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[alsa-devel] [PATCH 1/3] snd-meastro3: Add amp_gpio quirk for Compaq EVO N600C
by Hans de Goede 21 Apr '10

21 Apr '10

Without this quirk sound stops working after suspend resume. With this quirk, one still needs to manually unmute the master volume control after a suspend / / resume cycle. That is fixed in another patch in this set. Note that this patch was submitted to the alsa bug tracker a long time ago: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4319 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/maestro3.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 75283fb..491a2fc 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -884,6 +884,7 @@ static struct pci_device_id snd_m3_ids[] = { MODULE_DEVICE_TABLE(pci, snd_m3_ids); static struct snd_pci_quirk m3_amp_quirk_list[] __devinitdata = { + SND_PCI_QUIRK(0x0E11, 0x0094, "Compaq Evo N600c", 0x0c), SND_PCI_QUIRK(0x10f7, 0x833e, "Panasonic CF-28", 0x0d), SND_PCI_QUIRK(0x10f7, 0x833d, "Panasonic CF-72", 0x0d), SND_PCI_QUIRK(0x1033, 0x80f1, "NEC LM800J/7", 0x03), -- 1.7.0.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [REGRESSION] SB X-Fi Notebook stopped working in 1.0.23
by The Source 21 Apr '10

21 Apr '10

This device works fine with 1.0.22.1 driver, but doesn't with 1.0.23. snd-usb-audio & Ko are loaded but to no good. alsamixer doesn't list the device and no device files in /dev/snd. Reverting back to 1.0.22.1 solves the problem. System: ASUS N51Vf laptop, OS: Fedora 12 2.6.31.12-174.2.3.fc12.x86_64

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[PATCH] Add register_poll for alsaseq
by Hector Martin 21 Apr '10

21 Apr '10

Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- pyalsa/alsaseq.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 51 insertions(+), 0 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index de131a4..bbf6b2e 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -3236,6 +3236,53 @@ Sequencer_stop_queue(SequencerObject *self, Py_RETURN_NONE; } +PyDoc_STRVAR(Sequencer_registerpoll__doc__, +"register_poll(pollObj, input=False, output=False) -- Register poll file descriptors."); + +static PyObject * +Sequencer_registerpoll(SequencerObject *self, PyObject *args, PyObject *kwds) +{ + PyObject *pollObj, *reg, *t; + struct pollfd *pfd; + int i, count; + int input = 0; + int output = 0; + int mode = POLLIN|POLLOUT; + + static char * kwlist[] = { "pollObj", "input", "output", NULL }; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "O|ii", kwlist, &pollObj, &input, &output)) + return NULL; + + if (input && !output) + mode = POLLIN; + else if (!input && output) + mode = POLLOUT; + + count = snd_seq_poll_descriptors_count(self->handle, mode); + if (count <= 0) + Py_RETURN_NONE; + pfd = alloca(sizeof(struct pollfd) * count); + count = snd_seq_poll_descriptors(self->handle, pfd, count, mode); + if (count <= 0) + Py_RETURN_NONE; + + reg = PyObject_GetAttr(pollObj, PyString_InternFromString("register")); + + for (i = 0; i < count; i++) { + t = PyTuple_New(2); + if (t) { + PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); + PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); + Py_XDECREF(PyObject_CallObject(reg, t)); + Py_DECREF(t); + } + } + + Py_XDECREF(reg); + + Py_RETURN_NONE; +} @@ -3309,6 +3356,10 @@ static PyMethodDef Sequencer_methods[] = { (PyCFunction) Sequencer_stop_queue, METH_VARARGS | METH_KEYWORDS, Sequencer_stop_queue__doc__}, + {"register_poll", + (PyCFunction) Sequencer_registerpoll, + METH_VARARGS | METH_KEYWORDS, + Sequencer_registerpoll__doc__}, {NULL} }; -- 1.6.4.4 --------------010307040309030606040702--

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[PATCH] Add register_poll for alsaseq
by Hector Martin 21 Apr '10

21 Apr '10

--- pyalsa/alsaseq.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 53 insertions(+), 0 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index de131a4..fbab6da 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -3236,6 +3236,55 @@ Sequencer_stop_queue(SequencerObject *self, Py_RETURN_NONE; } +PyDoc_STRVAR(Sequencer_registerpoll__doc__, +"register_poll(pollObj, input=False, output=False) -- Register poll file descriptors."); + +static PyObject * +Sequencer_registerpoll(SequencerObject *self, PyObject *args, PyObject *kwds) +{ + PyObject *pollObj, *reg, *t; + struct pollfd *pfd; + int i, count; + int input = 0; + int output = 0; + int mode = POLLIN|POLLOUT; + + static char * kwlist[] = { "pollObj", "input", "output", NULL }; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "O|ii", kwlist, &pollObj, &input, &output)) + return NULL; + + if (input && !output) + mode = POLLIN; + else if (!input && output) + mode = POLLOUT; + + count = snd_seq_poll_descriptors_count(self->handle, mode); + if (count <= 0) + Py_RETURN_NONE; + pfd = malloc(sizeof(struct pollfd) * count); + if (pfd == NULL) + Py_RETURN_NONE; + count = snd_seq_poll_descriptors(self->handle, pfd, count, mode); + if (count <= 0) + Py_RETURN_NONE; + + reg = PyObject_GetAttr(pollObj, PyString_InternFromString("register")); + + for (i = 0; i < count; i++) { + t = PyTuple_New(2); + if (t) { + PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); + PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); + Py_XDECREF(PyObject_CallObject(reg, t)); + Py_DECREF(t); + } + } + + Py_XDECREF(reg); + + Py_RETURN_NONE; +} @@ -3309,6 +3358,10 @@ static PyMethodDef Sequencer_methods[] = { (PyCFunction) Sequencer_stop_queue, METH_VARARGS | METH_KEYWORDS, Sequencer_stop_queue__doc__}, + {"register_poll", + (PyCFunction) Sequencer_registerpoll, + METH_VARARGS | METH_KEYWORDS, + Sequencer_registerpoll__doc__}, {NULL} }; -- 1.6.4.4 --------------080100090701000909000606--

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[PATCH] Add register_poll for alsaseq
by Hector Martin 21 Apr '10

21 Apr '10

Signed-off-by: Hector Martin <hector(a)marcansoft.com> --- pyalsa/alsaseq.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 51 insertions(+), 0 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index de131a4..bbf6b2e 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -3236,6 +3236,53 @@ Sequencer_stop_queue(SequencerObject *self, Py_RETURN_NONE; } +PyDoc_STRVAR(Sequencer_registerpoll__doc__, +"register_poll(pollObj, input=False, output=False) -- Register poll file descriptors."); + +static PyObject * +Sequencer_registerpoll(SequencerObject *self, PyObject *args, PyObject *kwds) +{ + PyObject *pollObj, *reg, *t; + struct pollfd *pfd; + int i, count; + int input = 0; + int output = 0; + int mode = POLLIN|POLLOUT; + + static char * kwlist[] = { "pollObj", "input", "output", NULL }; + + if (!PyArg_ParseTupleAndKeywords(args, kwds, "O|ii", kwlist, &pollObj, &input, &output)) + return NULL; + + if (input && !output) + mode = POLLIN; + else if (!input && output) + mode = POLLOUT; + + count = snd_seq_poll_descriptors_count(self->handle, mode); + if (count <= 0) + Py_RETURN_NONE; + pfd = alloca(sizeof(struct pollfd) * count); + count = snd_seq_poll_descriptors(self->handle, pfd, count, mode); + if (count <= 0) + Py_RETURN_NONE; + + reg = PyObject_GetAttr(pollObj, PyString_InternFromString("register")); + + for (i = 0; i < count; i++) { + t = PyTuple_New(2); + if (t) { + PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); + PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); + Py_XDECREF(PyObject_CallObject(reg, t)); + Py_DECREF(t); + } + } + + Py_XDECREF(reg); + + Py_RETURN_NONE; +} @@ -3309,6 +3356,10 @@ static PyMethodDef Sequencer_methods[] = { (PyCFunction) Sequencer_stop_queue, METH_VARARGS | METH_KEYWORDS, Sequencer_stop_queue__doc__}, + {"register_poll", + (PyCFunction) Sequencer_registerpoll, + METH_VARARGS | METH_KEYWORDS, + Sequencer_registerpoll__doc__}, {NULL} }; -- 1.6.4.4 --------------060801080008070807030804--

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[alsa-devel] [PATCH] da7210: Fencepost error in reg cache read
by Phil Carmody 20 Apr '10

20 Apr '10

From: Phil Carmody <ext-phil.2.carmody(a)nokia.com> An index equal to the array size may not be accessed. Signed-off-by: Phil Carmody <ext-phil.2.carmody(a)nokia.com> --- sound/soc/codecs/da7210.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 366daf1..96932a8 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -165,7 +165,7 @@ static const u8 da7210_reg[] = { static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) { u8 *cache = codec->reg_cache; - BUG_ON(reg > ARRAY_SIZE(da7210_reg)); + BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); return cache[reg]; } -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 20 Apr '10

20 Apr '10

The following changes since commit d21e0f4cd16656f71207683ee27465600ad21625: Marek Vasut (1): ASoC: Zipit Z2 WM8750 ASoC driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Huang Weiyi (1): ASoC: wm2000: remove unused #include <linux/version.h> Mark Brown (2): Merge branch 'for-2.6.34' into for-2.6.35 ASoC: Add indirection for CODEC private data Sascha Hauer (5): ASoC: imx-ssi: honor IMX_SSI_DMA flag ASoC: imx-pcm-dma-mx2: restart DMA after an error ASoC: imx-ssi: Use a hrtimer in FIQ mode ASoC: imx-ssi: increase minimum periods to 4 ASoC: imx-ssi: do not call hrtimer_disable in trigger function include/sound/soc.h | 13 ++++++++- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/ak4104.c | 2 +- sound/soc/codecs/ak4535.c | 10 +++--- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/cs4270.c | 20 +++++++------- sound/soc/codecs/cx20442.c | 2 +- sound/soc/codecs/da7210.c | 2 +- sound/soc/codecs/ssm2602.c | 12 ++++---- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 10 +++--- sound/soc/codecs/tlv320dac33.c | 36 ++++++++++++------------ sound/soc/codecs/twl4030.c | 32 +++++++++++----------- sound/soc/codecs/uda134x.c | 16 +++++----- sound/soc/codecs/uda1380.c | 4 +- sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm8350.c | 28 ++++++++++---------- sound/soc/codecs/wm8400.c | 16 +++++----- sound/soc/codecs/wm8523.c | 10 +++--- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8711.c | 6 ++-- sound/soc/codecs/wm8731.c | 10 +++--- sound/soc/codecs/wm8750.c | 7 ++--- sound/soc/codecs/wm8753.c | 8 +++--- sound/soc/codecs/wm8776.c | 6 ++-- sound/soc/codecs/wm8900.c | 10 +++--- sound/soc/codecs/wm8903.c | 20 +++++++------- sound/soc/codecs/wm8904.c | 42 +++++++++++++++--------------- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8955.c | 16 +++++----- sound/soc/codecs/wm8960.c | 6 ++-- sound/soc/codecs/wm8961.c | 6 ++-- sound/soc/codecs/wm8971.c | 12 ++++---- sound/soc/codecs/wm8974.c | 2 +- sound/soc/codecs/wm8978.c | 12 ++++---- sound/soc/codecs/wm8988.c | 8 +++--- sound/soc/codecs/wm8990.c | 8 +++--- sound/soc/codecs/wm8993.c | 24 ++++++++-------- sound/soc/codecs/wm8994.c | 34 ++++++++++++------------ sound/soc/codecs/wm9081.c | 18 ++++++------ sound/soc/codecs/wm9713.c | 13 +++++---- sound/soc/codecs/wm_hubs.c | 6 ++-- sound/soc/imx/imx-pcm-dma-mx2.c | 15 ++++++++++- sound/soc/imx/imx-pcm-fiq.c | 55 ++++++++++++++++++++------------------ sound/soc/imx/imx-ssi.c | 3 +- 47 files changed, 303 insertions(+), 276 deletions(-)

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Re: [alsa-devel] No shutdown code in soc-core.c
by ANISH KUMAR 19 Apr '10

19 Apr '10

>Did you see Mark's reply on dev_pm_ops ? >i.e. >static const struct dev_pm_ops soc_pm_ops = { > .suspend = soc_suspend, > .resume = soc_resume, > .poweroff = soc_poweroff, >}; I missed out Mark reply.However i checked now this structure is not there in my code base.As i mentioned in my previous mail the flow of events when restart takes place. Anyway i thought something is amiss in this struct platform_driver soc_driver that is why i am mailing to the experts here otherwise my problem is fixed by having the code as mentioned in my previous mail. I just wanted to know the reason?Or Am i missing something probably the concept of turning off the audio driver?? Kerenl version used: 2.6.29 Regards, anish

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Re: [alsa-devel] [regression] Sound not transferred to docking station after resume from StR
by Takashi Iwai 19 Apr '10

19 Apr '10

At Mon, 19 Apr 2010 14:39:44 +0200, Frans Pop wrote: > > On Monday 19 April 2010, Takashi Iwai wrote: > > Or how about the patch below? > > That fixes the problem too. > > Please add my: > Reported-and-tested-by: Frans Pop <elendil(a)planet.nl> Thanks for a quick check. But I found that the patch isn't fully correct. The patch below should suffice instead of the previous one. Could you check it again? thanks, Takashi --- diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9cbd80c..afbe314 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -527,14 +527,6 @@ static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) ad198x_power_eapd(codec); return 0; } - -static int ad198x_resume(struct hda_codec *codec) -{ - ad198x_init(codec); - snd_hda_codec_resume_amp(codec); - snd_hda_codec_resume_cache(codec); - return 0; -} #endif static struct hda_codec_ops ad198x_patch_ops = { @@ -547,7 +539,6 @@ static struct hda_codec_ops ad198x_patch_ops = { #endif #ifdef SND_HDA_NEEDS_RESUME .suspend = ad198x_suspend, - .resume = ad198x_resume, #endif .reboot_notify = ad198x_shutup, };

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[alsa-devel] alsaproject.org web site messd up
by Justin Mattock 19 Apr '10

19 Apr '10

when I go to alsaproject.org, II get redirected to some thing saying that alsa has expired? is this real or fake? -- Justin P. Mattock

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[alsa-devel] Alsa-devel missing messages?
by Alan Horstmann 19 Apr '10

19 Apr '10

Hi, I seem to be getting no mail from Alsa-devel since Sat 17th, this was the last: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-April/026981.html Checking those archives over the previous few days it seems that I have been missing some sporadically, eg http://mailman.alsa-project.org/pipermail/alsa-devel/2010-April/026992.html appears not to have come to me. Are others experiencing the same? Is there a known problem? Other mail-lists to the same email address have come through. Alan

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[alsa-devel] [regression] Sound not transferred to docking station after resume from StR
by Frans Pop 19 Apr '10

19 Apr '10

Scenario: - suspend laptop while *un*docked, i.e. with laptop speakers in use - dock laptop - resume With 2.6.33 and earlier the sound would always come from the external speakers connected to the docking station. But with 2.6.34-rc4 I get sound still coming from the laptop's internal speakers, which is not correct. System: HP 2510p running Debian stable ("Lenny") Sound driver: snd_hda_intel; codec: AD1981 Attached my kernel config for .34, and dmesg output for both .33 and .34 showing: - boot (docked) - undocking (visible as several USB disconnects) - suspend - resume after docking I don't see any significant changes in dmesg output during resume. I'll do a bisect, unless someone has some idea what may be causing this.

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Re: [alsa-devel] No shutdown code in soc-core.c
by ANISH KUMAR 19 Apr '10

19 Apr '10

Why this shutdown filed is not there in the struct shown below: static struct platform_driver soc_driver { .shutdown = soc_shutdown, //soc-core.c } This shutdown call this codec_dev->shutdown function In my case i am turning off the amplifier using this call.Which solved my problem of "TAK" sound during reset. I agree that soc_codec_close() is calling the shutdown function but in my case there is a call to set the path after this which is turning on the amplifier and suddenly if we reset the phone it is causing "TAK" sound.Probably because AMPLIFIER was on. Anyway why this function call soc_shutdown() is there in soc-core.c if no one is calling it??? I know there will be a redundancy if we have shutdown field in the struct shown above as soc_codec_close call also calls codec shutdown code. But in my case it is solving my problem.

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[alsa-devel] [PATCH 4/4] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 19 Apr '10

19 Apr '10

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[alsa-devel] [PATCH 3/4] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 19 Apr '10

19 Apr '10

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[alsa-devel] [PATCH 2/4] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 19 Apr '10

19 Apr '10

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[alsa-devel] HDA Intel ALC259
by Davide Canali 19 Apr '10

19 Apr '10

Recently I bought a HP G62 120SL laptop. It has a HDA Intel ALC259 sound card. I had to follow several steps to make the audio system work properly: I write you the information here to allow you to add better support to this sound card, since I see that many users are having the same issues with this model. I'm using ubuntu lucid lynx (beta), with kernel: $ uname -a Linux nebula 2.6.32-21-generic #32-Ubuntu SMP Fri Apr 16 08:09:38 UTC 2010 x86_64 GNU/Linux As I bought the laptop, the headphones were working but there was no sound from the integrated speakers. After installing the package linux-backports-modules-alsa-lucid-generic, the situation was switched: internal speakers started to work, but the headphones produced no sound. After messing up a bit with HDAAnalyzer, I found a modification that allows both internal speakers and the audio out jack to work. The diff file is: Diff for codec 0/0 (0x10ec0270): --- +++ @@ -200,17 +200,17 @@ Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0321401f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP Unsolicited: tag=0x04, enabled=1 Connection: 2 - 0x0c 0x0d* + 0x0c* 0x0d Node 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Connection: 7 0x18* 0x19 0x1a 0x1b 0x1d 0x0b 0x12 Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] The only issue remaining now is audio input: sound recorded using the internal microphone, or an external one connected to the mic in jack results in a very low quality and scratchy audio. Further information about this sound card and the ALSA version I'm using: $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.22.1. Compiled on Apr 14 2010 for kernel 2.6.32-21-generic (SMP). $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd4400000 irq 34 $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC259 Analog [ALC259 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: INTEL HDMI 0 [INTEL HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 7: INTEL HDMI 1 [INTEL HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 $ aplay -L pulse Playback/recording through the PulseAudio sound server front:CARD=Intel,DEV=0 HDA Intel, ALC259 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC259 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC259 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC259 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC259 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC259 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=Intel,DEV=0 HDA Intel, INTEL HDMI 0 HDMI Audio Output $ sudo lshw -class multimedia *-multimedia description: Audio device product: 5 Series/3400 Series Chipset High Definition Audio vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 05 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: driver=HDA Intel latency=0 resources: irq:34 memory:d4400000-d4403fff Please let me know if you need further information. Thank you. Davide

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[alsa-devel] alsa-info script changed to version V
by wirechief 19 Apr '10

19 Apr '10

The alsa-info script no longer works from the command line. Newer version detected: To view the ChangeLog, please visit http://www.alsa-project.org/alsa-info.sh.changelog ALSA-Info script has been updated to v Please re-run the script wirechief@wirechief-home:~$ bash ./alsa-info.sh ./alsa-info.sh: line 1: !DOCTYPE: No such file or directory ./alsa-info.sh: line 2: PUBLIC: command not found ./alsa-info.sh: line 3: syntax error near unexpected token `newline' ./alsa-info.sh: line 3: `" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">' Is this permanant ? -- Reach out and share your life, care for others,

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[alsa-devel] Problem about cross-compiling alsa-driver for arm-linux (can't get .ko module files)
by Jie Jiang 18 Apr '10

18 Apr '10

Hi Javier, I have been trying to cross compile alsa-driver-1.0.22 for my arm-linux board. After the configuration and make process, there are some *.o object files generated in the "modules" directory. But the desired *.ko kernel module files are not generated. When I tried to insert these object files into the running kernel via insmod command, the insmod command returns -1 and complains "Invalid module format!". Here is my configure command: CC=gcc ./configure --with-cross=arm-linux- --with-kernel=/work/63/ipcam/Kernel/linux-2.6.19.2-in-use --with-build=/work/63/ipcam/Kernel/linux-2.6.19.2-in-use --with-debug=full --enable-dynamic-minors --host=arm-linux --build=i386-linux --with-redhat=no --with-suse=no --prefix=/work/63/ipcam/tmp The configure outputs are as follows: checking for arm-linux-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... yes 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 arm-linux-ranlib... arm-linux-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... /work/63/ipcam/Filesystem/rootfs/mx27ipcam-alsa/ipcam-alsa-driver-1.0.22 checking cross compile... arm-linux- checking for directory with kernel source... /work/63/ipcam/Kernel/linux-2.6.19.2-in-use checking for directory with kernel build... /work/63/ipcam/Kernel/linux-2.6.19.2-in-use checking for kernel linux/version.h... yes checking for kernel linux/autoconf.h... yes checking for kernel version... 0.0.0 checking for GCC version... Kernel compiler: Used compiler: arm-linux-gcc (GCC) 4.1.1 *** 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... unknown checking for existing ALSA module... unknown checking for Red Hat kernel... no 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/0.0.0/misc checking for verbose procfs... on checking for verbose printk... on checking for debug level... full checking for processor type... unknown 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... no Creating a dummy <linux/log2.h>... 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... no Creating <linux/pm_qos_params.h>... 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... no Creating a dummy <linux/gpio.h>... checking for kernel linux/bug.h... no Creating a dummy <linux/bug.h>... checking for kernel linux/math64.h... no Creating <linux/math64.h>... checking for kernel linux/regulator/consumer.h... no checking for kernel linux/dmi.h... yes checking for kernel linux/bitrev.h... no Creating a dummy <linux/bitrev.h>... checking for kernel linux/hrtimer.h... yes checking for exported symbol dump_stack... grep: /work/63/ipcam/Kernel/linux-2.6.19.2-in-use/kernel/ksyms.c: No such file or directory no checking for kernel module symbol versions... unknown checking for has ioport support... unknown checking for PCI support in kernel... unknown checking for I2C driver in kernel... unknown checking for firmware loader... unknown checking for SMP... unknown checking for ISA PnP driver in kernel... yes checking for PnP driver in kernel... yes checking for Kernel ISA-PnP support... unknown checking for Kernel ISA-PnP module support... unknown 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... no checking for pci_dev_present... no checking for msleep... yes checking for msleep_interrupt... yes checking for msecs_to_jiffies... yes checking for tty->count is the atomic type... no checking for io_remap_pfn_range... yes checking for kcalloc... yes checking for kstrdup... yes checking for kstrndup... no checking for kzalloc... yes checking for pci_ioremap_bar... no 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 old kill_fasync... no checking for dma_addr_t... yes checking for MUTEX macros... no checking for driver extra-version... checking for driver version... 1.0.22 checking for dynamic minor numbers... yes checking for sequencer support... yes checking for OSS/Free emulation... yes checking for OSS PCM plugin system inclusion... yes checking for RTC support... unknown checking for HPET support... unknown checking for Procfs support... unknown checking for USB support... unknown checking for USB module support... unknown checking for gfp_t... yes checking for GFP_DMA32... yes checking for page_to_pfn... yes checking for init_utsname... yes checking for is_power_of_2... no checking for __ffs... yes checking for PnP suspend/resume... yes checking for builtin _Bool support... yes checking for x86-compatible PC... unknown checking for kernel PCMCIA checking for PCMCIA support... unknown checking for PCMCIA module support... unknown checking for PC9800 support in kernel... unknown checking for parallel port support... unknown checking for parallel port module support... unknown checking for power management... unknown checking for cards to compile driver for... all checking for additonal options to compile driver for... all 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: include/config.h is unchanged config.status: creating include/config1.h config.status: include/config1.h is unchanged 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 And the generated Makefile.conf is: # # Configuration Makefile for ALSA driver # Copyright (c) 1994-98 by Jaroslav Kysela <perex(a)perex.cz> # NEW_KBUILD = mainsrcdir = . MAINSRCDIR = /work/63/ipcam/Filesystem/rootfs/mx27ipcam-alsa/ipcam-alsa-driver-1.0.22 CROSS_COMPILE ?= arm-linux- ARCH ?= ifdef NEW_KBUILD CC = $(CROSS_COMPILE)gcc CPP = $(CROSS_COMPILE)gcc -E export CROSS_COMPILE ARCH else AS = $(CROSS_COMPILE)as LD = $(CROSS_COMPILE)ld CC = $(CROSS_COMPILE)gcc CPP = $(CROSS_COMPILE)gcc -E AR = $(CROSS_COMPILE)ar MODFLAGS = -DMODULE=1 CFLAGS_KERNEL = GENKSYMS = /sbin/genksyms -k 0.0.0 endif prefix = /work/63/ipcam/tmp exec_prefix = ${prefix} sysconfdir = ${prefix}/etc bindir = ${exec_prefix}/bin sbindir = ${exec_prefix}/sbin mandir = ${prefix}/share/man moddir = /lib/modules/0.0.0/misc moddir_tree = c_opts = -O2 -Wdeclaration-after-statement -Wno-pointer-sign -DLINUX m_opts = processor = unknown msmp = kaversion = 0.0.0 kversion = 0 kpatchlevel = 0 ksublevel = 0 kextraversion = CONFIG_SND_VERSION = 1.0.22 CONFIG_SND_KERNELDIR = /work/63/ipcam/Kernel/linux-2.6.19.2-in-use CONFIG_MODULES = y CONFIG_SND_MVERSION = CONFIG_ISAPNP_KERNEL = y CONFIG_PNP_KERNEL = y CONFIG_PROC_FS = CONFIG_PM = CONFIG_AC97_BUS = CONFIG_HAS_DMA = y ifndef NEW_KBUILD INCLUDE = -I$(TOPDIR)/include -I/work/63/ipcam/Kernel/linux-2.6.19.2-in-use/include -I/work/63/ipcam/Kernel/linux-2.6.19.2-in-use/include CFLAGS = $(MODFLAGS) $(INCLUDE) $(c_opts) -Wall -Wstrict-prototypes \ -fomit-frame-pointer -Wno-trigraphs -O2 -fno-strict-aliasing -fno-common \ -pipe -DALSA_BUILD -nostdinc -iwithprefix include ifdef CONFIG_SND_MVERSION CFLAGS += -DMODVERSIONS -include $(CONFIG_SND_KERNELDIR)/include/linux/modversions.h endif CPPFLAGS = $(MODFLAGS) $(INCLUDE) $(c_opts) -DALSA_BUILD -nostdinc -iwithprefix include MAKE += $(m_opts) SNDVERSIONS = $(TOPDIR)/include/sndversions.h DEXPORT = $(TOPDIR)/include/modules else # 2.6 kbuild EXTRA_CFLAGS += -DALSA_BUILD LINUXINCLUDE := -I$(MAINSRCDIR)/include $(LINUXINCLUDE) CPPFLAGS := -I$(MAINSRCDIR)/include $(CPPFLAGS) KBUILD_CPPFLAGS := -I$(MAINSRCDIR)/include $(KBUILD_CPPFLAGS) # 2.6.24 complains this override... # CFLAGS := -I$(MAINSRCDIR)/include $(CFLAGS) MAKE_ADDS = endif ------------------------------------------------------------------------------------- >From the above outputs, it seems that the configure script can't determine the correct kernel version of my target kernel, and this will prevent the "NEW_KBUILD" variable from being defined in Makefile.conf. Also in Makefile.conf, the "ARCH" variable is not set. What's the correct way to configure and cross-compile alsa-driver to generate the desired *.ko kernel modules? Am I missing anything? Regards, Jie Jiang

1 0

[alsa-devel] [REGRESSION bisected] Sound goes too fast due to commit 7b3a177b0
by Éric Piel 17 Apr '10

17 Apr '10

Hello, Since 2.6.34-rc*, I have a regression on alsa which prevents the sound to be played correctly. When playing, the music goes too fast, skipping some parts. Typically, it's very easy to reproduce by doing: time mplayer -endpos 30 sound-file-which-lasts-more-than-thirty-sec.mp3 If the wall clock is less than 30s, you have the bug. With my intel-hda (AD1981), it's reliably reproducible: it gives ~27s, instead of the normal ~31s. After bisection, it turns out that it is commit 7b3a177b0d4f92b3431b8dca777313a07533a710, aka "ALSA: pcm_lib: fix "something must be really wrong" condition" which caused this regression. Reverting it on top of 2.6.34-rc3+ fixes the problem. Let me know if you need more info, Cheers, Eric PS: For the info, the bisection was especially hard to do because about 80 alsa commits around the faulty one were all applied on kernel versions which did not boot on my laptop. On top of the first known bad I had to revert all of them, and run an "invert bisection" on it... Enjoy the bisection result ;-)

3 13

[alsa-devel] [PATCH 1/4] pxa: add namespace on ssp
by Haojian Zhuang 17 Apr '10

17 Apr '10

3 6

[alsa-devel] No shutdown code in soc-core.c
by ANISH KUMAR 16 Apr '10

16 Apr '10

I have a doubt regarding this soc_driver struct which is defined in soc-core.c file.Why not soc_shutdown code is added in this struct??

3 2

Re: [alsa-devel] No shutdown code in soc-core.c
by ANISH KUMAR 16 Apr '10

16 Apr '10

>What are you trying to do ? There was a "TAK" noise because of non-presence of this code. This soc_shutdown calls the respective shutdown code in codec for prevention of this kind of strange noise.In my code it is turning off the Amplifier because of which i no longer get the "TAK" sound. So i am just wondering why this code is not present in the struct i mentioned in my previous mail.

2 1

[alsa-devel] ALSA packages release 1.0.23
by Jaroslav Kysela 16 Apr '10

16 Apr '10

Hello all, ALSA packages version 1.0.23 (except alsa-oss & alsa-python) are available for download. The list of changes is available at this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.22_v1.0.23 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] ctxfi mixer initialization bug
by Daniel Koukola 16 Apr '10

16 Apr '10

Hi, There seems to be some problem with the mixer initialization in the ctxfi driver. After the module is loaded, the analog capture doesn't work even though the “Digit-IO” mixer switch is set to off. You have to set it to on and then to off again to make the capture work. This was tested with a Sound Blaster X-Fi XtremeMusic card and the driver that comes with Fedora's 2.6.32.10-90.fc12.x86_64 kernel (reported version: 1.0.21). Dan

1 0

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

16 Apr '10

The following changes since commit 5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5: Daniel Mack (1): ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream 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 As mentioned last time I posted these the updates here are very large for 2.6.34 but the bulk of them are improvements to the newly added i.MX3x FIQ driver which make it able to play 44.1k and 48k which is sufficiently worthwhile that (especially in a new driver) I think it's worth the risk of making a fairly large change now. Huang Weiyi (1): ASoC: wm2000: remove unused #include <linux/version.h> Sascha Hauer (5): ASoC: imx-ssi: honor IMX_SSI_DMA flag ASoC: imx-pcm-dma-mx2: restart DMA after an error ASoC: imx-ssi: Use a hrtimer in FIQ mode ASoC: imx-ssi: increase minimum periods to 4 ASoC: imx-ssi: do not call hrtimer_disable in trigger function sound/soc/codecs/wm2000.c | 1 - sound/soc/imx/imx-pcm-dma-mx2.c | 15 ++++++++++- sound/soc/imx/imx-pcm-fiq.c | 55 ++++++++++++++++++++------------------ sound/soc/imx/imx-ssi.c | 3 +- 4 files changed, 45 insertions(+), 29 deletions(-)

2 1

Re: [alsa-devel] asus xonar d1
by Clemens Ladisch 16 Apr '10

16 Apr '10

Lars Schotte wrote: > so basically you are saying that, even though the d2x has higher SNR, Slightly higher; this doesn't matter in practice. > there is no point in buying that card because she only has more > christmas tree lights? Some people need digital inputs or MIDI. And the lights are useful to keep the jacks apart on badly lit LAN parties (according to Asus' marketing), or to part casemodders from their money (not an official Asus statement). > the second thing i am curious about is how were you able to wirte the > driver for it, Mostly reverse engineering. C-Media gives out the register datasheet only under NDA. I never signed that NDA. Regards, Clemens

1 0

[alsa-devel] X-Fi ac3 passthrough support
by Daniel L. Miller 16 Apr '10

16 Apr '10

Is AC3/DTS passthrough available on the X-Fi at this time? Using iecset, I have been unable to toggle the audio mode. -- Daniel

1 0

[alsa-devel] Defect Patch File for AD1981HD
by Dark.＠ Zzz 15 Apr '10

15 Apr '10

Hi guys, I don't know if it's the right place to send a mail, but a little defect in one of the driver patch files is bothering me for a long time. The patch file is for analog device (hda-intel, patch_analog.c), under the section of Patch for HP nx6320, the author took out the mic/line loopback contorl part: "#if 0 FIXME: analog mic/line loopback doesn't work with my tests... #endif". However I accidentally found out that the original code for loopback contorl is actually working perfectly. The thing is I need to listen to mic loopback all the time when I need to record something. So every time I update my kernel I need to download the newest alsa driver, modify then compile it by myself. I don't know if others noticed it, but since the mic loopback contorl section is functional, I'm thinking maybe some of you can do something about the source code. Thank You! Dark.@Zzz

1 0

[alsa-devel] Problem with sound on an HP HDX9200
by EasyLife21 x3dre 15 Apr '10

15 Apr '10

Hi Development Team, I have been working with wirechief & Thingol into trying to get my speaker system working with alsabut I have run into problems making sound work on the internal Altec Lansing speakers. To start I use sidux distro and I am running kernel 2.6.33-2 which since using this kernel, all soundonboard ceased to work through the speakers. Though through the ports at the front of the laptopthe headphone port plays sound. Which is the only way to hear music from the laptop to my headphones. When I was using 2.6.29 kernels, I never had a problem with sound and all the speakers worked well.So I think something must have been deprecated in the 2.6.3x series of kernels, that has killed my sound output. Below is the pins I used from windows on configuring sound to work. Though what is happening now isthat I hear sounds using speaker-test -Dplug:surround51 -c6 -twav but the sound is split between ; 0 - Front Left = sound through headphones jack port on front of laptop 4 - Center = no sound 1 - Front Right = sound through headphones jack port on front of laptop 3 - Rear Right = sound through laptop speakers 2 - Rear Left = sound through laptop speakers 5 - LFE = no sound echo 0x0a 0x0221201f > /sys/class/sound/hwC0D0/user_pin_configsecho 0x0b 0x0221201f > /sys/class/sound/hwC0D0/user_pin_configsecho 0x0c 0x9017001f > /sys/class/sound/hwC0D0/user_pin_configsecho 0x0d 0x01112014 > /sys/class/sound/hwC0D0/user_pin_configsecho 0x0e 0x02a12021 > /sys/class/sound/hwC0D0/user_pin_configsecho 0x0f 0x01116011 > /sys/class/sound/hwC0D0/user_pin_configsecho 0x10 0x01113012 > /sys/class/sound/hwC0D0/user_pin_configsecho 0x11 0x01114010 > /sys/class/sound/hwC0D0/user_pin_configsecho 0x12 0x95a30020 > /sys/class/sound/hwC0D0/user_pin_configsecho 0x21 0x20441030 > /sys/class/sound/hwC0D0/user_pin_configs echo 1 > /sys/class/sound/hwC0D0/reconfig http://paste.debian.net/68912/ (old alsa-info), http://pastebin.ca/1861801 (windows .ini file), http://pastebin.com/4ckCBq0z (user_pin_configs) http://paste.debian.net/68997/ (new alsa-info) www.alsa-project.org/db/?f=4d48c4a628c0a739a395db728b599fc310a0c534 tia Dre :) _________________________________________________________________ http://clk.atdmt.com/UKM/go/195013117/direct/01/ We want to hear all your funny, exciting and crazy Hotmail stories. Tell us now

1 0

Re: [alsa-devel] asus xonar d1
by Clemens Ladisch 15 Apr '10

15 Apr '10

Lars Schotte wrote: > does the d2x have more inputs and outputs? The D2(X) has an analog loopback path, digital inputs with loopback and monitoring, MIDI input/output, and, no outputs but most important, lots of colorful LEDs. Regards, Clemens

1 0

Re: [alsa-devel] no soundcards detected on MCP51 rev a
by Gregory T Flynn 15 Apr '10

