 for "Hall 2" 0500 for "Plate" 0600 for "Delay" 0700 for "Echo". 1.1.2 Duration -------------- There's a control "Duration". Changing it yields these relevant parts of the usbmon output: ffff8800608fed80 771218819 S Co:1:003:0 s 21 01 0300 0600 0002 2 = [VAL] ffff8800608fed80 771219174 C Co:1:003:0 0 2 > where [VAL] seems to range from 0000 (control knob left) to 007f (control knob right). 1.1.3 Feedback -------------- There's a control "Feedback", which is only adjustable in the effect programs "Delay" and "Echo". Changing it yields these relevant parts of the usbmon output: ffff8800662a6600 3120473733 S Co:1:006:0 s 21 01 0400 0600 0001 1 = [VAL] ffff8800662a6600 3120474160 C Co:1:006:0 0 1 > where [VAL] seems to range from 00 (control knob left) to 7f (control knob right). 1.1.4 Volume ------------ There's a control "Volume". Changing it yields these relevant parts of the usbmon output: ffff8800ac1ed900 2882138576 S Co:1:006:0 s 21 01 0200 0600 0001 1 = [VAL] ffff8800ac1ed900 2882138990 C Co:1:006:0 0 1 > where [VAL] seems to range from 00 (control knob left) to 7f (control knob right). 1.2 Sample rate --------------- 1.2.1 Sync Source ----------------- There's a drop down menu for selecting the sync source. The choices are "Internal" and "External" (<- SPDIF). When set to "External" there are lots of "Zi" lines in the log like these: ffff88006638d000 3325892013 C Zi:1:006:1 0:1:904:0 8 0:0:168 0:312:144 0:624:120 0:936:144 0:1248:168 1152 = a7e4fff5 d7ff6fe3 ffb0e0ff a9e9ff65 dcff0000 00000000 cce4fffa d7ff8ee3 ffff88004e611800 3325893010 C Zi:1:006:1 0:1:912:0 8 0:0:168 0:312:144 0:624:120 0:936:168 0:1248:144 1176 = d7e5ff0a d9ff6ce4 ff0be2ff 17eaff66 ddff0000 00000000 eee5ff32 d9ff59e4 Don't know if they are relevant. 1.2.1.1 From "Internal" to "External" ------------------------------------- Here's what happens when switching the "Sync Source" from "Internal" to "External": ffff8800662a60c0 3325542111 S Co:1:006:0 s 01 0b 0000 0001 0000 0 ffff8800662a60c0 3325543386 C Co:1:006:0 0 0 ffff88009159e740 3325594872 S Co:1:006:0 s 01 0b 0000 0002 0000 0 ffff88009159e740 3325596263 C Co:1:006:0 0 0 ffff8800a00d0900 3325633405 S Co:1:006:0 s 01 0b 0001 0001 0000 0 ffff8800a00d0900 3325633637 C Co:1:006:0 0 0 ffff88009159e440 3325644251 S Co:1:006:0 s 01 0b 0002 0002 0000 0 ffff88009159e440 3325644631 C Co:1:006:0 0 0 ffff8800a00d0480 3326127863 S Co:1:006:0 s 22 01 0100 0001 0003 3 = 80bb00 ffff8800a00d0480 3326128270 C Co:1:006:0 0 3 > ffff8800a00d0480 3326132492 S Co:1:006:0 s 22 01 0100 0081 0003 3 = 80bb00 ffff8800a00d0480 3326132895 C Co:1:006:0 0 3 > 1.2.1.2 From "External" to "Internal" ------------------------------------- Here's what happens when switching the "Sync Source" from "External" to "Internal": ffff8800a00d0480 3349133460 S Co:1:006:0 s 01 0b 0000 0001 0000 0 ffff8800a00d0480 3349134696 C Co:1:006:0 0 0 ffff88009159e740 3349186489 S Co:1:006:0 s 01 0b 0000 0002 0000 0 ffff88009159e740 3349187696 C Co:1:006:0 0 0 ffff88009159e440 3349239544 S Co:1:006:0 s 01 0b 0001 0001 0000 0 ffff88009159e440 3349239823 C Co:1:006:0 0 0 ffff8800a00d0900 3349255385 S Co:1:006:0 s 01 0b 0001 0002 0000 0 ffff8800a00d0900 3349256567 C Co:1:006:0 0 0 1.2.2 Sample Rate ----------------- When "Sync Source" is set to "Internal" you can choose a sample rate 44.1 kHz, 48 kHz, 88.2 kHz an 96 kHz. ffff88007a826a40 4164558574 S Co:1:003:0 s 22 01 0100 0001 0003 3 = 007701 ffff88007a826a40 4164559977 C Co:1:003:0 0 3 > ffff88007a826a40 4164563251 S Co:1:003:0 s 22 01 0100 0081 0003 3 = 007701 ffff88007a826a40 4164563597 C Co:1:003:0 0 3 > where [Val] is one of the following values: 44ac00 for 44.1 kHz 80bb00 for 48 kHz 885801 for 88.2 kHz 007701 for 96 kHz 1.3 Other settings ------------------ The other settings do not seem to change anything on the device but are only related to the driver or the GUI. (Well, maybe the "High Performance Mode" button changes something on the device, when the driver is loaded next time.) 2. The mixer (on Win**) ----------------------- The Fast Track Ultra 8R features 4 monitor mixers. Monitor 1/2, 3/4, 5/6 and 7/8. They are mixing to the corresponding outputs 1-8. Each mixer is represented by controls for "Analog In 1-8" and "Software Return 1-8" and "Master Out 1/2". For each of those 16 channels ("Analog In"s and "Software Return"s) there is a level slider, a pan knob, a mute button and a solo button. "Master Out" 1+2 are represented by a level slider. For "Monitor 1/2" and "Monitor 3/4" in addition to the controls mentioned above there's an "Effects Send" control knob for each "Analog In" and "Software Return", a control knob "Effects Return" for each of the "Master Out"s and a "Effects Bypass" button. 2.1 Effects Bypass ------------------ 2.1.1 Monitor 1/2 Bypass ------------------------ Setting the Monitor 1/2 to bypass the effect yields this: ffff8800ac1edcc0 734991273 S Co:1:006:0 s 21 01 0201 0700 0002 2 = 0080 ffff8800ac1edcc0 734991583 C Co:1:006:0 0 2 > ffff8800ac1edcc0 734999254 S Co:1:006:0 s 21 01 0202 0700 0002 2 = 0080 ffff8800ac1edcc0 734999581 C Co:1:006:0 0 2 > Reenabling the effects on Monitor 1/2 yields this: ffff8800662a63c0 1084925217 S Co:1:006:0 s 21 01 0201 0700 0002 2 = e7cd ffff8800662a63c0 1084925584 C Co:1:006:0 0 2 > ffff8800662a63c0 1084934745 S Co:1:006:0 s 21 01 0202 0700 0002 2 = e7cd ffff8800662a63c0 1084935078 C Co:1:006:0 0 2 > 2.1.2 Monitor 3/4 Bypass ------------------------ Setting the Monitor 3/4 to bypass the effect yields this: ffff8800ac1ed0c0 3024153559 S Co:1:006:0 s 21 01 0203 0700 0002 2 = 0080 ffff8800ac1ed0c0 3024153901 C Co:1:006:0 0 2 > ffff8800ac1ed0c0 3024158859 S Co:1:006:0 s 21 01 0204 0700 0002 2 = 0080 ffff8800ac1ed0c0 3024159277 C Co:1:006:0 0 2 > Reenabling the effects on Monitor 3/4 yields this: ffff8800662a6840 3034928368 S Co:1:006:0 s 21 01 0203 0700 0002 2 = e7cd ffff8800662a6840 3034928788 C Co:1:006:0 0 2 > ffff8800662a6840 3034936142 S Co:1:006:0 s 21 01 0204 0700 0002 2 = e7cd ffff8800662a6840 3034936540 C Co:1:006:0 0 2 > 2.2 Levels ---------- Changing a level for a channel yields this: ffff880093800e00 3649051254 S Co:1:003:0 s 21 01 [OutIn] 0500 0002 2 = [VAL] ffff880093800e00 3649051527 C Co:1:003:0 0 2 > where [VAL] ranges from 0080 to ffff and [OutIn] is a number composed of the Out-channel (01-08) and In ranges from 01 to 10 (Hex, Software returns starting at 09). Example: [OutIn] = 0209 for the Level of "Software Return 1" on Out 2. 2.2.1 Panning ------------- Panning is done by changing the respective Levels for 2 channels. When panning is in the middle position, the max value for each level seems to be "fdfc". Example: ffff8800ac1edd80 1908059496 S Co:1:003:0 s 21 01 0402 0500 0002 2 = fdfc ffff8800ac1edd80 1908059852 C Co:1:003:0 0 2 > ffff8800ac1edd80 1908064495 S Co:1:003:0 s 21 01 0302 0500 0002 2 = fdfc ffff8800ac1edd80 1908064850 C Co:1:003:0 0 2 > This sets the level of "Analog In 2" in "Monitor 3/4" to the maximum value while the panning control is in the middle position. 2.3 Solo and Mute ----------------- Solo and Mute are operated by changing the respective levels to minimum value. In other words: Not a seperate action from changing levels. 2.4 Effects Send ---------------- Changing the Effects send value for a Channel ffff8800ac1ed840 958480452 S Co:1:[MON]:0 s 21 01 09[CH] 0500 0002 2 = [VAL] ffff8800ac1ed840 958480800 C Co:1:[MON]:0 0 2 > where [VAL] ranges from e7cd (knob left, minimum) to daff (knob right, maximum), [MON] = 003 for Monitor 1/2 and 006 for Monitor 3/4 and [CH] is the channel number (0-10, Hex, Software returns starting at 09) 2.5 Effects Return ------------------ 2.5.1 Monitor 1/2 ----------------- Setting the "Effects Return 1" ('left') Value in Monitor 1/2 yields ffff8800608fe9c0 1220780329 S Co:1:003:0 s 21 01 0201 0700 0002 2 = [VAL] ffff8800608fe9c0 1220780648 C Co:1:003:0 0 2 > with [VAL] from e7cd to daff. For "Effect Return 2" ('right') it's ffff880093800080 1281633920 S Co:1:003:0 s 21 01 0202 0700 0002 2 = [VAL] ffff880093800080 1281634358 C Co:1:003:0 0 2 > 2.5.2 Monitor 3/4 ----------------- Setting the "Effects Return 1" ('left') Value in Monitor 3/4 yields ffff8800ac1ed240 45150924 S Co:1:006:0 s 21 01 0203 0700 0002 2 = [VAL] ffff8800ac1ed240 45151245 C Co:1:006:0 0 2 > with [VAL] from e7cd to daff. For "Effect Return 2" ('right') it's ffff8800a00d0480 31181802 S Co:1:006:0 s 21 01 0204 0700 0002 2 = [VAL] ffff8800a00d0480 31182181 C Co:1:006:0 0 2 > 2.6 Master Out -------------- 2.6.1 Monitor 1/2 ----------------- Changing the "Master Out 1" ('left') level in Monitor 1/2 gives this ffff880093800440 1385027420 S Co:1:003:0 s 21 01 0109 0500 0002 2 = [VAL] ffff880093800440 1385027770 C Co:1:003:0 0 2 > changing the "Master Out 2" ('right') level in Monitor 1/2 gives this ffff8800ac1ede40 1410769402 S Co:1:003:0 s 21 01 020a 0500 0002 2 = [VAL] ffff8800ac1ede40 1410769744 C Co:1:003:0 0 2 > with [VAL] ranging from e7cd to 0000 (loudest). 2.6.2 Monitor 3/4 ----------------- Changing the "Master Out 1" ('left') level in Monitor 3/4 gives this ffff8800662a6540 4152434840 S Co:1:006:0 s 21 01 030b 0500 0002 2 = [VAL] ffff8800662a6540 4152435226 C Co:1:006:0 0 2 > changing the "Master Out 2" ('right') level in Monitor 3/4 gives this ffff88008280ca40 14381909 S Co:1:006:0 s 21 01 040c 0500 0002 2 = [VAL] ffff88008280ca40 14382274 C Co:1:006:0 0 2 > with [VAL] ranging from e7cd to 0000 (loudest). 3. Startup ---------- I've attached a log file showing what happens when the device is attached to Windows. Please note, that this is done under VirtualBox, so there might be some noise at the beginning of the file. I hope, there's some information in the log on how the Windows control panel gathers the current mixer values from the device.