15 Apr '10

No dice, configured with '--with-debug=verbose': nuckin:~$ dmesg | grep hda hda_intel: Disable MSI for Nvidia chipset ALSA hda_intel.c:2518: chipset global capabilities = 0x770 ALSA hda_intel.c:909: codec_mask = 0xff ALSA hda_intel.c:702: azx_get_response timeout, polling the codec once: last cmd=0x000f0000 ALSA hda_intel.c:702: azx_get_response timeout, polling the codec once: last cmd=0x000f0000 ALSA hda_intel.c:712: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 ALSA hda_intel.c:1420: Codec #0 probe error; disabling it... ALSA hda_intel.c:1420: Codec #1 probe error; disabling it... ALSA hda_intel.c:1420: Codec #2 probe error; disabling it... ALSA hda_intel.c:1420: Codec #3 probe error; disabling it... ALSA hda_intel.c:1420: Codec #4 probe error; disabling it... ALSA hda_intel.c:1420: Codec #5 probe error; disabling it... ALSA hda_intel.c:1420: Codec #6 probe error; disabling it... ALSA hda_intel.c:1420: Codec #7 probe error; disabling it... ALSA hda_intel.c:909: codec_mask = 0xff ALSA hda_intel.c:751: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x000f0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0001 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0002 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0004 ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0001 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0002 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0004 ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_intel.c:1447: no codecs initialized upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.59 !!################################ !!Script ran on: Wed Apr 14 12:46:14 UTC 2010 !!Linux Distribution !!------------------ Arch Linux \r (\n) (\l) !!DMI Information !!--------------- Manufacturer: Micro-Star International Product Name: MEGA BOOK S430 !!Kernel Information !!------------------ Kernel release: 2.6.33-ARCH Operating System: GNU/Linux Architecture: x86_64 Processor: AMD Turion(tm) 64 X2 Mobile Technology TL-50 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.22.1 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- --- no soundcards --- !!PCI Soundcards installed in the system !!-------------------------------------- 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:10.1 0403: 10de:026c (rev a2) Subsystem: 1462:373d !!Loaded sound module options !!-------------------------- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 3 Apr 13 19:50 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Apr 13 19:50 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY aplay: device_list:223: no soundcards found... ARECORD arecord: device_list:223: no soundcards found... !!Amixer output !!------------- !!Alsactl output !!------------- --startcollapse-- --endcollapse-- !!All Loaded Modules !!------------------ Module ipt_MASQUERADE xt_state ipt_REJECT xt_tcpudp iptable_filter nf_nat_h323 nf_conntrack_h323 nf_nat_pptp nf_conntrack_pptp nf_conntrack_proto_gre nf_nat_proto_gre nf_nat_tftp nf_conntrack_tftp nf_nat_sip nf_conntrack_sip nf_nat_irc nf_conntrack_irc nf_nat_ftp nf_conntrack_ftp iptable_nat nf_nat nf_conntrack_ipv4 nf_conntrack nf_defrag_ipv4 ip_tables x_tables ipv6 ext2 joydev usbhid hid arc4 ecb rt73usb crc_itu_t rt2x00usb rt2x00lib led_class mac80211 cfg80211 rfkill snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_pcm_oss snd_mixer_oss snd_hda_intel snd_hda_codec snd_hwdep snd_pcm snd_timer snd ohci_hcd snd_page_alloc i2c_nforce2 edac_core battery ac ehci_hcd nvidia i2c_core edac_mce_amd usbcore video psmouse output forcedeth k8temp button thermal evdev serio_raw processor sg soundcore fuse rtc_cmos rtc_core rtc_lib ext4 mbcache jbd2 crc16 sr_mod sd_mod cdrom ata_generic pata_acpi pata_amd sata_nv libata scsi_mod !!ALSA/HDA dmesg !!------------------ ACPI: PCI Interrupt Link [LAZA] enabled at IRQ 23 HDA Intel 0000:00:10.1: PCI INT B -> Link[LAZA] -> GSI 23 (level, low) -> IRQ 23 hda_intel: Disable MSI for Nvidia chipset HDA Intel 0000:00:10.1: setting latency timer to 64 ALSA hda_intel.c:2518: chipset global capabilities = 0x770 usb 1-8: new high speed USB device using ehci_hcd and address 3 ALSA hda_intel.c:909: codec_mask = 0xff cfg80211: Calling CRDA to update world regulatory domain -- usbhid: USB HID core driver ALSA hda_intel.c:702: azx_get_response timeout, polling the codec once: last cmd=0x000f0000 ALSA hda_intel.c:702: azx_get_response timeout, polling the codec once: last cmd=0x000f0000 ALSA hda_intel.c:712: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 ALSA hda_intel.c:1420: Codec #0 probe error; disabling it... ALSA hda_intel.c:1420: Codec #1 probe error; disabling it... ALSA hda_intel.c:1420: Codec #2 probe error; disabling it... ALSA hda_intel.c:1420: Codec #3 probe error; disabling it... ALSA hda_intel.c:1420: Codec #4 probe error; disabling it... ALSA hda_intel.c:1420: Codec #5 probe error; disabling it... ALSA hda_intel.c:1420: Codec #6 probe error; disabling it... ALSA hda_intel.c:1420: Codec #7 probe error; disabling it... ALSA hda_intel.c:909: codec_mask = 0xff ALSA hda_intel.c:751: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x000f0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0001 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0002 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0xf0004 ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0000 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0001 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0002 ALSA hda_intel.c:815: send_cmd timeout: IRS=0x741, val=0x100f0004 ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_codec.c:1051: hda_codec: no AFG or MFG node found ALSA hda_intel.c:1447: no codecs initialized HDA Intel 0000:00:10.1: PCI INT B disabled EXT4-fs (sda4): mounted filesystem with ordered data mode > On Tue, Apr 13, 2010 at 7:19 PM, Daniel Chen <seven.steps(a)gmail.com> wrote: >> On Tue, Apr 13, 2010 at 5:06 PM, Gregory T Flynn <gflynn1(a)binghamton.edu> wrote: >>> Driver version: 1.0.21 >>> 00:10.1 0403: 10de:026c (rev a2) >>> Subsystem: 1462:373d >> >> Please try http://kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-snapsh… >> > > > > -- > - Greg Flynn > -- - Greg Flynn

2 7

[alsa-devel] USB transfer_buffer allocations on 64bit systems
by Daniel Mack 15 Apr '10

15 Apr '10

Hi, I was pointed to https://bugzilla.kernel.org/show_bug.cgi?id=15580 by Pedro Ribeiro, and unfortunately I'm pretty late in the game. I wasn't aware of that thread until yesterday. While the report is quite confusing, the reason seams pretty clear to me as I've been thru quite some time-demanding debugging of a very similar issue on Mac OS X. But I'm not totally sure whether we really hit the same issue here, so I'd like to have your opinions first. The problem is appearantly the way the transfer buffer is allocated in the drivers. In the snd-usb-caiaq driver, I used kzalloc() to get memory which works fine on 32bit systems. On x86_64, however, it seems that kzalloc() hands out memory beyond the 32bit addressable boundary, which the DMA controller of the 32bit PCI-connected EHCI controller is unable to write to or read from. Am I correct on this conclusion? Depending on the condition of the memory management, things might work or not, and especially right after a reboot, there's a better chance to get lower memory. The fix is to use usb_buffer_alloc() for that purpose which ensures memory that is suitable for DMA. And on x86_64, this also means that the upper 32 bits of the address returned are all 0's. If what I've stated is true, there are quite some more drivers affected by this issue. I collected a list of places where similar fixes are needed, and I can send patches if I get a thumbs-up. Pedro is currently testing a patch I sent out yesterday. Thanks, Daniel

11 88

[alsa-devel] [PATCH] ALSA: Add initial support for Thinkpad T410s HDA codec.
by Jens Taprogge 15 Apr '10

15 Apr '10

Dear ALSA Maintainers, attached please find a patch that adds support for at least the T410s HDA codec. Most likely it will also add support for the T410 and T510 based models. The patch was derived from Ideapad support. Support for the laptop's and docking-station output connectors as well as the docking-station microphone connector and the laptops internal devices has been tested. Since it has been developed without a data-sheet available, support for digital outputs and the laptop's microphone input may well be incorrect. Microphone mute functionality is not included: The microphone mute button seems to be reported through thinkpad_acpi key 0000101b. The mute button LED seems to be wired to thinkpad_acpi led number 15. Please apply. Signed-off-by: Jens Taprogge <jens.taprogge(a)taprogge.org> --- sound/pci/hda/patch_conexant.c | 153 ++++++++++++++++++++++++++++++++++++++- 1 files changed, 149 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 61682e1..2a488f6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -115,6 +115,7 @@ struct conexant_spec { unsigned int port_d_mode; unsigned int dell_vostro:1; unsigned int ideapad:1; + unsigned int thinkpad:1; unsigned int ext_mic_present; unsigned int recording; @@ -2031,6 +2032,9 @@ static void cxt5066_update_speaker(struct hda_codec *codec) /* Port D (HP/LO) */ pinctl = ((spec->hp_present & 2) && spec->cur_eapd) ? spec->port_d_mode : 0; + /* Mute if Port A is connected on Thinkpad */ + if (spec->thinkpad && (spec->hp_present & 1)) + pinctl = 0; snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); @@ -2211,6 +2215,50 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately + order is: external mic -> dock mic -> interal mic */ +static void cxt5066_thinkpad_automic(struct hda_codec *codec) +{ + unsigned int ext_present, dock_present; + + static struct hda_verb ext_mic_present[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0}, + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb dock_mic_present[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + + ext_present = snd_hda_jack_detect(codec, 0x1b); + dock_present = snd_hda_jack_detect(codec, 0x1a); + if (ext_present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else if (dock_present) { + snd_printdd("CXT5066: dock microphone detected\n"); + snd_hda_sequence_write(codec, dock_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + /* mute internal speaker if HP is plugged */ static void cxt5066_hp_automute(struct hda_codec *codec) { @@ -2223,7 +2271,8 @@ static void cxt5066_hp_automute(struct hda_codec *codec) /* Port D */ portD = snd_hda_jack_detect(codec, 0x1c); - spec->hp_present = !!(portA | portD); + spec->hp_present = !!(portA); + spec->hp_present |= portD ? 2 : 0; snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", portA, portD, spec->hp_present); cxt5066_update_speaker(codec); @@ -2274,6 +2323,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_thinkpad: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_thinkpad_automic(codec); + break; + } +} + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -2292,7 +2355,7 @@ static void cxt5066_set_mic_boost(struct hda_codec *codec) AC_VERB_SET_AMP_GAIN_MUTE, AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | cxt5066_analog_mic_boost.items[spec->mic_boost].index); - if (spec->ideapad) { + if (spec->ideapad | spec->thinkpad) { /* adjust the internal mic as well...it is not through 0x17 */ snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -2780,6 +2843,64 @@ static struct hda_verb cxt5066_init_verbs_ideapad[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_thinkpad[] = { + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */ + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port A: HP, Amp */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: Mic Dock */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port C: Mic */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port D: HP Dock, Amp */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, /* default to internal mic */ + + /* Audio input selector */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x2}, + {0x17, AC_VERB_SET_CONNECT_SEL, 1}, /* route ext mic */ + + /* SPDIF route: PCM */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x0}, + + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* internal microphone */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* enable int mic */ + + /* EAPD */ + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* enable unsolicited events for Port A, B, C and D */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5066_init_verbs_portd_lo[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { } /* end */ @@ -2798,6 +2919,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_vostro_automic(codec); else if (spec->ideapad) cxt5066_ideapad_automic(codec); + else if (spec->thinkpad) + cxt5066_thinkpad_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -2819,20 +2942,22 @@ static int cxt5066_olpc_init(struct hda_codec *codec) } enum { - CXT5066_LAPTOP, /* Laptops w/ EAPD support */ + CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ + CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_MODELS }; static const char *cxt5066_models[CXT5066_MODELS] = { - [CXT5066_LAPTOP] = "laptop", + [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", + [CXT5066_THINKPAD] = "thinkpad", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -2843,6 +2968,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD), {} }; @@ -2950,6 +3076,22 @@ static int patch_cxt5066(struct hda_codec *codec) /* input source automatically selected */ spec->input_mux = NULL; break; + case CXT5066_THINKPAD: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_thinkpad_event; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->init_verbs[0] = cxt5066_init_verbs_thinkpad; + spec->thinkpad = 1; + spec->port_d_mode = PIN_OUT; + spec->mic_boost = 2; /* default 20dB gain */ + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; } return 0; @@ -2969,6 +3111,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", .patch = patch_cxt5066 }, + { .id = 0x14f15069, .name = "CX20585", + .patch = patch_cxt5066 }, {} /* terminator */ }; @@ -2977,6 +3121,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); +MODULE_ALIAS("snd-hda-codec-id:14f15069"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -- 1.7.0.4

2 1

Re: [alsa-devel] [Cooker] Totem playing videos at fast-forward speed ?
by Herton Ronaldo Krzesinski 15 Apr '10

15 Apr '10

Em Ter 13 Abr 2010, às 18:57:51, Éric Piel escreveu: > Op 07-04-10 00:23, Herton Ronaldo Krzesinski schreef: > > Em Ter 06 Abr 2010, às 18:52:08, Éric Piel escreveu: > >> Op 06-04-10 22:29, Frank Griffin schreef: > >>> Anssi Hannula wrote: > >>>> Try mplayer: > >>>> mplayer file > >>>> > >>> > >>> Same speedup symptom as Totem > >>> > >>>> and mplayer without audio: > >>>> mplayer -ao null file > >>>> > >>> > >>> Back to normal speed. > >>> > >> > >> Hi, > >> I'm experiencing a similar problem on a machine with an intel HDA. It > >> happens only with a kernel 2.6.34-rc*. With a vanilla 2.6.33, everything > >> is fine. Is your kernel the normal cooker kernel? > >> > >> Does anyone know if there are some update to alsa in this kernel > >> compared to the vanilla kernel? > > > > yes, there is some pcm_lib changes applied from 2.6.34-rc* because of > > pulseaudio. Would be good to know if reverting these changes on 2.6.34 fixes > > the problem: > > ALSA: pcm_lib - fix xrun functionality > > ALSA: pcm_native - fix runtime->boundary calculation > > ALSA: pcm_lib - return back hw_ptr_interrupt > > ALSA: pcm_core: Fix wake_up() optimization > > ALSA: pcm_lib - fix wrong delta print for jiffies check > > ALSA: pcm_lib: fix "something must be really wrong" condition > > ALSA: pcm_lib - optimize wake_up() calls for PCM I/O > > ALSA: pcm_lib - cleanup & merge hw_ptr update functions > > ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions > > ALSA: pcm_lib.c - convert second xrun_debug() parameter to use defines > Hi, > I've done a bisection on a vanilla kernel, and it turned out to be > 7b3a177b0d4f92b3431b8dca777313a07533a710, aka ALSA: pcm_lib: fix > "something must be really wrong" condition . > > So I'd recommend to drop it from the mandriva kernel, and get confirmed > it is fixed. Ok adding alsa-devel to CC, may be the issue is known. Eric, from later messages you only experience the problem when bdl_pos_adj isn't set to 1 right? Anyway Frank said that his bdl_pos_adj is set to 32 on two machines (hda-intel), and on one of them the problem happens, while in the other not. I guess we could have a problem then on commit 7b3a177b0d4f92b3431b8dca777313a07533a710 > > See you, > Eric > -- []'s Herton

2 1

[alsa-devel] imx-ssi fixes
by Sascha Hauer 15 Apr '10

15 Apr '10

Just a resend of the patches I sent yesterday with an updated commit message. Sascha

3 4

[alsa-devel] [PATCH] ALSA: hda - via - fix master mute and automute with VT1812/VT2002P
by Herton Ronaldo Krzesinski 14 Apr '10

14 Apr '10

On VIA VT1812/VT2002P, the "Master Front Playback Switch" doesn't mute first line-out. This is an issue, for example I saw a laptop with VT1812 and only one line-out (main speaker) that doesn't mute main speaker, because the master switch doesn't act on first line-out, and first line-out nid in array is assigned to speaker as expected by autoconfig code. But there is one more issue also: main switches and automute code deal with mute on same Amp-Outs, which can cause conflicts, thus if you mute "Master Front Playback Switch" and remove headphone for example, it will not respect the mixer setting. To solve it, we can change the pin type instead of muting Amp-Out, which is done here. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_via.c | 23 +++++++++-------------- 1 files changed, 9 insertions(+), 14 deletions(-) Hi, I'm attaching the codec file of the laptop with the issue, this is kernel log relevant info: autoconfig: line_outs=1 (0x24/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=1 (0x33/0x0/0x0/0x0/0x0) mono: mono_out=0x0 inputs: mic=0x29, fmic=0x2b, line=0x0, fline=0x0, cd=0x0, aux=0x0 diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9ddc373..d6e6f40 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -295,8 +295,6 @@ static int analog_input_switch_put(struct snd_kcontrol *kcontrol, .put = analog_input_switch_put, \ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -static void via_hp_bind_automute(struct hda_codec *codec); - static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -343,13 +341,10 @@ static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, snd_hda_codec_amp_stereo( codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); - /* unmute */ - via_hp_bind_automute(codec); - } else { if (lmute) { /* Mute all left channels */ - for (i = 1; i < spec->autocfg.line_outs; i++) + for (i = 0; i < spec->autocfg.line_outs; i++) snd_hda_codec_amp_update( codec, spec->autocfg.line_out_pins[i], @@ -364,7 +359,7 @@ static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, } if (rmute) { /* mute all right channels */ - for (i = 1; i < spec->autocfg.line_outs; i++) + for (i = 0; i < spec->autocfg.line_outs; i++) snd_hda_codec_amp_update( codec, spec->autocfg.line_out_pins[i], @@ -2129,18 +2124,18 @@ static void via_hp_bind_automute(struct hda_codec *codec) if (!spec->hp_independent_mode) { /* Mute Line-Outs */ for (i = 0; i < spec->autocfg.line_outs; i++) - snd_hda_codec_amp_stereo( - codec, spec->autocfg.line_out_pins[i], - HDA_OUTPUT, 0, - HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + snd_hda_codec_write(codec, + spec->autocfg.line_out_pins[i], + 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + hp_present ? 0 : PIN_OUT); if (hp_present) present = hp_present; } /* Speakers */ for (i = 0; i < spec->autocfg.speaker_outs; i++) - snd_hda_codec_amp_stereo( - codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_write(codec, spec->autocfg.speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + present ? 0 : PIN_OUT); } -- 1.7.0.4

2 3

Re: [alsa-devel] asus xonar d1
by Clemens Ladisch 14 Apr '10

14 Apr '10

Lars Schotte wrote: > > What controls are missing? > > hmm. i dont know, i thought there will be more on such a card. because > on creatives and hd audio usually you have tons of controls and there > only 5 or so. The D1/DX hardware doesn't have that many controls because there aren't so many inputs or outputs. > and 3 of them is 7.1 output so its a bit confusing. There is one "Master" control with 8 channels. How this is displayed is up to the mixer application. (alsamixer isn't ideal in this respect, older versions even less so.) > to fedora 13 the controls are noticably faster, to my old debian lenny > even w/ the 2.6.32 kernel they lag. on fedora 13 w/ 2.6.33 they are > fast as usual, seems like you fixed that already ;-) . Yes, the driver now doesn't write registers that have not changed. In theory, this shouldn't make too much of a difference unless the mixer tries to set the same volume repeatedly. Regards, Clemens

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[alsa-devel] ALSA application programming: route audio from one PCM to another
by Stefan Schoenleitner 14 Apr '10

14 Apr '10

Hi, I finally managed to write an ALSA I/O plugin that does what I want. The plugin supports both playback and capture. Now I would like to write a simple audio application that takes audio samples * from the microphone and plays it back on my plugin and * from the plugin (capture) and plays it back on the speakers Hence as long as the application is running, it should do the above. * Is there a special ALSA way to route audio from one PCM to another ? * If not, I suppose it would just work if I open the plugin PCM and the hw PCM at the same time and copy audio frames between them ? cheers, stefan

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[alsa-devel] Procfs cleanup in ALSA core
by Takashi Iwai 14 Apr '10

14 Apr '10

Hi, the another fixes are clean-ups of procfs interface in ALSA core. The patches below are found in sound git tree topic/core-cleanup branch. (This branch also contains the BKL-cleanup branch, too). [PATCH 1/5] ALSA: info - Use standard types for info callbacks [PATCH 2/5] ALSA: info - Check file position validity in common layer [PATCH 3/5] ALSA: info - Implement common llseek for binary mode [PATCH 4/5] ALSA: core - Define llseek fops [PATCH 5/5] ALSA: Update the documentation for changes of proc files Takashi

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[alsa-devel] BKL removal from ALSA core
by Takashi Iwai 14 Apr '10

14 Apr '10

Hi, I recently cleaned up ALSA core code a bit. The first attemp is the removal of BKL, at least, in ALSA side. There are still in sound_core.c and OSS drivers, but who cares ;) The patches below are in sound git tree topic/bkl branch. [PATCH 1/3] ALSA: pcm - Remove BKL from async callback [PATCH 2/3] ALSA: info - Remove BKL [PATCH 3/3] ALSA: Remove BKL from open multiplexer Takashi

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[alsa-devel] [RFC] ALSA: pcm_lib - read/write IO improvement
by Jarkko Nikula 14 Apr '10

14 Apr '10

Hi I've noticed a problem with the ALSA read/write IO as it tries to pad buffer as full/empty as possible after each wake-up. Then there is no wake-up during the next period interrupt since avail < avail_min and effectively there could be very short time left for wake-up on an interrupt after that. What I've seen on OMAP3 and Intel HDA that typically avail_max is only a frame or two less than size of two periods. I.e. very near to xrun when using a buffer with two periods. I came up with two ideas. The first one won't do padding and the second one which does the padding as currently but checks are there available frames in the next period and does a wake-up if necessary (even if the avail < avail_min). With the first one, the avail_max is typically a few frames more than the size of one period. With the second one it's a few frames less than size of one period. So by not doing the padding the distance to xrun is about the same and the wake-up semantics remains the same. I understood from the thread below that not all applications like if the wake-up happens before avail >= avail_min what would happen with the second idea. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-December/023986.h… -- Jarkko ========================== RFC 1 ========================== From: Jarkko Nikula <jhnikula(a)gmail.com> Subject: [PATCH] ALSA: pcm_lib - Do not copy less than avail_min amount of frames in RW IO Copying less than avail_min amount of frames after each wake-up, i.e. by padding buffer as full/empty as possible by following the running HW pointer can actually cause xrun to be more possible than without padding. Padding causes that the avail is less than the avail_min when the next period interrupt occurs and thus no wake-up at that time. However, period interrupt following that one will see a situation where the avail is near the size of two periods. I.e. there could be only few frames left before xrun would happen if the buffer size is only two periods. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/core/pcm_lib.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a2ff861..a47a010 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1821,7 +1821,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); avail = snd_pcm_playback_avail(runtime); - if (!avail) { + if (avail < runtime->control->avail_min) { if (nonblock) { err = -EAGAIN; goto _end_unlock; @@ -2043,7 +2043,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (runtime->status->state == SNDRV_PCM_STATE_RUNNING) snd_pcm_update_hw_ptr(substream); avail = snd_pcm_capture_avail(runtime); - if (!avail) { + if (avail < runtime->control->avail_min) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); -- ========================== RFC 2 ========================== From: Jarkko Nikula <jhnikula(a)gmail.com> Subject: [PATCH] ALSA: pcm_lib - Do wake-up and copy if the next period has available frames It is possible that the next period has only a few frames before xrun would happen after next interrupt but wakeup does not happen now since the avail is less than the avail_min threshold. This patch adds a wake-up and copy check if the appl_ptr is less than end of next period so that application can transfer data already now. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/core/pcm_lib.c | 17 +++++++++++++---- 1 files changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a2ff861..424873d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -268,7 +268,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) int snd_pcm_update_state(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { - snd_pcm_uframes_t avail; + snd_pcm_uframes_t avail, next_end; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) avail = snd_pcm_playback_avail(runtime); @@ -287,7 +287,11 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) + next_end = runtime->hw_ptr_interrupt + 2 * runtime->period_size - 1; + if (next_end >=runtime->boundary) + next_end -= runtime->boundary; + if (avail >= runtime->control->avail_min || + runtime->control->appl_ptr <= next_end) wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); return 0; } @@ -1707,7 +1711,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, int is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; wait_queue_t wait; int err = 0; - snd_pcm_uframes_t avail = 0; + snd_pcm_uframes_t avail = 0, next_end; long tout; init_waitqueue_entry(&wait, current); @@ -1750,7 +1754,12 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, avail = snd_pcm_playback_avail(runtime); else avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->control->avail_min) + next_end = runtime->hw_ptr_interrupt + + 2 * runtime->period_size - 1; + if (next_end >=runtime->boundary) + next_end -= runtime->boundary; + if (avail >= runtime->control->avail_min || + runtime->control->appl_ptr <= next_end) break; } _endloop: --

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[alsa-devel] Help needed
by Akhilesh Soni 14 Apr '10

14 Apr '10

Hi, I'm using Osprey-440 4 channel Capture card in Centos 5.4 distribution with kernel 2.6.18-164.15.1.el5 #1 SMP Wed Mar 17 11:30:06 EDT 2010 x86_64 x86_64 x86_64 GNU/Linux I'm able to load the bttv driver for video capture and it created /dev/video0 /dev/video1 /dev/video2 /dev/video3 devices which I'm able to capture as well. But when I do modprobe snd-bt87x i'm unable to get the audio capture devices either /dev/audio or /dev/dsp Is there any parameters to be given in inserting this module for the card. Please guide me to resolve this issue. Is there any place where I can get the latest driver source for this card Thanks in advance. Akhilesh

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[alsa-devel] asus xonar d1
by Lars Schotte 14 Apr '10

14 Apr '10

i finally got the asus xonar d1 and installed the latest kernel from the backports for debian lenny and the card doesnt work at all and has unbelievably less controls in the mixer, so my guess is that the driver is ok, but alsa can not operate it. i am keeping on trying. i am about to download fedora 13 beta to test it there, maybe it helps. while the download is runing i am continuing to try and look what could be the problem. -- ------------------- Lars Schotte @ romy

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[alsa-devel] Laggy mixer with Xonar DX snd_virtuoso
by Flávio Martins 14 Apr '10

14 Apr '10

Hello Alsa team, I have a Xonar DX with a laggy volume control, you might have guessed already. Sound will interrupt if the volume change is big like from 100% to 50%, it will only skip a bit if the volume change is smaller. The interruptions/silence last more than 5 seconds easily. I can actually see the mixer slider going up and down on its own. I am seeking for your help and i'm using 1.0.22.1. Here goes the alsa-info.sh output, some weird things going on in dmesg http://www.alsa-project.org/db/?f=ee911fe73baa60aac51c62ff787ab8b8bd8565e8 Flávio Martins

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[alsa-devel] no soundcards detected on MCP51 rev a
by Gregory T Flynn 14 Apr '10

14 Apr '10

$ alsaconf will detect the soundcard but $ aplay -l aplay: device_list:223: no soundcards found... ALC883 Realtek lspci: 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) Alsa info result: upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.59 !!################################ !!Script ran on: Tue Apr 13 20:50:40 UTC 2010 !!Linux Distribution !!------------------ Arch Linux \r (\n) (\l) !!DMI Information !!--------------- Manufacturer: Micro-Star International Product Name: MEGA BOOK S430 !!Kernel Information !!------------------ Kernel release: 2.6.33-ARCH Operating System: GNU/Linux Architecture: x86_64 Processor: AMD Turion(tm) 64 X2 Mobile Technology TL-50 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- --- no soundcards --- !!PCI Soundcards installed in the system !!-------------------------------------- 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:10.1 0403: 10de:026c (rev a2) Subsystem: 1462:373d !!Loaded sound module options !!-------------------------- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 3 Apr 13 14:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Apr 13 14:37 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY aplay: device_list:223: no soundcards found... ARECORD arecord: device_list:223: no soundcards found... !!Amixer output !!------------- !!Alsactl output !!------------- --startcollapse-- --endcollapse-- !!All Loaded Modules !!------------------ Module ipt_MASQUERADE xt_state ipt_REJECT xt_tcpudp iptable_filter nf_nat_h323 nf_conntrack_h323 nf_nat_pptp nf_conntrack_pptp nf_conntrack_proto_gre nf_nat_proto_gre nf_nat_tftp nf_conntrack_tftp nf_nat_sip nf_conntrack_sip nf_nat_irc nf_conntrack_irc nf_nat_ftp nf_conntrack_ftp iptable_nat nf_nat nf_conntrack_ipv4 nf_conntrack nf_defrag_ipv4 ip_tables x_tables ipv6 ext2 joydev usbhid hid arc4 ecb rt73usb crc_itu_t rt2x00usb rt2x00lib led_class mac80211 cfg80211 rfkill snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_hda_intel snd_pcm_oss snd_hda_codec snd_hwdep snd_pcm snd_mixer_oss ohci_hcd snd_timer video ac battery ehci_hcd snd snd_page_alloc nvidia output usbcore thermal i2c_nforce2 i2c_core button k8temp processor forcedeth psmouse sg edac_core edac_mce_amd evdev serio_raw soundcore fuse rtc_cmos rtc_core rtc_lib ext4 mbcache jbd2 crc16 sr_mod sd_mod cdrom ata_generic pata_acpi pata_amd sata_nv libata scsi_mod !!ALSA/HDA dmesg !!------------------ ACPI: PCI Interrupt Link [LAZA] enabled at IRQ 23 HDA Intel 0000:00:10.1: PCI INT B -> Link[LAZA] -> GSI 23 (level, low) -> IRQ 23 hda_intel: Disable MSI for Nvidia chipset HDA Intel 0000:00:10.1: setting latency timer to 64 usb 1-8: new high speed USB device using ehci_hcd and address 3 -- usbhid: USB HID core driver hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 hda-intel: Codec #0 probe error; disabling it... hda-intel: Codec #1 probe error; disabling it... hda-intel: Codec #2 probe error; disabling it... hda-intel: Codec #3 probe error; disabling it... hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x000f0000 hda-intel: no codecs initialized HDA Intel 0000:00:10.1: PCI INT B disabled EXT4-fs (sda4): mounted filesystem with ordered data mode -- - Greg Flynn

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[alsa-devel] alsa-lib pcm_share.c snd_pcm_share_open() bug ?
by Mathieu Desnoyers 13 Apr '10

13 Apr '10

Hi, I was looking at snd_pcm_share_open() in alsa-lib-1.0.22, and stumbled on the following line: pcm->monotonic = pcm->monotonic; what is that supposed to do ? There is clearly something wrong there. Thanks, Mathieu -- Mathieu Desnoyers Operating System Efficiency R&D Consultant EfficiOS Inc. http://www.efficios.com

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[alsa-devel] [PATCH] Use thread-safe locale functions
by Rémi Denis-Courmont 13 Apr '10

13 Apr '10

setlocale() is not thread-safe. It can actually trigger a crash if another thread uses locale informations at the same time in the process. Library code should use POSIX newlocale/duplocale/uselocale/freelocale instead. Those functions only change the locale data for the calling thread. --- src/conf.c | 16 +++++++--------- 1 files changed, 7 insertions(+), 9 deletions(-) diff --git a/src/conf.c b/src/conf.c index 570c90f..ce252ab 100644 --- a/src/conf.c +++ b/src/conf.c @@ -499,22 +499,20 @@ static int safe_strtod(const char *str, double *val) { char *end; double v; - char *saved_locale; - char locstr[64]; /* enough? */ + locale_t saved_locale, c_locale; int err; if (!*str) return -EINVAL; - saved_locale = setlocale(LC_NUMERIC, NULL); - if (saved_locale) { - snprintf(locstr, sizeof(locstr), "%s", saved_locale); - setlocale(LC_NUMERIC, "C"); - } + c_locale = newlocale(LC_NUMERIC_MASK, "C", 0); + saved_locale = uselocale(c_locale); errno = 0; v = strtod(str, &end); err = -errno; - if (saved_locale) - setlocale(LC_NUMERIC, locstr); + if (c_locale != (locale_t)0) { + uselocale(saved_locale); + freelocale(c_locale); + } if (err) return err; if (*end) -- 1.7.0.4

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[alsa-devel] (no subject)
by Sascha Hauer 13 Apr '10

13 Apr '10

Hi, Here are further fixes for the i.MX ssi driver. Mark, please let me know whether I should combine 1/2 with the patch that actually broke fiq support. Sascha

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 13 Apr '10

13 Apr '10

The following changes since commit 5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5: Daniel Mack (1): ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream 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 The updates from Sascha are bug fixes for i.MX2x and a vast improvement in the performance of the FIQ based "DMA" on i.MX3x which allows it to handle sample rates over 16kHz. This code is newly added in 2.6.34. Huang Weiyi (1): ASoC: wm2000: remove unused #include <linux/version.h> Sascha Hauer (3): ASoC: imx-ssi: honor IMX_SSI_DMA flag ASoC: imx-pcm-dma-mx2: restart DMA after an error ASoC: imx-ssi: Use a hrtimer in FIQ mode sound/soc/codecs/wm2000.c | 1 - sound/soc/imx/imx-pcm-dma-mx2.c | 15 ++++++++++++- sound/soc/imx/imx-pcm-fiq.c | 45 ++++++++++++++++++-------------------- sound/soc/imx/imx-ssi.c | 3 +- 4 files changed, 37 insertions(+), 27 deletions(-)

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

13 Apr '10

The following changes since commit 53a61d967a480307db4c8f7950ddb8bb1a0fda9b: Mark Brown (1): Merge branch 'for-2.6.34' 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 Bill Gatliff (1): ASoC: Use SNDRV_PCM_RATE_8000_96000 macro for WM8731 Marek Vasut (2): ASoC: WM8750: Convert to new API ASoC: Zipit Z2 WM8750 ASoC driver sound/soc/codecs/wm8731.c | 6 +- sound/soc/codecs/wm8750.c | 295 ++++++++++++++++++--------------------- sound/soc/pxa/Kconfig | 8 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/spitz.c | 43 +++++- sound/soc/pxa/z2.c | 246 ++++++++++++++++++++++++++++++++ sound/soc/s3c24xx/jive_wm8750.c | 5 - 7 files changed, 432 insertions(+), 173 deletions(-) create mode 100644 sound/soc/pxa/z2.c

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 13 Apr '10

13 Apr '10

The following changes since commit fb48e3c6a4d8888aff61fbf567aadac7d206e973: Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak 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 This is rather large for a stable update due to the inclusion of Daniel's backported fix for corruption of DMA data due to use of dma_data by both capture and playback streams (which is large but straightforward) and my updates for WM8994, particularly DC servo handling. The WM8994 updates mostly affect that part (which was only introduced in 2.6.34) and are strongly recommended for all users, individually they are all relatively straightforward. Daniel Mack (1): ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream Janusz Krzysztofik (1): ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code Mark Brown (5): ASoC: Avoid wraparound in wm_hubs DC servo correction ASoC: Support second DC servo readback method for wm_hubs ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices ASoC: Only do WM8994 bias off transition from standby include/sound/soc-dai.h | 18 ++++++++- include/sound/soc.h | 1 + sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 6 +- sound/soc/codecs/wm8994.c | 56 ++++++++++++++----------- sound/soc/codecs/wm_hubs.c | 83 ++++++++++++++++++++++-------------- sound/soc/codecs/wm_hubs.h | 1 + sound/soc/davinci/davinci-i2s.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 3 +- sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/imx/imx-pcm-dma-mx2.c | 8 +++- sound/soc/imx/imx-ssi.c | 7 ++- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 3 +- sound/soc/omap/omap-pcm.c | 21 +++++---- sound/soc/pxa/pxa-ssp.c | 23 +++++----- sound/soc/pxa/pxa2xx-ac97.c | 17 +++++-- sound/soc/pxa/pxa2xx-i2s.c | 7 ++- sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/s3c24xx/s3c-ac97.c | 21 +++++---- sound/soc/s3c24xx/s3c-dma.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 13 ++++-- sound/soc/s3c24xx/s3c-pcm.c | 7 ++- sound/soc/s3c24xx/s3c24xx-i2s.c | 19 ++++---- sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/s6000/s6000-pcm.c | 40 +++++++++++++----- 26 files changed, 240 insertions(+), 138 deletions(-)

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[alsa-devel] [PATCH] ALSA: hda - Add quirk and model 'vaio' for ALC269-based Sony Vaio E Series
by Luca Bruno 13 Apr '10

13 Apr '10

Vaio E Series laptops need a slightly modified config for their ALC269. This patch adds a new 'vaio' model, plus initial quirk for them. It fixes alsa-bugs #4941, #4934 and #4945 Signed-off-by: Luca Bruno <lucab(a)debian.org> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++++- 2 files changed, 24 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1d38b0d..416f1f5 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -87,6 +87,7 @@ ALC269 eeepc-p901 ASUS Eeepc P901 S101 fujitsu FSC Amilo lifebook Fujitsu Lifebook S6420 + vaio Sony Vaio (E Series) auto auto-config reading BIOS (default) ALC662/663/272 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22b7c94..c2b78a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -137,6 +137,7 @@ enum { ALC269VB_DMIC, ALC269_FUJITSU, ALC269_LIFEBOOK, + ALC269_VAIO, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -13546,6 +13547,11 @@ static struct hda_verb alc269_quanta_fl1_verbs[] = { { } }; +static struct hda_verb alc269_vaio_verbs[] = { + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, + { } +}; + static struct hda_verb alc269_lifebook_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -14002,6 +14008,7 @@ static const char *alc269_models[ALC269_MODEL_LAST] = { [ALC269_DMIC] = "laptop-dmic", [ALC269_FUJITSU] = "fujitsu", [ALC269_LIFEBOOK] = "lifebook", + [ALC269_VAIO] = "vaio", [ALC269_AUTO] = "auto", }; @@ -14051,7 +14058,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), + SND_PCI_QUIRK(0x104d, 0x9071, "SONY Vaio", ALC269_VAIO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), @@ -14073,6 +14080,21 @@ static struct alc_config_preset alc269_presets[] = { .channel_mode = alc269_modes, .input_mux = &alc269_capture_source, }, + [ALC269_VAIO] = { + .mixers = { alc269_laptop_mixer }, + .cap_mixer = alc269_laptop_digital_capture_mixer, + .init_verbs = { alc269_init_verbs, + alc269_vaio_verbs, + alc269_laptop_dmic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .unsol_event = alc269_laptop_unsol_event, + .setup = alc269_laptop_dmic_setup, + .init_hook = alc269_laptop_inithook, + }, [ALC269_QUANTA_FL1] = { .mixers = { alc269_quanta_fl1_mixer }, .init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs }, -- 1.7.0.3

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Re: [alsa-devel] Bug#536896: linux-image-2.6.26-2-amd64: audio popping with gstreamer using apps when using amd64 kernel and i386 userspace
by Ben Hutchings 12 Apr '10

12 Apr '10

[Earlier messages can be found at http://bugs.debian.org/536896 ] On Mon, 2010-04-12 at 01:08 +0800, James Andrewartha wrote: > On Thu, 25 Feb 2010, Moritz Muehlenhoff wrote: > > > On Tue, Jul 14, 2009 at 09:57:54PM +0800, James Andrewartha wrote: > > > Package: linux-image-2.6.26-2-amd64 > > > Version: 2.6.26-17 > > > Severity: normal > > > > > > All audio from gstreamer-using apps (eg Totem and Banshee) pops when I use a > > > 2.6.{25,26,28,30} amd64 kernel and i386 userspace. 2.6.24-etchnhalf.1-amd64 > > > does not have this problem, nor does 2.6.26-2-686. It doesn't occur when using > > > non-gstreamer apps like mplayer or mpd, nor with various gst-launch pipelines > > > suggested by #gstreamer like gst-launch playbin uri=file:///home/trs80/a.mp3 > > > > Hi, > > The next release of Debian (6.0, code name Squeeze) will be based > > on 2.6.32. Please test the current 2.6.32 from unstable/testing and tell > > us whether the problem persists. If so, we should report it upstream > > to the kernel.org developers. > > > > The 2.6.32 kernel is available from packages.debian.org and can > > be installed in both Debian stable, testing and unstable > > installations. > > I've bisected it, and the bad patch is 130755108ba03461f69da990e54e02a254accd23: Thanks for taking the time to do this. > Author: Takashi Iwai <tiwai(a)suse.de> 2008-01-09 02:08:14 > Committer: Jaroslav Kysela <perex(a)perex.cz> 2008-02-01 01:29:47 > Parent: d948035a928400ae127c873fbf771389bee18949 ([ALSA] Remove PCM xfer_align sw params) > > [ALSA] PCM - clean up snd_pcm_lib_read/write > > Introduce a common helper function for snd_pcm_lib_read and snd_pcm_lib_write > for cleaning up the code. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> > > I don't know anything about this code, but I'm happy to deal with upstream > if you'd prefer me to. The above commit is supposed to be cleanup, but it has at least one semantic change: snd_pcm_mmap_control::avail_min no longer applies to non-blocking file handles. I don't know whether this is was an intentional or unintentional change, but it wasn't commented. I also don't know whether this can explain the popping, but I expect that it has changed the timing of audio I/O. Ben. -- Ben Hutchings Once a job is fouled up, anything done to improve it makes it worse.

2 2

[alsa-devel] [PATCH 3/4] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 12 Apr '10

12 Apr '10

2 1

Re: [alsa-devel] HDA Codec: Soundmax ADI AD1988 x Realtek ALC889
by Alex Austin 12 Apr '10

12 Apr '10

Not sure this is necessarily the forum for this, but it probably has more to do with motherboard signal trace layout than the codec, though I have certainly heard good things about SoundMAX codecs before. ADI makes products designed to go into hifi equipment, so has the engineering knowledge to get things right. Realtek makes audio chips just so they can have more chips on the board, if it already has rtl Ethernet and/or Wifi. On Apr 12, 2010 9:24 AM, "Dâniel Fraga" <fragabr(a)gmail.com> wrote: I upgraded my system from a Asus M2N-E (Soundmax AD1988 codec) motherboard to MSI X58 Pro-E (Realtek ALC889 codec). The problem is that I feel the sound worst than before. I feel lack of bass and treble. With AD1988 the sound was more "vivid", there was more bass and treble. With ALC889 it is "flat". The only advantage I noticed on ALC889 is the possibility of mixing multiple sounds at hardware level, what was impossible on AD1988. So, what are your opinion? Ps: I used on both the analog output with a Edifier E3100 2.1 speaker system. Ps2: could be a problem with a bad quality DAC of the Realtek ALC889? -- Linux 2.6.33: Man-Eating Seals of Antiquity _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

Re: [alsa-devel] ASoC updates for 2.6.34
by Mark Brown 12 Apr '10

12 Apr '10

Apologies for the quoting - I'm replying from my phone. I think in terms of code quality it'd be fine for stable but since it introduces an incompatible API change it's really a no go for the default stable release - any out of tree users would be broken by it which would be bad. It might be worth suggesting somewhere to publish them - if I get a chance I'll probably post something to my blog which should get good Google juice. Daniel Mack <daniel(a)caiaq.de> wrote: >On Mon, Apr 05, 2010 at 07:38:40PM +0100, Mark Brown wrote: >> The following changes since commit fb48e3c6a4d8888aff61fbf567aadac7d206e973: >> Graham Gower (1): >> ASoC: Fix passing platform_data to ac97 bus users and fix a leak >> >> 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 >> >> This is rather large for a stable update due to the inclusion of >> Daniel's backported fix for corruption of DMA data due to use of >> dma_data by both capture and playback streams (which is large but >> straightforward) > >Do you think it is safe to push this ASoC-DMA patch to stable@ as well? >I posted backported version already and I believe they're fine. > >Thanks, >Daniel >

1 0

[alsa-devel] HDA Codec: Soundmax ADI AD1988 x Realtek ALC889
by Dâniel Fraga 12 Apr '10

12 Apr '10

I upgraded my system from a Asus M2N-E (Soundmax AD1988 codec) motherboard to MSI X58 Pro-E (Realtek ALC889 codec). The problem is that I feel the sound worst than before. I feel lack of bass and treble. With AD1988 the sound was more "vivid", there was more bass and treble. With ALC889 it is "flat". The only advantage I noticed on ALC889 is the possibility of mixing multiple sounds at hardware level, what was impossible on AD1988. So, what are your opinion? Ps: I used on both the analog output with a Edifier E3100 2.1 speaker system. Ps2: could be a problem with a bad quality DAC of the Realtek ALC889? -- Linux 2.6.33: Man-Eating Seals of Antiquity

1 0

[alsa-devel] [PATCH 1/3] WM8750: Convert to new API
by Marek Vasut 12 Apr '10

12 Apr '10

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code around. Hugely inspired by WM8753 which was already converted. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- sound/soc/codecs/wm8750.c | 297 +++++++++++++++++++++------------------------ 1 files changed, 140 insertions(+), 157 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 3e440c0..49411d1 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -30,11 +30,6 @@ #include "wm8750.h" -/* codec private data */ -struct wm8750_priv { - unsigned int sysclk; -}; - /* * wm8750 register cache * We can't read the WM8750 register space when we @@ -54,6 +49,13 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; +/* codec private data */ +struct wm8750_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8750_reg)]; +}; + #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* @@ -696,25 +698,90 @@ static int wm8750_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8750_codec; + +static int wm8750_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8750_codec) { + dev_err(&pdev->dev, "WM8750 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8750_codec; + codec = wm8750_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); + wm8750_add_widgets(codec); + + return 0; + +err: + return ret; +} + +/* power down chip */ +static int wm8750_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_wm8750 = { + .probe = wm8750_probe, + .remove = wm8750_remove, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev, - enum snd_soc_control_type control) +static int wm8750_register(struct wm8750_priv *wm8750, + enum snd_soc_control_type control) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = &wm8750->codec; int reg, ret = 0; + if (wm8750_codec) { + dev_err(codec->dev, "Multiple WM8750 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "WM8750"; codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); - codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->private_data = wm8750; + codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1; + codec->reg_cache = &wm8750->reg_cache; + codec->private_data = wm8750; + + memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache)); ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); if (ret < 0) { @@ -728,13 +795,6 @@ static int wm8750_init(struct snd_soc_device *socdev, goto err; } - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto err; - } - /* charge output caps */ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -756,19 +816,37 @@ static int wm8750_init(struct snd_soc_device *socdev, reg = snd_soc_read(codec, WM8750_RINVOL); snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); - return ret; + wm8750_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + ret = snd_soc_register_dais(&wm8750_dai, 1); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); err: - kfree(codec->reg_cache); + kfree(wm8750); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; +static void wm8750_unregister(struct wm8750_priv *wm8750) +{ + wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(&wm8750_dai, 1); + snd_soc_unregister_codec(&wm8750->codec); + kfree(wm8750); + wm8750_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -782,24 +860,26 @@ static struct snd_soc_device *wm8750_socdev; static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; - ret = wm8750_init(socdev, SND_SOC_I2C); - if (ret < 0) - pr_err("failed to initialise WM8750\n"); + codec = &wm8750->codec; + codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8750); - return ret; + codec->dev = &i2c->dev; + + return wm8750_register(wm8750, SND_SOC_I2C); } static int wm8750_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8750_priv *wm8750 = i2c_get_clientdata(client); + wm8750_unregister(wm8750); return 0; } @@ -818,66 +898,31 @@ static struct i2c_driver wm8750_i2c_driver = { .remove = wm8750_i2c_remove, .id_table = wm8750_i2c_id, }; - -static int wm8750_add_i2c_device(struct platform_device *pdev, - const struct wm8750_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8750_i2c_driver); - return -ENODEV; -} #endif #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; + + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + codec = &wm8750->codec; codec->control_data = spi; + codec->dev = &spi->dev; - ret = wm8750_init(socdev, SND_SOC_SPI); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8750\n"); + dev_set_drvdata(&spi->dev, wm8750); - return ret; + return wm8750_register(wm8750, SND_SOC_SPI); } static int __devexit wm8750_spi_remove(struct spi_device *spi) { + struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev); + wm8750_unregister(wm8750); return 0; } @@ -892,93 +937,31 @@ static struct spi_driver wm8750_spi_driver = { }; #endif -static int wm8750_probe(struct platform_device *pdev) +static int __init wm8750_modinit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec; - struct wm8750_priv *wm8750; int ret; - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); - if (wm8750 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8750; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; - - ret = -ENODEV; - #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - ret = wm8750_add_i2c_device(pdev, setup); - } + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8750 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - ret = spi_register_driver(&wm8750_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } + ret = spi_register_driver(&wm8750_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8750 SPI driver: %d\n", ret); #endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return 0; } +module_init(wm8750_modinit); -/* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +static void __exit wm8750_exit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8750_spi_driver); #endif - kfree(codec->private_data); - kfree(codec); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); - -static int __init wm8750_modinit(void) -{ - return snd_soc_register_dai(&wm8750_dai); -} -module_init(wm8750_modinit); - -static void __exit wm8750_exit(void) -{ - snd_soc_unregister_dai(&wm8750_dai); } module_exit(wm8750_exit); -- 1.7.0

3 17

Re: [alsa-devel] wrong decibel data?
by Raymond Yau 10 Apr '10

10 Apr '10

2010/4/10 Werner Van Belle <werner(a)yellowcouch.org> > Raymond Yau wrote: > > The dynamic range of 16bit audio is 96dB > > http://en.wikipedia.org/wiki/Dynamic_range > > > Yes about that. I always wondered how they come up with 96dB ? > > A perceived doubling of volume is normally assumed to be +3dB, > (log_10(2)=0.3) which means that if you have 16 bit audio you have 16 > 'doublings', or in essence only 48 dB. Even worse, since the last bit is > a sign bit, you essentially can only achieve a dynamic range of 45dB ! > > Now, I know this is off topic, but I never heard any good explanation > why CD audio is suddenly 45 dB ? If anybody knows, please share your > thoughts ! > > Wkr, > > > http://en.wikipedia.org/wiki/Decibel When referring to measurements of amplitude it is usual to consider the ratio of the squares of *A*1 (measured amplitude) and *A*0 (reference amplitude). This is because in most applications power is proportional to the square of amplitude, in electrical circuit , dissipated power is typically proportional to the square of voltage or current

2 2

Re: [alsa-devel] wrong decibel data?
by Raymond Yau 10 Apr '10

10 Apr '10

2010/4/10 Werner Van Belle <werner(a)yellowcouch.org> > Raymond Yau wrote: > > The dynamic range of 16bit audio is 96dB > > http://en.wikipedia.org/wiki/Dynamic_range > > > Yes about that. I always wondered how they come up with 96dB ? > > A perceived doubling of volume is normally assumed to be +3dB, > (log_10(2)=0.3) which means that if you have 16 bit audio you have 16 > 'doublings', or in essence only 48 dB. Even worse, since the last bit is > a sign bit, you essentially can only achieve a dynamic range of 45dB ! > > Now, I know this is off topic, but I never heard any good explanation > why CD audio is suddenly 45 dB ? If anybody knows, please share your > thoughts ! > > Wkr, > > Even when you are using floating point number Floating point numbers provide a way to trade off signal-to-noise ratio for an increase in dynamic range. For n bit floating-point numbers, with n-m bits in the mantissa and m bits in the exponent DR is still a finite number http://en.wikipedia.org/wiki/Signal-to-noise_ratio

2 1

[alsa-devel] [PATCH] WM8750: Convert to new API
by Marek Vasut 09 Apr '10

09 Apr '10

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code around. Hugely inspired by WM8753 which was already converted. Also, this patch fixes the Jive and Spitz machine. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- sound/soc/codecs/wm8750.c | 295 ++++++++++++++++++--------------------- sound/soc/pxa/spitz.c | 43 +++++- sound/soc/s3c24xx/jive_wm8750.c | 5 - 3 files changed, 175 insertions(+), 168 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 3e440c0..eea7ba6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -30,11 +30,6 @@ #include "wm8750.h" -/* codec private data */ -struct wm8750_priv { - unsigned int sysclk; -}; - /* * wm8750 register cache * We can't read the WM8750 register space when we @@ -54,6 +49,13 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; +/* codec private data */ +struct wm8750_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8750_reg)]; +}; + #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* @@ -696,25 +698,90 @@ static int wm8750_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8750_codec; + +static int wm8750_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8750_codec) { + dev_err(&pdev->dev, "WM8750 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8750_codec; + codec = wm8750_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); + wm8750_add_widgets(codec); + + return 0; + +err: + return ret; +} + +/* power down chip */ +static int wm8750_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_wm8750 = { + .probe = wm8750_probe, + .remove = wm8750_remove, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev, - enum snd_soc_control_type control) +static int wm8750_register(struct wm8750_priv *wm8750, + enum snd_soc_control_type control) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = &wm8750->codec; int reg, ret = 0; + if (wm8750_codec) { + dev_err(codec->dev, "Multiple WM8750 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "WM8750"; codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); - codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->private_data = wm8750; + codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1; + codec->reg_cache = &wm8750->reg_cache; + codec->private_data = wm8750; + + memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache)); ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); if (ret < 0) { @@ -728,13 +795,6 @@ static int wm8750_init(struct snd_soc_device *socdev, goto err; } - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto err; - } - /* charge output caps */ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -756,19 +816,37 @@ static int wm8750_init(struct snd_soc_device *socdev, reg = snd_soc_read(codec, WM8750_RINVOL); snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); - return ret; + wm8750_codec = codec; + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(&wm8750_dai, 1); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); err: - kfree(codec->reg_cache); + kfree(wm8750); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; +static void wm8750_unregister(struct wm8750_priv *wm8750) +{ + wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(&wm8750_dai, 1); + snd_soc_unregister_codec(&wm8750->codec); + kfree(wm8750); + wm8750_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -782,24 +860,26 @@ static struct snd_soc_device *wm8750_socdev; static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; - i2c_set_clientdata(i2c, codec); + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + + codec = &wm8750->codec; codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8750); - ret = wm8750_init(socdev, SND_SOC_I2C); - if (ret < 0) - pr_err("failed to initialise WM8750\n"); + codec->dev = &i2c->dev; - return ret; + return wm8750_register(wm8750, SND_SOC_I2C); } static int wm8750_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8750_priv *wm8750 = i2c_get_clientdata(client); + wm8750_unregister(wm8750); return 0; } @@ -818,66 +898,31 @@ static struct i2c_driver wm8750_i2c_driver = { .remove = wm8750_i2c_remove, .id_table = wm8750_i2c_id, }; - -static int wm8750_add_i2c_device(struct platform_device *pdev, - const struct wm8750_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8750_i2c_driver); - return -ENODEV; -} #endif #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + + codec = &wm8750->codec; codec->control_data = spi; + codec->dev = &spi->dev; - ret = wm8750_init(socdev, SND_SOC_SPI); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8750\n"); + dev_set_drvdata(&spi->dev, wm8750); - return ret; + return wm8750_register(wm8750, SND_SOC_SPI); } static int __devexit wm8750_spi_remove(struct spi_device *spi) { + struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev); + wm8750_unregister(wm8750); return 0; } @@ -892,93 +937,31 @@ static struct spi_driver wm8750_spi_driver = { }; #endif -static int wm8750_probe(struct platform_device *pdev) +static int __init wm8750_modinit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec; - struct wm8750_priv *wm8750; int ret; - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); - if (wm8750 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8750; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; - - ret = -ENODEV; - #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - ret = wm8750_add_i2c_device(pdev, setup); - } + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8750 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - ret = spi_register_driver(&wm8750_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } + ret = spi_register_driver(&wm8750_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8750 SPI driver: %d\n", ret); #endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return 0; } +module_init(wm8750_modinit); -/* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +static void __exit wm8750_exit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8750_spi_driver); #endif - kfree(codec->private_data); - kfree(codec); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); - -static int __init wm8750_modinit(void) -{ - return snd_soc_register_dai(&wm8750_dai); -} -module_init(wm8750_modinit); - -static void __exit wm8750_exit(void) -{ - snd_soc_unregister_dai(&wm8750_dai); } module_exit(wm8750_exit); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c4cd2ac..1941a35 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -322,19 +322,44 @@ static struct snd_soc_card snd_soc_spitz = { .num_links = 1, }; -/* spitz audio private data */ -static struct wm8750_setup_data spitz_wm8750_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; - /* spitz audio subsystem */ static struct snd_soc_device spitz_snd_devdata = { .card = &snd_soc_spitz, .codec_dev = &soc_codec_dev_wm8750, - .codec_data = &spitz_wm8750_setup, }; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8750 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8750_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8750", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} + static struct platform_device *spitz_snd_device; static int __init spitz_init(void) @@ -344,6 +369,10 @@ static int __init spitz_init(void) if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) return -ENODEV; + ret = wm8750_i2c_setup(); + if (ret != 0) + return ret; + spitz_snd_device = platform_device_alloc("soc-audio", -1); if (!spitz_snd_device) return -ENOMEM; diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 59dc2c6..97d8ff3 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -152,15 +152,10 @@ static struct snd_soc_card snd_soc_machine_jive = { .num_links = 1, }; -/* jive audio private data */ -static struct wm8750_setup_data jive_wm8750_setup = { -}; - /* jive audio subsystem */ static struct snd_soc_device jive_snd_devdata = { .card = &snd_soc_machine_jive, .codec_dev = &soc_codec_dev_wm8750, - .codec_data = &jive_wm8750_setup, }; static struct platform_device *jive_snd_device; -- 1.7.0

3 2

[alsa-devel] Alsa Problems - O2 Joggler
by Daniel Case 08 Apr '10

08 Apr '10

Hi All, I have an O2 Joggler with the alsa-info output file located here: http://alsa-project.org/db/?f=4012bb42964e3c87ae7ff4e0561d9ee9d3eb5e2c The sound does not work unless I remove the following line from /etc/modprobe.d/alsa-base.conf # Power down HDA controllers after 10 idle seconds options snd-hda-intel power_save=10 power_save_controller=N I believe this means that the chipset is not able to wake up from power saving, and what I get is crackling sound. What can be done so that this issue is fixed (the device works with power saving)? The device came with a special version of Ubuntu Linux 8.04, and the source code (included Alsa) is available at http://lists.gpl-violations.org/pipermail/legal/2009-December/001755.html Thanks! -- ~Daniel

1 0

[alsa-devel] imx-ssi fixes
by Sascha Hauer 08 Apr '10

08 Apr '10

Hi all, Here are some bugfix patches for the imx-ssi driver. It should also fix the problems reported in FIQ mode. Sascha

4 14

[alsa-devel] [PATCH] WM8750: Convert to new API
by Marek Vasut 08 Apr '10

08 Apr '10

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code around. Hugely inspired by WM8753 which was already converted. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- sound/soc/codecs/wm8750.c | 295 ++++++++++++++++++--------------------- sound/soc/pxa/spitz.c | 43 +++++- sound/soc/s3c24xx/jive_wm8750.c | 40 +++++- 3 files changed, 210 insertions(+), 168 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 3e440c0..eea7ba6 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -30,11 +30,6 @@ #include "wm8750.h" -/* codec private data */ -struct wm8750_priv { - unsigned int sysclk; -}; - /* * wm8750 register cache * We can't read the WM8750 register space when we @@ -54,6 +49,13 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; +/* codec private data */ +struct wm8750_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8750_reg)]; +}; + #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* @@ -696,25 +698,90 @@ static int wm8750_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8750_codec; + +static int wm8750_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8750_codec) { + dev_err(&pdev->dev, "WM8750 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8750_codec; + codec = wm8750_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); + wm8750_add_widgets(codec); + + return 0; + +err: + return ret; +} + +/* power down chip */ +static int wm8750_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_wm8750 = { + .probe = wm8750_probe, + .remove = wm8750_remove, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev, - enum snd_soc_control_type control) +static int wm8750_register(struct wm8750_priv *wm8750, + enum snd_soc_control_type control) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = &wm8750->codec; int reg, ret = 0; + if (wm8750_codec) { + dev_err(codec->dev, "Multiple WM8750 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "WM8750"; codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); - codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->private_data = wm8750; + codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1; + codec->reg_cache = &wm8750->reg_cache; + codec->private_data = wm8750; + + memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache)); ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); if (ret < 0) { @@ -728,13 +795,6 @@ static int wm8750_init(struct snd_soc_device *socdev, goto err; } - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto err; - } - /* charge output caps */ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -756,19 +816,37 @@ static int wm8750_init(struct snd_soc_device *socdev, reg = snd_soc_read(codec, WM8750_RINVOL); snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); - return ret; + wm8750_codec = codec; + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(&wm8750_dai, 1); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); err: - kfree(codec->reg_cache); + kfree(wm8750); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; +static void wm8750_unregister(struct wm8750_priv *wm8750) +{ + wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(&wm8750_dai, 1); + snd_soc_unregister_codec(&wm8750->codec); + kfree(wm8750); + wm8750_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -782,24 +860,26 @@ static struct snd_soc_device *wm8750_socdev; static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; - i2c_set_clientdata(i2c, codec); + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + + codec = &wm8750->codec; codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8750); - ret = wm8750_init(socdev, SND_SOC_I2C); - if (ret < 0) - pr_err("failed to initialise WM8750\n"); + codec->dev = &i2c->dev; - return ret; + return wm8750_register(wm8750, SND_SOC_I2C); } static int wm8750_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8750_priv *wm8750 = i2c_get_clientdata(client); + wm8750_unregister(wm8750); return 0; } @@ -818,66 +898,31 @@ static struct i2c_driver wm8750_i2c_driver = { .remove = wm8750_i2c_remove, .id_table = wm8750_i2c_id, }; - -static int wm8750_add_i2c_device(struct platform_device *pdev, - const struct wm8750_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8750_i2c_driver); - return -ENODEV; -} #endif #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + + codec = &wm8750->codec; codec->control_data = spi; + codec->dev = &spi->dev; - ret = wm8750_init(socdev, SND_SOC_SPI); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8750\n"); + dev_set_drvdata(&spi->dev, wm8750); - return ret; + return wm8750_register(wm8750, SND_SOC_SPI); } static int __devexit wm8750_spi_remove(struct spi_device *spi) { + struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev); + wm8750_unregister(wm8750); return 0; } @@ -892,93 +937,31 @@ static struct spi_driver wm8750_spi_driver = { }; #endif -static int wm8750_probe(struct platform_device *pdev) +static int __init wm8750_modinit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec; - struct wm8750_priv *wm8750; int ret; - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); - if (wm8750 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8750; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; - - ret = -ENODEV; - #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - ret = wm8750_add_i2c_device(pdev, setup); - } + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8750 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - ret = spi_register_driver(&wm8750_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } + ret = spi_register_driver(&wm8750_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8750 SPI driver: %d\n", ret); #endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return 0; } +module_init(wm8750_modinit); -/* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +static void __exit wm8750_exit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8750_spi_driver); #endif - kfree(codec->private_data); - kfree(codec); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); - -static int __init wm8750_modinit(void) -{ - return snd_soc_register_dai(&wm8750_dai); -} -module_init(wm8750_modinit); - -static void __exit wm8750_exit(void) -{ - snd_soc_unregister_dai(&wm8750_dai); } module_exit(wm8750_exit); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index c4cd2ac..1941a35 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -322,19 +322,44 @@ static struct snd_soc_card snd_soc_spitz = { .num_links = 1, }; -/* spitz audio private data */ -static struct wm8750_setup_data spitz_wm8750_setup = { - .i2c_bus = 0, - .i2c_address = 0x1b, -}; - /* spitz audio subsystem */ static struct snd_soc_device spitz_snd_devdata = { .card = &snd_soc_spitz, .codec_dev = &soc_codec_dev_wm8750, - .codec_data = &spitz_wm8750_setup, }; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8750 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8750_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = 0x1b; + strlcpy(info.type, "wm8750", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} + static struct platform_device *spitz_snd_device; static int __init spitz_init(void) @@ -344,6 +369,10 @@ static int __init spitz_init(void) if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) return -ENODEV; + ret = wm8750_i2c_setup(); + if (ret != 0) + return ret; + spitz_snd_device = platform_device_alloc("soc-audio", -1); if (!spitz_snd_device) return -ENOMEM; diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 59dc2c6..b9d85c6 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -152,17 +152,43 @@ static struct snd_soc_card snd_soc_machine_jive = { .num_links = 1, }; -/* jive audio private data */ -static struct wm8750_setup_data jive_wm8750_setup = { -}; - /* jive audio subsystem */ static struct snd_soc_device jive_snd_devdata = { .card = &snd_soc_machine_jive, .codec_dev = &soc_codec_dev_wm8750, - .codec_data = &jive_wm8750_setup, }; +/* + * FIXME: This is a temporary bodge to avoid cross-tree merge issues. + * New drivers should register the wm8750 I2C device in the machine + * setup code (under arch/arm for ARM systems). + */ +static int wm8750_i2c_register(void) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + + memset(&info, 0, sizeof(struct i2c_board_info)); + strlcpy(info.type, "wm8750", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(0); + if (!adapter) { + printk(KERN_ERR "can't get i2c adapter 0\n"); + return -ENODEV; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + printk(KERN_ERR "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + return -ENODEV; + } + + return 0; +} + static struct platform_device *jive_snd_device; static int __init jive_init(void) @@ -172,6 +198,10 @@ static int __init jive_init(void) if (!machine_is_jive()) return 0; + ret = wm8750_i2c_setup(); + if (ret != 0) + return ret; + printk("JIVE WM8750 Audio support\n"); jive_snd_device = platform_device_alloc("soc-audio", -1); -- 1.7.0

2 1

[alsa-devel] [PATCH 10/16] ASoC: wm2000: remove unused #include <linux/version.h>
by Huang Weiyi 08 Apr '10

08 Apr '10

Remove unused #include <linux/version.h>('s) in sound/soc/codecs/wm2000.c Signed-off-by: Huang Weiyi <weiyi.huang(a)gmail.com> --- sound/soc/codecs/wm2000.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a34cbcf..002e289 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -23,7 +23,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/firmware.h> -- 1.6.1.3

3 2

[alsa-devel] [PATCH] ALSA: hda: Add support for Medion WIM2160
by Maurus Cuelenaere 07 Apr '10

07 Apr '10

This adds support for the Medion WIM2160 soundcard. There's no PCI quirk added because it has the same PCI id as the Medion MD2. Signed-off-by: Maurus Cuelenaere <mcuelenaere(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 53 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 53 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index da34095..79833e6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -226,6 +226,7 @@ enum { ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -8247,6 +8248,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc883_medion_wim2160_verbs[] = { + /* Unmute front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Set speaker pin to front mixer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Init headphone pin */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc883_medion_wim2160_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1a; + spec->autocfg.speaker_pins[0] = 0x15; +} + static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -8956,6 +8993,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -9578,6 +9616,21 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc883_medion_md2_setup, .init_hook = alc_automute_amp, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc883_medion_wim2160_setup, + .init_hook = alc_automute_amp, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, -- 1.7.0.3

2 10

[alsa-devel] User-space control elements in ext PCM plugin
by Marcus Weigelt 07 Apr '10

07 Apr '10

Hi, I´m having an external PCM plugin that should do some processing on the audio. That part works fine. Now I´d like to have some way to modify the way to give some parameters to the plugin to tell it how the processing is done (eg. set delay time, pan, attack time, ...). I´ve searched the web for days and hours, but could not really find something. On the alsa mailing list there was a similiar question three years ago, and the answer was, that one could create user-space control elements in the plugin. How is that done? I could not find examples or good documentation about it. Did I miss something out? Can anyone give me a hint, maybe a link to some docs about it or source of a PCM plugin creating/using user space control elements? Thanks a lot. Marcus -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

3 4

[alsa-devel] USB HID gadget driver (was: Re: USB transfer_buffer allocations on 64bit systems)
by Daniel Mack 07 Apr '10

07 Apr '10

On Wed, Apr 07, 2010 at 06:20:53PM +0900, suman wrote: > I am sorry if this is a lame question, I am in Ignorance. I have little time > to make a USB driver. 1. Please do not hijack email threads for unrelated subjects. Your question has nothing to do with the original topic and you copy the wrong people for your question. Start a new thread for that. 2. Do not reply on top. > I need to make a board work in device mode which acts as a keyboard and > mouse device, So when my device is connected to a windows system it > will recognize as a keyboard and a mouse device. I need a pointer for it. I > do not find much documentation on it other than gadget driver which belongs > to bulk class driver. > I know my USB controller supports OTG and supports both device and host > mode. You need to write your own gadget driver for that. See the ones that already exist in the tree and see which parts you can recycle for your purpose. The protocol you want to talk is HID. Daniel > If you have any pointer can you please give. I'm all ears. > > On Wed, Apr 7, 2010 at 6:06 PM, Daniel Mack <daniel(a)caiaq.de> wrote: > > > Hi, > > > > I was pointed to https://bugzilla.kernel.org/show_bug.cgi?id=15580 by > > Pedro Ribeiro, and unfortunately I'm pretty late in the game. I wasn't > > aware of that thread until yesterday. > > > > While the report is quite confusing, the reason seams pretty clear to me > > as I've been thru quite some time-demanding debugging of a very similar > > issue on Mac OS X. But I'm not totally sure whether we really hit the > > same issue here, so I'd like to have your opinions first. > > > > The problem is appearantly the way the transfer buffer is allocated in > > the drivers. In the snd-usb-caiaq driver, I used kzalloc() to get memory > > which works fine on 32bit systems. On x86_64, however, it seems that > > kzalloc() hands out memory beyond the 32bit addressable boundary, which > > the DMA controller of the 32bit PCI-connected EHCI controller is unable > > to write to or read from. Am I correct on this conclusion? > > > > Depending on the condition of the memory management, things might work > > or not, and especially right after a reboot, there's a better chance to > > get lower memory. > > > > The fix is to use usb_buffer_alloc() for that purpose which ensures > > memory that is suitable for DMA. And on x86_64, this also means that the > > upper 32 bits of the address returned are all 0's. > > > > If what I've stated is true, there are quite some more drivers affected > > by this issue. I collected a list of places where similar fixes are > > needed, and I can send patches if I get a thumbs-up. > > > > Pedro is currently testing a patch I sent out yesterday. > > > > Thanks, > > Daniel > > > > -- > > To unsubscribe from this list: send the line "unsubscribe linux-usb" in > > the body of a message to majordomo(a)vger.kernel.org > > More majordomo info at http://vger.kernel.org/majordomo-info.html > > > > > > -- > Rgds > Vijayendra Suman > > The only basis for a true and lasting unity of all humanity is the religion > of the heart. The religion of the heart is the religion of love. People can > only be united if they are free from jealousy, hatred and petty-mindedness. > Purify your heart first. Meet hatred with love and malice with goodwill. The > purer you make your heart, the greater will be the power of your love.

1 0

[alsa-devel] ali5451 - no sound after resuming from suspend
by Rami Autiomäki 07 Apr '10