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[alsa-devel] Getting pcm_usb_stream plugin to know its limits.
by michael＠cadilhac.name 22 Feb '10

22 Feb '10

Hi there everyone, I've been fighting for a few hours with a USB sound device (namely, Tascam US-144), to get it to work with Audacity (this not being the relevant part). At first, Audacity died in a painful segfault, without any explanation, when associated with the usb_stream plugin[1]. Audacity uses PortAudio, which checks for a few possible parameters of audio rate to find the list of accepted ones. It tries to set a parameter (snd_pcm_hw_params), and if it fails, closes the device, and continues its way. Suppose the `prepare` act of the usb_stream is aborted because of wrong parameters. Then, PortAudio tries to close the device, by calling snd_pcm_close. This in turn calls snd_pcm_drop, which calls the fast_op drop (that of ioplug), which is snd_pcm_ioplug_drop, which then calls the *stop* callback of the usb_stream (snd_pcm_us_stop). The stream not being properly initialized, us->uus->write_area is nil, and the memset fails. This is the modification I did for that purpose : Next, Audacity stopped segfaulting but still didn't work: it couldn't manage to configure the plugin properly. Here's how PortAudio searches for an acceptable audio rate. For each possible sample rate, it gets a full hw_params for the device: snd_pcm_hw_params_any( pcm, hwParams ); then set the sample rate to the hwParams. It then tries to set those params to the device: snd_pcm_hw_params( pcm, hwParams ); The period_size parameter is left untouched, and by default, the pcm inherits of a high value (maybe the highest). Now the problem is the following. pcm_usb_stream says that the maximum accepted period_size is 64*4096: (err = snd_pcm_ioplug_set_param_minmax(&us->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, 128, 64*4096) BUT the us122l driver[2] (from 2.6.32) explicitly prohibits period_size greater than 0x3000: if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 && (!high_speed || (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) || cfg->frame_size != 6 || cfg->period_frames > 0x3000) err = -EINVAL; This is now what happens in PortAudio: PortAudio tries each and every sound rate with the "default" period_size, this being greater than 0x3000. It follows that no sound rate will be accepted, because of the period_size, and the device fails to initialize. I'm not sure whose fault is this, and for the time being, I just did the following: Well, that's all folks. Happy new year to everyone. Footnotes: [1] Configured as pcm.!usb_stream { type usb_stream card "1" } which works with a fine-tuned aplay. [2] linux/sound/usb/usx2y/us122l.c -- Michaël `Micha' Cadilhac (LITQ, U. de Montréal) -- http://michael.cadilhac.name || La culture, c'est comme l'amour. || Faut y aller à petits coups au début pour bien en jouir plus tard. || -- Pierre Desproges

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[alsa-devel] [PATCH] ALSA: usbaudio Mbox support, output only
by Chris J Arges 22 Feb '10

22 Feb '10

This minimal patch has been tested on my laptop and currently output only works. This patch is based off of code that Clemens Ladisch posted on the alsa-devel mailing list way back in 2004. --chris j arges

2 1

[alsa-devel] Latency of mixer reconfiguration
by Lennart Poettering 22 Feb '10

22 Feb '10

Heya! So, as some of you might know in PA we extend the abilities of the hardware volume control in software. Meaning that if a hw mixer control only allows a limited range or granularity of a volume scale or does not distuingish between channels then PA will configure the hardware to the next higher available setting of the real volume it wants to achieve, and then attenuates the PCM stream to compensate for the rest. This actually works quite well but there is a certain probelm with latency: i.e. the mixer setting will be applied at a slightly different time than the PCM attenuation. Now the question I have is, how should I best deal with this? I currently assume that mixer volume changes are basically instantaneous when I call the respective function of ALSA. But are they really? How big is the latency at max? Do we need an API to query it? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] [PATCH] Typo. s/distrubs/disturbs/
by Paul Menzel 22 Feb '10

22 Feb '10

Signed-off-by: Paul Menzel <paulepanter(a)users.sourceforge.net> --- sound/pci/hda/hda_intel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 06f230f..051cf51 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1411,7 +1411,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) chip->codec_mask &= ~(1 << c); /* More badly, accessing to a non-existing * codec often screws up the controller chip, - * and distrubs the further communications. + * and disturbs the further communications. * Thus if an error occurs during probing, * better to reset the controller chip to * get back to the sanity state. -- 1.6.6.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Macmini 3,1 support
by Luke Yelavich 22 Feb '10

22 Feb '10

From: Luke Yelavich <luke.yelavich(a)canonical.com> BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989 Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The pinout is almost identical to the mb5 quirk, except for no microphone and the line-in mixer controls being on a different index. Everything works in 2ch mode, but as I am not sure what needs to be changed for 6ch mode, or whether the Mac Mini's chip supports 6ch mode, I have simply duplicated the code from the mb5 quirk for the mac mini chmode management. The new model parameter for this quirk is "macmini3". Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 136 ++++++++++++++++++++++++++ 2 files changed, 137 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 8f06f20..0bbdbb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -124,6 +124,7 @@ ALC882/883/885/888/889 asus-a7m ASUS A7M macpro MacPro support mb5 Macbook 5,1 + macmini3 Macmini 3,1 mbp3 Macbook Pro rev3 imac24 iMac 24'' with jack detection imac91 iMac 9,1 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0c22497..b5a6ba0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -211,6 +211,7 @@ enum { ALC885_MACPRO, ALC885_MBP3, ALC885_MB5, + ALC885_MACMINI3, ALC885_IMAC24, ALC885_IMAC91, ALC883_3ST_2ch_DIG, @@ -6751,6 +6752,14 @@ static struct hda_input_mux mb5_capture_source = { }, }; +static struct hda_input_mux macmini3_capture_source = { + .num_items = 2, + .items = { + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc883_3stack_6ch_intel = { .num_items = 4, .items = { @@ -6999,6 +7008,35 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = { { 6, alc885_mb5_ch6_init }, }; +/* + * 2ch + * Speakers/Woofer/HP = Front + * LineIn = Input + */ +static struct hda_verb alc885_macmini3_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } /* end */ +}; + +/* + * 6ch mode + * Speakers/HP = Front + * Woofer = LFE + * LineIn = Surround + */ +static struct hda_verb alc885_macmini3_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + { } /* end */ +}; + +static struct hda_channel_mode alc885_macmini3_6ch_modes[2] = { + { 2, alc885_mb5_ch2_init }, + { 6, alc885_mb5_ch6_init }, +}; + /* * 2ch mode @@ -7243,6 +7281,21 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc885_macmini3_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7617,6 +7670,53 @@ static struct hda_verb alc885_mb5_init_verbs[] = { { } }; +/* Macmini 3,1 */ +static struct hda_verb alc885_macmini3_init_verbs[] = { + /* DACs */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Surround mixer */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* LFE mixer */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* HP mixer */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Front Pin (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* LFE Pin (0x0e) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* HP Pin (0x0f) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Line In pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + { } +}; + /* Macbook Pro rev3 */ static struct hda_verb alc885_mbp3_init_verbs[] = { /* Front mixer: unmute input/output amp left and right (volume = 0) */ @@ -7800,6 +7900,18 @@ static void alc885_mb5_automute(struct hda_codec *codec) } +static void alc885_macmini3_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + static void alc885_mb5_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -7808,6 +7920,14 @@ static void alc885_mb5_unsol_event(struct hda_codec *codec, alc885_mb5_automute(codec); } +static void alc885_macmini3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -8974,6 +9094,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ASUS_A7M] = "asus-a7m", [ALC885_MACPRO] = "macpro", [ALC885_MB5] = "mb5", + [ALC885_MACMINI3] = "macmini3", [ALC885_MBP3] = "mbp3", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", @@ -9157,6 +9278,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { */ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), {} /* terminator */ }; @@ -9238,6 +9360,20 @@ static struct alc_config_preset alc882_presets[] = { .unsol_event = alc885_mb5_unsol_event, .init_hook = alc885_mb5_automute, }, + [ALC885_MACMINI3] = { + .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, + .init_verbs = { alc885_macmini3_init_verbs, + alc880_gpio1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .channel_mode = alc885_macmini3_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), + .input_mux = &macmini3_capture_source, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_macmini3_unsol_event, + .init_hook = alc885_macmini3_automute, + }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, .init_verbs = { alc882_macpro_init_verbs }, -- 1.6.6.1

2 1

[alsa-devel] [RESEND] [PATCH] ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10Q
by Daniel Chen 22 Feb '10

22 Feb '10

Attached the actual patch this time. -Dan

2 1

[alsa-devel] [PATCH 3/3] ASOC: SND_SOC_DAI: Get rid of 'runtime' field
by jassi brar 22 Feb '10

22 Feb '10

[PATCH 3/3] ASOC: SND_SOC_DAI: Get rid of 'runtime' field In order for having snd_soc_dai's shared among two or more dai_links, remove the relatively global 'runtime' field from the 'struct snd_soc_dai' Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- include/sound/soc-dai.h | 1 - sound/soc/soc-core.c | 2 -- 2 files changed, 0 insertions(+), 3 deletions(-)

1 0

[alsa-devel] [PATCH 2/3] ASOC: ATMEL: Getting cpu_dai method change
by jassi brar 22 Feb '10

22 Feb '10

[PATCH 2/3] ASOC: ATMEL: Getting cpu_dai method change Currently only the atmel driver make use of snd_soc_dai.runtime field. If the dais are to be shared among two or more dai_links, the field must be got rid of. So, in atmel driver reach the substream via dai_link->pcm so as to not depend of snd_soc_dai.runtime field. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/atmel/atmel-pcm.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-)

1 0

[alsa-devel] [PATCH 1/3] ASoC: PLATFORM: Pass dai_link as argument to suspend, resume
by jassi brar 22 Feb '10

22 Feb '10

[PATCH 1/3] ASoC: PLATFORM: Pass dai_link as argument to suspend, resume Passing pointer to relevant dai_link provides easier reach to the ASOC tree in suspend/resume of snd_soc_platform. It also provides direct access to the dai at the other end of the dai_link. Also, modified ATMEL pcm suspend/resume callbacks for the new API. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- include/sound/soc.h | 5 +++-- sound/soc/atmel/atmel-pcm.c | 6 ++++-- sound/soc/soc-core.c | 4 ++-- 3 files changed, 9 insertions(+), 6 deletions(-)

1 0

[alsa-devel] Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG (SB0820)
by dbm 21 Feb '10

21 Feb '10

Greetings, Just bought one of these to get around the fact the mobo involved wouldn't recognize any soundcard plugged into either PCI slots (onboard sound disabled in BIOS) -- it will however see a soundcard plugged into the pcie 1x slot..... 03:00.0 PCI bridge: Creative Labs Device 7006 (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=03, secondary=04, subordinate=04, sec-latency=32 Memory behind bridge: ee000000-ee0fffff Capabilities: [50] Power Management version 3 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/4 Enable- Capabilities: [80] Subsystem: Creative Labs Device 0010 Capabilities: [90] Express PCI/PCI-X Bridge, MSI 00 Capabilities: [100] Advanced Error Reporting <?> Kernel modules: shpchp 04:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Flags: bus master, medium devsel, latency 32, IRQ 16 Memory at ee000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel Does seemed to be claimed by snd-hda-intel (vid/pid 1102/0009), but subsystem descriptor of 1102/0018 is seemingly unknown by the drivers. Wild guess here is this should be claimed by ctxfi.ko? I can provide a list of hardware IC's fitted to the card -- most of them are *not* codec chips afaict.....that said, 2 oncard chips I cannot readily identify. My suspicion is it's like other cards covered by ctxfi.ko in that the card has no ac97-codec fitted and is just a plain ole' sblive type card....but this is only speculation... Anyone seens one of these, or have any idea where to start poking for clues (if any) to get it working? Cheers!