07 Apr '10

Hello, My laptop suspends and resumes find, but sound is not working after computer is suspended and resumed. I can fix sound by running as root "alsa force-reload". This bug has been some time ago reported to alsa bug tracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=959 I compiled vanilla kernel 2.6.33.2 from kernel.org, after suspending sound fails similarly with this kernel also. I looked my alsa-info logs (they are somewhere down there) before and after suspend, and files differ much at where it says "Current setup". -- Rami Autiomäki -- Package-specific info: ** Version: Linux version 2.6.32-3-686 (Debian 2.6.32-9) (maks(a)debian.org) (gcc version 4.3.4 (Debian 4.3.4-8) ) #1 SMP Thu Feb 25 06:14:20 UTC 2010 ** Command line: BOOT_IMAGE=/boot/vmlinuz-2.6.32-3-686 root=UUID=d58ac20c-4ecb-4cc6-91b5-acad8448af76 ro quiet noapic nolapic ** Tainted: P (1) * Proprietary module has been loaded. ** Kernel log: [ 732.060050] ali_stimer_read: stimer is not ready. [ 732.316048] ali_stimer_read: stimer is not ready. [ 732.572049] ali_stimer_read: stimer is not ready. [ 732.828049] ali_stimer_read: stimer is not ready. [ 733.084080] ali_stimer_read: stimer is not ready. [ 733.340048] ali_stimer_read: stimer is not ready. [ 733.596050] ali_stimer_read: stimer is not ready. [ 733.852459] ali_stimer_read: stimer is not ready. [ 734.108050] ali_stimer_read: stimer is not ready. [ 734.364048] ali_stimer_read: stimer is not ready. [ 734.620048] ali_stimer_read: stimer is not ready. [ 734.876053] ali_stimer_read: stimer is not ready. [ 735.132051] ali_stimer_read: stimer is not ready. [ 735.388049] ali_stimer_read: stimer is not ready. [ 735.644051] ali_stimer_read: stimer is not ready. [ 735.900050] ali_stimer_read: stimer is not ready. [ 736.156202] ali_stimer_read: stimer is not ready. [ 736.412074] ali_stimer_read: stimer is not ready. [ 736.668060] ali_stimer_read: stimer is not ready. [ 786.632050] ali_stimer_read: stimer is not ready. [ 786.888049] ali_stimer_read: stimer is not ready. [ 787.144051] ali_stimer_read: stimer is not ready. [ 787.400048] ali_stimer_read: stimer is not ready. [ 787.656097] ali_stimer_read: stimer is not ready. [ 787.912048] ali_stimer_read: stimer is not ready. [ 788.168049] ali_stimer_read: stimer is not ready. [ 788.424049] ali_stimer_read: stimer is not ready. [ 788.680049] ali_stimer_read: stimer is not ready. [ 788.936068] ali_stimer_read: stimer is not ready. [ 789.192047] ali_stimer_read: stimer is not ready. [ 789.448049] ali_stimer_read: stimer is not ready. [ 789.704049] ali_stimer_read: stimer is not ready. [ 789.960071] ali_stimer_read: stimer is not ready. [ 790.216049] ali_stimer_read: stimer is not ready. [ 790.472071] ali_stimer_read: stimer is not ready. [ 790.728049] ali_stimer_read: stimer is not ready. [ 790.984048] ali_stimer_read: stimer is not ready. [ 791.240053] ali_stimer_read: stimer is not ready. [ 791.496072] ali_stimer_read: stimer is not ready. [ 791.752059] ali_stimer_read: stimer is not ready. [ 792.008049] ali_stimer_read: stimer is not ready. [ 792.264082] ali_stimer_read: stimer is not ready. [ 792.520080] ali_stimer_read: stimer is not ready. [ 792.776110] ali_stimer_read: stimer is not ready. [ 793.032050] ali_stimer_read: stimer is not ready. [ 793.288138] ali_stimer_read: stimer is not ready. [ 793.544111] ali_stimer_read: stimer is not ready. [ 793.800114] ali_stimer_read: stimer is not ready. [ 794.056049] ali_stimer_read: stimer is not ready. [ 794.312049] ali_stimer_read: stimer is not ready. [ 794.568049] ali_stimer_read: stimer is not ready. [ 794.824049] ali_stimer_read: stimer is not ready. [ 795.080048] ali_stimer_read: stimer is not ready. [ 795.336073] ali_stimer_read: stimer is not ready. [ 795.592064] ali_stimer_read: stimer is not ready. [ 795.848049] ali_stimer_read: stimer is not ready. [ 796.104080] ali_stimer_read: stimer is not ready. [ 796.360068] ali_stimer_read: stimer is not ready. [ 796.616209] ali_stimer_read: stimer is not ready. [ 796.872049] ali_stimer_read: stimer is not ready. [ 797.128049] ali_stimer_read: stimer is not ready. [ 797.384049] ali_stimer_read: stimer is not ready. [ 797.640052] ali_stimer_read: stimer is not ready. [ 797.896113] ali_stimer_read: stimer is not ready. [ 798.152048] ali_stimer_read: stimer is not ready. [ 798.408048] ali_stimer_read: stimer is not ready. [ 798.664213] ali_stimer_read: stimer is not ready. [ 798.920083] ali_stimer_read: stimer is not ready. [ 799.176054] ali_stimer_read: stimer is not ready. [ 799.433427] ali_stimer_read: stimer is not ready. [ 799.688051] ali_stimer_read: stimer is not ready. [ 799.944082] ali_stimer_read: stimer is not ready. [ 800.200047] ali_stimer_read: stimer is not ready. [ 800.456050] ali_stimer_read: stimer is not ready. [ 800.712082] ali_stimer_read: stimer is not ready. [ 800.968109] ali_stimer_read: stimer is not ready. [ 801.224049] ali_stimer_read: stimer is not ready. [ 801.480090] ali_stimer_read: stimer is not ready. [ 801.736051] ali_stimer_read: stimer is not ready. [ 801.992071] ali_stimer_read: stimer is not ready. [ 802.248052] ali_stimer_read: stimer is not ready. [ 802.504109] ali_stimer_read: stimer is not ready. [ 802.760048] ali_stimer_read: stimer is not ready. [ 803.016049] ali_stimer_read: stimer is not ready. [ 803.272049] ali_stimer_read: stimer is not ready. [ 803.528052] ali_stimer_read: stimer is not ready. [ 803.784080] ali_stimer_read: stimer is not ready. [ 804.040048] ali_stimer_read: stimer is not ready. [ 804.296050] ali_stimer_read: stimer is not ready. [ 804.552049] ali_stimer_read: stimer is not ready. [ 804.808050] ali_stimer_read: stimer is not ready. [ 805.064049] ali_stimer_read: stimer is not ready. [ 805.320048] ali_stimer_read: stimer is not ready. [ 805.576049] ali_stimer_read: stimer is not ready. [ 805.832048] ali_stimer_read: stimer is not ready. [ 806.088049] ali_stimer_read: stimer is not ready. [ 806.344049] ali_stimer_read: stimer is not ready. [ 806.600077] ali_stimer_read: stimer is not ready. [ 1337.820270] reportbug[3574]: segfault at 78 ip b66212ab sp b5fc9460 error 4 in libpango-1.0.so.0.2600.2[b6602000+41000] [ 1370.688336] reportbug[3629]: segfault at 7c ip b65ea6f5 sp b5f905b0 error 4 in libpango-1.0.so.0.2600.2[b65c9000+41000] ** Loaded modules: Module Size Used by radeon 398727 2 ttm 26172 1 radeon drm_kms_helper 17247 1 radeon drm 107671 5 radeon,ttm,drm_kms_helper i2c_algo_bit 3497 1 radeon ppdev 4058 0 lp 5570 0 parport 22554 2 ppdev,lp sco 5837 2 bridge 32983 0 stp 996 1 bridge bnep 7444 2 rfcomm 25131 0 l2cap 21677 4 bnep,rfcomm crc16 1027 1 l2cap bluetooth 36327 6 sco,bnep,rfcomm,l2cap cisco_ipsec 572479 0 acpi_cpufreq 4943 0 speedstep_lib 2468 0 cpufreq_conservative 4018 0 cpufreq_stats 1940 0 cpufreq_powersave 602 0 cpufreq_userspace 1476 0 binfmt_misc 4907 1 uinput 4796 1 fuse 43750 1 loop 9753 0 arc4 974 2 ecb 1405 2 rt61pci 15720 0 snd_ali5451 11793 1 crc_itu_t 1035 1 rt61pci snd_ac97_codec 79140 1 snd_ali5451 ac97_bus 710 1 snd_ac97_codec rt2x00pci 3773 1 rt61pci snd_pcm_oss 28671 0 rt2x00lib 19025 2 rt61pci,rt2x00pci snd_mixer_oss 10461 1 snd_pcm_oss led_class 1757 1 rt2x00lib snd_pcm 47362 3 snd_ali5451,snd_ac97_codec,snd_pcm_oss mac80211 122750 2 rt2x00pci,rt2x00lib snd_seq_midi 3576 0 snd_rawmidi 12505 1 snd_seq_midi cfg80211 87581 2 rt2x00lib,mac80211 rfkill 10260 4 bluetooth,cfg80211 snd_seq_midi_event 3684 1 snd_seq_midi eeprom_93cx6 963 1 rt61pci snd_seq 35459 2 snd_seq_midi,snd_seq_midi_event snd_timer 12258 2 snd_pcm,snd_seq i2c_ali1535 3974 0 i2c_ali15x3 4202 0 i2c_core 12648 5 radeon,drm,i2c_algo_bit,i2c_ali1535,i2c_ali15x3 snd_seq_device 3673 3 snd_seq_midi,snd_rawmidi,snd_seq pcmcia 17442 0 snd 34363 11 snd_ali5451,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device container 1833 0 joydev 6771 0 ac 1640 0 battery 3782 0 shpchp 21220 0 alim1535_wdt 2110 0 soundcore 3450 1 snd processor 26571 1 acpi_cpufreq yenta_socket 16403 2 pci_hotplug 18065 1 shpchp snd_page_alloc 5041 1 snd_pcm rsrc_nonstatic 7057 1 yenta_socket pcmcia_core 20446 3 pcmcia,yenta_socket,rsrc_nonstatic pcspkr 1207 0 evdev 5609 23 psmouse 44413 0 serio_raw 2916 0 ext3 94192 2 jbd 32161 1 ext3 mbcache 3762 1 ext3 ide_cd_mod 21076 0 cdrom 26487 1 ide_cd_mod ide_gd_mod 17171 4 ide_pci_generic 1924 0 ata_generic 2015 0 libata 114408 1 ata_generic scsi_mod 101297 1 libata 8139cp 13417 0 ohci_hcd 16904 0 ehci_hcd 27574 0 alim15x3 4280 3 8139too 14945 0 video 14605 0 ali_agp 3046 1 usbcore 98126 3 ohci_hcd,ehci_hcd floppy 40923 0 output 1204 1 video mii 2714 2 8139cp,8139too ide_core 64146 4 ide_cd_mod,ide_gd_mod,ide_pci_generic,alim15x3 nls_base 4541 1 usbcore agpgart 19516 3 ttm,drm,ali_agp button 3598 0 thermal 9206 0 fan 2586 0 thermal_sys 9378 4 processor,video,thermal,fan ** PCI devices: 00:00.0 Host bridge [0600]: ALi Corporation M1671 Super P4 Northbridge [AGP4X,PCI and SDR/DDR] [10b9:1671] (rev 02) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ >SERR- <PERR+ INTx- Latency: 0 Region 0: Memory at f0000000 (32-bit, prefetchable) [size=128M] Capabilities: <access denied> Kernel driver in use: agpgart-ali 00:01.0 PCI bridge [0604]: ALi Corporation PCI to AGP Controller [10b9:5247] (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 00009000-00009fff Memory behind bridge: e0100000-e01fffff Prefetchable memory behind bridge: e8000000-efffffff Secondary status: 66MHz+ FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA+ MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- 00:02.0 USB Controller [0c03]: ALi Corporation USB 1.1 Controller [10b9:5237] (rev 03) (prog-if 10 [OHCI]) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (20000ns max), Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 10 Region 0: Memory at e0000000 (32-bit, non-prefetchable) [size=4K] Capabilities: <access denied> Kernel driver in use: ohci_hcd 00:04.0 Multimedia audio controller [0401]: ALi Corporation M5451 PCI AC-Link Controller Audio Device [10b9:5451] (rev 02) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR+ <PERR+ INTx- Latency: 64 (500ns min, 6000ns max) Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at 1000 [size=256] Region 1: Memory at e0001000 (32-bit, non-prefetchable) [size=4K] Capabilities: <access denied> Kernel driver in use: ALI 5451 00:06.0 Bridge [0680]: ALi Corporation M7101 Power Management Controller [PMU] [10b9:7101] Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O- Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Kernel driver in use: ali1535_smbus 00:07.0 ISA bridge [0601]: ALi Corporation M1533/M1535/M1543 PCI to ISA Bridge [Aladdin IV/V/V+] [10b9:1533] Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: <access denied> 00:08.0 Ethernet controller [0200]: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ [10ec:8139] (rev 10) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR+ INTx- Latency: 64 (8000ns min, 16000ns max) Interrupt: pin A routed to IRQ 10 Region 0: I/O ports at 1400 [size=256] Region 1: Memory at e0002000 (32-bit, non-prefetchable) [size=256] Capabilities: <access denied> Kernel driver in use: 8139too 00:0a.0 CardBus bridge [0607]: Texas Instruments PCI1410 PC card Cardbus Controller [104c:ac50] (rev 02) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 11 Region 0: Memory at 28000000 (32-bit, non-prefetchable) [size=4K] Bus: primary=00, secondary=02, subordinate=05, sec-latency=176 Memory window 0: 20000000-23fff000 (prefetchable) Memory window 1: 24000000-27fff000 I/O window 0: 00001c00-00001cff I/O window 1: 00002000-000020ff BridgeCtl: Parity- SERR- ISA- VGA- MAbort- >Reset- 16bInt- PostWrite+ 16-bit legacy interface ports at 0001 Kernel driver in use: yenta_cardbus 00:0f.0 IDE interface [0101]: ALi Corporation M5229 IDE [10b9:5229] (rev c4) (prog-if fa) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 (500ns min, 1000ns max) Interrupt: pin A routed to IRQ 255 Region 0: [virtual] Memory at 000001f0 (32-bit, non-prefetchable) [size=8] Region 1: [virtual] Memory at 000003f0 (type 3, non-prefetchable) [size=1] Region 2: [virtual] Memory at 00000170 (32-bit, non-prefetchable) [size=8] Region 3: [virtual] Memory at 00000370 (type 3, non-prefetchable) [size=1] Region 4: I/O ports at 1800 [size=16] Capabilities: <access denied> Kernel driver in use: ALI15x3_IDE 01:00.0 VGA compatible controller [0300]: ATI Technologies Inc Radeon Mobility M6 LY [1002:4c59] (prog-if 00 [VGA controller]) Subsystem: Hewlett-Packard Company Device [103c:0027] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping+ SERR+ FastB2B+ DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 66 (2000ns min), Cache Line Size: 32 bytes Interrupt: pin A routed to IRQ 11 Region 0: Memory at e8000000 (32-bit, prefetchable) [size=128M] Region 1: I/O ports at 9000 [size=256] Region 2: Memory at e0100000 (32-bit, non-prefetchable) [size=64K] [virtual] Expansion ROM at e0120000 [disabled] [size=128K] Capabilities: <access denied> 02:00.0 Network controller [0280]: RaLink RT2561/RT61 802.11g PCI [1814:0301] Subsystem: RaLink EW-7108PCg [1814:2561] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 128 bytes Interrupt: pin A routed to IRQ 11 Region 0: Memory at 24000000 (32-bit, non-prefetchable) [size=32K] Capabilities: <access denied> Kernel driver in use: rt61pci ** USB devices: Bus 001 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub -- System Information: Debian Release: squeeze/sid APT prefers testing APT policy: (500, 'testing') Architecture: i386 (i686) Kernel: Linux 2.6.32-3-686 (SMP w/1 CPU core) Locale: LANG=fi_FI.UTF-8, LC_CTYPE=fi_FI.UTF-8 (charmap=UTF-8) Shell: /bin/sh linked to /bin/dash *** /home/rami/Desktop/alsa-info_before_suspend.txt upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.59 !!################################ !!Script ran on: Mon Apr 5 16:31:34 UTC 2010 !!Linux Distribution !!------------------ Debian GNU/Linux squeeze/sid \n \l !!DMI Information !!--------------- Manufacturer: Hewlett-Packard Product Name: HP OmniBook PC !!Kernel Information !!------------------ Kernel release: 2.6.32-3-686 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_ali5451 !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [A5451 ]: ALI5451 - ALI 5451 ALI 5451 at 0x1000, irq 11 !!PCI Soundcards installed in the system !!-------------------------------------- 00:04.0 Multimedia audio controller: ALi Corporation M5451 PCI AC-Link Controller Audio Device (rev 02) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:04.0 0401: 10b9:5451 (rev 02) Subsystem: 103c:0027 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_ali5451 enable : N id : <NULL> index : -1 pcm_channels : 32 spdif : N !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: Realtek ALC200,200P rev 0 PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Flags: 0 Capabilities : -headphone out- DAC resolution : 18-bit ADC resolution : 18-bit 3D enhancement : Realtek 3D Stereo Enhancement Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic2 ADC/DAC loopback : off Extended ID : codec=0 rev=1 AMAP DSA=0 SPDIF VRA Extended status : SPCV SPDIF=3/4 VRA PCM front DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Category=0x2 Generation=1 Rate=48kHz 0:00 = 5950 0:02 = 0a0a 0:04 = 0000 0:06 = 000e 0:08 = 0000 0:0a = 000e 0:0c = 801f 0:0e = 801f 0:10 = 9f1f 0:12 = 8808 0:14 = 9f1f 0:16 = 9f1f 0:18 = 0808 0:1a = 0000 0:1c = 8000 0:1e = 0000 0:20 = 0100 0:22 = 0000 0:24 = 0000 0:26 = 800f 0:28 = 0605 0:2a = 0401 0:2c = bb80 0:2e = 0000 0:30 = 0000 0:32 = bb80 0:34 = 0000 0:36 = 0000 0:38 = 0000 0:3a = 2824 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0000 0:6e = 0013 0:70 = 8210 0:72 = 6aa9 0:74 = 0062 0:76 = 1f0f 0:78 = 0000 0:7a = 57c2 0:7c = 414c 0:7e = 4710 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 6 Apr 5 19:22 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 5 Apr 5 19:22 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 4 Apr 5 19:22 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 3 Apr 5 19:22 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Apr 5 19:22 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Apr 5 19:22 . drwxr-xr-x 3 root root 160 Apr 5 19:22 .. lrwxrwxrwx 1 root root 12 Apr 5 19:22 pci-0000:00:04.0 -> ../controlC0 !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: A5451 [ALI 5451], device 0: ALI 5451 [ALI 5451] Subdevices: 32/32 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Subdevice #8: subdevice #8 Subdevice #9: subdevice #9 Subdevice #10: subdevice #10 Subdevice #11: subdevice #11 Subdevice #12: subdevice #12 Subdevice #13: subdevice #13 Subdevice #14: subdevice #14 Subdevice #15: subdevice #15 Subdevice #16: subdevice #16 Subdevice #17: subdevice #17 Subdevice #18: subdevice #18 Subdevice #19: subdevice #19 Subdevice #20: subdevice #20 Subdevice #21: subdevice #21 Subdevice #22: subdevice #22 Subdevice #23: subdevice #23 Subdevice #24: subdevice #24 Subdevice #25: subdevice #25 Subdevice #26: subdevice #26 Subdevice #27: subdevice #27 Subdevice #28: subdevice #28 Subdevice #29: subdevice #29 Subdevice #30: subdevice #30 Subdevice #31: subdevice #31 ARECORD **** List of CAPTURE Hardware Devices **** card 0: A5451 [ALI 5451], device 0: ALI 5451 [ALI 5451] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [A5451] Card hw:0 'A5451'/'ALI 5451 at 0x1000, irq 11' Mixer name : 'Realtek ALC200,200P rev 0' Components : 'AC97a:414c4710' Controls : 37 Simple ctrls : 23 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 21 [68%] [-15.00dB] [on] Front Right: Playback 21 [68%] [-15.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 17 [55%] [-21.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control '3D Control - Switch',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 23 [74%] [0.00dB] [on] Front Right: Playback 23 [74%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 23 [74%] [0.00dB] [off] Capture [off] Front Right: Playback 23 [74%] [0.00dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic2' Simple mixer control 'Video',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 0 [0%] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 8 [53%] [-21.00dB] [on] Simple mixer control 'Aux',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined penum Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 0 [0%] [0.00dB] [off] Front Right: Capture 0 [0%] [0.00dB] [off] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.A5451 { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 21 value.1 21 } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Playback Switch' value true } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 31 value.1 31 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Mono Playback Switch' value true } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Mono Playback Volume' value 17 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Speaker Playback Switch' value true } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' comment.dbmin -4500 comment.dbmax 0 iface MIXER name 'PC Speaker Playback Volume' value 8 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Phone Playback Volume' value 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Playback Switch' value false } control.12 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value 0 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost (+20dB)' value false } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Playback Switch' value false } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'CD Playback Switch' value false } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'CD Playback Volume' value.0 23 value.1 23 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Video Playback Switch' value false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Video Playback Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Aux Playback Switch' value false } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Aux Playback Volume' value.0 0 value.1 0 } control.22 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PCM Playback Switch' value true } control.23 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'PCM Playback Volume' value.0 23 value.1 23 } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 2 comment.item.0 Mic comment.item.1 CD comment.item.2 Video comment.item.3 Aux comment.item.4 Line comment.item.5 Mix comment.item.6 'Mix Mono' comment.item.7 Phone iface MIXER name 'Capture Source' value.0 Mic value.1 Mic } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Capture Switch' value false } control.26 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name '3D Control - Switch' value false } control.28 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.29 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic2 } control.30 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' iface MIXER name '3D Control - Center' value 0 } control.31 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' iface MIXER name '3D Control - Depth' value 0 } control.32 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.33 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.34 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.36 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' iface MIXER name 'IEC958 Playback AC97-SPSA' value 0 } control.37 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module radeon ttm drm_kms_helper drm i2c_algo_bit ppdev lp parport sco bridge stp bnep rfcomm l2cap crc16 bluetooth cisco_ipsec acpi_cpufreq speedstep_lib cpufreq_conservative cpufreq_stats cpufreq_powersave cpufreq_userspace binfmt_misc uinput fuse loop arc4 ecb rt61pci snd_ali5451 crc_itu_t snd_ac97_codec ac97_bus rt2x00pci snd_pcm_oss rt2x00lib snd_mixer_oss led_class snd_pcm mac80211 snd_seq_midi snd_rawmidi cfg80211 rfkill snd_seq_midi_event eeprom_93cx6 snd_seq snd_timer i2c_ali1535 i2c_ali15x3 i2c_core snd_seq_device pcmcia snd container joydev ac battery shpchp alim1535_wdt soundcore processor yenta_socket pci_hotplug snd_page_alloc rsrc_nonstatic pcmcia_core pcspkr evdev psmouse serio_raw ext3 jbd mbcache ide_cd_mod cdrom ide_gd_mod ide_pci_generic ata_generic libata scsi_mod 8139cp ohci_hcd ehci_hcd alim15x3 8139too video ali_agp usbcore floppy output mii ide_core nls_base agpgart button thermal fan thermal_sys !!ALSA/HDA dmesg !!------------------ *** /home/rami/Desktop/alsa-info_after_suspend.txt upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.59 !!################################ !!Script ran on: Mon Apr 5 16:35:46 UTC 2010 !!Linux Distribution !!------------------ Debian GNU/Linux squeeze/sid \n \l !!DMI Information !!--------------- Manufacturer: Hewlett-Packard Product Name: HP OmniBook PC !!Kernel Information !!------------------ Kernel release: 2.6.32-3-686 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.22 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_ali5451 !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [A5451 ]: ALI5451 - ALI 5451 ALI 5451 at 0x1000, irq 11 !!PCI Soundcards installed in the system !!-------------------------------------- 00:04.0 Multimedia audio controller: ALi Corporation M5451 PCI AC-Link Controller Audio Device (rev 02) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:04.0 0401: 10b9:5451 (rev 02) Subsystem: 103c:0027 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_ali5451 enable : N id : <NULL> index : -1 pcm_channels : 32 spdif : N !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: Realtek ALC200,200P rev 0 PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Flags: 0 Capabilities : -headphone out- DAC resolution : 18-bit ADC resolution : 18-bit 3D enhancement : Realtek 3D Stereo Enhancement Current setup Mic gain : +20dB [+0dB] POP path : post 3D Sim. stereo : on 3D enhancement : on Loudness : on Mono output : Mic Mic select : Mic2 ADC/DAC loopback : on Extended ID : codec=3 rev=3 AMAP LDAC SDAC CDAC DSA=3 VRM SPDIF DRA VRA Extended status : PRL PRK PRJ PRI SPCV MADC LDAC SDAC CDAC SPDIF=10/11 VRM SPDIF DRA VRA PCM front DAC : 65535Hz PCM Surr DAC : 65535Hz PCM LFE DAC : 65535Hz PCM ADC : 65535Hz PCM MIC ADC : 65535Hz SPDIF Control : PRO Non-audio Preemph50/15 Category=0x7f Generation=1 Rate=32kHz DRS Validity Extended modem ID: codec=3 CID2 CID1 HSET LIN2 LIN1 Modem status : GPIO MREF ADC1 DAC1 ADC2 DAC2 HADC HDAC PRA(GPIO) PRB(res) PRC(ADC1) PRD(DAC1) PRE(ADC2) PRF(DAC2) PRG(HADC) PRH(HDAC) Line1 rate : 65535Hz Line2 rate : 65535Hz Headset rate : 65535Hz 0:00 = ffff 0:02 = ffff 0:04 = ffff 0:06 = ffff 0:08 = ffff 0:0a = ffff 0:0c = ffff 0:0e = ffff 0:10 = ffff 0:12 = ffff 0:14 = ffff 0:16 = ffff 0:18 = ffff 0:1a = ffff 0:1c = ffff 0:1e = ffff 0:20 = ffff 0:22 = ffff 0:24 = ffff 0:26 = ffff 0:28 = ffff 0:2a = ffff 0:2c = ffff 0:2e = ffff 0:30 = ffff 0:32 = ffff 0:34 = ffff 0:36 = ffff 0:38 = ffff 0:3a = ffff 0:3c = ffff 0:3e = ffff 0:40 = ffff 0:42 = ffff 0:44 = ffff 0:46 = ffff 0:48 = ffff 0:4a = ffff 0:4c = ffff 0:4e = ffff 0:50 = ffff 0:52 = ffff 0:54 = ffff 0:56 = ffff 0:58 = ffff 0:5a = ffff 0:5c = ffff 0:5e = ffff 0:60 = ffff 0:62 = ffff 0:64 = ffff 0:66 = ffff 0:68 = ffff 0:6a = ffff 0:6c = ffff 0:6e = ffff 0:70 = ffff 0:72 = ffff 0:74 = ffff 0:76 = ffff 0:78 = ffff 0:7a = ffff 0:7c = ffff 0:7e = ffff --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 6 Apr 5 19:22 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 5 Apr 5 19:22 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 4 Apr 5 19:33 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 3 Apr 5 19:22 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Apr 5 19:22 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Apr 5 19:22 . drwxr-xr-x 3 root root 160 Apr 5 19:22 .. lrwxrwxrwx 1 root root 12 Apr 5 19:22 pci-0000:00:04.0 -> ../controlC0 !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: A5451 [ALI 5451], device 0: ALI 5451 [ALI 5451] Subdevices: 32/32 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Subdevice #8: subdevice #8 Subdevice #9: subdevice #9 Subdevice #10: subdevice #10 Subdevice #11: subdevice #11 Subdevice #12: subdevice #12 Subdevice #13: subdevice #13 Subdevice #14: subdevice #14 Subdevice #15: subdevice #15 Subdevice #16: subdevice #16 Subdevice #17: subdevice #17 Subdevice #18: subdevice #18 Subdevice #19: subdevice #19 Subdevice #20: subdevice #20 Subdevice #21: subdevice #21 Subdevice #22: subdevice #22 Subdevice #23: subdevice #23 Subdevice #24: subdevice #24 Subdevice #25: subdevice #25 Subdevice #26: subdevice #26 Subdevice #27: subdevice #27 Subdevice #28: subdevice #28 Subdevice #29: subdevice #29 Subdevice #30: subdevice #30 Subdevice #31: subdevice #31 ARECORD **** List of CAPTURE Hardware Devices **** card 0: A5451 [ALI 5451], device 0: ALI 5451 [ALI 5451] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [A5451] Card hw:0 'A5451'/'ALI 5451 at 0x1000, irq 11' Mixer name : 'Realtek ALC200,200P rev 0' Components : 'AC97a:414c4710' Controls : 37 Simple ctrls : 23 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 21 [68%] [-15.00dB] [on] Front Right: Playback 21 [68%] [-15.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 17 [55%] [-21.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control '3D Control - Center',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control '3D Control - Depth',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 15 Mono: 0 [0%] Simple mixer control '3D Control - Switch',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 23 [74%] [0.00dB] [on] Front Right: Playback 23 [74%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'CD',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 23 [74%] [0.00dB] [off] Capture [off] Front Right: Playback 23 [74%] [0.00dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic2' Simple mixer control 'Video',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined penum Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 0 [0%] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 15 Mono: Playback 8 [53%] [-21.00dB] [on] Simple mixer control 'Aux',0 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch cswitch-joined penum Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 0 [0%] [0.00dB] [off] Front Right: Capture 0 [0%] [0.00dB] [off] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive penum Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.A5451 { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 21 value.1 21 } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Playback Switch' value true } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 31 value.1 31 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Mono Playback Switch' value true } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Mono Playback Volume' value 17 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Speaker Playback Switch' value true } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' comment.dbmin -4500 comment.dbmax 0 iface MIXER name 'PC Speaker Playback Volume' value 8 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Phone Playback Volume' value 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Playback Switch' value false } control.12 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value 0 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost (+20dB)' value false } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Playback Switch' value false } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'CD Playback Switch' value false } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'CD Playback Volume' value.0 23 value.1 23 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Video Playback Switch' value false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Video Playback Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Aux Playback Switch' value false } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Aux Playback Volume' value.0 0 value.1 0 } control.22 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PCM Playback Switch' value true } control.23 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'PCM Playback Volume' value.0 23 value.1 23 } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 2 comment.item.0 Mic comment.item.1 CD comment.item.2 Video comment.item.3 Aux comment.item.4 Line comment.item.5 Mix comment.item.6 'Mix Mono' comment.item.7 Phone iface MIXER name 'Capture Source' value.0 Mic value.1 Mic } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Capture Switch' value false } control.26 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name '3D Control - Switch' value false } control.28 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.29 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic2 } control.30 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' iface MIXER name '3D Control - Center' value 0 } control.31 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' iface MIXER name '3D Control - Depth' value 0 } control.32 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.33 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.34 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.36 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' iface MIXER name 'IEC958 Playback AC97-SPSA' value 0 } control.37 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module radeon ttm drm_kms_helper drm i2c_algo_bit ppdev lp parport sco bridge stp bnep rfcomm l2cap crc16 bluetooth cisco_ipsec acpi_cpufreq speedstep_lib cpufreq_conservative cpufreq_stats cpufreq_powersave cpufreq_userspace binfmt_misc uinput fuse loop arc4 ecb rt61pci snd_ali5451 crc_itu_t snd_ac97_codec ac97_bus rt2x00pci snd_pcm_oss rt2x00lib snd_mixer_oss led_class snd_pcm mac80211 snd_seq_midi snd_rawmidi cfg80211 rfkill snd_seq_midi_event eeprom_93cx6 snd_seq snd_timer i2c_ali1535 i2c_ali15x3 i2c_core snd_seq_device pcmcia snd container joydev ac battery shpchp alim1535_wdt soundcore processor yenta_socket pci_hotplug snd_page_alloc rsrc_nonstatic pcmcia_core pcspkr evdev psmouse serio_raw ext3 jbd mbcache ide_cd_mod cdrom ide_gd_mod ide_pci_generic ata_generic libata scsi_mod 8139cp ohci_hcd ehci_hcd alim15x3 8139too video ali_agp usbcore floppy output mii ide_core nls_base agpgart button thermal fan thermal_sys !!ALSA/HDA dmesg !!------------------

1 0

[alsa-devel] [PATCH]Support MCP89 and GT21x hdmi audio
by Wei Ni 07 Apr '10

07 Apr '10

Hi, Takashi I developed the hdmi audio driver for new chipset MCP89 and GT21x. The new HAD controller and codec support standard HDMI operation. I attached the patch file, please check it. Thanks Wei. ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[PATCH] ALSA: usb/caiaq: use usb_buffer_alloc()
by Daniel Mack 07 Apr '10

07 Apr '10

Use usb_buffer_alloc() and usb_buffer_free() for transfer buffers. We need DMA-coherent memory in this case as buffer contents are likely to be modified after the URB was submitted, because the URB buffers are mapped to the audio streams. On x86_64, buffers allocated with kmalloc() may be beyond the boundaries of 32bit accessible memory, and DMA bounce buffers will live at other locations, unaccessible by the driver, and hence outside of the audio buffer mapping. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Tested-by: Pedro Ribeiro <pedrib(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Alan Stern <stern(a)rowland.harvard.edu> Cc: Chris Wright <chrisw(a)sous-sol.org> Cc: David Woodhouse <dwmw2(a)infradead.org> Cc: Andi Kleen <andi(a)firstfloor.org> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Greg KH <gregkh(a)suse.de> Cc: iommu(a)lists.linux-foundation.org Cc: Kernel development list <linux-kernel(a)vger.kernel.org> Cc: USB list <linux-usb(a)vger.kernel.org> Cc: stable(a)kernel.org --- sound/usb/caiaq/audio.c | 57 ++++++++++++++++++++++++++-------------------- 1 files changed, 32 insertions(+), 25 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4328cad..adbeefd 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -552,46 +552,47 @@ static struct urb **alloc_urbs(struct snd_usb_caiaqdev *dev, int dir, int *ret) } for (i = 0; i < N_URBS; i++) { - urbs[i] = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); - if (!urbs[i]) { + struct urb *u = usb_alloc_urb(FRAMES_PER_URB, GFP_KERNEL); + if (!u) { log("unable to usb_alloc_urb(), OOM!?\n"); *ret = -ENOMEM; return urbs; } - urbs[i]->transfer_buffer = - kmalloc(FRAMES_PER_URB * BYTES_PER_FRAME, GFP_KERNEL); - if (!urbs[i]->transfer_buffer) { - log("unable to kmalloc() transfer buffer, OOM!?\n"); + urbs[i] = u; + u->dev = usb_dev; + u->pipe = pipe; + u->transfer_buffer_length = + FRAMES_PER_URB * BYTES_PER_FRAME; + u->context = &dev->data_cb_info[i]; + u->interval = 1; + u->transfer_flags = URB_ISO_ASAP; + u->number_of_packets = FRAMES_PER_URB; + u->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ? + read_completed : write_completed; + u->transfer_buffer = usb_alloc_coherent(usb_dev, + u->transfer_buffer_length, + GFP_KERNEL, &u->transfer_dma); + if (!u->transfer_buffer) { + log("usb_alloc_coherent() failed, OOM!?\n"); *ret = -ENOMEM; return urbs; } for (frame = 0; frame < FRAMES_PER_URB; frame++) { struct usb_iso_packet_descriptor *iso = - &urbs[i]->iso_frame_desc[frame]; + &u->iso_frame_desc[frame]; iso->offset = BYTES_PER_FRAME * frame; iso->length = BYTES_PER_FRAME; } - - urbs[i]->dev = usb_dev; - urbs[i]->pipe = pipe; - urbs[i]->transfer_buffer_length = FRAMES_PER_URB - * BYTES_PER_FRAME; - urbs[i]->context = &dev->data_cb_info[i]; - urbs[i]->interval = 1; - urbs[i]->transfer_flags = URB_ISO_ASAP; - urbs[i]->number_of_packets = FRAMES_PER_URB; - urbs[i]->complete = (dir == SNDRV_PCM_STREAM_CAPTURE) ? - read_completed : write_completed; } *ret = 0; return urbs; } -static void free_urbs(struct urb **urbs) +static void free_urbs(struct usb_device *usb_dev, struct urb **urbs) { int i; @@ -603,7 +604,10 @@ static void free_urbs(struct urb **urbs) continue; usb_kill_urb(urbs[i]); - kfree(urbs[i]->transfer_buffer); + usb_free_coherent(usb_dev, + urbs[i]->transfer_buffer_length, + urbs[i]->transfer_buffer, + urbs[i]->transfer_dma); usb_free_urb(urbs[i]); } @@ -613,6 +617,7 @@ static void free_urbs(struct urb **urbs) int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) { int i, ret; + struct usb_device *usb_dev = dev->chip.dev; dev->n_audio_in = max(dev->spec.num_analog_audio_in, dev->spec.num_digital_audio_in) / @@ -689,15 +694,15 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) dev->data_urbs_in = alloc_urbs(dev, SNDRV_PCM_STREAM_CAPTURE, &ret); if (ret < 0) { kfree(dev->data_cb_info); - free_urbs(dev->data_urbs_in); + free_urbs(usb_dev, dev->data_urbs_in); return ret; } dev->data_urbs_out = alloc_urbs(dev, SNDRV_PCM_STREAM_PLAYBACK, &ret); if (ret < 0) { kfree(dev->data_cb_info); - free_urbs(dev->data_urbs_in); - free_urbs(dev->data_urbs_out); + free_urbs(usb_dev, dev->data_urbs_in); + free_urbs(usb_dev, dev->data_urbs_out); return ret; } @@ -706,10 +711,12 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) void snd_usb_caiaq_audio_free(struct snd_usb_caiaqdev *dev) { + struct usb_device *usb_dev = dev->chip.dev; + debug("%s(%p)\n", __func__, dev); stream_stop(dev); - free_urbs(dev->data_urbs_in); - free_urbs(dev->data_urbs_out); + free_urbs(usb_dev, dev->data_urbs_in); + free_urbs(usb_dev, dev->data_urbs_out); kfree(dev->data_cb_info); } -- 1.7.1

1 0

Re: [alsa-devel] mx31 snd and mc13783 codec status
by Sascha Hauer 06 Apr '10

06 Apr '10

On Thu, Apr 01, 2010 at 06:01:38PM +0200, Valentin Longchamp wrote: > Hi Sascha, > > Sascha Hauer wrote: >> >> Find the latest version of my code here: >> >> The following changes since commit 01e77706cdde7c0b47e5ca1f4284a795504c7c40: >> Linus Torvalds (1): >> Merge branch 'for-linus' of git://gitorious.org/linux-omap-dss2/linux >> >> are available in the git repository at: >> >> git://git.pengutronix.de/git/sha/linux-2.6.git mc13783 >> >> Sascha Hauer (3): >> add a mc13783 codec driver >> add phycore-mc13783 sound support >> pcm038: add sound support >> >> arch/arm/mach-mx2/mach-pcm038.c | 23 ++- >> sound/soc/codecs/Kconfig | 4 + >> sound/soc/codecs/Makefile | 2 + >> sound/soc/codecs/mc13783.c | 727 +++++++++++++++++++++++++++++++++++++++ >> sound/soc/codecs/mc13783.h | 32 ++ >> sound/soc/imx/Kconfig | 9 + >> sound/soc/imx/Makefile | 3 + >> sound/soc/imx/phycore-mc13783.c | 160 +++++++++ >> 8 files changed, 959 insertions(+), 1 deletions(-) >> create mode 100644 sound/soc/codecs/mc13783.c >> create mode 100644 sound/soc/codecs/mc13783.h >> create mode 100644 sound/soc/imx/phycore-mc13783.c >> >>> And do you know if your initial mc13783 codec support coupled with >>> mx31 had some limitations ? Our setup is quite straightforward, we >>> have direct connection from the mx31 to the mc13783 on a single SSI. >> >> Our board uses both SSI channels of the MC13783 which we then mux into >> one channel in the DAM unit. I don't know how this affects you. >> > > I have struggled with the DAM unit (this thing is an awful bulk of > wires) and now I get some sound on the loudspeaker. > > The thing is that I get a less than a second sound loop (I use aplay to > test, so userspace app should be ok), as if the buffer that the fiq asm > interrupt (from ssi_fiq.S) copies to the SSI hardware never was updated. > > If I have understood the fiq behaviour correctly, you have a asm fiq > interrupt that does copy a larger tx buffer into the SSI hardware. > Besides it, you have the imx_ssi_timer_callback that checks when the tx > buffer was completely copied. If it is the case, then a new buffer tx > buffer is "issued" with the snd_pcm_period_elapsed call (and then > snd_pcm_update_hw_ptr0). Is this behaviour correct ? Yes. > > If then it looks like on my system, I have a problem with the > snd_pcm_update_hw_ptr0 call. You could try reverting b4e82b5b785670b68136765059d1afc65c0ae023. Though I have tested it on my boards it may have some implications. Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

3 4

Re: [alsa-devel] Help! Need to switch between normal speaker and BT headset dynamically
by ANISH KUMAR 06 Apr '10

06 Apr '10

As my question is related to the query what pradeep is asking and mark brown also suggested to me to use PA. I have some additional questions regarding PA which i would like to clarify before starting: 1.Does PA have good resampler code if my hardware doesn't support resampling? I have a system where we are using ALSA as the base and on top(application side) of it there is a thread running which is pumping data and this thread is started by java code.Combination of LINUX+JAVA. This thread can perform these tasks: 1.Resampling for some rates. 2.Talking to PV framework to get the decoded PCM data and pumping audio data to alsa. 3.Routing support using asound.conf file. Can i replace this thread with PA by writing some wrapper over PA?? What extra features PA is going to provide additionally if i start using PA other than what is listed above??As i know PA is kind of sound deamon which my thread is not. Is there any good resampler code which i can use without any licensing problem and which can be easily integrated???

2 1

[alsa-devel] Help! Need to switch between normal speaker and BT headset dynamically
by Pradeep Kumar Jilagam 06 Apr '10

06 Apr '10

Hi, I am trying to switch the ALSA audio output between default speaker and BT(BlueTooth) headset. For this, I have used the bluez stack for the BT headset scanning and detection. After this, I have updated the .asoundrc file with the corresponding details(like MAC address of the detected BT headset and the device type as "pcm.bluetooth"). And finally, I used the aplay utility to play the file. This approach works fine but this needs restart of the playback session to start routing on the BT headset. I am trying for a solution(application/plugin) that can dynamically set the BT headset as the default audio output device and start streaming to it automatically as soon as the BT headset is detected in the middle of audio playback. Is this possible to do and if yes, is there an ALSA application/plugin already available for this? Thanks, Pradeep

3 3

[alsa-devel] ~~Hi~~
by John Kha 06 Apr '10

06 Apr '10

http://sites.google.com/site/sfg56yhs/cqdj4h -- John C. Kha

1 0

[alsa-devel] Is this a bug in alsa? (mute + hdmi audio related)
by VDR User 06 Apr '10

06 Apr '10

I've got an htpc using the following: Debian linux (testing) kernel 2.6.33.2 (stable) VDR-1.7.14 with the vdr-xine-0.9.3 output plugin Nvidia GT220 video card xine-lib-1.2 w/vdpau alsa-driver-20100307.tar.gz w/Wei's reworked nvhdmi code I'm using an HDMI cable going from the video card to my tv, and audio is going over the cable as well. When I press mute in VDR, I see the mute icon appear but the audio is not muted. I'm not sure if this is a xine problem or alsa but I'm guessing alsa since my previous setup was all the same except I was using a different Nvidia video card, but still with vdpau, and audio/video going from DVI-OUT on the card, to a DVI->HDMI cable into my tv. Could the problem be within the alsa driver somewhere or am I barking up the wrong tree? Thanks for any help.

1 0

[alsa-devel] [PATCH 00/37] OF: move device node pointer into struct device.
by Grant Likely 06 Apr '10

06 Apr '10

Currently on OF enabled platforms, the device node pointer is stored in both device.archdata.of_node (.prom_node on sparc), and in of_device.node OF device tree support is being generalized to work on any platform, so instead of adding the of_node pointer to each platform's archdata individually, this patch series moves the of_node pointer into struct device proper and fixes up all users to reference the new location. The last 3 patches in this series remove the old .archdata.of_node, .archdata.prom_node and of_device.node instances. This series has been compile tested on powerpc, sparc and microblaze, (allmodconfig used on sparc and powerpc) and boot tested on an mpc5200 powerpc. All known (or at least all I could find) in-tree users of the old names have been removed. This series should also be fully bisectable. After I collect acks, I would like to get this whole series into linux-next to see if I've missed any references, but I can probably hold off merging the last three patches when the 2.6.35 merge window opens to give out of tree users a bit more time to adapt. In fact, I'd probably wait a few extra days after pushing out the bulk of this series before I push the last 3 patches into my linux-next branch to isolate the drivers that I missed while fixing. Cheers, g. --- Grant Likely (37): arch/sparc: Remove .archdata.of_node and use .of_node in struct device arch/microblaze: Remove .archdata.of_node and use .of_node in struct device arch/powerpc: Remove .archdata.of_node and use .of_node in struct device sound/sparc: use .dev.of_node instead of .node in struct of_device sound/soc: use .dev.of_node instead of .node in struct of_device sound/aoa: use .dev.of_node instead of .node in struct of_device drivers/of: use .of_node instead of .archdata.of_node in struct device drivers/cdrom: use .of_node instead of .archdata.of_node in struct device drivers/char: use .of_node instead of .archdata.of_node in struct device drivers/scsi: use .of_node instead of .archdata.of_node in struct device drivers/of: use .dev.of_node instead of .node in struct of_device drivers/mtd: use .dev.of_node instead of .node in struct of_device drivers/infiniband: use .dev.of_node instead of .node in struct of_device drivers/char: use .dev.of_node instead of .node in struct of_device drivers/i2c: use .dev.of_node instead of .node in struct of_device drivers/leds: use .dev.of_node instead of .node in struct of_device drivers/ata: use .dev.of_node instead of .node in struct of_device drivers/input: use .dev.of_node instead of .node in struct of_device drivers/serial: use .dev.of_node instead of .node in struct of_device drivers/pcmcia: use .dev.of_node instead of .node in struct of_device drivers/net: use .dev.of_node instead of .node in struct of_device drivers/sbus: use .dev.of_node instead of .node in struct of_device drivers/mmc: use .dev.of_node instead of .node in struct of_device drivers/spi: use .dev.of_node instead of .node in struct of_device drivers/macintosh: use .dev.of_node instead of .node in struct of_device drivers/usb: use .dev.of_node instead of .node in struct of_device drivers/scsi: use .dev.of_node instead of .node in struct of_device drivers/video: use .dev.of_node instead of .node in struct of_device drivers/atm: use .dev.of_node instead of .node in struct of_device drivers/watchdog: use .dev.of_node instead of .node in struct of_device drivers/serial: use .dev.of_node instead of .node in struct of_device arch/microblaze: use .dev.of_node instead of .node in struct of_device arch/powerpc: use .dev.of_node instead of .node in struct of_device arch/sparc: use .dev.of_node instead of .node in struct of_device of/sparc: use dev->of_node instead of dev->archdata.prom_node of: Eliminate dev_archdata_get_node() i2c/of: Allow device node to be passed via i2c_board_info arch/microblaze/include/asm/device.h | 14 ------------ arch/microblaze/include/asm/of_device.h | 1 - arch/microblaze/kernel/of_device.c | 12 ++++------ arch/microblaze/kernel/of_platform.c | 4 ++- arch/powerpc/include/asm/device.h | 22 +++++++------------ arch/powerpc/include/asm/macio.h | 2 +- arch/powerpc/include/asm/of_device.h | 1 - arch/powerpc/kernel/ibmebus.c | 4 ++- arch/powerpc/kernel/of_device.c | 12 ++++------ arch/powerpc/kernel/of_platform.c | 16 +++++++------- arch/powerpc/kernel/pci-common.c | 1 - arch/powerpc/kernel/vio.c | 17 +++++++-------- arch/powerpc/platforms/52xx/mpc52xx_gpio.c | 4 ++- arch/powerpc/platforms/52xx/mpc52xx_gpt.c | 16 +++++++------- arch/powerpc/platforms/52xx/mpc52xx_lpbfifo.c | 6 +++-- arch/powerpc/platforms/82xx/ep8248e.c | 6 +++-- arch/powerpc/platforms/83xx/suspend.c | 2 +- arch/powerpc/platforms/cell/axon_msi.c | 2 +- arch/powerpc/platforms/cell/iommu.c | 5 ++-- arch/powerpc/platforms/pasemi/gpio_mdio.c | 2 +- arch/powerpc/platforms/pasemi/setup.c | 4 ++- arch/powerpc/platforms/ps3/system-bus.c | 1 - arch/powerpc/platforms/pseries/iommu.c | 2 +- arch/powerpc/sysdev/axonram.c | 6 +++-- arch/powerpc/sysdev/bestcomm/bestcomm.c | 8 +++---- arch/powerpc/sysdev/fsl_msi.c | 14 ++++++------ arch/powerpc/sysdev/fsl_pmc.c | 2 +- arch/powerpc/sysdev/pmi.c | 2 +- arch/sparc/include/asm/device.h | 15 ------------- arch/sparc/include/asm/fb.h | 2 +- arch/sparc/include/asm/floppy_64.h | 4 ++- arch/sparc/include/asm/of_device.h | 1 - arch/sparc/include/asm/parport.h | 2 +- arch/sparc/kernel/auxio_64.c | 2 +- arch/sparc/kernel/central.c | 4 ++- arch/sparc/kernel/chmc.c | 10 ++++----- arch/sparc/kernel/ioport.c | 2 +- arch/sparc/kernel/of_device_32.c | 18 +++++++--------- arch/sparc/kernel/of_device_64.c | 29 ++++++++++++------------- arch/sparc/kernel/of_device_common.c | 4 ++- arch/sparc/kernel/pci.c | 13 ++++------- arch/sparc/kernel/pci_common.c | 9 ++++---- arch/sparc/kernel/pci_fire.c | 4 ++- arch/sparc/kernel/pci_msi.c | 18 ++++++++-------- arch/sparc/kernel/pci_psycho.c | 4 ++- arch/sparc/kernel/pci_sabre.c | 4 ++- arch/sparc/kernel/pci_schizo.c | 14 ++++++------ arch/sparc/kernel/pci_sun4v.c | 8 +++---- arch/sparc/kernel/power.c | 4 ++- arch/sparc/kernel/psycho_common.c | 2 +- arch/sparc/kernel/sbus.c | 16 +++++++------- arch/sparc/kernel/time_32.c | 2 +- arch/sparc/kernel/time_64.c | 6 +++-- drivers/ata/pata_macio.c | 2 +- drivers/ata/pata_of_platform.c | 2 +- drivers/atm/fore200e.c | 16 ++++++++------ drivers/cdrom/viocd.c | 2 +- drivers/char/hw_random/n2-drv.c | 2 +- drivers/char/hw_random/pasemi-rng.c | 2 +- drivers/char/ipmi/ipmi_si_intf.c | 4 ++- drivers/char/viotape.c | 2 +- drivers/gpio/pca953x.c | 2 +- drivers/i2c/busses/i2c-mpc.c | 17 ++++++++------- drivers/i2c/i2c-core.c | 3 +++ drivers/ide/pmac.c | 10 ++++----- drivers/infiniband/hw/ehca/ehca_main.c | 11 +++++---- drivers/input/serio/i8042-sparcio.h | 2 +- drivers/input/serio/xilinx_ps2.c | 8 +++---- drivers/leds/leds-gpio.c | 2 +- drivers/macintosh/macio_asic.c | 16 +++++++------- drivers/macintosh/macio_sysfs.c | 6 +++-- drivers/macintosh/rack-meter.c | 4 ++- drivers/macintosh/therm_pm72.c | 2 +- drivers/mmc/host/of_mmc_spi.c | 4 ++- drivers/mmc/host/sdhci-of-core.c | 2 +- drivers/mtd/maps/physmap_of.c | 6 +++-- drivers/mtd/maps/sun_uflash.c | 2 +- drivers/mtd/nand/fsl_elbc_nand.c | 6 +++-- drivers/mtd/nand/pasemi_nand.c | 2 +- drivers/net/can/sja1000/sja1000_of_platform.c | 4 ++- drivers/net/ehea/ehea_main.c | 14 ++++++------ drivers/net/greth.c | 3 ++- drivers/net/ibm_newemac/core.c | 13 ++++++----- drivers/net/ibm_newemac/debug.c | 9 ++++---- drivers/net/ibm_newemac/debug.h | 4 ++- drivers/net/ibm_newemac/mal.c | 28 ++++++++++++------------ drivers/net/ibm_newemac/rgmii.c | 12 +++++----- drivers/net/ibm_newemac/tah.c | 7 +++--- drivers/net/ibm_newemac/zmii.c | 9 ++++---- drivers/net/myri_sbus.c | 2 +- drivers/net/niu.c | 10 ++++----- drivers/net/phy/mdio-gpio.c | 6 +++-- drivers/net/sunbmac.c | 6 +++-- drivers/net/sunhme.c | 8 +++---- drivers/net/sunlance.c | 6 +++-- drivers/net/sunqe.c | 6 +++-- drivers/net/xilinx_emaclite.c | 8 +++---- drivers/of/device.c | 20 +++++++++-------- drivers/of/of_i2c.c | 4 ++- drivers/of/of_mdio.c | 5 ++-- drivers/of/of_spi.c | 1 - drivers/pcmcia/electra_cf.c | 2 +- drivers/sbus/char/bbc_envctrl.c | 4 ++- drivers/sbus/char/bbc_i2c.c | 4 ++- drivers/sbus/char/display7seg.c | 2 +- drivers/sbus/char/envctrl.c | 2 +- drivers/sbus/char/flash.c | 4 ++- drivers/sbus/char/uctrl.c | 2 +- drivers/scsi/ibmvscsi/ibmvfc.c | 2 +- drivers/scsi/ibmvscsi/ibmvscsi.c | 2 +- drivers/scsi/qlogicpti.c | 10 ++++----- drivers/scsi/sun_esp.c | 16 +++++++------- drivers/serial/apbuart.c | 2 +- drivers/serial/mpc52xx_uart.c | 8 +++---- drivers/serial/nwpserial.c | 2 +- drivers/serial/of_serial.c | 4 ++- drivers/serial/pmac_zilog.c | 2 +- drivers/serial/sunhv.c | 2 +- drivers/serial/sunsab.c | 6 +++-- drivers/serial/sunsu.c | 6 +++-- drivers/serial/sunzilog.c | 8 +++---- drivers/serial/uartlite.c | 6 +++-- drivers/spi/mpc52xx_psc_spi.c | 11 +++++---- drivers/spi/mpc52xx_spi.c | 14 ++++++------ drivers/spi/spi_mpc8xxx.c | 6 +++-- drivers/usb/host/ehci-ppc-of.c | 2 +- drivers/usb/host/isp1760-if.c | 2 +- drivers/usb/host/ohci-ppc-of.c | 2 +- drivers/video/cg6.c | 2 +- drivers/video/ffb.c | 2 +- drivers/watchdog/cpwd.c | 2 +- include/linux/i2c.h | 4 +++ sound/aoa/fabrics/layout.c | 2 +- sound/aoa/soundbus/core.c | 8 +++---- sound/aoa/soundbus/i2sbus/control.c | 2 +- sound/aoa/soundbus/i2sbus/core.c | 4 ++- sound/aoa/soundbus/sysfs.c | 4 ++- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/sparc/cs4231.c | 6 +++-- sound/sparc/dbri.c | 2 +- 140 files changed, 422 insertions(+), 458 deletions(-) -- Signature

10 52

[alsa-devel] good USB card with S/PDIF support in ALSA?
by pl bossart 05 Apr '10

05 Apr '10

Hey y'all, I'd like to connect my Fedora laptop to my audio receiver and of course I have no S/PDIF output... Does anyone know of a good USB device, that works well with ALSA and supports compressed multichannel data pass-through? I have seen reports of issues on this mailing list with M-Audio, any other alternatives? Thanks. Pierre

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 05 Apr '10

05 Apr '10

The following changes since commit 2c9504228fe4e4cc65f3debf6d2a6a0f44cf4c2d: Mark Brown (1): Merge branch 'for-2.6.34' 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 The MFD updates for WM8994 are here because the only real consumer of the interrupts on that device is the CODEC driver so it'd create cross tree issues to try to merge the consumer side separately to the controller; Samuel has acked the interrupt controller patch. Daniel Mack (1): ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream Janusz Krzysztofik (1): ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code Mark Brown (9): ASoC: Avoid wraparound in wm_hubs DC servo correction ASoC: Support second DC servo readback method for wm_hubs ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices ASoC: Only do WM8994 bias off transition from standby mfd: Add WM8994 interrupt controller support gpiolib: Implement gpio_to_irq for WM8994 GPIO controller ASoC: Implement interrupt based WM8994 microphone detection Merge branch 'for-2.6.34' into for-2.6.35 Tejun Heo (1): mfd: update gfp/slab.h includes drivers/gpio/wm8994-gpio.c | 12 ++ drivers/mfd/Kconfig | 4 +- drivers/mfd/Makefile | 2 +- drivers/mfd/davinci_voicecodec.c | 1 + drivers/mfd/wm8994-core.c | 43 +++++- drivers/mfd/wm8994-irq.c | 310 ++++++++++++++++++++++++++++++++++++++ include/linux/mfd/wm8994/core.h | 53 +++++++- include/linux/mfd/wm8994/pdata.h | 1 + include/sound/soc.h | 2 +- sound/soc/codecs/wm8994.c | 193 ++++++++++++++++++++---- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 83 ++++++---- sound/soc/codecs/wm_hubs.h | 1 + sound/soc/omap/omap-pcm.c | 17 +- 14 files changed, 648 insertions(+), 77 deletions(-) create mode 100644 drivers/mfd/wm8994-irq.c

2 1

[alsa-devel] snd-hda-intel: Is there a way to specify a quirk for the default bdl_pos_adj value for the given chipset (PCI ID)?
by Aleksander Adamowski 05 Apr '10

05 Apr '10

Hi! I'm working on a patch against GIT for snd-hda-intel that would let my card's capture work properly without losing ability to capture audio (that's an old story available in this thread: http://thread.gmane.org/gmane.linux.alsa.devel/66775). I know how to make position_fix=1 permanent for the given PCI ID (the snd_pci_quirk position_fix_list struct array in hda_intel.c), but there seems to be no way of making a similar quirk for bdl_pos_adj - am I looking in the wrong place? My work on a patch goes in parallel with determining the minimal set of modifications that will solve the problem. Currently, I've determined that the combination of 3 module options described in this forum post solves my problem: http://www.linuxhardware.org/forums/viewtopic.php?p=66&sid=f6e8953338fd94c7… I'm working on isolating the minimum set of these that solves the problem (which requires a couple of hours to be reproduce, so that takes time). I've observed that neither position_fix=1, nor bdl_pos_adj=0 solve the problem by themselves in isolation. I'm unsure whether model=3stack-6ch is necessary at all, because patch_realtek.c already has the model mapping for my combination of PCI IDs and it maps to a different model: SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), Once the minimization is accomplished, I want to prepare a patch against GIT (git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git is the repo to work against, as far as I understand). But since it seems that bdl_pos_adj=0 is required for proper operation, I need a method to map the default for that option in the sources, if there is such a method. I'd be grateful for any pointers. -- Best Regards, Aleksander Adamowski http://olo.org.pl

1 0

[alsa-devel] [PATCH] Enable amplifiers on Acer Inspire 6530G
by Tony Vroon 05 Apr '10

05 Apr '10

After more tests it appears that EAPD needs to be enabled on both the 0x14 and 0x15 NIDs to enable the main speaker and headphone amplifiers. The maximum volume setting is now equal to what the machine achieves under other operating systems. Disabling Front or LFE playback triggers EAPD and disables the amplifier. As such, these two playback switches have been removed from the mixer. Signed-off-by: Tony Vroon <tony(a)linx.net> --- Takashi, I would appreciate it if this could make it for 2.6.34; it could be explained as a regression in the earlier 6530g changes. --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1621,6 +1621,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { */ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Route to built-in subwoofer as well as speakers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, /* Front Mic: set to PIN_IN (empty by default) */ @@ -1632,10 +1637,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Enable speaker output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, /* Enable headphone output */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -8398,9 +8405,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),

2 1

[alsa-devel] USB mixer on MixVibes U46DJ
by Mark Hills 05 Apr '10

05 Apr '10

The audio input and output of this USB soundcard works well, but I could not make much sense of the hardware mixer. The hardware is relatively simple: 2 stereo pairs in, 3 stereo pairs out (or 2 stereo pairs out when operating at 48kHz) for DJs. The mixer presented by 'alsamixer' is not a good match for the hardware. I hoped to create a mapping to fix this, but then it was more difficult because it seems that the mixer controls behave inconsistently. For example, to mute the monitoring of inputs 1-4 one must use: $ amixer cset numid=4 off,off $ amixer cset numid=5 off $ amixer cset numid=4 on,on This, on its own, does not work: $ amixer cset numid=5 It was trial-and-error which got me to this conclusion. Is this common behaviour for a USB device like this, and can I map around it? Or does it mean the hardware is buggy? The mixer presented by the Windows software is nothing special, with normal controls (see http://www.pogo.org.uk/~mark/tmp/u46.png), and it behaves consistently. So it is possible. How can I get this device to sensible mixer controls? Thanks! $ amixer -c 1 contents numid=1,iface=MIXER,name='U46DJ Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=2,iface=MIXER,name='U46DJ Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='U46DJ Capture Volume' ; type=INTEGER,access=rw---R--,values=4,min=0,max=15,step=0 : values=0,0,0,0 | dBminmax-min=0.00dB,max=22.50dB numid=4,iface=MIXER,name='U46DJ Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=5,iface=MIXER,name='U46DJ Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=7,iface=MIXER,name='U46DJ Playback Switch',index=2 ; type=BOOLEAN,access=rw------,values=4 : values=on,on,on,on numid=8,iface=MIXER,name='U46DJ Playback Switch',index=3 ; type=BOOLEAN,access=rw------,values=1 : values=on numid=6,iface=MIXER,name='U46DJ Playback Volume' ; type=INTEGER,access=rw---R--,values=4,min=0,max=31,step=0 : values=23,23,23,23 | dBminmax-min=-34.50dB,max=12.00dB numid=9,iface=MIXER,name='U46DJ Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=6,min=0,max=31,step=0 : values=31,31,31,31,31,31 | dBminmax-min=-46.50dB,max=0.00dB $ lsusb -v -d 0a92:00e1 Bus 003 Device 009: ID 0a92:00e1 EGO SYStems, Inc. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.01 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0a92 EGO SYStems, Inc. idProduct 0x00e1 bcdDevice 1.00 iManufacturer 1 ESI iProduct 2 U46DJ iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 264 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 2 U46DJ AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 117 bInCollection 2 baInterfaceNr( 0) 2 baInterfaceNr( 1) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0201 Microphone bAssocTerminal 6 bNrChannels 4 wChannelConfig 0x000f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) iChannelNames 0 iTerminal 2 U46DJ AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 7 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x03 Mute Volume bmaControls( 2) 0x02 Volume bmaControls( 3) 0x03 Mute Volume bmaControls( 4) 0x02 Volume iFeature 2 U46DJ AudioControl Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 7 bNrInPins 1 baSource( 0) 4 iSelector 2 U46DJ AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0101 USB Streaming bAssocTerminal 4 bSourceID 5 iTerminal 2 U46DJ AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 3 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 iTerminal 2 U46DJ AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 8 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x03 Mute Volume bmaControls( 2) 0x02 Volume bmaControls( 3) 0x03 Mute Volume bmaControls( 4) 0x02 Volume iFeature 2 U46DJ AudioControl Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 9 bNrInPins 2 baSourceID( 0) 8 baSourceID( 1) 1 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 bmControls 0x00 bmControls 0x00 iMixer 0 junk at descriptor end: 00 00 00 00 00 02 AudioControl Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 9 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x03 Mute Volume bmaControls( 2) 0x02 Volume bmaControls( 3) 0x03 Mute Volume bmaControls( 4) 0x02 Volume bmaControls( 5) 0x03 Mute Volume bmaControls( 6) 0x03 Mute Volume iFeature 2 U46DJ AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 1 bSourceID 2 iTerminal 2 U46DJ Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 2 U46DJ Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 2 U46DJ AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 6 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0220 1x 544 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 2 U46DJ Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 2 U46DJ AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 6 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 4 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x01a0 1x 416 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Devices bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 2 U46DJ HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 52 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 20 Device Status: 0x0001 Self Powered -- Mark

1 0

[alsa-devel] [PATCH] ALSA: hda: Add model quirk for Sony VAIO VGN-SZ650N
by Daniel Chen 05 Apr '10

05 Apr '10

Hi, Attached is a patch against sound-2.6 (applicable to both 2.6.32.11 and 2.6.33.2) that adds the original reporter's SSID for a quirk entry, allowing for reliably audible sound. Best, -Dan

2 3

[alsa-devel] [PATCH] WM8750: Convert to new API
by Marek Vasut 05 Apr '10

05 Apr '10

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code around. Hugely inspired by WM8753 which was already converted. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- sound/soc/codecs/wm8750.c | 297 +++++++++++++++++++++------------------------ 1 files changed, 140 insertions(+), 157 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index ee08408..f01513e 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,11 +29,6 @@ #include "wm8750.h" -/* codec private data */ -struct wm8750_priv { - unsigned int sysclk; -}; - /* * wm8750 register cache * We can't read the WM8750 register space when we @@ -53,6 +48,13 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; +/* codec private data */ +struct wm8750_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8750_reg)]; +}; + #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* @@ -695,25 +697,90 @@ static int wm8750_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8750_codec; + +static int wm8750_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8750_codec) { + dev_err(&pdev->dev, "WM8750 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8750_codec; + codec = wm8750_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); + wm8750_add_widgets(codec); + + return 0; + +err: + return ret; +} + +/* power down chip */ +static int wm8750_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_wm8750 = { + .probe = wm8750_probe, + .remove = wm8750_remove, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev, - enum snd_soc_control_type control) +static int wm8750_register(struct wm8750_priv *wm8750, + enum snd_soc_control_type control) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = &wm8750->codec; int reg, ret = 0; + if (wm8750_codec) { + dev_err(codec->dev, "Multiple WM8750 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "WM8750"; codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); - codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->private_data = wm8750; + codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1; + codec->reg_cache = &wm8750->reg_cache; + codec->private_data = wm8750; + + memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache)); ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); if (ret < 0) { @@ -727,13 +794,6 @@ static int wm8750_init(struct snd_soc_device *socdev, goto err; } - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto err; - } - /* charge output caps */ wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -755,19 +815,37 @@ static int wm8750_init(struct snd_soc_device *socdev, reg = snd_soc_read(codec, WM8750_RINVOL); snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); - return ret; + wm8750_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + ret = snd_soc_register_dais(&wm8750_dai, 1); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); err: - kfree(codec->reg_cache); + kfree(wm8750); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; +static void wm8750_unregister(struct wm8750_priv *wm8750) +{ + wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(&wm8750_dai, 1); + snd_soc_unregister_codec(&wm8750->codec); + kfree(wm8750); + wm8750_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -781,24 +859,26 @@ static struct snd_soc_device *wm8750_socdev; static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; - ret = wm8750_init(socdev, SND_SOC_I2C); - if (ret < 0) - pr_err("failed to initialise WM8750\n"); + codec = &wm8750->codec; + codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8750); - return ret; + codec->dev = &i2c->dev; + + return wm8750_register(wm8750, SND_SOC_I2C); } static int wm8750_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8750_priv *wm8750 = i2c_get_clientdata(client); + wm8750_unregister(wm8750); return 0; } @@ -817,66 +897,31 @@ static struct i2c_driver wm8750_i2c_driver = { .remove = wm8750_i2c_remove, .id_table = wm8750_i2c_id, }; - -static int wm8750_add_i2c_device(struct platform_device *pdev, - const struct wm8750_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8750_i2c_driver); - return -ENODEV; -} #endif #if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; + + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + codec = &wm8750->codec; codec->control_data = spi; + codec->dev = &spi->dev; - ret = wm8750_init(socdev, SND_SOC_SPI); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8750\n"); + dev_set_drvdata(&spi->dev, wm8750); - return ret; + return wm8750_register(wm8750, SND_SOC_SPI); } static int __devexit wm8750_spi_remove(struct spi_device *spi) { + struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev); + wm8750_unregister(wm8750); return 0; } @@ -891,93 +936,31 @@ static struct spi_driver wm8750_spi_driver = { }; #endif -static int wm8750_probe(struct platform_device *pdev) +static int __init wm8750_modinit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec; - struct wm8750_priv *wm8750; int ret; - - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); - if (wm8750 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8750; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; - - ret = -ENODEV; - #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - ret = wm8750_add_i2c_device(pdev, setup); - } + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8750 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - ret = spi_register_driver(&wm8750_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } + ret = spi_register_driver(&wm8750_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8750 SPI driver: %d\n", ret); #endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; + return 0; } +module_init(wm8750_modinit); -/* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +static void __exit wm8750_exit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8750_spi_driver); #endif - kfree(codec->private_data); - kfree(codec); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); - -static int __init wm8750_modinit(void) -{ - return snd_soc_register_dai(&wm8750_dai); -} -module_init(wm8750_modinit); - -static void __exit wm8750_exit(void) -{ - snd_soc_unregister_dai(&wm8750_dai); } module_exit(wm8750_exit); -- 1.7.0

3 5

[alsa-devel] Nvidia GTS 250M HDMI audio output
by Dark Shadow 04 Apr '10

04 Apr '10

What can I do to help get this working on my laptop. So far it sort of shows up in Alsa mixer but with no controls and a strange chip name "Nvidia ID d" I am not a programmer but if you tell me any commands I will give the output. Here is the output of alsa-info.sh http://www.alsa-project.org/db/?f=1f21b5d7ee864d157cbddd3bd93de527f5aa19bf Alsa the *reg* and *debug* did not exist but *codec* did so here is the output. Here is cat /proc/asound/card1/codec#* $ cat /proc/asound/card1/codec#* Codec: Nvidia ID d Address: 0 Function Id: 0x1 Vendor Id: 0x10de000d Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Codec: Nvidia ID d Address: 1 Function Id: 0x1 Vendor Id: 0x10de000d Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Codec: Nvidia ID d Address: 2 Function Id: 0x1 Vendor Id: 0x10de000d Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Codec: Nvidia ID d Address: 3 Function Id: 0x1 Vendor Id: 0x10de000d Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x72b1: 8-Channels Digital Stripe CP Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Unsolicited: tag=00, enabled=0 Node 0x05 [Pin Complex] wcaps 0x407381: 8-Channels Digital CP Pincap 0x09000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04

1 2

[alsa-devel] [PATCH - powermac] Reverse HP detection on G4 DA
by Risto Suominen 04 Apr '10

04 Apr '10

From: Risto Suominen <Risto.Suominen(a)gmail.com> Reverse headphone detection bit on PowerMac G4 Digital Audio (Tumbler). Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.32.9 (probably any 2.6), ALSA 1.0.22.1. The patch is available here, too: http://ristosu.wippiespace.com/pub/alsa-tumbler-1.0.22.1-p15.diff --- a/alsa-kernel/ppc/tumbler.c.org 2009-12-28 16:21:33.000000000 +0200 +++ b/alsa-kernel/ppc/tumbler.c 2010-03-16 19:59:01.000000000 +0200 @@ -30,6 +30,7 @@ #include <linux/kmod.h> #include <linux/slab.h> #include <linux/interrupt.h> +#include <linux/string.h> #include <sound/core.h> #include <asm/io.h> #include <asm/irq.h> @@ -46,6 +47,8 @@ #define DBG(fmt...) #endif +#define IS_G4DA (machine_is_compatible("PowerMac3,4")) + /* i2c address for tumbler */ #define TAS_I2C_ADDR 0x34 @@ -1134,7 +1137,7 @@ static long tumbler_find_device(const ch gp->inactive_val = (*base) ? 0x4 : 0x5; } else { const u32 *prop = NULL; - gp->active_state = 0; + gp->active_state = IS_G4DA && !strcmp(device, "keywest-gpio15"); gp->active_val = 0x4; gp->inactive_val = 0x5; /* Here are some crude hacks to extract the GPIO polarity and

2 1

[alsa-devel] [PATCH 5/5] Echoaudio - add suspend/resume support
by Oliver Lupton 04 Apr '10

04 Apr '10

Just thought I'd mention that I've tested these five patches applied to a fedora 12 2.6.32.9 kernel with an Indigo IO card and it appears to now work fine when resuming from standby (which it didn't previously -- I had to remove/reinsert the card and close all applications with the device open) Cheers, Olli

2 1

Re: [alsa-devel] [PATCH 3/3] sound/pci: Eliminate use after free
by Takashi Iwai 04 Apr '10

04 Apr '10

At Fri, 2 Apr 2010 14:47:59 +0200 (CEST), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > Use the call to snd_card_free in the error handling code at the end of the > function, as in the other error cases. > > A simplified version of the semantic patch that finds this problem is as > follows: (http://coccinelle.lip6.fr/) > > // <smpl> > @@ > expression E,E2; > @@ > > snd_card_free(E) > ... > ( > E = E2 > | > * E > ) > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Applied now. Thanks. Takashi

1 0

Re: [alsa-devel] [patch] i2c: cleanup: change parameter to pointer
by Takashi Iwai 04 Apr '10

04 Apr '10

At Fri, 2 Apr 2010 14:29:23 +0300, Dan Carpenter wrote: > > We actually pass an array of 7 chars not 5. > This silences a smatch warning. > > Signed-off-by: Dan Carpenter <error27(a)gmail.com> Thanks. I applied together with the fx in the declaration part in include/sound/ak4113.h. Takashi > --- > Compile tested only. :/ > > diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c > index fff62cc..971a84a 100644 > --- a/sound/i2c/other/ak4113.c > +++ b/sound/i2c/other/ak4113.c > @@ -70,7 +70,7 @@ static int snd_ak4113_dev_free(struct snd_device *device) > } > > int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, > - ak4113_write_t *write, const unsigned char pgm[5], > + ak4113_write_t *write, const unsigned char *pgm, > void *private_data, struct ak4113 **r_ak4113) > { > struct ak4113 *chip; >

1 0

[alsa-devel] [PATCH - powermac] Add debug log
by Risto Suominen 04 Apr '10

04 Apr '10

From: Risto Suominen <Risto.Suominen(a)gmail.com> Add some debug log in tumbler.c. Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.32.9 (probably any 2.6), ALSA 1.0.22.1. This patch is available here, too: http://ristosu.wippiespace.com/pub/alsa-tumbler-1.0.22.1-p16d.diff It has no effect, unless one changes '#undef DEBUG' to '#define DEBUG' in the beginning of the file (and possibly in tumbler.patch, too). --- a/alsa-kernel/ppc/tumbler.c-p15 2010-03-16 19:59:01.000000000 +0200 +++ b/alsa-kernel/ppc/tumbler.c 2010-03-23 21:34:46.000000000 +0200 @@ -246,6 +246,7 @@ static int tumbler_set_master_volume(str snd_printk(KERN_ERR "failed to set volume \n"); return -EINVAL; } + DBG("(I) succeeded to set volume (%u, %u)\n", left_vol, right_vol); return 0; } @@ -356,6 +357,7 @@ static int tumbler_set_drc(struct pmac_t snd_printk(KERN_ERR "failed to set DRC\n"); return -EINVAL; } + DBG("(I) succeeded to set DRC (%u, %u)\n", val[0], val[1]); return 0; } @@ -392,6 +394,7 @@ static int tumbler_set_drc(struct pmac_t snd_printk(KERN_ERR "failed to set DRC\n"); return -EINVAL; } + DBG("(I) succeeded to set DRC (%u, %u)\n", val[0], val[1]); return 0; }

1 0

[alsa-devel] [PATCH - powermac] Lineout detection on G4 DA
by Risto Suominen 04 Apr '10

04 Apr '10

From: Risto Suominen <Risto.Suominen(a)gmail.com> Lineout (Pro Speaker) detection on PowerMac G4 Digital Audio (Tumbler). Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.32.9 (probably any 2.6), ALSA 1.0.22.1. The patch is available here, too: http://ristosu.wippiespace.com/pub/alsa-tumbler-1.0.22.1-p16.diff This has not been tested. Hopefully someone with that kind of machine and those speakers can do it. --- a/alsa-kernel/ppc/tumbler.c-p15 2010-03-16 19:59:01.000000000 +0200 +++ b/alsa-kernel/ppc/tumbler.c 2010-03-23 21:42:03.000000000 +0200 @@ -1137,7 +1141,8 @@ static long tumbler_find_device(const ch gp->inactive_val = (*base) ? 0x4 : 0x5; } else { const u32 *prop = NULL; - gp->active_state = IS_G4DA && !strcmp(device, "keywest-gpio15"); + gp->active_state = IS_G4DA + && !strncmp(device, "keywest-gpio1", 13); gp->active_val = 0x4; gp->inactive_val = 0x5; /* Here are some crude hacks to extract the GPIO polarity and @@ -1315,6 +1320,9 @@ static int __devinit tumbler_init(struct if (irq <= NO_IRQ) irq = tumbler_find_device("line-output-detect", NULL, &mix->line_detect, 1); + if (IS_G4DA && irq <= NO_IRQ) + irq = tumbler_find_device("keywest-gpio16", + NULL, &mix->line_detect, 1); mix->lineout_irq = irq; tumbler_reset_audio(chip);

1 0

[alsa-devel] [RFC][PATCH] ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
by Janusz Krzysztofik 02 Apr '10

02 Apr '10

With recent (2.6.34) chnages in PCM handling, capture stopped working on my OMAP1510 based Amstrad Delta videophone. Using 2.6.34-rc2, I was able to correct the problem in 3 different ways: 1. reverting commit 7b3a177b0d4f92b3431b8dca777313a07533a710, 2. enabling additional jiffies check with echo 4 >/proc/asound/card0/pcm0c0/xrun_debug 3. applying the patch below. Since I wasn't able to reproduce the problem on my i686 PC, I guess the problem is probably machine specific. The patch reuses the method for software emulation of missing hardware pointer, already implemented for playback on OMAP1510. It's possible that event if a hardware pointer is available for capture on this machine, its behaviour may be not compatible with what upper layer expects. If you think the problem may be more general and should be solved differently, on a higher level, I can try to work more on it if you give me a hint. If the patch gets accepted, I suggest it goes as a fix in the current release cycle. Created and tested against linux-2.6.34-rc2. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- git/sound/soc/omap/omap-pcm.c.orig 2010-03-25 16:00:13.000000000 +0100 +++ git/sound/soc/omap/omap-pcm.c 2010-03-28 05:16:42.000000000 +0200 @@ -60,12 +60,11 @@ static void omap_pcm_dma_irq(int ch, u16 struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { + if ((cpu_is_omap1510())) { /* * OMAP1510 doesn't fully support DMA progress counter * and there is no software emulation implemented yet, - * so have to maintain our own playback progress counter + * so have to maintain our own progress counters * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); @@ -189,8 +188,7 @@ static int omap_pcm_prepare(struct snd_p dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else @@ -248,14 +246,15 @@ static snd_pcm_uframes_t omap_pcm_pointe dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (cpu_is_omap1510()) { + offset = prtd->period_index * runtime->period_size; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else if (!(cpu_is_omap1510())) { + } else { ptr = omap_get_dma_src_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else - offset = prtd->period_index * runtime->period_size; + } if (offset >= runtime->buffer_size) offset = 0;

3 8

[alsa-devel] [PATCH 4/4] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 02 Apr '10

02 Apr '10

2 3

Re: [alsa-devel] mx27_wm9712.c memory leak
by David Lee 02 Apr '10

02 Apr '10

2010/3/31 David Lee <david.lee.csj(a)gmail.com> > Dear alsaers: > > I am doing a test for imx27 platform with wm9712 as its sound card. When my > programe that used for recording audio is running in background, I use top > command to watch system condition. But the memory used and catchedd is > rising as time past.I check my application programe , I dont think there is > any wrong.So I think there's some wrong in audio drivers. > Pleae check attachment for the driver code and my programe . > > Any suggestion will be greatly appreciate. > > Best regards. > > > david sever of GOD >

1 0

[alsa-devel] alsactl restore: unknown hardware: ymf724f
by Angel Tsankov 02 Apr '10

02 Apr '10

Hello, I run 'alsactl restore' on a machine with 2 sound cards -- a built-in Intel Corporation 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller (rev 02) and a non-built-in Yamaha Corporation YMF-724F [DS-1 Audio Controller] (rev 03) -- and get the following message: Unknown hardware: "YMF724F" "SigmaTel STAC9700,83,84" "AC97a:83847600" "0x1073" "0x000d" Hardware is initialized using a guess method As a consequence the volume levels of the Yamaha card do not get restored to the levels stored in /etc/asound.state. The volume levels of the built-in card however are properly restored. The asound.state file has been created by executing 'alsactl store'. The kernel has been built with support for ALSA. I've built and installed the kernel modules for both cards (not the ones in the alsa-driver package but those that come with kernel version 2.6.30.2). Any ideas why alsactl cannot find the hardware it has previously identified as "YMF724F", "SigmaTel STAC9700,83,84", and so on? Regards, Angel Tsankov

4 26

[alsa-devel] Memory corruption in ASoC
by Daniel Mack 01 Apr '10

01 Apr '10

Hi, I've been hunting a serious memory corruption in the ASoC core for some time now and finally managed to figure out where the actual problem lies. We're currently using a 2.6.32.10, on a PXA303 based series of devices (see sound/soc/pxa/raumfeld.c). So here's the demo program, reduced to the max: http://caiaq.de/download/tmp/alsa.c The thread launched to call getifaddrs() every 10ms is actually just there because a) this was the way the bug was originally triggered in our application and b) this seems to be a good way to put the kernel's memory management under fire, so you don't have to wait long for a crash. Important to mention is that the bug is only triggered when snd_pcm_set_params() is called for both streams, and only if more than one device is opened. This sources smash my kernel more or less instantly with a backtrace similar to the one at the end of this mail. You might also end up getting hung tasks, and the kernel detector which complains about them after awhile also points out code somewhere in the netlink area (which is of course innocent, we clearly deal with a memory corruption here). So - look at these simplified sniplets: // called for stream #0 pxa_ssp_hw_params(...) { // sound/soc/pxa/pxa-ssp.c ... cpu_dai->dma_data = ssp_get_dma_params(...) ... } // called for stream #1 pxa_ssp_startup(...) { // sound/soc/pxa/pxa-ssp.c ... if (cpu_dai->dma_data) { kfree(cpu_dai->dma_data); ... } } // called for stream #0 __pxa2xx_pcm_hw_free(...) { // sound/arm/pxa2xx-pcm-lib.c ... if (rtd && rtd->params && rtd->params->drcmr) *rtd->params->drcmr = 0; // <-- BOOM ... } So the first stream (PLAYBACK) already exported its dma_data which is now freed by the code initializing the second stream (CAPTURE). This corrupts all existing users of course, and in this particular case, the cleanup in __pxa2xx_pcm_hw_free() dereferences a pointer which is bogus. What I really don't understand is why this didn't crash a lot earlier for many more users. So how is this supposed to be fixed? Should dma_data become a member of some per-stream instance? I believe that also other platforms than PXA are actually affected - am I right? Daniel [ 42.158232] Unable to handle kernel NULL pointer dereference at virtual address 00000008 [ 42.166289] pgd = c7f14000 [ 42.168970] [00000008] *pgd=a7eff031, *pte=00000000, *ppte=00000000 [ 42.175187] Internal error: Oops: 17 [#1] [ 42.179163] last sysfs file: /sys/devices/virtual/vc/vcsa1/dev [ 42.184950] Modules linked in: [ 42.187983] CPU: 0 Not tainted (2.6.32.10 #99) [ 42.192769] PC is at netlink_dump_start+0x154/0x190 [ 42.197609] LR is at 0x306874 [ 42.200552] pc : [<c029dc38>] lr : [<00306874>] psr: 20000093 [ 42.200562] sp : c7ecdd10 ip : 00000000 fp : c053c648 [ 42.211954] r10: c7844000 r9 : c7ec8200 r8 : 0000045e [ 42.217141] r7 : c7edc140 r6 : c7ec8600 r5 : 00000000 r4 : 00000000 [ 42.223620] r3 : 20000093 r2 : 20000013 r1 : c79df600 r0 : 0000006c [ 42.230102] Flags: nzCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment user [ 42.237276] Control: 0000397f Table: a7f14018 DAC: 00000015 [ 42.242979] Process alsa (pid: 1118, stack limit = 0xc7ecc278) [ 42.248769] Stack: (0xc7ecdd10 to 0xc7ece000) [ 42.253099] dd00: c0296b94 c7ec8200 00000006 00000000 [ 42.261227] dd20: c7ec8600 c0298118 00000000 c7ecde40 c7ecdd5c c7ed8a80 c005030c c7ec8200 [ 42.269359] dd40: c7ed8a80 c0298040 c7ecdd7c c7ec8600 00000000 00000038 c053c648 c029ea10 [ 42.277493] dd60: c7ed8a80 00000014 c7ed8a80 c0298030 c7844000 c029e6e0 00000014 7fffffff [ 42.285625] dd80: 00000014 00000000 c7ed8a80 c7ec8600 c7ecdf5c c7672580 c7ecde80 00000000 [ 42.293748] dda0: 00000014 c029efb0 00000000 c005118c 00000000 00000000 0000045e 00000000 [ 42.301872] ddc0: 00000000 00000000 00000000 00000000 00000000 c7ecdf5c 00000014 be1ff9ec [ 42.309997] dde0: c7ecc000 0000000c be1ffa84 c027f730 c0069880 c7ecddf4 00000000 00000001 [ 42.318120] de00: ffffffff 00000000 00000000 00000000 00000000 00000000 c7a10f00 00000000 [ 42.326247] de20: 00000000 00000000 be1ff9ec c7a10f00 c0069880 c7ecde34 c7ecde34 00120000 [ 42.334379] de40: 00000014 be1ff9ec c7ecde80 00001000 c7672580 c0050518 c7ecdd68 c7ecdf5c [ 42.342503] de60: 00000000 be1ffa2c c7672580 be1ffa28 be1ffa44 be1ffa44 0000000c 00000000 [ 42.350628] de80: 0000000c c02802c4 c7672580 00000014 c7672580 c0280420 c7ecddb8 c7ecdf5c [ 42.358751] dea0: 00000000 00000000 c027ea8c 00000000 c7ecdedc 00000014 be1ff9ec c7ecdedc [ 42.366876] dec0: 0000000c c7672580 00000000 c7ecdedc 00000014 c027fa08 c00440c4 00000010 [ 42.375001] dee0: 00000000 00000000 0000000c c02802c4 0000000c c7672580 c7ecdf04 c0280690 [ 42.383124] df00: c7ecdf08 00000010 0000045e 00000000 00000000 c76747d8 c7674780 be1fea1c [ 42.391249] df20: 00000fec c7ecdf4c 00000005 c7ec9f80 c7672580 00000000 c03fd590 c027f3d8 [ 42.399373] df40: c03fd590 c0532ab0 c7e8ad80 00000000 00000000 c047c88f 00000005 c7ecdedc [ 42.407497] df60: 0000000c c7ecdf78 00000001 00000000 00000000 00000000 be1ff9ec 00000000 [ 42.415620] df80: 00000001 00000000 00000000 be1ff9ec 0000000c 0000000c 00000122 c00440c4 [ 42.423746] dfa0: 00000000 c0043f40 be1ff9ec 0000000c 00000005 be1ff9d8 00000014 00000000 [ 42.431870] dfc0: be1ff9ec 0000000c 0000000c 00000122 be1ff9ec 00001000 00000000 be1ffa84 [ 42.439995] dfe0: 00000000 be1ff9c0 4011e8a4 40123848 60000010 00000005 00000000 00000000 [ 42.448139] [<c029dc38>] (netlink_dump_start+0x154/0x190) from [<c0298118>] (rtnetlink_rcv_msg+0xd8/0x220) [ 42.457739] [<c0298118>] (rtnetlink_rcv_msg+0xd8/0x220) from [<c029ea10>] (netlink_rcv_skb+0x4c/0xb0) [ 42.466904] [<c029ea10>] (netlink_rcv_skb+0x4c/0xb0) from [<c0298030>] (rtnetlink_rcv+0x1c/0x2c) [ 42.475640] [<c0298030>] (rtnetlink_rcv+0x1c/0x2c) from [<c029e6e0>] (netlink_unicast+0x22c/0x2d0) [ 42.484550] [<c029e6e0>] (netlink_unicast+0x22c/0x2d0) from [<c029efb0>] (netlink_sendmsg+0x260/0x274) [ 42.493806] [<c029efb0>] (netlink_sendmsg+0x260/0x274) from [<c027f730>] (sock_sendmsg+0xa0/0xc4) [ 42.502630] [<c027f730>] (sock_sendmsg+0xa0/0xc4) from [<c027fa08>] (sys_sendto+0xb0/0xd4) [ 42.510866] [<c027fa08>] (sys_sendto+0xb0/0xd4) from [<c0043f40>] (ret_fast_syscall+0x0/0x28) [ 42.519344] Code: ebfffe2e e10f2000 e3823080 e121f003 (e5943008) [ 42.525679] ---[ end trace 3209e05a50ffc8c1 ]---