1 0

[alsa-devel] [PATCH] ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10Q
by Daniel Chen 20 Feb '10

20 Feb '10

Hi, Attached please find a patch correcting the quirk for Toshiba Satellite L40-10Q. Best, -Dan

1 0

[alsa-devel] [PATCH v3 2/3] ASoC: OMAP4: Add support for McPDM
by Candelaria Villareal, Jorge 20 Feb '10

20 Feb '10

McPDM is the interface between Phoenix audio codec and the OMAP4430 processor. It enables data to be transfered to/from Phoenix at sample rates of 88.4 or 96 KHz. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> --- sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 490 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++++ 4 files changed, 646 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 61952aa..b96e9d8 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -6,6 +6,9 @@ config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP +config SND_OMAP_SOC_MCPDM + tristate + config SND_OMAP_SOC_N810 tristate "SoC Audio support for Nokia N810" depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3db8a6c..bf8b388 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,9 +1,11 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcpdm-objs := mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o +obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o # OMAP Machine Support snd-soc-n810-objs := n810.o diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c new file mode 100644 index 0000000..f418cac --- /dev/null +++ b/sound/soc/omap/mcpdm.c @@ -0,0 +1,490 @@ +/* + * mcpdm.c -- McPDM interface driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Copyright (C) 2009 - Texas Instruments, Inc. + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/wait.h> +#include <linux/interrupt.h> +#include <linux/err.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/irq.h> + +#include "mcpdm.h" + +static struct omap_mcpdm *mcpdm; + +static inline void omap_mcpdm_write(u16 reg, u32 val) +{ + __raw_writel(val, mcpdm->io_base + reg); +} + +static inline int omap_mcpdm_read(u16 reg) +{ + return __raw_readl(mcpdm->io_base + reg); +} + +static void omap_mcpdm_reg_dump(void) +{ + dev_dbg(mcpdm->dev, "***********************\n"); + dev_dbg(mcpdm->dev, "IRQSTATUS_RAW: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS_RAW)); + dev_dbg(mcpdm->dev, "IRQSTATUS: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQSTATUS)); + dev_dbg(mcpdm->dev, "IRQENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_SET)); + dev_dbg(mcpdm->dev, "IRQENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQENABLE_CLR)); + dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n", + omap_mcpdm_read(MCPDM_IRQWAKE_EN)); + dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_SET)); + dev_dbg(mcpdm->dev, "DMAENABLE_CLR: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAENABLE_CLR)); + dev_dbg(mcpdm->dev, "DMAWAKEEN: 0x%04x\n", + omap_mcpdm_read(MCPDM_DMAWAKEEN)); + dev_dbg(mcpdm->dev, "CTRL: 0x%04x\n", + omap_mcpdm_read(MCPDM_CTRL)); + dev_dbg(mcpdm->dev, "DN_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_DATA)); + dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n", + omap_mcpdm_read(MCPDM_UP_DATA)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_DN)); + dev_dbg(mcpdm->dev, "FIFO_CTRL_UP: 0x%04x\n", + omap_mcpdm_read(MCPDM_FIFO_CTRL_UP)); + dev_dbg(mcpdm->dev, "DN_OFFSET: 0x%04x\n", + omap_mcpdm_read(MCPDM_DN_OFFSET)); + dev_dbg(mcpdm->dev, "***********************\n"); +} + +/* + * Takes the McPDM module in and out of reset state. + * Uplink and downlink can be reset individually. + */ +static void omap_mcpdm_reset_capture(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +static void omap_mcpdm_reset_playback(int reset) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (links & MCPDM_DOWNLINK) { + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; + } + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Enables the transfer through the PDM interface to/from the Phoenix + * codec by enabling the corresponding UP or DN channels. + */ +void omap_mcpdm_start(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Disables the transfer through the PDM interface to/from the Phoenix + * codec by disabling the corresponding UP or DN channels. + */ +void omap_mcpdm_stop(int stream) +{ + int ctrl = omap_mcpdm_read(MCPDM_CTRL); + + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; + + omap_mcpdm_write(MCPDM_CTRL, ctrl); +} + +/* + * Configures McPDM uplink for audio recording. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + int ctrl; + + if (!uplink) + return -EINVAL; + + mcpdm->uplink = uplink; + + /* Enable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + + /* Configure DMA controller */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= uplink->format & PDMOUTFORMAT; + + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Configures McPDM downlink for audio playback. + * This function should be called before omap_mcpdm_start. + */ +int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + int ctrl; + + if (!downlink) + return -EINVAL; + + mcpdm->downlink = downlink; + + /* Enable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; + + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + + /* Enable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + + /* Set pdm out format */ + ctrl = omap_mcpdm_read(MCPDM_CTRL); + ctrl &= ~PDMOUTFORMAT; + ctrl |= downlink->format & PDMOUTFORMAT; + + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); + + omap_mcpdm_write(MCPDM_CTRL, ctrl); + + return 0; +} + +/* + * Cleans McPDM uplink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) +{ + int irq_mask = 0; + + if (!uplink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); + + /* Clear Downlink channels */ + mcpdm->up_channels = 0; + + mcpdm->uplink = NULL; + + return 0; +} + +/* + * Cleans McPDM downlink configuration. + * This function should be called when the stream is closed. + */ +int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) +{ + int irq_mask = 0; + + if (!downlink) + return -EINVAL; + + /* Disable irq request generation */ + irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; + omap_mcpdm_write(MCPDM_IRQENABLE_CLR, irq_mask); + + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); + + /* clear Downlink channels */ + mcpdm->dn_channels = 0; + + mcpdm->downlink = NULL; + + return 0; +} + +static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id) +{ + struct omap_mcpdm *mcpdm_irq = dev_id; + int irq_status; + + irq_status = omap_mcpdm_read(MCPDM_IRQSTATUS); + + /* Acknowledge irq event */ + omap_mcpdm_write(MCPDM_IRQSTATUS, irq_status); + + if (irq & MCPDM_DN_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "DN FIFO error %x\n", irq_status); + omap_mcpdm_reset_playback(1); + omap_mcpdm_playback_open(mcpdm_irq->downlink); + omap_mcpdm_reset_playback(0); + } + + if (irq & MCPDM_DN_IRQ) { + dev_dbg(mcpdm_irq->dev, "DN write request\n"); + } + + if (irq & MCPDM_UP_IRQ_FULL) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ_EMPTY) { + dev_err(mcpdm_irq->dev, "UP FIFO error %x\n", irq_status); + omap_mcpdm_reset_capture(1); + omap_mcpdm_capture_open(mcpdm_irq->uplink); + omap_mcpdm_reset_capture(0); + } + + if (irq & MCPDM_UP_IRQ) { + dev_dbg(mcpdm_irq->dev, "UP write request\n"); + } + + return IRQ_HANDLED; +} + +int omap_mcpdm_request(void) +{ + int ret; + + clk_enable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + + if (!mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is in use\n"); + spin_unlock(&mcpdm->lock); + return -EBUSY; + } + mcpdm->free = 0; + + spin_unlock(&mcpdm->lock); + + /* Disable lines while request is ongoing */ + omap_mcpdm_write(MCPDM_CTRL, 0x00); + + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, + 0, "McPDM", (void *)mcpdm); + if (ret) { + dev_err(mcpdm->dev, "Request for McPDM IRQ failed\n"); + return ret; + } + + return 0; +} + +void omap_mcpdm_free(void) +{ + clk_disable(mcpdm->clk); + + spin_lock(&mcpdm->lock); + if (mcpdm->free) { + dev_err(mcpdm->dev, "McPDM interface is already free\n"); + spin_unlock(&mcpdm->lock); + return; + } + mcpdm->free = 1; + spin_unlock(&mcpdm->lock); + + free_irq(mcpdm->irq, (void *)mcpdm); +} + +/* Enable/disable DC offset cancelation for the analog + * headset path (PDM channels 1 and 2). + */ +int omap_mcpdm_set_offset(int offset1, int offset2) +{ + int offset; + + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; + + offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + + /* offset cancellation for channel 1 */ + if (offset1) + offset |= DN_OFST_RX1_EN; + else + offset &= ~DN_OFST_RX1_EN; + + /* offset cancellation for channel 2 */ + if (offset2) + offset |= DN_OFST_RX2_EN; + else + offset &= ~DN_OFST_RX2_EN; + + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); + + return 0; +} + +static int __devinit omap_mcpdm_probe(struct platform_device *pdev) +{ + struct resource *res; + int ret = 0; + + mcpdm = kzalloc(sizeof(struct omap_mcpdm), GFP_KERNEL); + if (!mcpdm) { + ret = -ENOMEM; + goto exit; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) { + dev_err(&pdev->dev, "no resource\n"); + goto err_resource; + } + + spin_lock_init(&mcpdm->lock); + mcpdm->free = 1; + mcpdm->io_base = ioremap(res->start, resource_size(res)); + if (!mcpdm->io_base) { + ret = -ENOMEM; + goto err_resource; + } + + mcpdm->irq = platform_get_irq(pdev, 0); + + mcpdm->clk = clk_get(&pdev->dev, "pdm_ck"); + if (IS_ERR(mcpdm->clk)) { + ret = PTR_ERR(mcpdm->clk); + dev_err(&pdev->dev, "unable to get pdm_ck: %d\n", ret); + goto err_clk; + } + + mcpdm->dev = &pdev->dev; + platform_set_drvdata(pdev, mcpdm); + + return 0; + +err_clk: + iounmap(mcpdm->io_base); +err_resource: + kfree(mcpdm); +exit: + return ret; +} + +static int __devexit omap_mcpdm_remove(struct platform_device *pdev) +{ + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + + clk_disable(mcpdm_ptr->clk); + clk_put(mcpdm_ptr->clk); + + iounmap(mcpdm_ptr->io_base); + + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; + + kfree(mcpdm_ptr); + + return 0; +} + +static struct platform_driver omap_mcpdm_driver = { + .probe = omap_mcpdm_probe, + .remove = __devexit_p(omap_mcpdm_remove), + .driver = { + .name = "omap-mcpdm", + }, +}; + +static struct platform_device *omap_mcpdm_device; + +static int __init omap_mcpdm_init(void) +{ + int ret; + + ret = platform_driver_register(&omap_mcpdm_driver); + if (ret) + goto error; + return 0; + +error: + return ret; +} +arch_initcall(omap_mcpdm_init); diff --git a/sound/soc/omap/mcpdm.h b/sound/soc/omap/mcpdm.h new file mode 100644 index 0000000..41e50cc --- /dev/null +++ b/sound/soc/omap/mcpdm.h @@ -0,0 +1,151 @@ +/* + * mcpdm.h -- Defines for McPDM driver + * + * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +/* McPDM registers */ + +#define MCPDM_REVISION 0x00 +#define MCPDM_SYSCONFIG 0x10 +#define MCPDM_IRQSTATUS_RAW 0x24 +#define MCPDM_IRQSTATUS 0x28 +#define MCPDM_IRQENABLE_SET 0x2C +#define MCPDM_IRQENABLE_CLR 0x30 +#define MCPDM_IRQWAKE_EN 0x34 +#define MCPDM_DMAENABLE_SET 0x38 +#define MCPDM_DMAENABLE_CLR 0x3C +#define MCPDM_DMAWAKEEN 0x40 +#define MCPDM_CTRL 0x44 +#define MCPDM_DN_DATA 0x48 +#define MCPDM_UP_DATA 0x4C +#define MCPDM_FIFO_CTRL_DN 0x50 +#define MCPDM_FIFO_CTRL_UP 0x54 +#define MCPDM_DN_OFFSET 0x58 + +/* + * MCPDM_IRQ bit fields + * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR + */ + +#define MCPDM_DN_IRQ (1 << 0) +#define MCPDM_DN_IRQ_EMPTY (1 << 1) +#define MCPDM_DN_IRQ_ALMST_EMPTY (1 << 2) +#define MCPDM_DN_IRQ_FULL (1 << 3) + +#define MCPDM_UP_IRQ (1 << 8) +#define MCPDM_UP_IRQ_EMPTY (1 << 9) +#define MCPDM_UP_IRQ_ALMST_FULL (1 << 10) +#define MCPDM_UP_IRQ_FULL (1 << 11) + +#define MCPDM_DOWNLINK_IRQ_MASK 0x00F +#define MCPDM_UPLINK_IRQ_MASK 0xF00 + +/* + * MCPDM_DMAENABLE bit fields + */ + +#define DMA_DN_ENABLE 0x1 +#define DMA_UP_ENABLE 0x2 + +/* + * MCPDM_CTRL bit fields + */ + +#define PDM_UP1_EN 0x0001 +#define PDM_UP2_EN 0x0002 +#define PDM_UP3_EN 0x0004 +#define PDM_DN1_EN 0x0008 +#define PDM_DN2_EN 0x0010 +#define PDM_DN3_EN 0x0020 +#define PDM_DN4_EN 0x0040 +#define PDM_DN5_EN 0x0080 +#define PDMOUTFORMAT 0x0100 +#define CMD_INT 0x0200 +#define STATUS_INT 0x0400 +#define SW_UP_RST 0x0800 +#define SW_DN_RST 0x1000 +#define PDM_UP_MASK 0x007 +#define PDM_DN_MASK 0x0F8 +#define PDM_CMD_MASK 0x200 +#define PDM_STATUS_MASK 0x400 + + +#define PDMOUTFORMAT_LJUST (0 << 8) +#define PDMOUTFORMAT_RJUST (1 << 8) + +/* + * MCPDM_FIFO_CTRL bit fields + */ + +#define UP_THRES_MAX 0xF +#define DN_THRES_MAX 0xF + +/* + * MCPDM_DN_OFFSET bit fields + */ + +#define DN_OFST_RX1_EN 0x0001 +#define DN_OFST_RX2_EN 0x0100 + +#define DN_OFST_RX1 1 +#define DN_OFST_RX2 9 +#define DN_OFST_MAX 0x1F + +#define MCPDM_UPLINK 1 +#define MCPDM_DOWNLINK 2 + +struct omap_mcpdm_link { + int irq_mask; + int threshold; + int format; + int channels; +}; + +struct omap_mcpdm_platform_data { + unsigned long phys_base; + u16 irq; +}; + +struct omap_mcpdm { + struct device *dev; + unsigned long phys_base; + void __iomem *io_base; + u8 free; + int irq; + + spinlock_t lock; + struct omap_mcpdm_platform_data *pdata; + struct clk *clk; + struct omap_mcpdm_link *downlink; + struct omap_mcpdm_link *uplink; + struct completion irq_completion; + + int dn_channels; + int up_channels; +}; + +extern void omap_mcpdm_start(int stream); +extern void omap_mcpdm_stop(int stream); +extern int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink); +extern int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink); +extern int omap_mcpdm_request(void); +extern void omap_mcpdm_free(void); +extern int omap_mcpdm_set_offset(int offset1, int offset2); -- 1.6.0.4