5 24

[alsa-devel] line in recording w/ alsa - suggestions for a snd card?
by Lars Schotte 01 Apr '10

01 Apr '10

hi folks, i ve given up on that problems w/ recording because it leads to LQ recordings in the best case. so does anybody have a suggestion for a sound card PCI-internal that works w/ ALSA and records HQ from line-in that can be used for digitalizing? frankly, as long as it would work money isnt important but on the other hand it doesnt need to be 192 kHz one for >100 eur. so as long as the card does record HQ in 44/48 kHz would be perfectly OK and the cheaper, the better. thanks. -- ------------------- Lars Schotte @ romy

3 5

[alsa-devel] CA0106
by Lars Schotte 01 Apr '10

01 Apr '10

hi, Clemens wrote that " The CA0106 is supported, and > used on several cheap Sound Blaster cards, e.g. the SB 5.1 VX " CA0106 is supported, but I looked at the website and there is: http://www.alsa-project.org/main/index.php/Matrix:Vendor-Creative_Labs CA0106 Details [PCI] Digital/Analog input does not work yet. Needs more development work. so ... that seems like line in doesnt work yet, but on the unofficial http://alsa.opensrc.org/index.php/Ca0106 is " Capture from MIC an LINE IN input. " so looks like that already works, but maybe recently. that could be relevant to me as far as not buy a VX when I want to make it work on debian stable w/o changes, but however, I could install the git version or a snapshot. its just not clear all. -- ------------------- Lars Schotte @ romy

1 0

[alsa-devel] C-Media CMI8738-LX noise in recording from line in
by Lars Schotte 01 Apr '10

01 Apr '10

I have this Sweex Sorround sound card PCI and I want to use it for recording / digitalizing / magnetic and other media types through line-in. I have to use that card (or another) because the built-in sound card doesnt work /is broken/. The soundcard has following chipset - C-Media CMI8738-LX. So I try to record from the line in input, which works for listening, so line in input is played good through line out then. I dont know if there is a passthrought or sth. however, when I record it, using audacity, or even alsarecord / it doesnt matter / the sound is reported and also actually is too silent and there is a lot of noise. So with that noise, this card is for digitalizing useless. this soundcard has physically no shared connector with the rear, or bass or whatever. so there are 5 connectors, line in, mic in, front, rear and cent/sub. each for himself. maybe thats the case, that the old ones did have shared one and therefore it breaks the recording or whatever, just a theory. because a few years ago i had an old cmipci and that one worked fine. just to be clear, there is sound, so it records, the problem is that its too silent and there is a lot of noise in it. i write this note because I saw some reports that suggest that it produces _only_ noise in recording. thats not the case here. -- ------------------- Lars Schotte @ romy

4 5

[alsa-devel] start threshold in alsa plugin
by Stefan Schoenleitner 01 Apr '10

01 Apr '10

Hi, in my io-plugin code I would like to know the start threshold value. AFAIK the start threshold is set to a default value that depends on the configured buffer size SND_PCM_IOPLUG_HW_BUFFER_BYTES. In my case the buffer size is 160*2*1 .. 160*2*2 bytes which is 1..2 periods as one period has 160 samples and each sample has 16 bit (2 bytes). * How exactly is the default threshold value derived from the buffer size ? I also tried to get the start threshold value from the PCM: ------------------------------------------------------------------------------ if (snd_pcm_sw_params_malloc(&sw_params)<0) return -1; if ((err=snd_pcm_sw_params_current(plug_info->io.pcm, sw_params))<0) { printf("Unable to determine current swparams: %s\n", snd_strerror(err)); return -err; } if (snd_pcm_sw_params_get_start_threshold(sw_params, &param)<0) return -1; DBG("start threshold: %lu", param); ------------------------------------------------------------------------------ Unfortunately the call to snd_pcm_sw_params_current() fails (i/o error). * Do you know why that call fails ? * Is this the right way to get the start threshold value from a pcm ? I'm calling the above code at the end of the plugin entry function SND_PCM_PLUGIN_DEFINE_FUNC() after all hw constraints (including the buffer size) have been set. Maybe the plugin entry function is just not the right place to retrieve the sw configuration from the PCM as it is too early and the sw config of the pcm has not yet been set up ? * Should I use a callback like sw_params to retrieve the default config ? cheers, Stefan

1 0

[alsa-devel] Regressions: MSI vs HDA-Intel
by Torsten Kaiser 01 Apr '10

01 Apr '10

I have two systems that use the alsa hda-intel driver, both show regressions in relation to MSI. System 1: Athlon(tm) X2 BE-2400 with an AMD RS690/SB600 chipset 00:14.2 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (Mainboard is a MSI MS-7368) After upgrading from 2.6.31 to 2.6.33 I get a delay during bootup: [ 1.155925] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 1.159143] input: AT Translated Set 2 keyboard as /devices/platform/i8042/serio0/input/input2 [ 1.232595] firewire_core: created device fw0: GUID 00dc10000129c48f, S400 [ 1.252677] hda_codec: ALC888: BIOS auto-probing. [ 1.259226] HDA Intel 0000:01:05.2: PCI INT B -> GSI 19 (level, low) -> IRQ 19 [ 1.260293] HDA Intel 0000:01:05.2: irq 27 for MSI/MSI-X [ 1.314712] input: GenPS/2 Genius Mouse as /devices/platform/i8042/serio1/input/input3 [ 4.322508] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 [ 5.332508] hda-intel: No response from codec, disabling MSI: last cmd=0x000f0000 [ 6.342508] hda-intel: Codec #0 probe error; disabling it... [ 7.392508] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x000f0000 [ 7.430464] ALSA device list: [ 7.431549] #0: HDA ATI SB at 0xfe7f4000 irq 16 [ 7.432656] #1: HDA ATI HDMI at 0xfe9e8000 irq 27 2.6.34-rc2 does not boot on this system, something related to the new bootmem allocator, so I can't say if this might already be fixed for 2.6.34. Should the system be blacklisted like the two Asus Athlon X2 boards, or is this some other bug? System 2: Opteron(tm) 2218 with an nVidia MCP55 chipset 00:06.1 Audio device [0403]: nVidia Corporation MCP55 High Definition Audio [10de:0371] (rev a2) (Mainboard is an ASUS KFN5-D SLI) Commit 80c43ed724797627d8f86855248c497a6161a214 disabled MSI for this system, but the MSI mode was working fine for all 2.6.33 versions I used. Should the generic blacklist disable MSI for all Athlon X2 instead of all nNvidia chipsets? If you need more information, please ask, I will try to provide anything you need. Thanks, Torsten

3 7

[alsa-devel] Handling of the DXS controls in the via82xx driver
by Soeren D. Schulze 01 Apr '10

01 Apr '10

Hi everyone, before my latest upgrade to Linux 2.6.32 (yes, sorry, it's Debian), I had four levers in my alsamixer: VIA DXS, VIA DXS 1, VIA DXS 2 and VIA DXS 3. They controlled the hardware mixing when accessing the sound device multiple times. Now they have disappeared. I think I can pretty certainly trace this down to commit 2fb930b53f513cbc4c102d415d2923a8a7091337 to Linux [http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.32.y.git;a=commi…], which "move[s] DXS volume controls to PCM interface". As I'm not an ALSA developer, I don't realize the intended effect of this change, but the actual effect is that I cannot change the DXS volumes any more. mplayer still controls the global PCM volume. Well, in fact, there is one way. While an mplayer is running, I can do: amixer cset numid=47,iface=PCM,name='PCM Playback Volume' 24,24 When restarting mplayer, the volume is reset to 31,31 (maximum). Commit 3d00941371a765779c4e3509214c7e5793cce1fe [http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.33.y.git;a=commi…] seems to be responsible for this. The bad thing about it is that 31,31 is a really bad setting for my poorly engineered sound card. It distorts a lot. I suspect that the real *bug* is that mplayer does not use the per-channel DXS controls even though it's supposed to (which seems to be why the DXS controls are called "PCM Playback Volume"). Personally, rather than fixing this bug, I would just revert to the former state. I was always happy with all the DXS levers set to 75%. Thank you Sören

3 6

[PATCH] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 31 Mar '10

31 Mar '10

Since basic SSP features are shared between PXA2xx and PXA168, the difference is focused on clock generating. Now split ssp code into two parts. One is for general ssp feature. The other is for pxa2xx parts. Update the SSP timing configuration in hw_params(). The SSP timing is verified on I2S and Left J interface of PXA168 with 16-bit sample stream. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 8 +- sound/soc/pxa/Makefile | 4 +- sound/soc/pxa/magician.c | 38 ++-- sound/soc/pxa/pxa-ssp.c | 728 +++++++++++--------------------------------- sound/soc/pxa/pxa-ssp.h | 52 ++-- sound/soc/pxa/pxa2xx-ssp.c | 318 +++++++++++++++++++ sound/soc/pxa/pxa2xx-ssp.h | 48 +++ sound/soc/pxa/raumfeld.c | 14 +- sound/soc/pxa/zylonite.c | 6 +- 9 files changed, 603 insertions(+), 613 deletions(-) create mode 100644 sound/soc/pxa/pxa2xx-ssp.c create mode 100644 sound/soc/pxa/pxa2xx-ssp.h diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 78e6121..7be1d5f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -21,7 +21,7 @@ config SND_PXA2XX_SOC_AC97 config SND_PXA2XX_SOC_I2S tristate -config SND_PXA_SOC_SSP +config SND_PXA2XX_SOC_SSP tristate select PXA_SSP @@ -113,7 +113,7 @@ config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_WM9713 help Say Y if you want to add support for SoC audio on the @@ -122,7 +122,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 help @@ -132,7 +132,7 @@ config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN select SND_PXA2XX_SOC_I2S - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_UDA1380 help Say Y if you want to add support for SoC audio on the diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index f3e08fd..33c1579 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -2,12 +2,12 @@ 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-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-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_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 4c8d99a..da6ff83 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -34,7 +34,7 @@ #include "../codecs/uda1380.h" #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define MAGICIAN_MIC 0 #define MAGICIAN_MIC_EXT 1 @@ -89,7 +89,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int acps, acds, width, rate; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int div4 = PXA2XX_SSP_CLK_SCDB_4; int ret = 0; rate = params_rate(params); @@ -106,11 +106,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_16; break; default: /* 32 */ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; } break; case 11025: @@ -118,11 +118,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4; break; default: /* 32 */ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; } break; case 22050: @@ -130,11 +130,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 44100: @@ -142,11 +142,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 48000: @@ -154,11 +154,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 96000: @@ -167,12 +167,12 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; break; default: /* 32 */ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; + div4 = PXA2XX_SSP_CLK_SCDB_1; break; } break; @@ -195,20 +195,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set audio clock as clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; /* set the SSP audio system clock ACDS divider */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); + PXA2XX_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; /* set the SSP audio system clock SCDB divider4 */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); + PXA2XX_SSP_AUDIO_DIV_SCDB, div4); if (ret < 0) return ret; @@ -427,7 +427,7 @@ static struct snd_soc_dai_link magician_dai[] = { { .name = "uda1380", .stream_name = "UDA1380 Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], .init = magician_uda1380_init, .ops = &magician_playback_ops, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed54bef..bd5e800 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -5,6 +5,9 @@ * Author: Liam Girdwood * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * + * Copyright 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 as published by the * Free Software Foundation; either version 2 of the License, or (at your @@ -14,44 +17,20 @@ * o Test network mode for > 16bit sample size */ -#include <linux/init.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/soc.h> #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include <plat/ssp.h> #include "pxa2xx-pcm.h" #include "pxa-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 -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -63,7 +42,7 @@ static void dump_registers(struct ssp_device *ssp) pxa_ssp_read_reg(ssp, SSACD)); } -static void pxa_ssp_enable(struct ssp_device *ssp) +void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +50,7 @@ static void pxa_ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void pxa_ssp_disable(struct ssp_device *ssp) +void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -84,7 +63,7 @@ struct pxa2xx_pcm_dma_data { char name[20]; }; -static struct pxa2xx_pcm_dma_params * +struct pxa2xx_pcm_dma_params * pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -105,6 +84,102 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) return &dma->params; } +EXPORT_SYMBOL_GPL(pxa_ssp_get_dma_params); + +/* + * 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 pxa_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; + + pxa_ssp_disable(ssp); + + /* reset port settings */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + 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; + + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_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 pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -193,212 +268,88 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) #define pxa_ssp_resume NULL #endif -/** - * ssp_set_clkdiv - set SSP clock divider - * @div: serial clock rate divider - */ -static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { - sscr0 &= ~0x0000ff00; - sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ - } else { - sscr0 &= ~0x000fff00; - sscr0 |= (div - 1) << 8; /* 1..4096 */ - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); -} - -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* - * Set the SSP ports SYSCLK. + * While configuring SSP timing, two formula on SSP clocks should be followed. + * + * 1) (frame delay + frame width) <= (start delay + dummy start + * + data size + dummy stop) + * + * 2) 1 <= frame width < (dummy start + data size + dummy stop) */ -static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) +static int pxa_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 val; + int width = snd_pcm_format_physical_width(params_format(params)); + int channels = params_channels(params); + int dma_32b = 0, stream_out = 0, data_size; + u32 sscr0, sspsp; - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + /* generate correct DMA params */ + kfree(cpu_dai->dma_data); - dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", - cpu_dai->id, clk_id, freq); + if ((width == 32) || (channels == 1)) + dma_32b = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream_out = 1; + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, dma_32b, stream_out); - switch (clk_id) { - case PXA_SSP_CLK_NET_PLL: - sscr0 |= SSCR0_MOD; - break; - case PXA_SSP_CLK_PLL: - /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) - priv->sysclk = 1843200; - else - priv->sysclk = 13000000; - break; - case PXA_SSP_CLK_EXT: - priv->sysclk = freq; - sscr0 |= SSCR0_ECS; - break; - case PXA_SSP_CLK_NET: - priv->sysclk = freq; - sscr0 |= SSCR0_NCS | SSCR0_MOD; + /* clear selected SSP bits */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + + /* data_size should only be 16-bit or 32-bit because of DMA. + * 24-bit data is treated as 32-bit data. + */ + data_size = width * channels; + switch (data_size) { + case 16: + sscr0 |= SSCR0_DataSize(16); break; - case PXA_SSP_CLK_AUDIO: - priv->sysclk = 0; - pxa_ssp_set_scr(ssp, 1); - sscr0 |= SSCR0_ACS; + case 32: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; default: - return -ENODEV; + dev_err(&ssp->pdev->dev, "Invalid data size:%d\n", data_size); + return -EINVAL; } - /* The SSP clock must be disabled when changing SSP clock mode - * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) - clk_disable(ssp->clk); - val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; - pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) - clk_enable(ssp->clk); - - return 0; -} - -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - val &= ~SSACD_SCDX8; -#endif - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; -#if defined(CONFIG_PXA3xx) - case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; -#endif - default: - return -EINVAL; + pxa_ssp_disable(ssp); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); + sspsp &= ~SSPSP_TIMING_MASK; + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + if (channels == 1) { + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_EDMYSTOP(((width - 1) >> 2) & 7); + sspsp |= SSPSP_DMYSTOP((width - 1) & 3); + sspsp |= SSPSP_SFRMDLY(width << 1); + sspsp |= SSPSP_SFRMWDTH(width); + } else { + /* channels == 2 */ + sspsp |= SSPSP_SFRMWDTH(width); + /* Don't set any timing while FSRT is set */ + sspsp |= SSPSP_FSRT; } - pxa_ssp_write_reg(ssp, SSACD, val); break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); + case SND_SOC_DAIFMT_LEFT_J: + if (channels == 1) { + sspsp |= SSPSP_EDMYSTOP((width >> 2) & 7); + sspsp |= SSPSP_DMYSTOP(width & 3); + } + sspsp |= SSPSP_SFRMWDTH(width); break; - default: - return -ENODEV; } - return 0; -} - -/* - * Configure the PLL frequency pxa27x and (afaik - pxa320 only) - */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; - -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif - - switch (freq_out) { - case 5622000: - break; - case 11345000: - ssacd |= (0x1 << 4); - break; - case 12235000: - ssacd |= (0x2 << 4); - break; - case 14857000: - ssacd |= (0x3 << 4); - break; - case 32842000: - ssacd |= (0x4 << 4); - break; - case 48000000: - ssacd |= (0x5 << 4); - break; - case 0: - /* Disable */ - break; - - default: -#ifdef CONFIG_PXA3xx - /* PXA3xx has a clock ditherer which can be used to generate - * a wider range of frequencies - calculate a value for it. - */ - if (cpu_is_pxa3xx()) { - u32 val; - u64 tmp = 19968; - tmp *= 1000000; - do_div(tmp, freq_out); - val = tmp; - - val = (val << 16) | 64; - pxa_ssp_write_reg(ssp, SSACDD, val); - - ssacd |= (0x6 << 4); - - dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %uHz\n", - val, freq_out); - break; - } -#endif - - return -EINVAL; - } + /* update SSP register at the same time */ + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_enable(ssp); - pxa_ssp_write_reg(ssp, SSACD, ssacd); + dump_registers(ssp); return 0; } @@ -458,213 +409,6 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, 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 pxa_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; - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { - dev_err(&ssp->pdev->dev, - "can't change hardware dai format: stream is in use"); - return -EINVAL; - } - - /* reset port settings */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); - sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - sspsp = 0; - - 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; - - case SND_SOC_DAIFMT_DSP_A: - sspsp |= SSPSP_FSRT; - case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; - sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - break; - - default: - return -EINVAL; - } - - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - pxa_ssp_write_reg(ssp, SSCR1, sscr1); - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - - dump_registers(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; - - return 0; -} - -/* - * Set the SSP audio DMA parameters and sample size. - * Can be called multiple times by oss emulation. - */ -static int pxa_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 chn = params_channels(params); - u32 sscr0; - u32 sspsp; - int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; - - /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); - - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) - return 0; - - /* clear selected SSP bits */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; -#endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = pxa_ssp_read_reg(ssp, SSPSP); - - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - - dump_registers(ssp); - - return 0; -} - static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -760,128 +504,22 @@ static void pxa_ssp_remove(struct platform_device *pdev, pxa_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_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 pxa_ssp_dai_ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, -}; - -struct snd_soc_dai pxa_ssp_dai[] = { - { - .name = "pxa2xx-ssp1", - .id = 0, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { .name = "pxa2xx-ssp2", - .id = 1, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp3", - .id = 2, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp4", - .id = 3, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(pxa_ssp_dai); - -static int __init pxa_ssp_init(void) -{ - return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); -} -module_init(pxa_ssp_init); - -static void __exit pxa_ssp_exit(void) +int pxa_ssp_register_dai(struct snd_soc_dai *dai) { - snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); + struct snd_soc_dai_ops *ops = dai->ops; + + ops->startup = pxa_ssp_startup; + ops->shutdown = pxa_ssp_shutdown; + ops->trigger = pxa_ssp_trigger; + ops->hw_params = pxa_ssp_hw_params; + ops->set_fmt = pxa_ssp_set_dai_fmt; + ops->set_tdm_slot = pxa_ssp_set_dai_tdm_slot; + ops->set_tristate = pxa_ssp_set_dai_tristate; + + dai->probe = pxa_ssp_probe; + dai->remove = pxa_ssp_remove; + dai->suspend = pxa_ssp_suspend; + dai->resume = pxa_ssp_resume; + + return snd_soc_register_dai(dai); } -module_exit(pxa_ssp_exit); - -/* Module information */ -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..af4a09b 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -9,39 +9,25 @@ #ifndef _PXA_SSP_H #define _PXA_SSP_H -/* pxa DAI SSP IDs */ -#define PXA_DAI_SSP1 0 -#define PXA_DAI_SSP2 1 -#define PXA_DAI_SSP3 2 -#define PXA_DAI_SSP4 3 - -/* SSP clock sources */ -#define PXA_SSP_CLK_PLL 0 -#define PXA_SSP_CLK_EXT 1 -#define PXA_SSP_CLK_NET 2 -#define PXA_SSP_CLK_AUDIO 3 -#define PXA_SSP_CLK_NET_PLL 4 - -/* SSP audio dividers */ -#define PXA_SSP_AUDIO_DIV_ACDS 0 -#define PXA_SSP_AUDIO_DIV_SCDB 1 -#define PXA_SSP_DIV_SCR 2 - -/* SSP ACDS audio dividers values */ -#define PXA_SSP_CLK_AUDIO_DIV_1 0 -#define PXA_SSP_CLK_AUDIO_DIV_2 1 -#define PXA_SSP_CLK_AUDIO_DIV_4 2 -#define PXA_SSP_CLK_AUDIO_DIV_8 3 -#define PXA_SSP_CLK_AUDIO_DIV_16 4 -#define PXA_SSP_CLK_AUDIO_DIV_32 5 - -/* SSP divider bypass */ -#define PXA_SSP_CLK_SCDB_4 0 -#define PXA_SSP_CLK_SCDB_1 1 -#define PXA_SSP_CLK_SCDB_8 2 - -#define PXA_SSP_PLL_OUT 0 +/* + * SSP audio 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 +}; -extern struct snd_soc_dai pxa_ssp_dai[4]; +extern void pxa_ssp_enable(struct ssp_device *ssp); +extern void pxa_ssp_disable(struct ssp_device *ssp); +extern struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out); +extern int pxa_ssp_register_dai(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c new file mode 100644 index 0000000..eff5334 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -0,0 +1,318 @@ +/* + * pxa2xx-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * Copyright 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ssp.h" +#include "pxa-ssp.h" + +/** + * pxa_ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * pxa_ssp_get_clkdiv - get SSP clock divider + */ +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa2xx_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; + int val; + + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA2XX_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA2XX_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA2XX_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_AUDIO: + priv->sysclk = 0; + pxa_ssp_set_scr(ssp, 1); + sscr0 |= SSCR0_ACS; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (!cpu_is_pxa3xx()) + clk_disable(ssp->clk); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + switch (div_id) { + case PXA2XX_SSP_AUDIO_DIV_ACDS: + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_AUDIO_DIV_SCDB: + val = pxa_ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; +#endif + switch (div) { + case PXA2XX_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA2XX_SSP_CLK_SCDB_4: + break; +#if defined(CONFIG_PXA3xx) + case PXA2XX_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; +#endif + default: + return -EINVAL; + } + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_DIV_SCR: + pxa_ssp_set_scr(ssp, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa2xx_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; + +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + pxa_ssp_write_reg(ssp, SSACDD, 0); +#endif + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: +#ifdef CONFIG_PXA3xx + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64; + pxa_ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq_out); + break; + } +#endif + + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +static struct snd_soc_dai_ops pxa2xx_ssp_dai_ops = { + .set_sysclk = pxa2xx_ssp_set_dai_sysclk, + .set_clkdiv = pxa2xx_ssp_set_dai_clkdiv, + .set_pll = pxa2xx_ssp_set_dai_pll, +}; + +#define PXA2XX_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA2XX_SSP_DAI(_id) \ +{ \ + .name = "pxa2xx-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .ops = &pxa2xx_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa2xx_ssp_dai[] = { + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP1), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP2), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP3), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP4), +}; +EXPORT_SYMBOL_GPL(pxa2xx_ssp_dai); + +static int __init pxa2xx_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa2xx_ssp_init); + +static void __exit pxa2xx_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + snd_soc_unregister_dai(dai); + } +} +module_exit(pxa2xx_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA2xx SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-ssp.h b/sound/soc/pxa/pxa2xx-ssp.h new file mode 100644 index 0000000..c24e0b5 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.h @@ -0,0 +1,48 @@ + +/* + * ASoC PXA 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 __PXA2XX_SOC_SSP_H +#define __PXA2XX_SOC_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA2XX_DAI_SSP1 0 +#define PXA2XX_DAI_SSP2 1 +#define PXA2XX_DAI_SSP3 2 +#define PXA2XX_DAI_SSP4 3 + +/* SSP clock sources */ +#define PXA2XX_SSP_CLK_PLL 0 +#define PXA2XX_SSP_CLK_EXT 1 +#define PXA2XX_SSP_CLK_NET 2 +#define PXA2XX_SSP_CLK_AUDIO 3 +#define PXA2XX_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA2XX_SSP_AUDIO_DIV_ACDS 0 +#define PXA2XX_SSP_AUDIO_DIV_SCDB 1 +#define PXA2XX_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA2XX_SSP_CLK_AUDIO_DIV_1 0 +#define PXA2XX_SSP_CLK_AUDIO_DIV_2 1 +#define PXA2XX_SSP_CLK_AUDIO_DIV_4 2 +#define PXA2XX_SSP_CLK_AUDIO_DIV_8 3 +#define PXA2XX_SSP_CLK_AUDIO_DIV_16 4 +#define PXA2XX_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA2XX_SSP_CLK_SCDB_4 0 +#define PXA2XX_SSP_CLK_SCDB_1 1 +#define PXA2XX_SSP_CLK_SCDB_8 2 + +#define PXA2XX_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa2xx_ssp_dai[]; + +#endif /* __PXA2XX_SOC_SSP_H */ diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 7e3f416..e66dc12 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -29,7 +29,7 @@ #include "../codecs/cs4270.h" #include "../codecs/ak4104.h" #include "pxa2xx-pcm.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define GPIO_SPDIF_RESET (38) #define GPIO_MCLK_RESET (111) @@ -138,11 +138,11 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -170,7 +170,7 @@ static int raumfeld_line_resume(struct platform_device *pdev) static struct snd_soc_dai_link raumfeld_line_dai = { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &cs4270_dai, .ops = &raumfeld_cs4270_ops, }; @@ -232,11 +232,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -250,7 +250,7 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { static struct snd_soc_dai_link raumfeld_spdif_dai = { .name = "ak4104", .stream_name = "Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP2], .codec_dai = &ak4104_dai, .ops = &raumfeld_ak4104_ops, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index dd678ae..c65e0db 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -25,7 +25,7 @@ #include "../codecs/wm9713.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" /* * There is a physical switch SW15 on the board which changes the MCLK @@ -125,7 +125,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP3], .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], .ops = &zylonite_voice_ops, }, -- 1.5.6.5

1 0

[PATCH] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 31 Mar '10

31 Mar '10

Since basic SSP features are shared between PXA2xx and PXA168, the difference is focused on clock generating. Now split ssp code into two parts. One is for general ssp feature. The other is for pxa2xx parts. Update the SSP timing configuration in hw_params(). The SSP timing is verified on I2S and Left J interface of PXA168 with 16-bit sample stream. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 8 +- sound/soc/pxa/Makefile | 4 +- sound/soc/pxa/magician.c | 38 ++-- sound/soc/pxa/pxa-ssp.c | 728 +++++++++++--------------------------------- sound/soc/pxa/pxa-ssp.h | 52 ++-- sound/soc/pxa/pxa2xx-ssp.c | 318 +++++++++++++++++++ sound/soc/pxa/pxa2xx-ssp.h | 48 +++ sound/soc/pxa/raumfeld.c | 14 +- sound/soc/pxa/zylonite.c | 6 +- 9 files changed, 603 insertions(+), 613 deletions(-) create mode 100644 sound/soc/pxa/pxa2xx-ssp.c create mode 100644 sound/soc/pxa/pxa2xx-ssp.h diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 78e6121..7be1d5f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -21,7 +21,7 @@ config SND_PXA2XX_SOC_AC97 config SND_PXA2XX_SOC_I2S tristate -config SND_PXA_SOC_SSP +config SND_PXA2XX_SOC_SSP tristate select PXA_SSP @@ -113,7 +113,7 @@ config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE select SND_PXA2XX_SOC_AC97 - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_WM9713 help Say Y if you want to add support for SoC audio on the @@ -122,7 +122,7 @@ config SND_SOC_ZYLONITE config SND_SOC_RAUMFELD tristate "SoC Audio support Raumfeld audio adapter" depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_CS4270 select SND_SOC_AK4104 help @@ -132,7 +132,7 @@ config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN select SND_PXA2XX_SOC_I2S - select SND_PXA_SOC_SSP + select SND_PXA2XX_SOC_SSP select SND_SOC_UDA1380 help Say Y if you want to add support for SoC audio on the diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index f3e08fd..33c1579 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -2,12 +2,12 @@ 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-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-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_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 4c8d99a..da6ff83 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -34,7 +34,7 @@ #include "../codecs/uda1380.h" #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define MAGICIAN_MIC 0 #define MAGICIAN_MIC_EXT 1 @@ -89,7 +89,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int acps, acds, width, rate; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int div4 = PXA2XX_SSP_CLK_SCDB_4; int ret = 0; rate = params_rate(params); @@ -106,11 +106,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_16; break; default: /* 32 */ /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; } break; case 11025: @@ -118,11 +118,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_4; break; default: /* 32 */ /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; } break; case 22050: @@ -130,11 +130,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 44100: @@ -142,11 +142,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 48000: @@ -154,11 +154,11 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; } break; case 96000: @@ -167,12 +167,12 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, switch (width) { case 16: /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_1; break; default: /* 32 */ /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; + acds = PXA2XX_SSP_CLK_AUDIO_DIV_2; + div4 = PXA2XX_SSP_CLK_SCDB_1; break; } break; @@ -195,20 +195,20 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set audio clock as clock source */ - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, SND_SOC_CLOCK_OUT); if (ret < 0) return ret; /* set the SSP audio system clock ACDS divider */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); + PXA2XX_SSP_AUDIO_DIV_ACDS, acds); if (ret < 0) return ret; /* set the SSP audio system clock SCDB divider4 */ ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); + PXA2XX_SSP_AUDIO_DIV_SCDB, div4); if (ret < 0) return ret; @@ -427,7 +427,7 @@ static struct snd_soc_dai_link magician_dai[] = { { .name = "uda1380", .stream_name = "UDA1380 Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK], .init = magician_uda1380_init, .ops = &magician_playback_ops, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index ed54bef..bd5e800 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -5,6 +5,9 @@ * Author: Liam Girdwood * Mark Brown <broonie(a)opensource.wolfsonmicro.com> * + * Copyright 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 as published by the * Free Software Foundation; either version 2 of the License, or (at your @@ -14,44 +17,20 @@ * o Test network mode for > 16bit sample size */ -#include <linux/init.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/soc.h> #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include <plat/ssp.h> #include "pxa2xx-pcm.h" #include "pxa-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 -}; - static void dump_registers(struct ssp_device *ssp) { dev_dbg(&ssp->pdev->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", @@ -63,7 +42,7 @@ static void dump_registers(struct ssp_device *ssp) pxa_ssp_read_reg(ssp, SSACD)); } -static void pxa_ssp_enable(struct ssp_device *ssp) +void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +50,7 @@ static void pxa_ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void pxa_ssp_disable(struct ssp_device *ssp) +void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -84,7 +63,7 @@ struct pxa2xx_pcm_dma_data { char name[20]; }; -static struct pxa2xx_pcm_dma_params * +struct pxa2xx_pcm_dma_params * pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -105,6 +84,102 @@ pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) return &dma->params; } +EXPORT_SYMBOL_GPL(pxa_ssp_get_dma_params); + +/* + * 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 pxa_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; + + pxa_ssp_disable(ssp); + + /* reset port settings */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + sspsp = 0; + + 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; + + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP; + sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; + /* See hw_params() */ + break; + + case SND_SOC_DAIFMT_DSP_A: + sspsp |= SSPSP_FSRT; + case SND_SOC_DAIFMT_DSP_B: + sscr0 |= SSCR0_MOD | SSCR0_PSP; + sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + break; + + default: + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_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 pxa_ssp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -193,212 +268,88 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) #define pxa_ssp_resume NULL #endif -/** - * ssp_set_clkdiv - set SSP clock divider - * @div: serial clock rate divider - */ -static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { - sscr0 &= ~0x0000ff00; - sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ - } else { - sscr0 &= ~0x000fff00; - sscr0 |= (div - 1) << 8; /* 1..4096 */ - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); -} - -/** - * pxa_ssp_get_clkdiv - get SSP clock divider - */ -static u32 pxa_ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* - * Set the SSP ports SYSCLK. + * While configuring SSP timing, two formula on SSP clocks should be followed. + * + * 1) (frame delay + frame width) <= (start delay + dummy start + * + data size + dummy stop) + * + * 2) 1 <= frame width < (dummy start + data size + dummy stop) */ -static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) +static int pxa_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 val; + int width = snd_pcm_format_physical_width(params_format(params)); + int channels = params_channels(params); + int dma_32b = 0, stream_out = 0, data_size; + u32 sscr0, sspsp; - u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + /* generate correct DMA params */ + kfree(cpu_dai->dma_data); - dev_dbg(&ssp->pdev->dev, - "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", - cpu_dai->id, clk_id, freq); + if ((width == 32) || (channels == 1)) + dma_32b = 1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream_out = 1; + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, dma_32b, stream_out); - switch (clk_id) { - case PXA_SSP_CLK_NET_PLL: - sscr0 |= SSCR0_MOD; - break; - case PXA_SSP_CLK_PLL: - /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) - priv->sysclk = 1843200; - else - priv->sysclk = 13000000; - break; - case PXA_SSP_CLK_EXT: - priv->sysclk = freq; - sscr0 |= SSCR0_ECS; - break; - case PXA_SSP_CLK_NET: - priv->sysclk = freq; - sscr0 |= SSCR0_NCS | SSCR0_MOD; + /* clear selected SSP bits */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + + /* data_size should only be 16-bit or 32-bit because of DMA. + * 24-bit data is treated as 32-bit data. + */ + data_size = width * channels; + switch (data_size) { + case 16: + sscr0 |= SSCR0_DataSize(16); break; - case PXA_SSP_CLK_AUDIO: - priv->sysclk = 0; - pxa_ssp_set_scr(ssp, 1); - sscr0 |= SSCR0_ACS; + case 32: + sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; default: - return -ENODEV; + dev_err(&ssp->pdev->dev, "Invalid data size:%d\n", data_size); + return -EINVAL; } - /* The SSP clock must be disabled when changing SSP clock mode - * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) - clk_disable(ssp->clk); - val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; - pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) - clk_enable(ssp->clk); - - return 0; -} - -/* - * Set the SSP clock dividers. - */ -static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - int val; - - switch (div_id) { - case PXA_SSP_AUDIO_DIV_ACDS: - val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - pxa_ssp_write_reg(ssp, SSACD, val); - break; - case PXA_SSP_AUDIO_DIV_SCDB: - val = pxa_ssp_read_reg(ssp, SSACD); - val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - val &= ~SSACD_SCDX8; -#endif - switch (div) { - case PXA_SSP_CLK_SCDB_1: - val |= SSACD_SCDB; - break; - case PXA_SSP_CLK_SCDB_4: - break; -#if defined(CONFIG_PXA3xx) - case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) - val |= SSACD_SCDX8; - else - return -EINVAL; - break; -#endif - default: - return -EINVAL; + pxa_ssp_disable(ssp); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); + sspsp &= ~SSPSP_TIMING_MASK; + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + if (channels == 1) { + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_EDMYSTOP(((width - 1) >> 2) & 7); + sspsp |= SSPSP_DMYSTOP((width - 1) & 3); + sspsp |= SSPSP_SFRMDLY(width << 1); + sspsp |= SSPSP_SFRMWDTH(width); + } else { + /* channels == 2 */ + sspsp |= SSPSP_SFRMWDTH(width); + /* Don't set any timing while FSRT is set */ + sspsp |= SSPSP_FSRT; } - pxa_ssp_write_reg(ssp, SSACD, val); break; - case PXA_SSP_DIV_SCR: - pxa_ssp_set_scr(ssp, div); + case SND_SOC_DAIFMT_LEFT_J: + if (channels == 1) { + sspsp |= SSPSP_EDMYSTOP((width >> 2) & 7); + sspsp |= SSPSP_DMYSTOP(width & 3); + } + sspsp |= SSPSP_SFRMWDTH(width); break; - default: - return -ENODEV; } - return 0; -} - -/* - * Configure the PLL frequency pxa27x and (afaik - pxa320 only) - */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->ssp; - u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; - -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) - pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif - - switch (freq_out) { - case 5622000: - break; - case 11345000: - ssacd |= (0x1 << 4); - break; - case 12235000: - ssacd |= (0x2 << 4); - break; - case 14857000: - ssacd |= (0x3 << 4); - break; - case 32842000: - ssacd |= (0x4 << 4); - break; - case 48000000: - ssacd |= (0x5 << 4); - break; - case 0: - /* Disable */ - break; - - default: -#ifdef CONFIG_PXA3xx - /* PXA3xx has a clock ditherer which can be used to generate - * a wider range of frequencies - calculate a value for it. - */ - if (cpu_is_pxa3xx()) { - u32 val; - u64 tmp = 19968; - tmp *= 1000000; - do_div(tmp, freq_out); - val = tmp; - - val = (val << 16) | 64; - pxa_ssp_write_reg(ssp, SSACDD, val); - - ssacd |= (0x6 << 4); - - dev_dbg(&ssp->pdev->dev, - "Using SSACDD %x to supply %uHz\n", - val, freq_out); - break; - } -#endif - - return -EINVAL; - } + /* update SSP register at the same time */ + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_enable(ssp); - pxa_ssp_write_reg(ssp, SSACD, ssacd); + dump_registers(ssp); return 0; } @@ -458,213 +409,6 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, 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 pxa_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; - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { - dev_err(&ssp->pdev->dev, - "can't change hardware dai format: stream is in use"); - return -EINVAL; - } - - /* reset port settings */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); - sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - sspsp = 0; - - 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; - - case SND_SOC_DAIFMT_DSP_A: - sspsp |= SSPSP_FSRT; - case SND_SOC_DAIFMT_DSP_B: - sscr0 |= SSCR0_MOD | SSCR0_PSP; - sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; - break; - - default: - return -EINVAL; - } - - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - pxa_ssp_write_reg(ssp, SSCR1, sscr1); - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - - dump_registers(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; - - return 0; -} - -/* - * Set the SSP audio DMA parameters and sample size. - * Can be called multiple times by oss emulation. - */ -static int pxa_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 chn = params_channels(params); - u32 sscr0; - u32 sspsp; - int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; - - /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); - - /* Network mode with one active slot (ttsa == 1) can be used - * to force 16-bit frame width on the wire (for S16_LE), even - * with two channels. Use 16-bit DMA transfers for this case. - */ - cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, - ((chn == 2) && (ttsa != 1)) || (width == 32), - substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) - return 0; - - /* clear selected SSP bits */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - /* bit size */ - sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) - sscr0 |= SSCR0_FPCKE; -#endif - sscr0 |= SSCR0_DataSize(16); - break; - case SNDRV_PCM_FORMAT_S24_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8)); - break; - case SNDRV_PCM_FORMAT_S32_LE: - sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); - break; - } - pxa_ssp_write_reg(ssp, SSCR0, sscr0); - - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = pxa_ssp_read_reg(ssp, SSPSP); - - if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - pxa_ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - - /* When we use a network mode, we always require TDM slots - * - complain loudly and fail if they've not been set up yet. - */ - if ((sscr0 & SSCR0_MOD) && !ttsa) { - dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n"); - return -EINVAL; - } - - dump_registers(ssp); - - return 0; -} - static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -760,128 +504,22 @@ static void pxa_ssp_remove(struct platform_device *pdev, pxa_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_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 pxa_ssp_dai_ops = { - .startup = pxa_ssp_startup, - .shutdown = pxa_ssp_shutdown, - .trigger = pxa_ssp_trigger, - .hw_params = pxa_ssp_hw_params, - .set_sysclk = pxa_ssp_set_dai_sysclk, - .set_clkdiv = pxa_ssp_set_dai_clkdiv, - .set_pll = pxa_ssp_set_dai_pll, - .set_fmt = pxa_ssp_set_dai_fmt, - .set_tdm_slot = pxa_ssp_set_dai_tdm_slot, - .set_tristate = pxa_ssp_set_dai_tristate, -}; - -struct snd_soc_dai pxa_ssp_dai[] = { - { - .name = "pxa2xx-ssp1", - .id = 0, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { .name = "pxa2xx-ssp2", - .id = 1, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp3", - .id = 2, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, - { - .name = "pxa2xx-ssp4", - .id = 3, - .probe = pxa_ssp_probe, - .remove = pxa_ssp_remove, - .suspend = pxa_ssp_suspend, - .resume = pxa_ssp_resume, - .playback = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 8, - .rates = PXA_SSP_RATES, - .formats = PXA_SSP_FORMATS, - }, - .ops = &pxa_ssp_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(pxa_ssp_dai); - -static int __init pxa_ssp_init(void) -{ - return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); -} -module_init(pxa_ssp_init); - -static void __exit pxa_ssp_exit(void) +int pxa_ssp_register_dai(struct snd_soc_dai *dai) { - snd_soc_unregister_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); + struct snd_soc_dai_ops *ops = dai->ops; + + ops->startup = pxa_ssp_startup; + ops->shutdown = pxa_ssp_shutdown; + ops->trigger = pxa_ssp_trigger; + ops->hw_params = pxa_ssp_hw_params; + ops->set_fmt = pxa_ssp_set_dai_fmt; + ops->set_tdm_slot = pxa_ssp_set_dai_tdm_slot; + ops->set_tristate = pxa_ssp_set_dai_tristate; + + dai->probe = pxa_ssp_probe; + dai->remove = pxa_ssp_remove; + dai->suspend = pxa_ssp_suspend; + dai->resume = pxa_ssp_resume; + + return snd_soc_register_dai(dai); } -module_exit(pxa_ssp_exit); - -/* Module information */ -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..af4a09b 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -9,39 +9,25 @@ #ifndef _PXA_SSP_H #define _PXA_SSP_H -/* pxa DAI SSP IDs */ -#define PXA_DAI_SSP1 0 -#define PXA_DAI_SSP2 1 -#define PXA_DAI_SSP3 2 -#define PXA_DAI_SSP4 3 - -/* SSP clock sources */ -#define PXA_SSP_CLK_PLL 0 -#define PXA_SSP_CLK_EXT 1 -#define PXA_SSP_CLK_NET 2 -#define PXA_SSP_CLK_AUDIO 3 -#define PXA_SSP_CLK_NET_PLL 4 - -/* SSP audio dividers */ -#define PXA_SSP_AUDIO_DIV_ACDS 0 -#define PXA_SSP_AUDIO_DIV_SCDB 1 -#define PXA_SSP_DIV_SCR 2 - -/* SSP ACDS audio dividers values */ -#define PXA_SSP_CLK_AUDIO_DIV_1 0 -#define PXA_SSP_CLK_AUDIO_DIV_2 1 -#define PXA_SSP_CLK_AUDIO_DIV_4 2 -#define PXA_SSP_CLK_AUDIO_DIV_8 3 -#define PXA_SSP_CLK_AUDIO_DIV_16 4 -#define PXA_SSP_CLK_AUDIO_DIV_32 5 - -/* SSP divider bypass */ -#define PXA_SSP_CLK_SCDB_4 0 -#define PXA_SSP_CLK_SCDB_1 1 -#define PXA_SSP_CLK_SCDB_8 2 - -#define PXA_SSP_PLL_OUT 0 +/* + * SSP audio 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 +}; -extern struct snd_soc_dai pxa_ssp_dai[4]; +extern void pxa_ssp_enable(struct ssp_device *ssp); +extern void pxa_ssp_disable(struct ssp_device *ssp); +extern struct pxa2xx_pcm_dma_params * +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out); +extern int pxa_ssp_register_dai(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/pxa/pxa2xx-ssp.c b/sound/soc/pxa/pxa2xx-ssp.c new file mode 100644 index 0000000..eff5334 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.c @@ -0,0 +1,318 @@ +/* + * pxa2xx-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 2005,2008 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * Copyright 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode for > 16bit sample size + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.h> +#include <plat/ssp.h> + +#include "pxa2xx-pcm.h" +#include "pxa2xx-ssp.h" +#include "pxa-ssp.h" + +/** + * pxa_ssp_set_clkdiv - set SSP clock divider + * @div: serial clock rate divider + */ +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + sscr0 &= ~0x0000ff00; + sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ + } else { + sscr0 &= ~0x000fff00; + sscr0 |= (div - 1) << 8; /* 1..4096 */ + } + pxa_ssp_write_reg(ssp, SSCR0, sscr0); +} + +/** + * pxa_ssp_get_clkdiv - get SSP clock divider + */ +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) +{ + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + u32 div; + + if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + div = ((sscr0 >> 8) & 0xff) * 2 + 2; + else + div = ((sscr0 >> 8) & 0xfff) + 1; + return div; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int pxa2xx_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; + int val; + + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & + ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + + dev_dbg(&ssp->pdev->dev, + "pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %u\n", + cpu_dai->id, clk_id, freq); + + switch (clk_id) { + case PXA2XX_SSP_CLK_NET_PLL: + sscr0 |= SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_PLL: + /* Internal PLL is fixed */ + if (cpu_is_pxa25x()) + priv->sysclk = 1843200; + else + priv->sysclk = 13000000; + break; + case PXA2XX_SSP_CLK_EXT: + priv->sysclk = freq; + sscr0 |= SSCR0_ECS; + break; + case PXA2XX_SSP_CLK_NET: + priv->sysclk = freq; + sscr0 |= SSCR0_NCS | SSCR0_MOD; + break; + case PXA2XX_SSP_CLK_AUDIO: + priv->sysclk = 0; + pxa_ssp_set_scr(ssp, 1); + sscr0 |= SSCR0_ACS; + break; + default: + return -ENODEV; + } + + /* The SSP clock must be disabled when changing SSP clock mode + * on PXA2xx. On PXA3xx it must be enabled when doing so. */ + if (!cpu_is_pxa3xx()) + clk_disable(ssp->clk); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); + if (!cpu_is_pxa3xx()) + clk_enable(ssp->clk); + + return 0; +} + +/* + * Set the SSP clock dividers. + */ +static int pxa2xx_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + int val; + + switch (div_id) { + case PXA2XX_SSP_AUDIO_DIV_ACDS: + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_AUDIO_DIV_SCDB: + val = pxa_ssp_read_reg(ssp, SSACD); + val &= ~SSACD_SCDB; +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + val &= ~SSACD_SCDX8; +#endif + switch (div) { + case PXA2XX_SSP_CLK_SCDB_1: + val |= SSACD_SCDB; + break; + case PXA2XX_SSP_CLK_SCDB_4: + break; +#if defined(CONFIG_PXA3xx) + case PXA2XX_SSP_CLK_SCDB_8: + if (cpu_is_pxa3xx()) + val |= SSACD_SCDX8; + else + return -EINVAL; + break; +#endif + default: + return -EINVAL; + } + pxa_ssp_write_reg(ssp, SSACD, val); + break; + case PXA2XX_SSP_DIV_SCR: + pxa_ssp_set_scr(ssp, div); + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Configure the PLL frequency pxa27x and (afaik - pxa320 only) + */ +static int pxa2xx_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; + +#if defined(CONFIG_PXA3xx) + if (cpu_is_pxa3xx()) + pxa_ssp_write_reg(ssp, SSACDD, 0); +#endif + + switch (freq_out) { + case 5622000: + break; + case 11345000: + ssacd |= (0x1 << 4); + break; + case 12235000: + ssacd |= (0x2 << 4); + break; + case 14857000: + ssacd |= (0x3 << 4); + break; + case 32842000: + ssacd |= (0x4 << 4); + break; + case 48000000: + ssacd |= (0x5 << 4); + break; + case 0: + /* Disable */ + break; + + default: +#ifdef CONFIG_PXA3xx + /* PXA3xx has a clock ditherer which can be used to generate + * a wider range of frequencies - calculate a value for it. + */ + if (cpu_is_pxa3xx()) { + u32 val; + u64 tmp = 19968; + tmp *= 1000000; + do_div(tmp, freq_out); + val = tmp; + + val = (val << 16) | 64; + pxa_ssp_write_reg(ssp, SSACDD, val); + + ssacd |= (0x6 << 4); + + dev_dbg(&ssp->pdev->dev, + "Using SSACDD %x to supply %uHz\n", + val, freq_out); + break; + } +#endif + + return -EINVAL; + } + + pxa_ssp_write_reg(ssp, SSACD, ssacd); + + return 0; +} + +static struct snd_soc_dai_ops pxa2xx_ssp_dai_ops = { + .set_sysclk = pxa2xx_ssp_set_dai_sysclk, + .set_clkdiv = pxa2xx_ssp_set_dai_clkdiv, + .set_pll = pxa2xx_ssp_set_dai_pll, +}; + +#define PXA2XX_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA2XX_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA2XX_SSP_DAI(_id) \ +{ \ + .name = "pxa2xx-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 8, \ + .rates = PXA2XX_SSP_RATES, \ + .formats = PXA2XX_SSP_FORMATS, \ + }, \ + .ops = &pxa2xx_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa2xx_ssp_dai[] = { + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP1), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP2), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP3), + PXA2XX_SSP_DAI(PXA2XX_DAI_SSP4), +}; +EXPORT_SYMBOL_GPL(pxa2xx_ssp_dai); + +static int __init pxa2xx_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa2xx_ssp_init); + +static void __exit pxa2xx_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa2xx_ssp_dai); i++) { + dai = &pxa2xx_ssp_dai[i]; + snd_soc_unregister_dai(dai); + } +} +module_exit(pxa2xx_ssp_exit); + +/* Module information */ +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_DESCRIPTION("PXA2xx SSP/PCM SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/pxa2xx-ssp.h b/sound/soc/pxa/pxa2xx-ssp.h new file mode 100644 index 0000000..c24e0b5 --- /dev/null +++ b/sound/soc/pxa/pxa2xx-ssp.h @@ -0,0 +1,48 @@ + +/* + * ASoC PXA 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 __PXA2XX_SOC_SSP_H +#define __PXA2XX_SOC_SSP_H + +/* pxa DAI SSP IDs */ +#define PXA2XX_DAI_SSP1 0 +#define PXA2XX_DAI_SSP2 1 +#define PXA2XX_DAI_SSP3 2 +#define PXA2XX_DAI_SSP4 3 + +/* SSP clock sources */ +#define PXA2XX_SSP_CLK_PLL 0 +#define PXA2XX_SSP_CLK_EXT 1 +#define PXA2XX_SSP_CLK_NET 2 +#define PXA2XX_SSP_CLK_AUDIO 3 +#define PXA2XX_SSP_CLK_NET_PLL 4 + +/* SSP audio dividers */ +#define PXA2XX_SSP_AUDIO_DIV_ACDS 0 +#define PXA2XX_SSP_AUDIO_DIV_SCDB 1 +#define PXA2XX_SSP_DIV_SCR 2 + +/* SSP ACDS audio dividers values */ +#define PXA2XX_SSP_CLK_AUDIO_DIV_1 0 +#define PXA2XX_SSP_CLK_AUDIO_DIV_2 1 +#define PXA2XX_SSP_CLK_AUDIO_DIV_4 2 +#define PXA2XX_SSP_CLK_AUDIO_DIV_8 3 +#define PXA2XX_SSP_CLK_AUDIO_DIV_16 4 +#define PXA2XX_SSP_CLK_AUDIO_DIV_32 5 + +/* SSP divider bypass */ +#define PXA2XX_SSP_CLK_SCDB_4 0 +#define PXA2XX_SSP_CLK_SCDB_1 1 +#define PXA2XX_SSP_CLK_SCDB_8 2 + +#define PXA2XX_SSP_PLL_OUT 0 + +extern struct snd_soc_dai pxa2xx_ssp_dai[]; + +#endif /* __PXA2XX_SOC_SSP_H */ diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index 7e3f416..e66dc12 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -29,7 +29,7 @@ #include "../codecs/cs4270.h" #include "../codecs/ak4104.h" #include "pxa2xx-pcm.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" #define GPIO_SPDIF_RESET (38) #define GPIO_MCLK_RESET (111) @@ -138,11 +138,11 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -170,7 +170,7 @@ static int raumfeld_line_resume(struct platform_device *pdev) static struct snd_soc_dai_link raumfeld_line_dai = { .name = "CS4270", .stream_name = "CS4270", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP1], .codec_dai = &cs4270_dai, .ops = &raumfeld_cs4270_ops, }; @@ -232,11 +232,11 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA2XX_SSP_DIV_SCR, 4); if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -250,7 +250,7 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { static struct snd_soc_dai_link raumfeld_spdif_dai = { .name = "ak4104", .stream_name = "Playback", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP2], .codec_dai = &ak4104_dai, .ops = &raumfeld_ak4104_ops, }; diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index dd678ae..c65e0db 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -25,7 +25,7 @@ #include "../codecs/wm9713.h" #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" -#include "pxa-ssp.h" +#include "pxa2xx-ssp.h" /* * There is a physical switch SW15 on the board which changes the MCLK @@ -125,7 +125,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; @@ -176,7 +176,7 @@ static struct snd_soc_dai_link zylonite_dai[] = { { .name = "WM9713 Voice", .stream_name = "WM9713 Voice", - .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP3], + .cpu_dai = &pxa2xx_ssp_dai[PXA2XX_DAI_SSP3], .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], .ops = &zylonite_voice_ops, }, -- 1.5.6.5

1 0

[PATCH] [ARM] pxa: add namespace on ssp
by Haojian Zhuang 31 Mar '10

31 Mar '10

In order to prevent code ambiguous, add namespace on functions in ssp driver. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/plat-pxa/include/plat/ssp.h | 12 ++-- arch/arm/plat-pxa/ssp.c | 22 +++--- drivers/spi/pxa2xx_spi.c | 8 +- sound/soc/pxa/pxa-ssp.c | 132 +++++++++++++++++----------------- 4 files changed, 87 insertions(+), 87 deletions(-) diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h index d16d79a..fe43150 100644 --- a/arch/arm/plat-pxa/include/plat/ssp.h +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -159,28 +159,28 @@ struct ssp_device { }; /** - * ssp_write_reg - Write to a SSP register + * pxa_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) +static inline void pxa_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 + * pxa_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) +static inline u32 pxa_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 *); +struct ssp_device *pxa_ssp_request(int port, const char *label); +void pxa_ssp_free(struct ssp_device *); #endif /* __ASM_ARCH_SSP_H */ diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index cfebcd8..511805e 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -37,7 +37,7 @@ static DEFINE_MUTEX(ssp_lock); static LIST_HEAD(ssp_list); -struct ssp_device *ssp_request(int port, const char *label) +struct ssp_device *pxa_ssp_request(int port, const char *label) { struct ssp_device *ssp = NULL; @@ -58,9 +58,9 @@ struct ssp_device *ssp_request(int port, const char *label) return ssp; } -EXPORT_SYMBOL(ssp_request); +EXPORT_SYMBOL(pxa_ssp_request); -void ssp_free(struct ssp_device *ssp) +void pxa_ssp_free(struct ssp_device *ssp) { mutex_lock(&ssp_lock); if (ssp->use_count) { @@ -70,9 +70,9 @@ void ssp_free(struct ssp_device *ssp) dev_err(&ssp->pdev->dev, "device already free\n"); mutex_unlock(&ssp_lock); } -EXPORT_SYMBOL(ssp_free); +EXPORT_SYMBOL(pxa_ssp_free); -static int __devinit ssp_probe(struct platform_device *pdev) +static int __devinit pxa_ssp_probe(struct platform_device *pdev) { const struct platform_device_id *id = platform_get_device_id(pdev); struct resource *res; @@ -164,7 +164,7 @@ err_free: return ret; } -static int __devexit ssp_remove(struct platform_device *pdev) +static int __devexit pxa_ssp_remove(struct platform_device *pdev) { struct resource *res; struct ssp_device *ssp; @@ -196,9 +196,9 @@ static const struct platform_device_id ssp_id_table[] = { { }, }; -static struct platform_driver ssp_driver = { - .probe = ssp_probe, - .remove = __devexit_p(ssp_remove), +static struct platform_driver pxa_ssp_driver = { + .probe = pxa_ssp_probe, + .remove = __devexit_p(pxa_ssp_remove), .driver = { .owner = THIS_MODULE, .name = "pxa2xx-ssp", @@ -208,12 +208,12 @@ static struct platform_driver ssp_driver = { static int __init pxa_ssp_init(void) { - return platform_driver_register(&ssp_driver); + return platform_driver_register(&pxa_ssp_driver); } static void __exit pxa_ssp_exit(void) { - platform_driver_unregister(&ssp_driver); + platform_driver_unregister(&pxa_ssp_driver); } arch_initcall(pxa_ssp_init); diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c index 4a64da7..2fd7d5a 100644 --- a/drivers/spi/pxa2xx_spi.c +++ b/drivers/spi/pxa2xx_spi.c @@ -1464,7 +1464,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) platform_info = dev->platform_data; - ssp = ssp_request(pdev->id, pdev->name); + ssp = pxa_ssp_request(pdev->id, pdev->name); if (ssp == NULL) { dev_err(&pdev->dev, "failed to request SSP%d\n", pdev->id); return -ENODEV; @@ -1474,7 +1474,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) master = spi_alloc_master(dev, sizeof(struct driver_data) + 16); if (!master) { dev_err(&pdev->dev, "cannot alloc spi_master\n"); - ssp_free(ssp); + pxa_ssp_free(ssp); return -ENOMEM; } drv_data = spi_master_get_devdata(master); @@ -1603,7 +1603,7 @@ out_error_irq_alloc: out_error_master_alloc: spi_master_put(master); - ssp_free(ssp); + pxa_ssp_free(ssp); return status; } @@ -1647,7 +1647,7 @@ static int pxa2xx_spi_remove(struct platform_device *pdev) free_irq(ssp->irq, drv_data); /* Release SSP */ - ssp_free(ssp); + pxa_ssp_free(ssp); /* Disconnect from the SPI framework */ spi_unregister_master(drv_data->master); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4ca9245..ed54bef 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -55,15 +55,15 @@ struct ssp_priv { 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)); + pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1), + pxa_ssp_read_reg(ssp, SSTO)); dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", - ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), - ssp_read_reg(ssp, SSACD)); + pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR), + pxa_ssp_read_reg(ssp, SSACD)); } -static void ssp_enable(struct ssp_device *ssp) +static void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +71,7 @@ static void ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void ssp_disable(struct ssp_device *ssp) +static void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -85,7 +85,7 @@ struct pxa2xx_pcm_dma_data { }; static struct pxa2xx_pcm_dma_params * -ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -117,7 +117,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, if (!cpu_dai->active) { clk_enable(ssp->clk); - ssp_disable(ssp); + pxa_ssp_disable(ssp); } if (cpu_dai->dma_data) { @@ -136,7 +136,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) { - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); } @@ -161,7 +161,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->to = __raw_readl(ssp->mmio_base + SSTO); priv->psp = __raw_readl(ssp->mmio_base + SSPSP); - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); return 0; } @@ -181,7 +181,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) __raw_writel(priv->psp, ssp->mmio_base + SSPSP); if (cpu_dai->active) - ssp_enable(ssp); + pxa_ssp_enable(ssp); else clk_disable(ssp->clk); @@ -197,9 +197,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_device *ssp, u32 div) +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; @@ -208,15 +208,15 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) sscr0 &= ~0x000fff00; sscr0 |= (div - 1) << 8; /* 1..4096 */ } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); } /** - * ssp_get_clkdiv - get SSP clock divider + * pxa_ssp_get_clkdiv - get SSP clock divider */ -static u32 ssp_get_scr(struct ssp_device *ssp) +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) @@ -236,7 +236,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; int val; - u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, @@ -264,7 +264,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(ssp, 1); + pxa_ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -275,8 +275,8 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) clk_disable(ssp->clk); - val = ssp_read_reg(ssp, SSCR0) | sscr0; - ssp_write_reg(ssp, SSCR0, val); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) clk_enable(ssp->clk); @@ -295,11 +295,11 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case PXA_SSP_AUDIO_DIV_ACDS: - val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - ssp_write_reg(ssp, SSACD, val); + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_AUDIO_DIV_SCDB: - val = ssp_read_reg(ssp, SSACD); + val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) @@ -322,10 +322,10 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - ssp_write_reg(ssp, SSACD, val); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(ssp, div); + pxa_ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -342,11 +342,11 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) - ssp_write_reg(ssp, SSACDD, 0); + pxa_ssp_write_reg(ssp, SSACDD, 0); #endif switch (freq_out) { @@ -384,7 +384,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val = tmp; val = (val << 16) | 64; - ssp_write_reg(ssp, SSACDD, val); + pxa_ssp_write_reg(ssp, SSACDD, val); ssacd |= (0x6 << 4); @@ -398,7 +398,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, return -EINVAL; } - ssp_write_reg(ssp, SSACD, ssacd); + pxa_ssp_write_reg(ssp, SSACD, ssacd); return 0; } @@ -413,7 +413,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); /* set slot width */ @@ -430,10 +430,10 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, sscr0 |= SSCR0_SlotsPerFrm(slots); /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, tx_mask); - ssp_write_reg(ssp, SSRSA, rx_mask); + pxa_ssp_write_reg(ssp, SSTSA, tx_mask); + pxa_ssp_write_reg(ssp, SSRSA, rx_mask); } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); return 0; } @@ -448,12 +448,12 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr1; - sscr1 = ssp_read_reg(ssp, SSCR1); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1); if (tristate) sscr1 &= ~SSCR1_TTE; else sscr1 |= SSCR1_TTE; - ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); return 0; } @@ -477,14 +477,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { dev_err(&ssp->pdev->dev, "can't change hardware dai format: stream is in use"); return -EINVAL; } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -536,9 +536,9 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); dump_registers(ssp); @@ -567,7 +567,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -577,20 +577,20 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx @@ -606,13 +606,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { + if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -646,7 +646,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_DMYSTRT(1); } - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; @@ -677,45 +677,45 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(ssp); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); + val = pxa_ssp_read_reg(ssp, SSSR); + pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(ssp); + pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); break; default: @@ -737,7 +737,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; @@ -757,7 +757,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->ssp); + pxa_ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.5.6.5

1 0

[PATCH] [ARM] pxa: add namespace on ssp
by Haojian Zhuang 31 Mar '10

31 Mar '10

In order to prevent code ambiguous, add namespace on functions in ssp driver. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/plat-pxa/include/plat/ssp.h | 12 ++-- arch/arm/plat-pxa/ssp.c | 22 +++--- drivers/spi/pxa2xx_spi.c | 8 +- sound/soc/pxa/pxa-ssp.c | 132 +++++++++++++++++----------------- 4 files changed, 87 insertions(+), 87 deletions(-) diff --git a/arch/arm/plat-pxa/include/plat/ssp.h b/arch/arm/plat-pxa/include/plat/ssp.h index d16d79a..fe43150 100644 --- a/arch/arm/plat-pxa/include/plat/ssp.h +++ b/arch/arm/plat-pxa/include/plat/ssp.h @@ -159,28 +159,28 @@ struct ssp_device { }; /** - * ssp_write_reg - Write to a SSP register + * pxa_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) +static inline void pxa_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 + * pxa_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) +static inline u32 pxa_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 *); +struct ssp_device *pxa_ssp_request(int port, const char *label); +void pxa_ssp_free(struct ssp_device *); #endif /* __ASM_ARCH_SSP_H */ diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index cfebcd8..511805e 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -37,7 +37,7 @@ static DEFINE_MUTEX(ssp_lock); static LIST_HEAD(ssp_list); -struct ssp_device *ssp_request(int port, const char *label) +struct ssp_device *pxa_ssp_request(int port, const char *label) { struct ssp_device *ssp = NULL; @@ -58,9 +58,9 @@ struct ssp_device *ssp_request(int port, const char *label) return ssp; } -EXPORT_SYMBOL(ssp_request); +EXPORT_SYMBOL(pxa_ssp_request); -void ssp_free(struct ssp_device *ssp) +void pxa_ssp_free(struct ssp_device *ssp) { mutex_lock(&ssp_lock); if (ssp->use_count) { @@ -70,9 +70,9 @@ void ssp_free(struct ssp_device *ssp) dev_err(&ssp->pdev->dev, "device already free\n"); mutex_unlock(&ssp_lock); } -EXPORT_SYMBOL(ssp_free); +EXPORT_SYMBOL(pxa_ssp_free); -static int __devinit ssp_probe(struct platform_device *pdev) +static int __devinit pxa_ssp_probe(struct platform_device *pdev) { const struct platform_device_id *id = platform_get_device_id(pdev); struct resource *res; @@ -164,7 +164,7 @@ err_free: return ret; } -static int __devexit ssp_remove(struct platform_device *pdev) +static int __devexit pxa_ssp_remove(struct platform_device *pdev) { struct resource *res; struct ssp_device *ssp; @@ -196,9 +196,9 @@ static const struct platform_device_id ssp_id_table[] = { { }, }; -static struct platform_driver ssp_driver = { - .probe = ssp_probe, - .remove = __devexit_p(ssp_remove), +static struct platform_driver pxa_ssp_driver = { + .probe = pxa_ssp_probe, + .remove = __devexit_p(pxa_ssp_remove), .driver = { .owner = THIS_MODULE, .name = "pxa2xx-ssp", @@ -208,12 +208,12 @@ static struct platform_driver ssp_driver = { static int __init pxa_ssp_init(void) { - return platform_driver_register(&ssp_driver); + return platform_driver_register(&pxa_ssp_driver); } static void __exit pxa_ssp_exit(void) { - platform_driver_unregister(&ssp_driver); + platform_driver_unregister(&pxa_ssp_driver); } arch_initcall(pxa_ssp_init); diff --git a/drivers/spi/pxa2xx_spi.c b/drivers/spi/pxa2xx_spi.c index 4a64da7..2fd7d5a 100644 --- a/drivers/spi/pxa2xx_spi.c +++ b/drivers/spi/pxa2xx_spi.c @@ -1464,7 +1464,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) platform_info = dev->platform_data; - ssp = ssp_request(pdev->id, pdev->name); + ssp = pxa_ssp_request(pdev->id, pdev->name); if (ssp == NULL) { dev_err(&pdev->dev, "failed to request SSP%d\n", pdev->id); return -ENODEV; @@ -1474,7 +1474,7 @@ static int __init pxa2xx_spi_probe(struct platform_device *pdev) master = spi_alloc_master(dev, sizeof(struct driver_data) + 16); if (!master) { dev_err(&pdev->dev, "cannot alloc spi_master\n"); - ssp_free(ssp); + pxa_ssp_free(ssp); return -ENOMEM; } drv_data = spi_master_get_devdata(master); @@ -1603,7 +1603,7 @@ out_error_irq_alloc: out_error_master_alloc: spi_master_put(master); - ssp_free(ssp); + pxa_ssp_free(ssp); return status; } @@ -1647,7 +1647,7 @@ static int pxa2xx_spi_remove(struct platform_device *pdev) free_irq(ssp->irq, drv_data); /* Release SSP */ - ssp_free(ssp); + pxa_ssp_free(ssp); /* Disconnect from the SPI framework */ spi_unregister_master(drv_data->master); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 4ca9245..ed54bef 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -55,15 +55,15 @@ struct ssp_priv { 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)); + pxa_ssp_read_reg(ssp, SSCR0), pxa_ssp_read_reg(ssp, SSCR1), + pxa_ssp_read_reg(ssp, SSTO)); dev_dbg(&ssp->pdev->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", - ssp_read_reg(ssp, SSPSP), ssp_read_reg(ssp, SSSR), - ssp_read_reg(ssp, SSACD)); + pxa_ssp_read_reg(ssp, SSPSP), pxa_ssp_read_reg(ssp, SSSR), + pxa_ssp_read_reg(ssp, SSACD)); } -static void ssp_enable(struct ssp_device *ssp) +static void pxa_ssp_enable(struct ssp_device *ssp) { uint32_t sscr0; @@ -71,7 +71,7 @@ static void ssp_enable(struct ssp_device *ssp) __raw_writel(sscr0, ssp->mmio_base + SSCR0); } -static void ssp_disable(struct ssp_device *ssp) +static void pxa_ssp_disable(struct ssp_device *ssp) { uint32_t sscr0; @@ -85,7 +85,7 @@ struct pxa2xx_pcm_dma_data { }; static struct pxa2xx_pcm_dma_params * -ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) +pxa_ssp_get_dma_params(struct ssp_device *ssp, int width4, int out) { struct pxa2xx_pcm_dma_data *dma; @@ -117,7 +117,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, if (!cpu_dai->active) { clk_enable(ssp->clk); - ssp_disable(ssp); + pxa_ssp_disable(ssp); } if (cpu_dai->dma_data) { @@ -136,7 +136,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) { - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); } @@ -161,7 +161,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->to = __raw_readl(ssp->mmio_base + SSTO); priv->psp = __raw_readl(ssp->mmio_base + SSPSP); - ssp_disable(ssp); + pxa_ssp_disable(ssp); clk_disable(ssp->clk); return 0; } @@ -181,7 +181,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) __raw_writel(priv->psp, ssp->mmio_base + SSPSP); if (cpu_dai->active) - ssp_enable(ssp); + pxa_ssp_enable(ssp); else clk_disable(ssp->clk); @@ -197,9 +197,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_device *ssp, u32 div) +static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; @@ -208,15 +208,15 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) sscr0 &= ~0x000fff00; sscr0 |= (div - 1) << 8; /* 1..4096 */ } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); } /** - * ssp_get_clkdiv - get SSP clock divider + * pxa_ssp_get_clkdiv - get SSP clock divider */ -static u32 ssp_get_scr(struct ssp_device *ssp) +static u32 pxa_ssp_get_scr(struct ssp_device *ssp) { - u32 sscr0 = ssp_read_reg(ssp, SSCR0); + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) @@ -236,7 +236,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; int val; - u32 sscr0 = ssp_read_reg(ssp, SSCR0) & + u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); dev_dbg(&ssp->pdev->dev, @@ -264,7 +264,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(ssp, 1); + pxa_ssp_set_scr(ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -275,8 +275,8 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) clk_disable(ssp->clk); - val = ssp_read_reg(ssp, SSCR0) | sscr0; - ssp_write_reg(ssp, SSCR0, val); + val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; + pxa_ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) clk_enable(ssp->clk); @@ -295,11 +295,11 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case PXA_SSP_AUDIO_DIV_ACDS: - val = (ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); - ssp_write_reg(ssp, SSACD, val); + val = (pxa_ssp_read_reg(ssp, SSACD) & ~0x7) | SSACD_ACDS(div); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_AUDIO_DIV_SCDB: - val = ssp_read_reg(ssp, SSACD); + val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) @@ -322,10 +322,10 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, default: return -EINVAL; } - ssp_write_reg(ssp, SSACD, val); + pxa_ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(ssp, div); + pxa_ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -342,11 +342,11 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->ssp; - u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; + u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) if (cpu_is_pxa3xx()) - ssp_write_reg(ssp, SSACDD, 0); + pxa_ssp_write_reg(ssp, SSACDD, 0); #endif switch (freq_out) { @@ -384,7 +384,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val = tmp; val = (val << 16) | 64; - ssp_write_reg(ssp, SSACDD, val); + pxa_ssp_write_reg(ssp, SSACDD, val); ssacd |= (0x6 << 4); @@ -398,7 +398,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, return -EINVAL; } - ssp_write_reg(ssp, SSACD, ssacd); + pxa_ssp_write_reg(ssp, SSACD, ssacd); return 0; } @@ -413,7 +413,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); /* set slot width */ @@ -430,10 +430,10 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, sscr0 |= SSCR0_SlotsPerFrm(slots); /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, tx_mask); - ssp_write_reg(ssp, SSRSA, rx_mask); + pxa_ssp_write_reg(ssp, SSTSA, tx_mask); + pxa_ssp_write_reg(ssp, SSRSA, rx_mask); } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); return 0; } @@ -448,12 +448,12 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, struct ssp_device *ssp = priv->ssp; u32 sscr1; - sscr1 = ssp_read_reg(ssp, SSCR1); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1); if (tristate) sscr1 &= ~SSCR1_TTE; else sscr1 |= SSCR1_TTE; - ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); return 0; } @@ -477,14 +477,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { dev_err(&ssp->pdev->dev, "can't change hardware dai format: stream is in use"); return -EINVAL; } /* reset port settings */ - sscr0 = ssp_read_reg(ssp, SSCR0) & + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & (SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); sspsp = 0; @@ -536,9 +536,9 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - ssp_write_reg(ssp, SSCR0, sscr0); - ssp_write_reg(ssp, SSCR1, sscr1); - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); dump_registers(ssp); @@ -567,7 +567,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sscr0; u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); - int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; /* generate correct DMA params */ if (cpu_dai->dma_data) @@ -577,20 +577,20 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + cpu_dai->dma_data = pxa_ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); /* we can only change the settings if the port is not in use */ - if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) + if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; /* clear selected SSP bits */ - sscr0 = ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); - ssp_write_reg(ssp, SSCR0, sscr0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); /* bit size */ - sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: #ifdef CONFIG_PXA3xx @@ -606,13 +606,13 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16)); break; } - ssp_write_reg(ssp, SSCR0, sscr0); + pxa_ssp_write_reg(ssp, SSCR0, sscr0); switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); + sspsp = pxa_ssp_read_reg(ssp, SSPSP); - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { + if ((pxa_ssp_get_scr(ssp) == 4) && (width == 16)) { /* This is a special case where the bitclk is 64fs * and we're not dealing with 2*32 bits of audio * samples. @@ -646,7 +646,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_DMYSTRT(1); } - ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); break; default: break; @@ -677,45 +677,45 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(ssp); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); + val = pxa_ssp_read_reg(ssp, SSSR); + pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(ssp); + pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = ssp_read_reg(ssp, SSCR1); + val = pxa_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); + pxa_ssp_write_reg(ssp, SSCR1, val); break; default: @@ -737,7 +737,7 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio"); if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; @@ -757,7 +757,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->ssp); + pxa_ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.5.6.5

1 0

[alsa-devel] [PATCH 0/4] pxa ssp patches
by Haojian Zhuang 31 Mar '10

31 Mar '10

These patches are based on ssp_cleanup of pxa git tree. They're verified on pxa168 aspenite platform. Support I2S and Left J interface.