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Re: [alsa-devel] [patch] oxygen: clean up. make precedence explicit
by Clemens Ladisch 19 Feb '10

19 Feb '10

Bernd Petrovitsch wrote: > On Fre, 2010-02-19 at 14:29 +0300, Dan Carpenter wrote: > > On Fri, Feb 19, 2010 at 11:33:30AM +0100, Bernd Petrovitsch wrote: > > Basically often when people write: > > if (!foo == bar) { ... > > > > What they mean is: > > if (!(foo == bar)) { ... But there are also cases where they mean what they've written. > Ugh. The IMHO better way is > if (foo != bar) { ... In my case, the driver compares an "enabled" variable against a "disabled" one; negating the comparison operator would obfuscate the logic. > > But if they really do mean the original code they could just write > > this so it's clear to everyone: > > if ((!foo) == bar) { ... This is unnatural (especially in a simple example like this) because the parens haven't been needed at all before smatch. !foo==bar is always identical to !(foo==bar) for boolean values; to avoid false positives, you could output the warning only when the code is trying to manipulate non-boolean values. IMO the message would be justified if it said "using suspicious boolean operations on non-boolean types". (In fact, my driver uses types long and u8 in this expression, so I will clean it up.) Regards, Clemens

2 1

[alsa-devel] bug report: using snd_BUG_ON() instead of WARN_ON()
by Dan Carpenter 19 Feb '10

19 Feb '10

Smatch found a couple places try use the return value for snd_BUG_ON() without realizing it's always zero or that it can be defined away entirely under certain configs. sound/core/info_oss.c 46 if (snd_BUG_ON(dev < 0 || dev >= SNDRV_OSS_INFO_DEV_COUNT)) 47 return -ENXIO; 48 if (snd_BUG_ON(num < 0 || num >= SNDRV_CARDS)) 49 return -ENXIO; sound/drivers/opl3/opl3_midi.c +652 snd_opl3_kill_voice(34) warn: buffer overflow 'opl3->voices' 18 <= 20 opl3_midi.c checks the range with snd_BUG_ON() and then adds 3 so it possibly goes out of bounds. I'm not sure the situation there. sound/core/seq/seq_midi.c +403 snd_seq_midisynth_register_port(126) error: buffer overflow 'client->ports_per_device' 8 <= 8 sound/core/seq/seq_midi.c +404 snd_seq_midisynth_register_port(127) error: buffer overflow 'client->ports' 8 <= 8 sound/core/info_oss.c +52 snd_oss_info_register(10) error: buffer overflow 'snd_sndstat_strings[num]' 6 <= 6 sound/core/info_oss.c +52 snd_oss_info_register(10) error: buffer overflow 'snd_sndstat_strings' 32 <= 32 sound/core/info_oss.c +63 snd_oss_info_register(21) error: buffer overflow 'snd_sndstat_strings[num]' 6 <= 6 sound/core/info_oss.c +63 snd_oss_info_register(21) error: buffer overflow 'snd_sndstat_strings' 32 <= 32 sound/pci/cs46xx/dsp_spos_scb_lib.c +1497 cs46xx_dsp_destroy_pcm_channel(28) error: buffer overflow 'ins->src_scb_slots' 14 <= 14 regards, dan carpenter

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Re: [alsa-devel] [patch] oxygen: clean up. make precedence explicit
by Dan Carpenter 19 Feb '10

19 Feb '10

On Fri, Feb 19, 2010 at 11:33:30AM +0100, Bernd Petrovitsch wrote: > On Fre, 2010-02-19 at 13:10 +0300, Dan Carpenter wrote: > > On Fri, Feb 19, 2010 at 09:29:05AM +0100, Clemens Ladisch wrote: > > > > This doesn't change anything, but I think it makes the code clearer. > > > > It silences a smatch warning: > > > > sound/pci/oxygen/oxygen_mixer.c +91 dac_mute_put(7) warn: add some parenthesis here? > > > > > > That message doesn't say why some parentheses should be added. > > > And it's a question; how do I give it the answer "no"? :-) > > > > > > > - changed = !value->value.integer.value[0] != chip->dac_mute; > > > > + changed = (!value->value.integer.value[0]) != chip->dac_mute; > > > > > > This doesn't look any clearer to me; I don't think that the unary > > > negation operator could be thought to have lower precedence than "!=". > > > > Well, it's hard to argue that it's more ambiguous. :P > But it doesn't make the code clearer - unless you are a C novice. Unary > operators generally bind stronger than others - be it "+", "-", "!", > "~", "*". > I would expect kernel programmers to know that (and I don't assume > in-depth knowledge of operator precedence rules). > > > > Why does smatch warn about this combination? Do such errors actually > > > happen: > > > > Yep. I have made some myself when writing smatch. > > > > For example here are some related bugs in the current kernel. > > > > drivers/staging/rtl8192u/ieee80211/ieee80211_wx.c > > 721 if (!ext->ext_flags & IW_ENCODE_EXT_GROUP_KEY && > Well, I see potential bugs here and the if() should have been > a) if (!(ext->ext_flags & IW_ENCODE_EXT_GROUP_KEY) && Yep. This is clearly what the code should say. The problem in the original code is that IW_ENCODE_EXT_GROUP_KEY is not equal to either 1 or to 0. (So that means the condition in the original code is always false). > b) if (!ext->ext_flags && IW_ENCODE_EXT_GROUP_KEY && > So you one has to look at the driver for the correct fix (and perhaps > both of above are wrong). > > And I don't see what parenthesis around a logical negations can help > with the above error example. > Basically often when people write: if (!foo == bar) { ... What they mean is: if (!(foo == bar)) { ... But if they really do mean the original code they could just write this so it's clear to everyone: if ((!foo) == bar) { ... To me it's like "==" vs "=". Of course, every programmer knows the what the difference is but it helps to have gcc warn about adding the extra parenthesis. Maybe I suck, but it definitely has helped me in then past. I don't have strong feelings about this btw. The original code in oxygyn_mixer works fine. I just was making some changes to smatch and that was a new warning today. There is no method to my madness. regards, dan carpenter > Bernd > -- > Bernd Petrovitsch Email : bernd(a)petrovitsch.priv.at > LUGA : http://www.luga.at

1 1

[alsa-devel] alsamixer crashes with SIGFPE
by Florian Faber 19 Feb '10

19 Feb '10

Hi, there are several spots in alsamixer's mixer_display.c that cause SIGFPE through division by zero if (max-min) is zero: bar_height = ((volumes[c] - min) * volume_height + max - min - 1) / (max - min); value = ((volumes[0] - min) * 100 + (max - min) / 2) / (max - min); etc. Flo -- Machines can do the work, so people have time to think. public key DA43FEF4 x-hkp://wwwkeys.eu.pgp.net

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[alsa-devel] [PATCH v3 1/3] ASoC: OMAP: data_type and sync_mode configurable in audio dma
by Candelaria Villareal, Jorge 19 Feb '10

19 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Allow client drivers to set the data_type (16, 32) and the sync_mode (element, packet, etc) of the audio dma transferences. McBSP dai driver configures it for a data type of 16 bits and element sync mode. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 ++ sound/soc/omap/omap-pcm.c | 15 ++++++++------- sound/soc/omap/omap-pcm.h | 4 +++- 3 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 6bbbd2a..d297256 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -287,6 +287,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; if (mcbsp_data->configured) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9db2770..825db38 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, @@ -149,6 +150,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) struct omap_runtime_data *prtd = runtime->private_data; struct omap_pcm_dma_data *dma_data = prtd->dma_data; struct omap_dma_channel_params dma_params; + int bytes; /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -156,11 +158,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) return 0; memset(&dma_params, 0, sizeof(dma_params)); - /* - * Note: Regardless of interface data formats supported by OMAP McBSP - * or EAC blocks, internal representation is always fixed 16-bit/sample - */ - dma_params.data_type = OMAP_DMA_DATA_TYPE_S16; + dma_params.data_type = dma_data->data_type; dma_params.trigger = dma_data->dma_req; dma_params.sync_mode = dma_data->sync_mode; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -170,6 +168,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = runtime->dma_addr; dma_params.dst_start = dma_data->port_addr; dma_params.dst_port = OMAP_DMA_PORT_MPUI; + dma_params.dst_fi = dma_data->packet_size; } else { dma_params.src_amode = OMAP_DMA_AMODE_CONSTANT; dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC; @@ -177,6 +176,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.src_start = dma_data->port_addr; dma_params.dst_start = runtime->dma_addr; dma_params.src_port = OMAP_DMA_PORT_MPUI; + dma_params.src_fi = dma_data->packet_size; } /* * Set DMA transfer frame size equal to ALSA period size and frame @@ -184,7 +184,8 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) * we can transfer the whole ALSA buffer with single DMA transfer but * still can get an interrupt at each period bounary */ - dma_params.elem_count = snd_pcm_lib_period_bytes(substream) / 2; + bytes = snd_pcm_lib_period_bytes(substream); + dma_params.elem_count = bytes >> dma_data->data_type; dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index 38a821d..b19975d 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -29,8 +29,10 @@ struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ unsigned long port_addr; /* transmit/receive register */ - int sync_mode; /* DMA sync mode */ void (*set_threshold)(struct snd_pcm_substream *substream); + int data_type; /* data type 8,16,32 */ + int sync_mode; /* DMA sync mode */ + int packet_size; /* packet size only in PACKET mode */ }; extern struct snd_soc_platform omap_soc_platform; -- 1.6.0.4

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[alsa-devel] Since kernel-2.6.32, alsactl fails to restore defaults
by Pacho Ramos 19 Feb '10

19 Feb '10

I originally reported this issue downstream to: http://bugs.gentoo.org/show_bug.cgi?id=299362 Later to kernel: http://bugzilla.kernel.org/show_bug.cgi?id=15107 And they told me that this is an alsactl issue, since it's not exiting with 0 when it notices that the device doesn't need special quirks: # alsactl -f /var/lib/alsa/asound.state restore Unknown hardware: "VIA8237" "Realtek ALC658D" "AC97a:414c4781" "0x147b" "0x1415" Hardware is initialized using a guess method # echo $? 157 What could I do? Thanks for helping me, if you need more information, simply ask me for it :-)

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[alsa-devel] [PATCH v3 3/3] ASoC: OMAP4: Add McPDM platform driver
by Candelaria Villareal, Jorge 19 Feb '10

19 Feb '10

From: Misael Lopez Cruz <x0052729(a)ti.com> McPDM platform driver is configured to use sDMA in order to transfer to/from memory. Support for interfacing with ABE will be added later. McPDM dai currently supports up to 4 downlink channels and 2 uplink channels simultaneously, as well as 88.2 and 96 KHz, and a sample size of 32 bits. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya <x0080101(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/omap-mcpdm.c | 251 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++++ 3 files changed, 281 insertions(+), 1 deletions(-) create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index bf8b388..d6382fa 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,7 +1,7 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-mcbsp-objs := omap-mcbsp.o -snd-soc-omap-mcpdm-objs := mcpdm.o +snd-soc-omap-mcpdm-objs := mcpdm.o omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c new file mode 100644 index 0000000..25f19e4 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.c @@ -0,0 +1,251 @@ +/* + * omap-mcpdm.c -- OMAP ALSA SoC DAI driver using McPDM port + * + * Copyright (C) 2009 Texas Instruments + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * Contact: Jorge Eduardo Candelaria <x0107209(a)ti.com> + * Margarita Olaya <magi.olaya(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/control.h> +#include <plat/dma.h> +#include <plat/mcbsp.h> +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" + +struct omap_mcpdm_data { + struct omap_mcpdm_link *links; + int active; +}; + +static struct omap_mcpdm_link omap_mcpdm_links[] = { + /* downlink */ + { + .irq_mask = MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, + /* uplink */ + { + .irq_mask = MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL, + .threshold = 1, + .format = PDMOUTFORMAT_LJUST, + }, +}; + +static struct omap_mcpdm_data mcpdm_data = { + .links = omap_mcpdm_links, + .active = 0, +}; + +/* + * Stream DMA parameters + */ +static struct omap_pcm_dma_data omap_mcpdm_dai_dma_params[] = { + { + .name = "Audio playback", + .dma_req = OMAP44XX_DMA_MCPDM_DL, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_DN_DATA, + }, + { + .name = "Audio capture", + .dma_req = OMAP44XX_DMA_MCPDM_UP, + .data_type = OMAP_DMA_DATA_TYPE_S32, + .sync_mode = OMAP_DMA_SYNC_PACKET, + .packet_size = 16, + .port_addr = OMAP44XX_MCPDM_L3_BASE + MCPDM_UP_DATA, + }, +}; + +static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int err = 0; + + if (!cpu_dai->active) + err = omap_mcpdm_request(); + + return err; +} + +static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (!cpu_dai->active) + omap_mcpdm_free(); +} + +static int omap_mcpdm_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + int stream = substream->stream; + int err = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!mcpdm_priv->active++) + omap_mcpdm_start(stream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!--mcpdm_priv->active) + omap_mcpdm_stop(stream); + break; + default: + err = -EINVAL; + } + + return err; +} + +static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int channels, err, link_mask = 0; + + cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + + channels = params_channels(params); + switch (channels) { + case 4: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 3; + case 3: + if (stream == SNDRV_PCM_STREAM_CAPTURE) + /* up to 2 channels for capture */ + return -EINVAL; + link_mask |= 1 << 2; + case 2: + link_mask |= 1 << 1; + case 1: + link_mask |= 1 << 0; + break; + default: + /* unsupported number of channels */ + return -EINVAL; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mcpdm_links[stream].channels = link_mask << 3; + err = omap_mcpdm_playback_open(&mcpdm_links[stream]); + } else { + mcpdm_links[stream].channels = link_mask << 0; + err = omap_mcpdm_capture_open(&mcpdm_links[stream]); + } + + return err; +} + +static int omap_mcpdm_dai_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcpdm_data *mcpdm_priv = cpu_dai->private_data; + struct omap_mcpdm_link *mcpdm_links = mcpdm_priv->links; + int stream = substream->stream; + int err; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + err = omap_mcpdm_playback_close(&mcpdm_links[stream]); + else + err = omap_mcpdm_capture_close(&mcpdm_links[stream]); + + return err; +} + +static struct snd_soc_dai_ops omap_mcpdm_dai_ops = { + .startup = omap_mcpdm_dai_startup, + .shutdown = omap_mcpdm_dai_shutdown, + .trigger = omap_mcpdm_dai_trigger, + .hw_params = omap_mcpdm_dai_hw_params, + .hw_free = omap_mcpdm_dai_hw_free, +}; + +#define OMAP_MCPDM_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define OMAP_MCPDM_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai omap_mcpdm_dai = { + .name = "omap-mcpdm", + .id = -1, + .playback = { + .channels_min = 1, + .channels_max = 4, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = OMAP_MCPDM_RATES, + .formats = OMAP_MCPDM_FORMATS, + }, + .ops = &omap_mcpdm_dai_ops, + .private_data = &mcpdm_data, +}; +EXPORT_SYMBOL_GPL(omap_mcpdm_dai); + +static int __init snd_omap_mcpdm_init(void) +{ + return snd_soc_register_dai(&omap_mcpdm_dai); +} +module_init(snd_omap_mcpdm_init); + +static void __exit snd_omap_mcpdm_exit(void) +{ + snd_soc_unregister_dai(&omap_mcpdm_dai); +} +module_exit(snd_omap_mcpdm_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("OMAP PDM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcpdm.h b/sound/soc/omap/omap-mcpdm.h new file mode 100644 index 0000000..73b80d5 --- /dev/null +++ b/sound/soc/omap/omap-mcpdm.h @@ -0,0 +1,29 @@ +/* + * omap-mcpdm.h + * + * Copyright (C) 2009 Texas Instruments + * + * Contact: Misael Lopez Cruz <x0052729(a)ti.com> + * + * 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. + * + * 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., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __OMAP_MCPDM_H__ +#define __OMAP_MCPDM_H__ + +extern struct snd_soc_dai omap_mcpdm_dai; + +#endif /* End of __OMAP_MCPDM_H__ */ -- 1.6.0.4

2 1

Re: [alsa-devel] [patch] oxygen: clean up. make precedence explicit
by Clemens Ladisch 19 Feb '10

19 Feb '10

> This doesn't change anything, but I think it makes the code clearer. > It silences a smatch warning: > sound/pci/oxygen/oxygen_mixer.c +91 dac_mute_put(7) warn: add some parenthesis here? That message doesn't say why some parentheses should be added. And it's a question; how do I give it the answer "no"? :-) > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; This doesn't look any clearer to me; I don't think that the unary negation operator could be thought to have lower precedence than "!=". Why does smatch warn about this combination? Do such errors actually happen? Regards, Clemens

2 1

[alsa-devel] [patch] oxygen: clean up. make precedence explicit
by Dan Carpenter 19 Feb '10

19 Feb '10

This doesn't change anything, but I think it makes the code clearer. It silences a smatch warning: sound/pci/oxygen/oxygen_mixer.c +91 dac_mute_put(7) warn: add some parenthesis here? Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- Compile tested. diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index f375b8a..28590b9 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, int changed; mutex_lock(&chip->mutex); - changed = !value->value.integer.value[0] != chip->dac_mute; + changed = (!value->value.integer.value[0]) != chip->dac_mute; if (changed) { chip->dac_mute = !value->value.integer.value[0]; chip->model.update_dac_mute(chip);

1 0

[alsa-devel] [PATCH v3 0/3] McPDM driver patches series
by Candelaria Villareal, Jorge 19 Feb '10

19 Feb '10

McPDM module is the interface between the OMAP and TWL6030 audio codec. The interface supports the following: * Up to 5 downlink channels and 3 uplink channels * 88.2 and 96 KHz * 32 bit sample size Currently, driver is set to support up to 4 downlink channels. If 2 channels are selected, output will go to the headset, and if 4 channels are selected, output will go to the headset and speakers. Functionality to select the output at runtime will be added afterwards. Command and status mode is not supported in current patches. Changes from v2: - Functions have been renamed using playback/capture convention -- Jorge Eduardo Candelaria (1): ASoC: OMAP4: Add support for McPDM Misael Lopez Cruz (2): ASoC: OMAP: data_type and sync_mode configurable in audio dma ASoC: OMAP4: Add McPDM platform driver sound/soc/omap/Kconfig | 3 + sound/soc/omap/Makefile | 2 + sound/soc/omap/mcpdm.c | 490 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/mcpdm.h | 151 +++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 + sound/soc/omap/omap-mcpdm.c | 251 ++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 29 +++ sound/soc/omap/omap-pcm.c | 15 +- sound/soc/omap/omap-pcm.h | 4 +- 9 files changed, 939 insertions(+), 8 deletions(-) create mode 100644 sound/soc/omap/mcpdm.c create mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.c create mode 100644 sound/soc/omap/omap-mcpdm.h

1 0

[alsa-devel] Troubles using the external SDK
by Hans-Kristian Arntzen 19 Feb '10

19 Feb '10

I am trying to make an alsa output plugin that streams the audio over network (something like esd, just even more simplistic) using the external SDK, however, I am having trouble implementing some things correctly it seems, and the docs weren't descriptive enough. So far, it works quite well with music playback that doesn't really care about audio blocking, etc. The main problem is audio sync and blocking with more complex applications like MPlayer. As far as I can see in the MPlayer source, to retain audio sync, it uses snd_pcm_delay() for delay, and snd_pcm_status_get_avail() to figure out how much it can write without audio blocking. This is fine, but how can snd_pcm_status_get_avail() get the data it needs from the plugins? It doesn't seem to call on any callbacks to do this. It seems so far that snd_pcm_status_get_avail() always returns buffer_size, since I seem to get working audio, but the video starts to stutter. The stuttering goes away when I force a really small maximum buffer size (we're talking 512-ish bytes), but this is really, really bad, and doesn't work too well on my netbook (performance/timing issues I guess). Is it possible to define a snd_pcm_status_get_avail() callback in the SDK in some way? Anyways. I want to know how you'd do this. The OSS driver for example doesn't even seem to define a delay() callback, which strikes me as odd. How can it know about the delay? Also, I want to know the reasoning behind the pointer() function. It seems to be called all the time. I don't know what it really does, and what good it does. Is it supposed to represent where in the buffer we are? As in, it's supposed to stay somewhere between 0 and buffer_size? If so, my experience with it doesn't make sense at all. If I return that actual value, the programs simply stop up and loop endlessly over the pointer() function. The temporary "hack" (as in probably very wrong) was to return the total amount of data that's been sent since the start. Can anyone help a frustrated programmer? :P -- Themaister

1 0

[alsa-devel] [PATCHv2 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 18 Feb '10

18 Feb '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. Since previous review round, I have corrected both the pointed issues in ASoC patch. In addition, I rebased this patch on top of Janusz Krzysztofik's McBSP register cache patches in linux-omap-2.6.git Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 63 +++++ arch/arm/plat-omap/mcbsp.c | 394 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 176 ++++++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 636 insertions(+), 1 deletions(-)

7 11

[alsa-devel] [PATCHv2 0/2] McBSP: OMAP3: Add sidetone feature
by Ilkka Koskinen 18 Feb '10

18 Feb '10

The first patch implements McBSP sidetone feature on OMAP3 while the second one provides ALSA interface for using it. The patch set is based on the patch Eduardo Valentin sent to alsa-devel mailing list in October 2009. Channel gain control and enabling the sidetone have been moved to ALSA interface as suggested in comments. These patches apply on top of Janusz Krzysztofik's McBSP register cache patches in linux-omap-2.6.git Since previous review round, I have combined McBSP port specific omap_mcbsp_st_set_mode() and omap_mcbsp_st_put_mode() functions to corresponding ones as suggested by Peter Ujfalusi. Eero Nurkkala (1): McBSP: OMAP3: Add sidetone feature Ilkka Koskinen (1): ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone arch/arm/mach-omap2/mcbsp.c | 2 + arch/arm/plat-omap/include/plat/mcbsp.h | 63 +++++ arch/arm/plat-omap/mcbsp.c | 394 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 176 ++++++++++++++ sound/soc/omap/omap-mcbsp.h | 2 + 5 files changed, 636 insertions(+), 1 deletions(-)

4 7

[alsa-devel] [LOCKDEP] 33-rc8 Running aplay with pulse as the default
by Ed Tomlinson 18 Feb '10

18 Feb '10

Hi, Inorder to get skype working with linux I have the following in .asoundrc for my normal userid. --- pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse --- Try #1 Starting as root if I do: su - <id> aplay sound.wav <traceback inlined below> aplay sound.wav (silence) killall pulseaudio aplay sound.wav (works as expected now going thru pulseaudio) Try #2 Starting as root if I do: aplay sound.wav (works as expected using alsa alone) su - <id> aplay sound.wav <another lockdep traceback> aplay sound.wav (works as expected now going thru pulseaudio) With .32 there were no tracebacks (lockdep was enabled) and try #1 would work where I now get silence. Ideas? Ed Tomlinson aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC1200 Analog [ALC1200 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC1200 Digital [ALC1200 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 "ALC1200 Analog" is what has speakers connected The em28xx in the traceback is an input source from a usb hdtv dongle. pavucontrol setup has been verified. [ 91.070620] [ 91.070621] ======================================================= [ 91.071378] [ INFO: possible circular locking dependency detected ] [ 91.071378] 2.6.33-rc8-crc #106 [ 91.071378] ------------------------------------------------------- [ 91.071378] pulseaudio/2717 is trying to acquire lock: [ 91.071378] (&dev->lock){+.+.+.}, at: [<ffffffffa02797f1>] snd_em28xx_capture_open+0x61/0x190 [em28xx_alsa] [ 91.071378] [ 91.071378] but task is already holding lock: [ 91.071378] (&pcm->open_mutex){+.+.+.}, at: [<ffffffffa010ecd2>] snd_pcm_open+0x182/0x450 [snd_pcm] [ 91.071378] [ 91.071378] which lock already depends on the new lock. [ 91.071378] [ 91.071378] [ 91.071378] the existing dependency chain (in reverse order) is: [ 91.071378] [ 91.071378] -> #3 (&pcm->open_mutex){+.+.+.}: [ 91.071378] [<ffffffff81083215>] __lock_acquire+0xfc5/0x1550 [ 91.071378] [<ffffffff8108383c>] lock_acquire+0x9c/0x140 [ 91.071378] [<ffffffff814b96c1>] __mutex_lock_common+0x61/0x610 [ 91.071378] [<ffffffff814b9d53>] mutex_lock_nested+0x43/0x50 [ 91.071378] [<ffffffffa010ad32>] snd_pcm_release+0x42/0xb0 [snd_pcm] [ 91.071378] [<ffffffff8112115d>] __fput+0x15d/0x290 [ 91.071378] [<ffffffff811212ad>] fput+0x1d/0x30 [ 91.071378] [<ffffffff810fe001>] remove_vma+0x51/0x80 [ 91.071378] [<ffffffff810ffa1d>] do_munmap+0x2fd/0x390 [ 91.071378] [<ffffffff810ffb06>] sys_munmap+0x56/0x80 [ 91.071378] [<ffffffff81002adb>] system_call_fastpath+0x16/0x1b [ 91.071378] [ 91.071378] -> #2 (&mm->mmap_sem){++++++}: [ 91.071378] [<ffffffff81083215>] __lock_acquire+0xfc5/0x1550 [ 91.071378] [<ffffffff8108383c>] lock_acquire+0x9c/0x140 [ 91.071378] [<ffffffff810f62c7>] might_fault+0xa7/0xd0 [ 91.071378] [<ffffffff8113102b>] filldir+0x7b/0xe0 [ 91.071378] [<ffffffff8118aa21>] sysfs_readdir+0xf1/0x1b0 [ 91.071378] [<ffffffff81131268>] vfs_readdir+0xb8/0xe0 [ 91.071378] [<ffffffff8113144d>] sys_getdents+0xad/0x110 [ 91.071378] [<ffffffff81002adb>] system_call_fastpath+0x16/0x1b [ 91.071378] [ 91.071378] -> #1 (sysfs_mutex){+.+.+.}: [ 91.071378] [<ffffffff81083215>] __lock_acquire+0xfc5/0x1550 [ 91.071378] [<ffffffff8108383c>] lock_acquire+0x9c/0x140 [ 91.071378] [<ffffffff814b96c1>] __mutex_lock_common+0x61/0x610 [ 91.071378] [<ffffffff814b9d53>] mutex_lock_nested+0x43/0x50 [ 91.071378] [<ffffffff8118b082>] sysfs_addrm_start+0x22/0x30 [ 91.071378] [<ffffffff8118bc58>] create_dir+0x58/0xb0 [ 91.514885] [<ffffffff8118bd0a>] sysfs_create_dir+0x5a/0x70 [ 91.514885] [<ffffffff8123ebbd>] kobject_add_internal+0xbd/0x1f0 [ 91.514885] [<ffffffff8123ee08>] kobject_add_varg+0x38/0x60 [ 91.514885] [<ffffffff8123eee4>] kobject_add+0x44/0x70 [ 91.514885] [<ffffffff81392510>] device_add+0xb0/0x5e0 [ 91.514885] [<ffffffff81392a5e>] device_register+0x1e/0x30 [ 91.514885] [<ffffffff8140a50d>] i2c_register_adapter+0x12d/0x250 [ 91.514885] [<ffffffff8140a7f1>] i2c_add_adapter+0xb1/0xd0 [ 91.514885] [<ffffffffa0231ba4>] em28xx_i2c_register+0x104/0x520 [em28xx] [ 91.514885] [<ffffffffa0233daa>] em28xx_usb_probe+0x68a/0xb90 [em28xx] [ 91.514885] [<ffffffff813eb4ab>] usb_probe_interface+0xeb/0x1b0 [ 91.514885] [<ffffffff81395546>] driver_probe_device+0xc6/0x1d0 [ 91.514885] [<ffffffff813956eb>] __driver_attach+0x9b/0xa0 [ 91.514885] [<ffffffff8139492c>] bus_for_each_dev+0x6c/0xa0 [ 91.514885] [<ffffffff8139535e>] driver_attach+0x1e/0x20 [ 91.514885] [<ffffffff81394c11>] bus_add_driver+0xe1/0x280 [ 91.514885] [<ffffffff81395a58>] driver_register+0x98/0x140 [ 91.514885] [<ffffffff813ea06c>] usb_register_driver+0xdc/0x1a0 [ 91.514885] [<ffffffffa0248023>] 0xffffffffa0248023 [ 91.514885] [<ffffffff810001dc>] do_one_initcall+0x3c/0x1d0 [ 91.514885] [<ffffffff81092d75>] sys_init_module+0xe5/0x250 [ 91.514885] [<ffffffff81002adb>] system_call_fastpath+0x16/0x1b [ 91.514885] [ 91.514885] -> #0 (&dev->lock){+.+.+.}: [ 91.514885] [<ffffffff81083668>] __lock_acquire+0x1418/0x1550 [ 91.514885] [<ffffffff8108383c>] lock_acquire+0x9c/0x140 [ 91.514885] [<ffffffff814b96c1>] __mutex_lock_common+0x61/0x610 [ 91.514885] [<ffffffff814b9d53>] mutex_lock_nested+0x43/0x50 [ 91.514885] [<ffffffffa02797f1>] snd_em28xx_capture_open+0x61/0x190 [em28xx_alsa] [ 91.514885] [<ffffffffa010adee>] snd_pcm_open_substream+0x4e/0x90 [snd_pcm] [ 91.514885] [<ffffffffa010ece9>] snd_pcm_open+0x199/0x450 [snd_pcm] [ 91.514885] [<ffffffffa010efd4>] snd_pcm_capture_open+0x34/0x40 [snd_pcm] [ 91.514885] [<ffffffffa00c5778>] snd_open+0x198/0x4e0 [snd] [ 91.514885] [<ffffffff8112384d>] chrdev_open+0x17d/0x320 [ 91.514885] [<ffffffff8111d358>] __dentry_open+0x1a8/0x400 [ 91.514885] [<ffffffff8111d6b4>] nameidata_to_filp+0x54/0x70 [ 91.514885] [<ffffffff8112e7f1>] do_filp_open+0x841/0xc00 [ 91.514885] [<ffffffff8111e954>] do_sys_open+0xa4/0x180 [ 91.514885] [<ffffffff8111ea70>] sys_open+0x20/0x30 [ 91.514885] [<ffffffff81002adb>] system_call_fastpath+0x16/0x1b [ 91.514885] [ 91.514885] other info that might help us debug this: [ 91.514885] [ 91.514885] 1 lock held by pulseaudio/2717: [ 91.514885] #0: (&pcm->open_mutex){+.+.+.}, at: [<ffffffffa010ecd2>] snd_pcm_open+0x182/0x450 [snd_pcm] [ 91.514885] [ 91.514885] stack backtrace: [ 91.514885] Pid: 2717, comm: pulseaudio Not tainted 2.6.33-rc8-crc #106 [ 91.514885] Call Trace: [ 91.514885] [<ffffffff81080fd9>] print_circular_bug+0xe9/0xf0 [ 91.514885] [<ffffffff81083668>] __lock_acquire+0x1418/0x1550 [ 91.514885] [<ffffffff81081dc8>] ? debug_check_no_locks_freed+0xc8/0x150 [ 91.514885] [<ffffffff8108383c>] lock_acquire+0x9c/0x140 [ 91.514885] [<ffffffffa02797f1>] ? snd_em28xx_capture_open+0x61/0x190 [em28xx_alsa] [ 91.514885] [<ffffffff814b96c1>] __mutex_lock_common+0x61/0x610 [ 91.514885] [<ffffffffa02797f1>] ? snd_em28xx_capture_open+0x61/0x190 [em28xx_alsa] [ 91.514885] [<ffffffffa02797f1>] ? snd_em28xx_capture_open+0x61/0x190 [em28xx_alsa] [ 91.514885] [<ffffffffa0109c70>] ? snd_pcm_hw_rule_muldivk+0x0/0xa0 [snd_pcm] [ 91.514885] [<ffffffff8107fd54>] ? lockdep_init_map+0x44/0x130 [ 91.514885] [<ffffffff814b9d53>] mutex_lock_nested+0x43/0x50 [ 91.514885] [<ffffffffa02797f1>] snd_em28xx_capture_open+0x61/0x190 [em28xx_alsa] [ 91.514885] [<ffffffffa010adee>] snd_pcm_open_substream+0x4e/0x90 [snd_pcm] [ 91.514885] [<ffffffffa010ece9>] snd_pcm_open+0x199/0x450 [snd_pcm] [ 91.514885] [<ffffffff81081cfd>] ? trace_hardirqs_on+0xd/0x10 [ 91.514885] [<ffffffff810434f0>] ? default_wake_function+0x0/0x20 [ 91.514885] [<ffffffff814b965e>] ? mutex_unlock+0xe/0x10 [ 91.514885] [<ffffffffa010efd4>] snd_pcm_capture_open+0x34/0x40 [snd_pcm] [ 91.514885] [<ffffffffa00c5778>] snd_open+0x198/0x4e0 [snd] [ 91.514885] [<ffffffff8112384d>] chrdev_open+0x17d/0x320 [ 91.514885] [<ffffffff8111d358>] __dentry_open+0x1a8/0x400 [ 91.514885] [<ffffffff811236d0>] ? chrdev_open+0x0/0x320 [ 91.514885] [<ffffffff8111d6b4>] nameidata_to_filp+0x54/0x70 [ 91.514885] [<ffffffff8112e7f1>] do_filp_open+0x841/0xc00 [ 91.514885] [<ffffffff814bf981>] ? sub_preempt_count+0x51/0x60 [ 91.514885] [<ffffffff814bbb7c>] ? _raw_spin_unlock+0x5c/0x70 [ 91.514885] [<ffffffff8111e954>] do_sys_open+0xa4/0x180 [ 91.514885] [<ffffffff8111ea70>] sys_open+0x20/0x30 [ 91.514885] [<ffffffff81002adb>] system_call_fastpath+0x16/0x1b [ 92.643423] hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj.

2 12

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 17 Feb '10

17 Feb '10

The following changes since commit 7833ae0edf50b0eb303e95b1bec5fbd63a1e2672: Peter Ujfalusi (1): ASoC: tlv320dac33: Correct the OSCSET calculation are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Mark Brown (1): ASoC: Make pmdown_time a long Peter Ujfalusi (1): ASoC: TWL4030: Use codec defaults for Headset initial configuration include/sound/soc.h | 2 +- sound/soc/codecs/twl4030.c | 4 ++-- sound/soc/soc-core.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-)

2 1

[PATCH] ALSA: usbaudio Mbox support, output only
by Chris J Arges 17 Feb '10

17 Feb '10

Signed-off-by: Chris J Arges <christopherarges(a)gmail.com> --- sound/usb/usbquirks.h | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e691eba..14552d2 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2215,6 +2215,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Digidesign Mbox */ +{ + /* Thanks to Clemens Ladisch <clemens(a)ladisch.de> */ + USB_DEVICE(0x0dba, 0x1000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "MBox", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .format = SNDRV_PCM_FORMAT_S24_3BE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x02, + .ep_attr = 0x01, + .maxpacksize = 0x130, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- 1.6.3.3 --=-g9t4QPaQL9hW42/Pyv3o--

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[alsa-devel] What is the required accuracy of audio sampling rate?
by Yiliang Bao 17 Feb '10

17 Feb '10

Hi, I have a PCI-e audio card that does not give very accurate sampling rate (about -0.2% error, i.e., if I set the sampling rate to 8K, I got only 7984 samples per second on average). It caused various synchronization problems in applications like gstreamer. Since there will be some difference from one clock domain to another, I am wondering if there is any defined tolerance on the sampling rate, and where the right place is to correct this kind of error. I would really appreciate if someone can point me to the related documents/links. Thanks, Yiliang

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[alsa-devel] [PATCH] Echoaudio, fix Guru Meditation #00000005.48454C50
by Giuliano Pochini 17 Feb '10

17 Feb '10

(resend: I mistyped the list address) There is a small window between the hw_params() callback and when runtime->frame_bits is set by ALSA middle layer. When another substream is already running, if an interrupt is delivered during that window the irq handler calls pcm_pointer() which does a division by zero. The patch below makes the irq handler skip substreams that are initialized but not started yet. Cc to Clemens Ladisch because he proposed an alternate fix. For more information, please read the original thread in the linux-kernel mailing list: http://lkml.org/lkml/2010/2/2/187 Short description: This patch fixes a division by zero error in the irq handler. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c__orig 2010-02-16 22:21:36.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-02-16 22:21:45.000000000 +0100 @@ -1864,7 +1864,9 @@ static irqreturn_t snd_echo_interrupt(in /* The hardware doesn't tell us which substream caused the irq, thus we have to check all running substreams. */ for (ss = 0; ss < DSP_MAXPIPES; ss++) { - if ((substream = chip->substream[ss])) { + substream = chip->substream[ss]; + if (substream && ((struct audiopipe *)substream->runtime-> + private_data)->state == PIPE_STATE_STARTED) { period = pcm_pointer(substream) / substream->runtime->period_size; if (period != chip->last_period[ss]) { -- Giuliano.

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[alsa-devel] [PATCH] ASoC: TWL4030: Use codec defaults for Headset initial configuration
by Peter Ujfalusi 17 Feb '10

17 Feb '10

Disable the amplifiers for the headset outputs, and do not select routings by default to the headset outputs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- The obvious side effect of this patch is that on boards which are using the headset output the playback will not work 'out of box'. The user need to specifically enable the routings for the boards. Board that came to my mind is the Beagle, and probably it's clones as well. sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 277862e..6f5d4af 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -75,8 +75,8 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_BTPGA (0x1F) */ 0x00, /* REG_BTSTPGA (0x20) */ 0x00, /* REG_EAR_CTL (0x21) */ - 0x24, /* REG_HS_SEL (0x22) */ - 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_SEL (0x22) */ + 0x00, /* REG_HS_GAIN_SET (0x23) */ 0x00, /* REG_HS_POPN_SET (0x24) */ 0x00, /* REG_PREDL_CTL (0x25) */ 0x00, /* REG_PREDR_CTL (0x26) */ -- 1.7.0

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[alsa-devel] Is Intel 82801G (ICH7 family) with VIA VT1708B 8-Ch supported?
by Zaar Hai 17 Feb '10

17 Feb '10

Good day, dear ALSA developers! I've got several boards based on Intel G41 chipset, which has ICH7 south bridge with VT1708B codec on it. While VT1708B seems to be supported since 2007, on all my computers the card is recognized, but mixer does not work: $ alsamixer alsamixer: function snd_mixer_load failed: Invalid argument I've tried it on Debian, OpenSuse and Ubuntu (including nightly builds), both 32 and 64 bit version, on 4 computers (all have the same hardware) - result is the same: card recognized, sound is played, but mixer (volume channels, etc.) does not. So, I've thought, that since G41 is rather new, may be this ICH7 with VT1708B combo is not supported by alsa yet? [1]. http://pastebin.com/m50dcbff0 - contents of /proc/asound/card0/codec#0 [2]. http://pastebin.com/m42320afc - lspci output [3]. http://pastebin.com/m443eaa12 - aplay -l output P.S. If you think this message rather belongs to users and not devel list, just tell me to go there. I've just thought that supported/not supported question should better be asked devel list. -- Zaar

1 1

[alsa-devel] Understanding SND_PCM_INFO_ [BATCH, BLOCK_TRANSFER, DOUBLE]
by Eliot Blennerhassett 17 Feb '10

17 Feb '10

I've been looking at these info flags, and whether they cause the driver to be accessed any differently by the upper layers. So far I can find hardly any examples where they are used in any significant way - what am I missing if anything? SND_/SNDRV_ _PCM_INFO_BATCH double buffering skips the check enabled by XRUN_DEBUG_JIFFIESCHECK alsa-lib returns it in snd_pcm_hw_params_is_batch() _PCM_INFO_BLOCK_TRANSFER /* hardware transfer block of samples */ translated by pcm_oss.c into ~DSP_CAP_REALTIME lib: snd_pcm_hw_params_is_block_transfer() _PCM_INFO_DOUBLE /* double buffering needed for PCM start/stop */ snd_pcm_hw_params_is_double() I don't see any of these functions called by aplay... -- Eliot

1 0

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 16 Feb '10

16 Feb '10

The following changes since commit 3a66d3877eaa4ab9818000a15c07326adaa9ca79: Mark Brown (1): ASoC: Add WM2000 driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Mark Brown (2): ASoC: Make pmdown_time a per-card setting ASoC: Make pmdown_time runtime configurable Peter Ujfalusi (2): ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playback ASoC: tlv320dac33: Correct the OSCSET calculation include/sound/soc.h | 2 ++ sound/soc/codecs/tlv320dac33.c | 5 ++++- sound/soc/soc-core.c | 31 ++++++++++++++++++++++++++++++- 3 files changed, 36 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Strange behaviour in snd-usb-cayak with Audio4DJ
by Pedro Ribeiro 16 Feb '10

16 Feb '10

Hi all, I'm trying to setup my Audio4DJ usb soundcard. It has 4 channels (2 stereo ones). ALSA information script output is here http://www.alsa-project.org/db/?f=fe4cc3eda2df49c6055f55d4cef935524124d1a1 and my .asoundrc is here http://pastebin.com/m3f70572d Currently my .asoundrc has 4 devices: - Audio4DJ_A (combines channels 1 and 2) - Audio4DJ_B (combines channels 3 and 4) - Audio4DJ (combines Audio4DJ_A and Audio4DJ_B) - ttable is a device for JACK according to this http://alsa-project.org/main/index.php/Asoundrc#Virtual_multi_channel_devic… My goal is to have a separate ALSA soundcard for each pair of channels, an ALSA soundcard for 4 channels and a JACK device for all 4 channels. Running speaker-test -c2 -D plughw:Audio4DJ,0,X -twav -l1 with X=0 and X=1 correctly outputs to each channel. Also, pcm.Audio4DJ_A and pcm.Audio4DJ_B correctly work through both ALSA and JACK for each pair of channels. However, if I try to use the pcm.Audio4DJ (which combines both pcm.Audio4DJ_A and pcm.Audio4DJ_B), I get correct output through channels 1 and 2, but channels 3 and 4 have a lot of garbage - you can hear the track playing but it is at a very high pitch and distorted. JACK doesn't even start with the Audio4DJ device: # jackd -R --timeout 4500 -d alsa -d Audio4DJ -p 128 -n 3 -r 44100 -i 4 -o apparent rate = 44100 creating alsa driver ... Audio4DJ|Audio4DJ|128|3|44100|4|4|nomon|swmeter|-|32bit configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 3 periods ALSA: final selected sample format for capture: 32bit float little-endian ALSA: use 3 periods for capture ALSA: final selected sample format for playback: 32bit float little-endian ALSA: use 3 periods for playback ALSA: could not complete read of 0 frames: error = -32 cycle execution failure, exiting DRIVER NT: could not run driver cycle jack main caught signal 12 (crashes immediately) The ttable device locks up the system for a few seconds and then JACK crashes: # jackd -R --timeout 4500 -d alsa -d ttable -p 128 -n 3 -r 44100 -i 4 -o apparent rate = 44100 creating alsa driver ... ttable|ttable|128|3|44100|4|4|nomon|swmeter|-|32bit configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 3 periods ALSA: final selected sample format for capture: 32bit integer little-endian ALSA: use 3 periods for capture ALSA: final selected sample format for playback: 32bit integer little-endian ALSA: use 3 periods for playback jackd watchdog: timeout - killing jackd Without a .asoundrc it doesn't even work. Can someone please tell me if this is expected behaviour or a bug? Thanks, Pedro

2 3

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: small fixes
by Peter Ujfalusi 16 Feb '10

16 Feb '10

Hello, Two small fixes for the tlv320dac33 codec driver: Clearing the FIFFLUSH flag before playback start The register cache has the bit set after the playback is stopped, so clear it before the playback. OSCSET calculation in case of 44.1KHz sampling rate was off by 1 (0x1588 instead of 0x1589, which is the value should be used according to TI). --- Peter Ujfalusi (2): ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playback ASoC: tlv320dac33: Correct the OSCSET calculation sound/soc/codecs/tlv320dac33.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-)

3 4

Re: [alsa-devel] [PATCH][ALSA] usbmidi support for Access Music synths.
by Clemens Ladisch 16 Feb '10

16 Feb '10

Sebastien Alaiwan wrote: > here's a patch that adds MIDI support through USB for one of the > Access Music synths, the VirusTI. Thanks! Please run the checkpatch script on your patch. > + u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; // "midi send" enable > + > + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), seq, > + ARRAY_SIZE(seq), &actual_length, 1000); There are architectures where doing DMA with data on the stack is not possible. Have a look at send_bulk_static_data. > /* > + * Detects the endpoints for Access Music Virus TI > + */ > +static int snd_usbmidi_detect_two_cables_per_endpoint(struct snd_usb_midi* umidi, > + struct snd_usb_midi_endpoint_info* endpoints) > +{ > + int err, i; > + > + err = snd_usbmidi_detect_endpoints(umidi, endpoints, MIDI_MAX_ENDPOINTS); > + for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { > + if (endpoints[i].in_ep) { > + printk("Creating two in cables for input EP 0x%.2X\n", endpoints[i].in_ep); If this is just for debugging, use snd_printdd. > + endpoints[i].in_cables = 0x0003; > + } > + if (endpoints[i].out_ep) { > + printk("Creating two out cables for output EP 0x%.2X\n", endpoints[i].out_ep); > + endpoints[i].out_cables = 0x0003; > + } > + } > + return err; > +} Does this device use more than one endpoint per direction? If not, this could be done with QUIRK_MIDI_FIXED_ENDPOINT. > +++ b/sound/usb/usbquirks.h > @@ -2105,6 +2105,28 @@ > } > }, > > +/* Access Music devices */ > +{ > + /* VirusTI Desktop */ > + USB_DEVICE_VENDOR_SPEC(0x133e, 0x0815), Please keep the list sorted (as much as possible). > + .vendor_name = "AccessMusic", > + .product_name = "Virus TI", These are needed only if the device hasn't vendor/product strings. > + // "Virus TI MIDI" and "Virus TI Synth" If these are port names, put them into the snd_usbmidi_port_info array. Regards, Clemens

2 6

[alsa-devel] Lynx card driver
by Andrew Gillis 16 Feb '10

16 Feb '10

I need a driver for the Lynx series of cards. http://www.alsa-project.org/main/index.php/Matrix:Vendor-Lynx_Studio_Techno… Would anybody be interested in doing this work as a paid contract? -Andrew -- Andrew Gillis Lead Developer VortexBox vortexbox.org

3 3

[alsa-devel] [PATCH 0/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 15 Feb '10

15 Feb '10

The following patch set adds suspend/resume support to all echoaudio drivers. It has been tested on an IndigoIOx and a Gina3G. Rawmidi device has been tested partially. PCM and controls interfaces have been tested more and I think it's reasonably safe. I also have a Gina24 and a Gina3G, but I didn't test suspend/resume with them because I couldn't manage to make PM test system work on my pmac. Echoaudio soundcards do not support power management and Dx PCI states. Anyway, the card can enter a "comatose" state by sending a command to the DSP that is supposed to turn parts of the hardware off and reduce power consumption. Doing that causes the loss of the card's internal state, thus it is impossible, upon resume, to make the card continue from the exact point where it was suspended. The driver restores all controls and PCM settings and the applications have to call prepare() to restart the substreams. Rawmidi is restarted automatically, but some data in the i/o buffers may be lost. -- Giuliano.

2 4

[alsa-devel] [PATCH 5/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

5/5 Patchin' patcher: This patch updates alsa-driver echoaudio.patch . Short description: This patch updates alsa-driver echoaudio.patch . Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> --- alsa-driver-1.0.22.1/pci/echoaudio/echoaudio.patch__orig 2009-12-28 16:15:54.000000000 +0100 +++ alsa-driver-1.0.22.1/pci/echoaudio/echoaudio.patch 2010-01-29 23:35:16.000000000 +0100 @@ -1,6 +1,6 @@ ---- ../../alsa-kernel/pci/echoaudio/echoaudio.c 2006-08-05 14:41:12.000000000 +0200 -+++ echoaudio.c 2006-10-07 20:30:53.000000000 +0200 -@@ -1926,6 +1926,7 @@ +--- ../../alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:17:31.000000000 +0100 ++++ echoaudio.c 2010-01-29 23:28:45.000000000 +0100 +@@ -1943,6 +1943,7 @@ static __devinit int snd_echo_create(str struct echoaudio *chip; int err; size_t sz; @@ -8,7 +8,7 @@ static struct snd_device_ops ops = { .dev_free = snd_echo_dev_free, }; -@@ -1988,7 +1989,12 @@ +@@ -2014,7 +2015,12 @@ static __devinit int snd_echo_create(str chip->comm_page_phys = chip->commpage_dma_buf.addr; chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; @@ -19,6 +19,28 @@ + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device); +#endif + err = init_hw(chip, chip->pci->device, subsystem_device); - if (err) { - DE_INIT(("init_hw err=%d\n", err)); - snd_echo_free(chip); + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { +@@ -2245,6 +2251,7 @@ static int snd_echo_resume(struct pci_de + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; ++ u16 subsystem_device; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); +@@ -2252,7 +2259,12 @@ static int snd_echo_resume(struct pci_de + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + +- err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); ++#if LINUX_VERSION_CODE >= KERNEL_VERSION(2, 3, 0) ++ subsystem_device = pci->subsystem_device; ++#else ++ pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &subsystem_device); ++#endif ++ err = init_hw(chip, chip->pci->device, subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); -- Giuliano.

1 0

[alsa-devel] [PATCH 4/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

4/5 Add suspend support #2: This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). P.S. I'll fix "rettings" later. Short description: This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:09:09.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.c 2010-01-29 23:11:43.000000000 +0100 @@ -753,6 +753,8 @@ static int pcm_trigger(struct snd_pcm_su spin_lock(&chip->lock); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + DE_ACT(("pcm_trigger resume\n")); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: DE_ACT(("pcm_trigger start\n")); @@ -776,6 +778,8 @@ static int pcm_trigger(struct snd_pcm_su err = start_transport(chip, channelmask, chip->pipe_cyclic_mask); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + DE_ACT(("pcm_trigger suspend\n")); case SNDRV_PCM_TRIGGER_STOP: DE_ACT(("pcm_trigger stop\n")); for (i = 0; i < DSP_MAXPIPES; i++) { @@ -1951,18 +1955,27 @@ static __devinit int snd_echo_create(str return err; pci_set_master(pci); - /* allocate a chip-specific data */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - pci_disable_device(pci); - return -ENOMEM; + /* Allocate chip if needed */ + if (!*rchip) { + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + pci_disable_device(pci); + return -ENOMEM; + } + DE_INIT(("chip=%p\n", chip)); + spin_lock_init(&chip->lock); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + atomic_set(&chip->opencount, 0); + mutex_init(&chip->mode_mutex); + chip->can_set_rate = 1; + } else { + /* If this was called from the resume function, chip is + * already allocated and it contains current card settings. + */ + chip = *rchip; } - DE_INIT(("chip=%p\n", chip)); - - spin_lock_init(&chip->lock); - chip->card = card; - chip->pci = pci; - chip->irq = -1; /* PCI resource allocation */ chip->dsp_registers_phys = pci_resource_start(pci, 0); @@ -2002,7 +2015,9 @@ static __devinit int snd_echo_create(str chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); - if (err) { + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { DE_INIT(("init_hw err=%d\n", err)); snd_echo_free(chip); return err; @@ -2013,9 +2028,6 @@ static __devinit int snd_echo_create(str snd_echo_free(chip); return err; } - atomic_set(&chip->opencount, 0); - mutex_init(&chip->mode_mutex); - chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ return 0; @@ -2048,6 +2060,7 @@ static int __devinit snd_echo_probe(stru snd_card_set_dev(card, &pci->dev); + chip = NULL; /* Tells snd_echo_create to allocate chip */ if ((err = snd_echo_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; @@ -2187,6 +2200,112 @@ ctl_error: +#if defined(CONFIG_PM) + +static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + + DE_INIT(("suspend start\n")); + snd_pcm_suspend_all(chip->analog_pcm); + snd_pcm_suspend_all(chip->digital_pcm); + +#ifdef ECHOCARD_HAS_MIDI + /* This call can sleep */ + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 0); +#endif + spin_lock_irq(&chip->lock); + if (wait_handshake(chip)) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + clear_handshake(chip); + if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + spin_unlock_irq(&chip->lock); + + chip->dsp_code = NULL; + free_irq(chip->irq, chip); + chip->irq = -1; + pci_save_state(pci); + pci_disable_device(pci); + + DE_INIT(("suspend done\n")); + return 0; +} + + + +static int snd_echo_resume(struct pci_dev *pci) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); + commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + + err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); + snd_echo_free(chip); + return err; + } + DE_INIT(("resume init OK\n")); + + /* Temporarily set chip->pipe_alloc_mask=0 otherwise + * restore_dsp_settings() fails. + */ + pipe_alloc_mask = chip->pipe_alloc_mask; + chip->pipe_alloc_mask = 0; + err = restore_dsp_rettings(chip); + chip->pipe_alloc_mask = pipe_alloc_mask; + if (err < 0) { + kfree(commpage_bak); + return err; + } + DE_INIT(("resume restore OK\n")); + + memcpy(&commpage->audio_format, &commpage_bak->audio_format, + sizeof(commpage->audio_format)); + memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr, + sizeof(commpage->sglist_addr)); + memcpy(&commpage->midi_output, &commpage_bak->midi_output, + sizeof(commpage->midi_output)); + kfree(commpage_bak); + + if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, + ECHOCARD_NAME, chip)) { + snd_echo_free(chip); + snd_printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + DE_INIT(("resume irq=%d\n", chip->irq)); + +#ifdef ECHOCARD_HAS_MIDI + if (chip->midi_input_enabled) + enable_midi_input(chip, TRUE); + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 1); +#endif + + DE_INIT(("resume done\n")); + return 0; +} + +#endif /* CONFIG_PM */ + + + static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2209,6 +2328,10 @@ static struct pci_driver driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), +#ifdef CONFIG_PM + .suspend = snd_echo_suspend, + .resume = snd_echo_resume, +#endif /* CONFIG_PM */ }; Only in alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/: echoaudio.c.orig diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -497,9 +497,6 @@ static int load_firmware(struct echoaudi if ((box_type = load_asic(chip)) < 0) return box_type; /* error */ - if ((err = restore_dsp_rettings(chip)) < 0) - return err; - return box_type; } @@ -659,51 +656,106 @@ static void get_audio_meters(struct echo static int restore_dsp_rettings(struct echoaudio *chip) { - int err; + int i, o, err; DE_INIT(("restore_dsp_settings\n")); if ((err = check_asic_status(chip)) < 0) return err; - /* @ Gina20/Darla20 only. Should be harmless for other cards. */ + /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; chip->comm_page->handshake = 0xffffffff; - if ((err = set_sample_rate(chip, chip->sample_rate)) < 0) + /* Restore output busses */ + for (i = 0; i < num_busses_out(chip); i++) { + err = set_output_gain(chip, i, chip->output_gain[i]); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_VMIXER + for (i = 0; i < num_pipes_out(chip); i++) + for (o = 0; o < num_busses_out(chip); o++) { + err = set_vmixer_gain(chip, o, i, + chip->vmixer_gain[o][i]); + if (err < 0) + return err; + } + if (update_vmixer_level(chip) < 0) + return -EIO; +#endif /* ECHOCARD_HAS_VMIXER */ + +#ifdef ECHOCARD_HAS_MONITOR + for (o = 0; o < num_busses_out(chip); o++) + for (i = 0; i < num_busses_in(chip); i++) { + err = set_monitor_gain(chip, o, i, + chip->monitor_gain[o][i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_MONITOR */ + +#ifdef ECHOCARD_HAS_INPUT_GAIN + for (i = 0; i < num_busses_in(chip); i++) { + err = set_input_gain(chip, i, chip->input_gain[i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_INPUT_GAIN */ + + err = update_output_line_level(chip); + if (err < 0) return err; - if (chip->meters_enabled) - if (send_vector(chip, DSP_VC_METERS_ON) < 0) - return -EIO; + err = update_input_line_level(chip); + if (err < 0) + return err; -#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK - if (set_input_clock(chip, chip->input_clock) < 0) + err = set_sample_rate(chip, chip->sample_rate); + if (err < 0) + return err; + + if (chip->meters_enabled) { + err = send_vector(chip, DSP_VC_METERS_ON); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH + if (set_digital_mode(chip, chip->digital_mode) < 0) return -EIO; #endif -#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH - if (set_output_clock(chip, chip->output_clock) < 0) +#ifdef ECHOCARD_HAS_DIGITAL_IO + if (set_professional_spdif(chip, chip->professional_spdif) < 0) return -EIO; #endif - if (update_output_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_PHANTOM_POWER + if (set_phantom_power(chip, chip->phantom_power) < 0) return -EIO; +#endif - if (update_input_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK + /* set_input_clock() also restores automute setting */ + if (set_input_clock(chip, chip->input_clock) < 0) return -EIO; +#endif -#ifdef ECHOCARD_HAS_VMIXER - if (update_vmixer_level(chip) < 0) +#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH + if (set_output_clock(chip, chip->output_clock) < 0) return -EIO; #endif if (wait_handshake(chip) < 0) return -EIO; clear_handshake(chip); + if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) + return -EIO; DE_INIT(("restore_dsp_rettings done\n")); - return send_vector(chip, DSP_VC_UPDATE_FLAGS); + return 0; } @@ -920,9 +972,6 @@ static int init_dsp_comm_page(struct ech chip->card_name = ECHOCARD_NAME; chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->digital_mode = DIGITAL_MODE_NONE; - chip->input_clock = ECHO_CLOCK_INTERNAL; - chip->output_clock = ECHO_CLOCK_WORD; chip->asic_loaded = FALSE; memset(chip->comm_page, 0, sizeof(struct comm_page)); @@ -933,7 +982,6 @@ static int init_dsp_comm_page(struct ech chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); - chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE); @@ -944,50 +992,21 @@ static int init_dsp_comm_page(struct ech -/* This function initializes the several volume controls for busses and pipes. -This MUST be called after the DSP is up and running ! */ +/* This function initializes the chip structure with default values, ie. all + * muted and internal clock source. Then it copies the settings to the DSP. + * This MUST be called after the DSP is up and running ! + */ static int init_line_levels(struct echoaudio *chip) { - int st, i, o; - DE_INIT(("init_line_levels\n")); - - /* Mute output busses */ - for (i = 0; i < num_busses_out(chip); i++) - if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; - -#ifdef ECHOCARD_HAS_VMIXER - /* Mute the Vmixer */ - for (i = 0; i < num_pipes_out(chip); i++) - for (o = 0; o < num_busses_out(chip); o++) - if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_vmixer_level(chip))) - return st; -#endif /* ECHOCARD_HAS_VMIXER */ - -#ifdef ECHOCARD_HAS_MONITOR - /* Mute the monitor mixer */ - for (o = 0; o < num_busses_out(chip); o++) - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_MONITOR */ - -#ifdef ECHOCARD_HAS_INPUT_GAIN - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_input_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_INPUT_GAIN */ - - return 0; + memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); + memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); + memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); + memset(chip->vmixer_gain, ECHOGAIN_MUTED, sizeof(chip->vmixer_gain)); + chip->input_clock = ECHO_CLOCK_INTERNAL; + chip->output_clock = ECHO_CLOCK_WORD; + chip->sample_rate = 44100; + return restore_dsp_rettings(chip); } diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:08:36.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echoaudio.h 2010-01-29 23:11:43.000000000 +0100 @@ -472,6 +472,8 @@ static void free_firmware(const struct f #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); +static void snd_echo_midi_output_trigger( + struct snd_rawmidi_substream *substream, int up); static int midi_service_irq(struct echoaudio *chip); static int __devinit snd_echo_midi_create(struct snd_card *card, struct echoaudio *chip); -- Giuliano.

1 0

[alsa-devel] [PATCH 3/5] Echoaudio - add suspend/resume support
by Giuliano Pochini 14 Feb '10

14 Feb '10

3/5 Add suspend support #1: This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Short description: Move the controls init code outside the init_hw() function because is must not be called during resume. Signed-off-by: Giuliano Pochini <pochini(a)shiny.it> diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -57,15 +57,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + /* The Darla20 has no external clock sources */ static u32 detect_input_clocks(const struct echoaudio *chip) { diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/darla24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -56,15 +56,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echo3g_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/echo3g_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -97,20 +97,6 @@ static int init_hw(struct echoaudio *chi chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_phantom_power(chip, 0); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); DE_INIT(("init_hw done\n")); return err; @@ -118,6 +104,18 @@ static int init_hw(struct echoaudio *chi +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->non_audio_spdif = FALSE; + chip->bad_board = FALSE; + chip->phantom_power = FALSE; + return init_line_levels(chip); +} + + + static int set_phantom_power(struct echoaudio *chip, char on) { u32 control_reg = le32_to_cpu(chip->comm_page->control_register); diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -62,17 +62,20 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/gina24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -57,9 +57,6 @@ static int init_hw(struct echoaudio *chi ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | ECHO_CLOCK_BIT_ADAT; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { @@ -81,19 +78,22 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodj_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodj_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodjx_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigodjx_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_express_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_express_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigo_express_dsp.c 2009-12-28 15:21:33.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigo_express_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -61,6 +61,7 @@ static int set_sample_rate(struct echoau control_reg |= clock; if (control_reg != old_control_reg) { + DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoio_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoio_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoiox_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/indigoiox_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla20_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla20_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -64,17 +64,20 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla24_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/layla24_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -61,9 +61,6 @@ static int init_hw(struct echoaudio *chi ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; if ((err = load_firmware(chip)) < 0) return err; @@ -72,17 +69,22 @@ static int init_hw(struct echoaudio *chi if ((err = init_line_levels(chip)) < 0) return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mia_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mia_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -66,15 +66,19 @@ static int init_hw(struct echoaudio *chi return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip))) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff -dup alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mona_dsp.c alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c --- alsa-driver-1.0.22.1__orig//alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-29 23:08:13.000000000 +0100 +++ alsa-driver-1.0.22.1/alsa-kernel/pci/echoaudio/mona_dsp.c 2010-01-29 23:11:43.000000000 +0100 @@ -67,28 +67,26 @@ static int init_hw(struct echoaudio *chi else chip->dsp_code_to_load = FW_MONA_301_DSP; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - if ((err = load_firmware(chip)) < 0) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; -- Giuliano.

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