1 0

[alsa-devel] [PATCH 2/4] ASoC: split pxa ssp for reusing code
by Haojian Zhuang 31 Mar '10

31 Mar '10

1 0

[alsa-devel] pulseaudio module-udev-detect failure on 2.6.34-rc1
by Wu Fengguang 31 Mar '10

31 Mar '10

Greetings, Pulseaudio can no longer detect sound cards on kernel 2.6.34-rc1. Note that 2.6.33 is OK. I noticed that the related sysfs directories/files changed. Note that the card0/ directory is gone in new kernel: 2.6.33 wfg@hp ~% ll /sys/class/sound total 0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 timer -> ../../devices/virtual/sound/timer lrwxrwxrwx 1 root root 0 2010-03-31 12:23 pcmC0D1p -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/pcmC0D1p lrwxrwxrwx 1 root root 0 2010-03-31 12:23 pcmC0D0p -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/pcmC0D0p lrwxrwxrwx 1 root root 0 2010-03-31 12:23 pcmC0D0c -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/pcmC0D0c lrwxrwxrwx 1 root root 0 2010-03-31 12:23 mixer5 -> ../../devices/platform/thinkpad_acpi/sound/card5/mixer5 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 mixer -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/mixer lrwxrwxrwx 1 root root 0 2010-03-31 12:23 hwC0D0 -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/hwC0D0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 dsp -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/dsp lrwxrwxrwx 1 root root 0 2010-03-31 12:23 controlC5 -> ../../devices/platform/thinkpad_acpi/sound/card5/controlC5 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 controlC0 -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/controlC0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 card5 -> ../../devices/platform/thinkpad_acpi/sound/card5 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 card0 -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0 lrwxrwxrwx 1 root root 0 2010-03-31 12:23 audio -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/audio lrwxrwxrwx 1 root root 0 2010-03-31 12:23 adsp -> ../../devices/pci0000:00/0000:00:1b.0/sound/card0/adsp 2.6.34-rc1 wfg@hp ~% ll /sys/class/sound total 0 drwxr-xr-x 3 root root 0 2010-03-31 19:27 timer drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D7p drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D3p drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D0p drwxr-xr-x 3 root root 0 2010-03-31 19:27 pcmC0D0c drwxr-xr-x 3 root root 0 2010-03-31 19:27 hwC0D3 drwxr-xr-x 3 root root 0 2010-03-31 19:27 hwC0D0 drwxr-xr-x 3 root root 0 2010-03-31 19:27 controlC0 Any ideas? Thanks, Fengguang

2 2

[alsa-devel] [PATCH] ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
by Daniel Chen 31 Mar '10

31 Mar '10

Hi, Attached is a patch against sound-2.6 master HEAD (applicable to 2.6.32.y and 2.6.33.y) that clamps the 0 dB offset accordingly. Best, -Dan

2 1

Re: [alsa-devel] mx31 snd and mc13783 codec status
by Mark Brown 31 Mar '10

31 Mar '10

On Tue, Mar 30, 2010 at 06:23:18PM +0200, Valentin Longchamp wrote: > I have quickly hacked something based on what you had done for > pcm037 with the MC13783 codec in some of your first sound patches. I > know that this is not ready for mainline and I'm still struggling to > debug my code, but I would like to have results quite soon so that > my colleague could see what sounds (especially regarding to the > volume) our system is capable to produce. If you're struggling to find time to do full support for the CODEC a good technique is often to get just very basic fixed function support (say just DAC to headphone support) implemented and merged, then go back and implement I'd also suggest looking at the current Freescale BSPs - obviously this device is used on a lot of Freescale reference boards and I believe there's a reasonably mainlineish driver in the BSPs they have for current devices.

2 1

Re: [alsa-devel] mx31 snd and mc13783 codec status
by Sascha Hauer 31 Mar '10

31 Mar '10

Hi Valentin, On Tue, Mar 30, 2010 at 06:23:18PM +0200, Valentin Longchamp wrote: > Hi Sascha, > > For one of our experiments where our robots need to produce sound, I > would need a sound driver for mx31moboard platform. > > I have quickly hacked something based on what you had done for pcm037 > with the MC13783 codec in some of your first sound patches. I know that > this is not ready for mainline and I'm still struggling to debug my > code, but I would like to have results quite soon so that my colleague > could see what sounds (especially regarding to the volume) our system is > capable to produce. > > I would like to know if by any chance you have some updates on this code > ? Find the latest version of my code here: The following changes since commit 01e77706cdde7c0b47e5ca1f4284a795504c7c40: Linus Torvalds (1): Merge branch 'for-linus' of git://gitorious.org/linux-omap-dss2/linux are available in the git repository at: git://git.pengutronix.de/git/sha/linux-2.6.git mc13783 Sascha Hauer (3): add a mc13783 codec driver add phycore-mc13783 sound support pcm038: add sound support arch/arm/mach-mx2/mach-pcm038.c | 23 ++- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 727 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 32 ++ sound/soc/imx/Kconfig | 9 + sound/soc/imx/Makefile | 3 + sound/soc/imx/phycore-mc13783.c | 160 +++++++++ 8 files changed, 959 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/phycore-mc13783.c > And do you know if your initial mc13783 codec support coupled with > mx31 had some limitations ? Our setup is quite straightforward, we have > direct connection from the mx31 to the mc13783 on a single SSI. Our board uses both SSI channels of the MC13783 which we then mux into one channel in the DAM unit. I don't know how this affects you. Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 30 Mar '10

30 Mar '10

The following changes since commit 923a00427a141a6d1c83f1b151221a2458b2de8d: Tejun Heo (1): ASoC: update gfp/slab.h includes 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 Barry Song (1): ASoC: ad193x: move codec register/unregister to bus probe/remove Dan Carpenter (1): ASoC: wm8994: playback => capture Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak Kuninori Morimoto (4): ASoC: fsi: ensures process inside master lock ASoC: fsi: IRQ related process had be united ASoC: fsi: Add FIFO size calculate ASoC: fsi: Add FSI2 device support Mark Brown (2): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.35 Merge branch 'for-2.6.34' into for-2.6.35 sound/soc/codecs/ac97.c | 15 ++-- sound/soc/codecs/ad193x.c | 102 ++++++++++-------------- sound/soc/codecs/wm8994.c | 2 +- sound/soc/sh/Kconfig | 3 +- sound/soc/sh/fsi.c | 191 ++++++++++++++++++++++++++++++++------------- sound/soc/soc-core.c | 3 +- 6 files changed, 192 insertions(+), 124 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 30 Mar '10

30 Mar '10

The following changes since commit 3cc4e53f86dab635166929bfa47cc68d59b28c26: Mark Brown (1): ASoC: Remove BROKEN from i.MX audio after dependencies merged 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 A couple of fairly straightforward bug fixes. The ac97 patch is mildly invasive but it tested OK for me (and should be very obvious if it fails). Dan Carpenter (1): ASoC: wm8994: playback => capture Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak sound/soc/codecs/ac97.c | 15 +++++++++------ sound/soc/codecs/wm8994.c | 2 +- sound/soc/soc-core.c | 3 ++- 3 files changed, 12 insertions(+), 8 deletions(-)

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[alsa-devel] digiface vs. multiface vs. jack vs. other cards
by Fernando Lopez-Lezcano 30 Mar '10

30 Mar '10

At CCRMA we have several studios with linux computers in them (in addition to computers sprinkled throughout the building). Some of the studios have digiface interfaces, some multiface. All of them are connected to the sound system or mixer digitally through adat lightpipes. Workstations have Delta 66 or Gina3G soundcards. It would be nice for a user to be able to play, say, a stereo Ardour session in any of them without any changes to the session or routing. The digifaces send channels 1-24 out the adat ports. The multifaces on the other hand send 9-16 on the adat port. Deltas and Ginas send out stuff on channels 1-4 or 1-6. You can see the multiface interfaces are a problem. To get sound out of them you have to send on channels 9-16 which is different from _all_ others! I tried several workarounds without much success: a) fix this at the alsa level. Should be the way to go, right? Create an /etc/asound.conf that uses ttable to translate channels for the multifaces so that the digital out is actually sent through channels 1-8 (and create the same name plug interface in all other computers that does nothing so that the user can set that name as the default in qjackctl and will get the same behavior in all computers). Works, but not quite. When I try to use that "fake" plug interface with jack2 I get a warning that I'm using a plug interface (sort of fair) and jack only opens it with _2_ i/o channels (even though the ttable is 18 channels wide!). If I _force_ input and output channels in jack to 18 it does work fine. But now it will _not_ work on other computers that have less (or more) than 18 channels. Argh. b) fix it in hdspmixer. Well, it turns out that hdspmixer does not read the default matrix gains from a file so I can't really override the default. AFAICT the gains are calculated algorithmically in the code and there is no command line option to load presets from a file. Or I could write a small script that will set the right gains for 1-8 to go out through 9-16 but if a user starts hdspmixer that gets undone with no warning. And, of course: c) give up and connect an 8 channel analog snake to the analog outputs of the multifaces and do away with a digital connection to the mixer... Anyone has any other suggestions?? This should be easy to fix, right? :-) -- Fernando

2 1

[alsa-devel] [PATCH] ad193x: move codec register/unregister to bus probe/remove
by Barry Song 29 Mar '10

29 Mar '10

The way i've factored out the bus probe and removal functions so that there's no code in the individual I2C and SPI functions means that the register() and unregister() functions could just be squashed into the bus_probe() and bus_remove() functions. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad193x.c | 102 +++++++++++++++++++------------------------- 1 files changed, 44 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index d034464..e6fe8ac 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -34,8 +34,6 @@ static const u8 ad193x_reg[AD193X_NUM_REGS] = { static struct snd_soc_codec *ad193x_codec; struct snd_soc_codec_device soc_codec_dev_ad193x; -static int ad193x_register(struct ad193x_priv *ad193x, int bus_type); -static void ad193x_unregister(struct ad193x_priv *ad193x); /* * AD193X volume/mute/de-emphasis etc. controls @@ -289,69 +287,23 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) { struct snd_soc_codec *codec; struct ad193x_priv *ad193x; + int ret; + + if (ad193x_codec) { + dev_err(dev, "Another ad193x is registered\n"); + return -EINVAL; + } ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - codec = &ad193x->codec; - codec->control_data = ctrl_data; - codec->dev = dev; - dev_set_drvdata(dev, ad193x); - return ad193x_register(ad193x, bus_type); -} - -static int ad193x_bus_remove(struct device *dev) -{ - struct ad193x_priv *ad193x = dev_get_drvdata(dev); - - ad193x_unregister(ad193x); - return 0; -} - -static struct snd_soc_dai_ops ad193x_dai_ops = { - .hw_params = ad193x_hw_params, - .digital_mute = ad193x_mute, - .set_tdm_slot = ad193x_set_tdm_slot, - .set_fmt = ad193x_set_dai_fmt, -}; - -/* codec DAI instance */ -struct snd_soc_dai ad193x_dai = { - .name = "AD193X", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad193x_dai_ops, -}; -EXPORT_SYMBOL_GPL(ad193x_dai); - -static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) -{ - int ret; - struct snd_soc_codec *codec = &ad193x->codec; - - if (ad193x_codec) { - dev_err(codec->dev, "Another ad193x is registered\n"); - return -EINVAL; - } - + codec = &ad193x->codec; mutex_init(&codec->mutex); + codec->control_data = ctrl_data; + codec->dev = dev; codec->private_data = ad193x; codec->reg_cache = ad193x->reg_cache; codec->reg_cache_size = AD193X_NUM_REGS; @@ -412,14 +364,48 @@ static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) return 0; } -static void ad193x_unregister(struct ad193x_priv *ad193x) +static int ad193x_bus_remove(struct device *dev) { + struct ad193x_priv *ad193x = dev_get_drvdata(dev); + snd_soc_unregister_dai(&ad193x_dai); snd_soc_unregister_codec(&ad193x->codec); kfree(ad193x); ad193x_codec = NULL; + + return 0; } +static struct snd_soc_dai_ops ad193x_dai_ops = { + .hw_params = ad193x_hw_params, + .digital_mute = ad193x_mute, + .set_tdm_slot = ad193x_set_tdm_slot, + .set_fmt = ad193x_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad193x_dai = { + .name = "AD193X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad193x_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad193x_dai); + static int ad193x_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); -- 1.5.6.3

3 2

[alsa-devel] Volume levels with Geode LX/CS5536
by Valentin 29 Mar '10

29 Mar '10

Hi, the maximum volume level (100%) on my soundcard for the PCM mixer is 12.0dB instead of 0.0dB. Is there a way for me to fix this, or help fixing it? crimsun on freenode/#alsa told me to ask you guys :) My alsa-info.sh output is at http://0au.de/~apo/p/ai.txt Cheers, Valentin

2 1

[alsa-devel] bug in spi and i2c dual buses codecs
by Barry Song 29 Mar '10

29 Mar '10

Hi Mark/Liam, I think there is really a bug in spi and i2c dual buses codecs drivers, for example, for any codec with two buses, if you config kernel like this: SPI MASTER built-in I2C module Codec built-in You will get a link error like this: sound/built-in.o: In function `i2c_add_driver': include/linux/i2c.h:423: undefined reference to `i2c_register_driver' sound/built-in.o: In function `ad193x_modexit': sound/soc/codecs/ad193x.c:537: undefined reference to `i2c_del_driver' make: *** [.tmp_vmlinux1] Error 1 I remember that is also one reason why we have splitted this kind of driver into three parts(codec core, i2c bus, spi bus). In fact, if any bus is compiled as module and code is built-in, link error will be reported. Thanks Barry

3 4

[alsa-devel] Soundcard Detection problem (ALSA regression)
by Gregory T Flynn 29 Mar '10

29 Mar '10

So far I have tested the current stable drive and now the current snapshot, both to no avail. The last ALSA driver version that worked on this sound card was 1.0.18. General Info from the 1.0.18 driver: !!PCI Soundcards installed in the system !!-------------------------------------- 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:10.1 0403: 10de:026c (rev a2) Subsystem: 1462:373d Currently I'm running Ubuntu 10.04 beta1 and this sounds regression (which was also in 9.10) is my only problem with the new system. So it would be awesome to get solved and packed into the beta before release. more details in bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4967 -- - Greg Flynn

1 0

[alsa-devel] sound: patch_realtek: truncating bits
by Dan Carpenter 29 Mar '10

29 Mar '10

Smatch complains that we are truncating bits here: sound/pci/hda/patch_realtek.c +12459 alc268_aspire_one_speaker_automute(6) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13482 alc269_quanta_fl1_speaker_automute(6) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13511 alc269_lifebook_speaker_automute(11) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13646 alc269_speaker_automute(8) warn: value 28800 can't fit into 255 'bits' The code looks like this: 13643 unsigned char bits; 13644 13645 present = snd_hda_jack_detect(codec, nid); 13646 bits = present ? AMP_IN_MUTE(0) : 0; "bits" is declared as an unsigned char but AMP_IN_MUTE(0) is 0x7080 so we are only using the last 0x80. I couldn't figure out if it was intended. This was first introduced by this commit which change "bits" from unsigned int to unsigned char. commit 60db6b53fb43421beb2ff3fe3e63412bf81620aa Author: Kailang Yang <kailang(a)realtek.com> Date: Tue Aug 26 13:13:00 2008 +0200 ALSA: hda - Add support of Quanta FL1 Added the support of Quanta FL1 with ALC269 code chip. Also a bit space clean-ups. If this is intentional, it would be cleaner to explicitly cast or mask away the bits we don't care about. regards, dan carpenter

2 1

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for ga-ma770-ud3 board
by Daniel Chen 29 Mar '10

29 Mar '10

Hi, Attached is a patch against current sound-2.6 master HEAD (also applicable to 2.6.32.y and 2.6.33.y) that makes the OR's audio more tolerable with PulseAudio. Best, -Dan

2 1

[alsa-devel] [PATCH] ALSA: ac97: Add Toshiba P500 to ac97 jack sense blacklist
by Daniel Chen 29 Mar '10

29 Mar '10

Hi, Attached is a patch against sound-2.6 (also applicable to 2.6.32.y and 2.6.33.y) adding the OR's SSID to ad1981_jacks_blacklist for audible audio by default. Best, -Dan

2 1

[alsa-devel] [PATCH 2/4 v2] ASoC: fsi: Add FIFO size calculate
by Kuninori Morimoto 29 Mar '10

29 Mar '10

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o move calculate area to fsi_fifo_init o modify fifo calculation way o Add tiny document for fifo calculation sound/soc/sh/fsi.c | 70 ++++++++++++++++++++++++++++++++-------------------- 1 files changed, 43 insertions(+), 27 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7c295df..ae88865 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -46,8 +46,9 @@ #define MUTE 0x020C #define CLK_RST 0x0210 #define SOFT_RST 0x0214 +#define FIFO_SZ 0x0218 #define MREG_START INT_ST -#define MREG_END SOFT_RST +#define MREG_END FIFO_SZ /* DO_FMT */ /* DI_FMT */ @@ -85,6 +86,11 @@ #define IR (1 << 4) /* Interrupt Reset */ #define FSISR (1 << 0) /* Software Reset */ +/* FIFO_SZ */ +#define OUT_SZ_MASK 0x7 +#define BO_SZ_SHIFT 8 +#define AO_SZ_SHIFT 0 + #define FSI_RATES SNDRV_PCM_RATE_8000_96000 #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) @@ -384,9 +390,42 @@ static void fsi_clk_ctrl(struct fsi_priv *fsi, int enable) fsi_master_mask_set(master, CLK_RST, val, 0); } -static void fsi_fifo_init(struct fsi_priv *fsi, int is_play) +static void fsi_fifo_init(struct fsi_priv *fsi, + int is_play, + struct snd_soc_dai *dai) { - u32 ctrl; + struct fsi_master *master = fsi_get_master(fsi); + u32 ctrl, shift, i; + + /* get on-chip RAM capacity */ + shift = fsi_master_read(master, FIFO_SZ); + shift >>= fsi_is_port_a(fsi) ? AO_SZ_SHIFT : BO_SZ_SHIFT; + shift &= OUT_SZ_MASK; + fsi->fifo_max = 256 << shift; + dev_dbg(dai->dev, "fifo = %d words\n", fsi->fifo_max); + + /* + * The maximum number of sample data varies depending + * on the number of channels selected for the format. + * + * FIFOs are used in 4-channel units in 3-channel mode + * and in 8-channel units in 5- to 7-channel mode + * meaning that more FIFOs than the required size of DPRAM + * are used. + * + * ex) if 256 words of DP-RAM is connected + * 1 channel: 256 (256 x 1 = 256) + * 2 channels: 128 (128 x 2 = 256) + * 3 channels: 64 ( 64 x 3 = 192) + * 4 channels: 64 ( 64 x 4 = 256) + * 5 channels: 32 ( 32 x 5 = 160) + * 6 channels: 32 ( 32 x 6 = 192) + * 7 channels: 32 ( 32 x 7 = 224) + * 8 channels: 32 ( 32 x 8 = 256) + */ + for (i = 1; i < fsi->chan; i <<= 1) + fsi->fifo_max >>= 1; + dev_dbg(dai->dev, "%d channel %d store\n", fsi->chan, fsi->fifo_max); ctrl = is_play ? DOFF_CTL : DIFF_CTL; @@ -689,29 +728,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, dev_err(dai->dev, "unknown format.\n"); return -EINVAL; } - - switch (fsi->chan) { - case 1: - fsi->fifo_max = 256; - break; - case 2: - fsi->fifo_max = 128; - break; - case 3: - case 4: - fsi->fifo_max = 64; - break; - case 5: - case 6: - case 7: - case 8: - fsi->fifo_max = 32; - break; - default: - dev_err(dai->dev, "channel size error.\n"); - return -EINVAL; - } - fsi_reg_write(fsi, reg, data); /* @@ -725,7 +741,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, fsi_irq_clear_status(fsi); /* fifo init */ - fsi_fifo_init(fsi, is_play); + fsi_fifo_init(fsi, is_play, dai); return ret; } -- 1.6.3.3

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[alsa-devel] [for 2.6.34][PATCH] ALSA: pcm_lib - fix xrun functionality
by Jarkko Nikula 27 Mar '10

27 Mar '10

The commit 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 broke the interrupt time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG is not set. This is because the xrun() is null defined without it. Fix this by letting the function xrun() to be always defined as it was before. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_lib.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b546ac2..a2ff861 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -148,6 +148,9 @@ static void pcm_debug_name(struct snd_pcm_substream *substream, #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) +#else +#define xrun_debug(substream, mask) 0 +#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ @@ -169,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ @@ -255,8 +259,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ -#define xrun_debug(substream, mask) 0 -#define xrun(substream) do { } while (0) #define hw_ptr_error(substream, fmt, args...) do { } while (0) #define xrun_log(substream, pos) do { } while (0) #define xrun_log_show(substream) do { } while (0) -- 1.7.0

4 4

Re: [alsa-devel] The volume buttons on Lenovo Thinkpad X61S (was: Re: [Alsa-user] Console beep on Lenovo Thinkpad X61S)
by Jaroslav Kysela 26 Mar '10

26 Mar '10

On Fri, 26 Mar 2010, Lars Bjørndal wrote: > The volume buttons up, down and mute, doesn't work from within the > console. How could that be fixed? Please note that I use Fedora 12, > and I don't use X. It's a bit different thing. The ACPI events are translated as key-press events to the /dev/input/event* interface. So you need an application (daemon) watching for these key-press events and modifying the volume using the ALSA mixer interface. The GUI programs do this. Here is a short program in python reading input events and translating them to amixer calls: =========== cut here =================== #!/usr/bin/python import struct import os # use 'evtest' program to determine right /dev/input device and code acpievents = "/dev/input/event1" fmt = 'iihhi' fd = open(acpievents, "rb") event = fd.read(16) while event: (time1, time2, type, code, value) = struct.unpack(fmt, event) if type == 1: if code == 115: # keypress VOLUMEUP os.system("amixer -q -c 0 set PCM 10%+") elif code == 114: # keypress VOLUMEDOWN os.system("amixer -q -c 0 set PCM 10%-") event = fd.read(16) fd.close() ================ cut here ============= Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] Speakertest repeats sounds on ATI HD3200 HDMI (Fedora 12 x86_64, 2.6.33.1 kernel.org)
by Udo van den Heuvel 26 Mar '10

26 Mar '10

Hello, I am getting repeated sounds when using speakertest: FronFronFronFronFronFronFronFronFronFron... RontRontRontRont... etc. Seeh ttps://bugtrack.alsa-project.org/alsa-bug/view.php?id=4965 for details about the bug. It's a Jetway NC81-LF w/ ATI HD3200 onboard video. HDMI connection to the TV. Video works great, adding audio gives stuttering effects. So we started debugging. I have a speakertest-strace: http://pastebin.com/Ad7j8Yc5. I start the test and the audio goes like: FronFronFronFronFronFronFronFronFronFronFronFronFronFronFronFron the strace sits and I pastebinned it. Then the speaker-test continues and the audio goes like RontRontRontRontRontRont See http://pastebin.com/TNpsv1bv for the strace. Any idea what might be wrong here? Kind regards, Udo

1 0

[alsa-devel] [PATCH] WM8750: Convert to new API
by Marek Vasut 26 Mar '10

26 Mar '10

Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code around. Hugely inspired by WM8753 which was already converted. Signed-off-by: Marek Vasut <marek.vasut(a)gmail.com> --- sound/soc/codecs/wm8750.c | 411 +++++++++++++++++++++++++-------------------- 1 files changed, 232 insertions(+), 179 deletions(-) diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 475c67a..c7ee6d2 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -29,13 +29,6 @@ #include "wm8750.h" -#define WM8750_VERSION "0.12" - -/* codec private data */ -struct wm8750_priv { - unsigned int sysclk; -}; - /* * wm8750 register cache * We can't read the WM8750 register space when we @@ -55,6 +48,59 @@ static const u16 wm8750_reg[] = { 0x0079, 0x0079, 0x0079, /* 40 */ }; +/* codec private data */ +struct wm8750_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8750_reg)]; +}; + +/* + * read wm8750 register cache + */ +static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg < 1 || reg >= (ARRAY_SIZE(wm8750_reg) + 1)) + return -1; + return cache[reg - 1]; +} + +/* + * write wm8750 register cache + */ +static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg < 1 || reg >= (ARRAY_SIZE(wm8750_reg) + 1)) + return; + cache[reg - 1] = value; +} + +/* + * write to the WM8750 register space + */ +static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8750 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8750_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + #define wm8750_reset(c) snd_soc_write(c, WM8750_RESET, 0) /* @@ -688,6 +734,11 @@ static int wm8750_resume(struct platform_device *pdev) for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) { if (i == WM8750_RESET) continue; + + /* No point in writing hardware default values back */ + if (cache[i] == wm8750_reg[i]) + continue; + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); data[1] = cache[i] & 0x00ff; codec->hw_write(codec->control_data, data, 2); @@ -706,31 +757,111 @@ static int wm8750_resume(struct platform_device *pdev) return 0; } +static struct snd_soc_codec *wm8750_codec; + +static int wm8750_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (!wm8750_codec) { + dev_err(&pdev->dev, "WM8750 codec not yet registered\n"); + return -EINVAL; + } + + socdev->card->codec = wm8750_codec; + codec = wm8750_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8750: failed to create pcms\n"); + goto err; + } + + snd_soc_add_controls(codec, wm8750_snd_controls, + ARRAY_SIZE(wm8750_snd_controls)); + wm8750_add_widgets(codec); + + return 0; + +err: + return ret; +} + +/* + * This function forces any delayed work to be queued and run. + */ +static int run_delayed_work(struct delayed_work *dwork) +{ + int ret; + + /* cancel any work waiting to be queued. */ + ret = cancel_delayed_work(dwork); + + /* if there was any work waiting then we run it now and + * wait for it's completion */ + if (ret) { + schedule_delayed_work(dwork, 0); + flush_scheduled_work(); + } + return ret; +} + +/* power down chip */ +static int wm8750_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_wm8750 = { + .probe = wm8750_probe, + .remove = wm8750_remove, + .suspend = wm8750_suspend, + .resume = wm8750_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); + /* * initialise the WM8750 driver * register the mixer and dsp interfaces with the kernel */ -static int wm8750_init(struct snd_soc_device *socdev, - enum snd_soc_control_type control) +static int wm8750_register(struct wm8750_priv *wm8750) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_codec *codec = &wm8750->codec; int reg, ret = 0; + if (wm8750_codec) { + dev_err(codec->dev, "Multiple WM8750 devices not supported\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "WM8750"; codec->owner = THIS_MODULE; + codec->read = wm8750_read_reg_cache; + codec->write = wm8750_write; + codec->bias_level = SND_SOC_BIAS_STANDBY; codec->set_bias_level = wm8750_set_bias_level; codec->dai = &wm8750_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); - codec->reg_cache = kmemdup(wm8750_reg, sizeof(wm8750_reg), GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->private_data = wm8750; + codec->reg_cache_size = ARRAY_SIZE(wm8750->reg_cache) + 1; + codec->reg_cache = &wm8750->reg_cache; + codec->private_data = wm8750; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); - goto err; - } + memcpy(codec->reg_cache, wm8750_reg, sizeof(wm8750->reg_cache)); + INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work); ret = wm8750_reset(codec); if (ret < 0) { @@ -738,13 +869,6 @@ static int wm8750_init(struct snd_soc_device *socdev, goto err; } - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to create pcms\n"); - goto err; - } - /* charge output caps */ wm8750_set_bias_level(codec, SND_SOC_BIAS_PREPARE); codec->bias_level = SND_SOC_BIAS_STANDBY; @@ -768,19 +892,39 @@ static int wm8750_init(struct snd_soc_device *socdev, reg = snd_soc_read(codec, WM8750_RINVOL); snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100); - snd_soc_add_controls(codec, wm8750_snd_controls, - ARRAY_SIZE(wm8750_snd_controls)); - wm8750_add_widgets(codec); - return ret; + wm8750_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(&wm8750_dai, 1); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; +err_codec: + run_delayed_work(&codec->delayed_work); + snd_soc_unregister_codec(codec); err: - kfree(codec->reg_cache); + kfree(wm8750); return ret; } -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ -static struct snd_soc_device *wm8750_socdev; +static void wm8750_unregister(struct wm8750_priv *wm8750) +{ + wm8750_set_bias_level(&wm8750->codec, SND_SOC_BIAS_OFF); + run_delayed_work(&wm8750->codec.delayed_work); + snd_soc_unregister_dais(&wm8750_dai, 1); + snd_soc_unregister_codec(&wm8750->codec); + kfree(wm8750); + wm8750_codec = NULL; +} #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -794,24 +938,27 @@ static struct snd_soc_device *wm8750_socdev; static int wm8750_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; - ret = wm8750_init(socdev, SND_SOC_I2C); - if (ret < 0) - pr_err("failed to initialise WM8750\n"); + codec = &wm8750->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->control_data = i2c; + i2c_set_clientdata(i2c, wm8750); - return ret; + codec->dev = &i2c->dev; + + return wm8750_register(wm8750); } static int wm8750_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + struct wm8750_priv *wm8750 = i2c_get_clientdata(client); + wm8750_unregister(wm8750); return 0; } @@ -830,66 +977,56 @@ static struct i2c_driver wm8750_i2c_driver = { .remove = wm8750_i2c_remove, .id_table = wm8750_i2c_id, }; +#endif -static int wm8750_add_i2c_device(struct platform_device *pdev, - const struct wm8750_setup_data *setup) +#if defined(CONFIG_SPI_MASTER) +static int wm8750_spi_write(struct spi_device *spi, const char *data, int len) { - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; - ret = i2c_add_driver(&wm8750_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } + if (len <= 0) + return 0; - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8750", I2C_NAME_SIZE); + msg[0] = data[0]; + msg[1] = data[1]; - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } + spi_message_init(&m); + memset(&t, 0, (sizeof t)); - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } + t.tx_buf = &msg[0]; + t.len = len; - return 0; + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); -err_driver: - i2c_del_driver(&wm8750_i2c_driver); - return -ENODEV; + return len; } -#endif -#if defined(CONFIG_SPI_MASTER) static int __devinit wm8750_spi_probe(struct spi_device *spi) { - struct snd_soc_device *socdev = wm8750_socdev; - struct snd_soc_codec *codec = socdev->card->codec; - int ret; + struct snd_soc_codec *codec; + struct wm8750_priv *wm8750; + + wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); + if (wm8750 == NULL) + return -ENOMEM; + codec = &wm8750->codec; + codec->hw_write = (hw_write_t)wm8750_spi_write; codec->control_data = spi; + codec->dev = &spi->dev; - ret = wm8750_init(socdev, SND_SOC_SPI); - if (ret < 0) - dev_err(&spi->dev, "failed to initialise WM8750\n"); + dev_set_drvdata(&spi->dev, wm8750); - return ret; + return wm8750_register(wm8750); } static int __devexit wm8750_spi_remove(struct spi_device *spi) { + struct wm8750_priv *wm8750 = dev_get_drvdata(&spi->dev); + wm8750_unregister(wm8750); return 0; } @@ -904,115 +1041,31 @@ static struct spi_driver wm8750_spi_driver = { }; #endif -static int wm8750_probe(struct platform_device *pdev) +static int __init wm8750_modinit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8750_setup_data *setup = socdev->codec_data; - struct snd_soc_codec *codec; - struct wm8750_priv *wm8750; int ret; - - pr_info("WM8750 Audio Codec %s", WM8750_VERSION); - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - wm8750 = kzalloc(sizeof(struct wm8750_priv), GFP_KERNEL); - if (wm8750 == NULL) { - kfree(codec); - return -ENOMEM; - } - - codec->private_data = wm8750; - socdev->card->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - wm8750_socdev = socdev; - INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work); - - ret = -ENODEV; - #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - ret = wm8750_add_i2c_device(pdev, setup); - } + ret = i2c_add_driver(&wm8750_i2c_driver); + if (ret != 0) + pr_err("Failed to register WM8750 I2C driver: %d\n", ret); #endif #if defined(CONFIG_SPI_MASTER) - if (setup->spi) { - ret = spi_register_driver(&wm8750_spi_driver); - if (ret != 0) - printk(KERN_ERR "can't add spi driver"); - } + ret = spi_register_driver(&wm8750_spi_driver); + if (ret != 0) + pr_err("Failed to register WM8750 SPI driver: %d\n", ret); #endif - - if (ret != 0) { - kfree(codec->private_data); - kfree(codec); - } - return ret; -} - -/* - * This function forces any delayed work to be queued and run. - */ -static int run_delayed_work(struct delayed_work *dwork) -{ - int ret; - - /* cancel any work waiting to be queued. */ - ret = cancel_delayed_work(dwork); - - /* if there was any work waiting then we run it now and - * wait for it's completion */ - if (ret) { - schedule_delayed_work(dwork, 0); - flush_scheduled_work(); - } - return ret; + return 0; } +module_init(wm8750_modinit); -/* power down chip */ -static int wm8750_remove(struct platform_device *pdev) +static void __exit wm8750_exit(void) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; - - if (codec->control_data) - wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF); - run_delayed_work(&codec->delayed_work); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8750_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8750_spi_driver); #endif - kfree(codec->private_data); - kfree(codec); - - return 0; -} - -struct snd_soc_codec_device soc_codec_dev_wm8750 = { - .probe = wm8750_probe, - .remove = wm8750_remove, - .suspend = wm8750_suspend, - .resume = wm8750_resume, -}; -EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); - -static int __init wm8750_modinit(void) -{ - return snd_soc_register_dai(&wm8750_dai); -} -module_init(wm8750_modinit); - -static void __exit wm8750_exit(void) -{ - snd_soc_unregister_dai(&wm8750_dai); } module_exit(wm8750_exit); -- 1.7.0

3 2

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, devel now at v2.6.34-rc2-55-g0bf0e5a
by Takashi Iwai 26 Mar '10

26 Mar '10

At Fri, 26 Mar 2010 10:37:56 +0100 (CET), noreply-git(a)alsa-project.org wrote: > > commit cd508fe58b07499403e806b558c7f15c90442292 > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Fri Mar 26 10:28:46 2010 +0100 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Fri Mar 26 10:37:39 2010 +0100 > > ALSA: hda-intel - add special 'hwio' model to bypass initialization > > Using the 'model=hwio' option, the driver bypasses any codec > initialization and the reset procedure for codecs is also > bypassed. This mode is usefull to enable direct access using > hwdep interface (using hdaverb or hda-analyzer tools) and > retain codec setup from BIOS. What is the merit over probe_only=1 option...? Takashi

2 2

[alsa-devel] [PATCH] ALSA: ac97: Add IBM ThinkPad R40e to Headphone/Line Jack Sense blacklist
by Daniel Chen 26 Mar '10

26 Mar '10

Hi, Attached is a patch against current sound-2.6 master HEAD (also applicable to 2.6.32.y and 2.6.33.y) that adds the IBM ThinkPad R40e to the Headphone/Line Jack Sense blacklist, which is required for audible audio. Best, -Dan

2 1

[alsa-devel] TASCAM US-1641
by Jan Depner 26 Mar '10

26 Mar '10

A couple of weeks ago I purchased a TASCAM US-1641 on the off chance that I might be able to get the thing working on Linux (or give up and use it in Windows ;-) I have built the 2.6.33 kernel and started playing around in Karsten Wiese's US-122L module to see what I could get to happen. I did this on the off chance that TASCAM just increased the number of ports but used the same protocol for communication that they used for the US-122L. I added an entry for the US-1641 in us122l.h (#define USB_ID_US1641 0x8017) and then added an entry to the snd_us122l_usb_id_table in us122l.c. After recompiling and installing the modules I get the following in my dmesg output when I turn the device on: usb 1-2: new high speed USB device using ehci_hcd and address 3 couldn't __get_free_pages() snd-usb-us122l: probe of 1-2:1.1 failed with error -22 I don't have any problem with spending a lot of time on this (actually, it's kind of fascinating) and I am proficient in C but I could use some help. I'm hoping that someone on this list might have some information that could help me in trying to get this thing working (I've even got UsbSnoop installed on my Windows partition). Any information that could help would be greatly appreciated. Cheers, Jan Depner

4 6

[alsa-devel] [PATCH] [PATCH] [patch 1/2]AC97: add AC97 support for AT91.
by Sedji Gaouaou 25 Mar '10

25 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 | 339 ++++++++++++++++++++++++++++++++++++++------------- 2 files changed, 255 insertions(+), 86 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..564a474 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; } @@ -449,7 +490,7 @@ static int atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -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, 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; } @@ -608,15 +721,50 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) return retval; } +static struct ac97_pcm at91_ac97_pcm_defs[] __devinitdata = { + /* Playback */ + { + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } }, + }, + /* PCM in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } } + }, + /* Mic in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = (1<<AC97_SLOT_MIC), + } } + }, +}; + static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) { struct snd_pcm *pcm; struct snd_pcm_hardware hw = atmel_ac97c_hw; - int capture, playback, retval; + int capture, playback, retval, err; 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 +923,12 @@ 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); @@ -844,43 +997,52 @@ 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); @@ -897,20 +1059,22 @@ 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 +1098,12 @@ 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 +1115,12 @@ 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 +1145,16 @@ 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 1

[alsa-devel] Problem adding support for digital mixer in codec
by Candelaria Villarreal, Jorge 25 Mar '10

25 Mar '10

Hi, I am trying to add support for a digital mixer. The digital mixer is connected like the following example: Path 1 -------\ Path 2 --------\ MIXER ------> DAC----> ... Path 3 --------/ Path 4 -------/ However, I think ALSA does not support this configuration, as the DAC should be the first element in a route. I tried adding the DAC to the route first, and connected its output to the mixer inputs, like this: {"Mixer", "Multimedia Ext Switch", "HSDAC Left"}, {"Mixer", "Multimedia Ext Switch", "HSDAC Right"}, {"Mixer", "Voice Switch", "HSDAC Left"}, {"Mixer", "Voice Switch", "HSDAC Right"}, {"Mixer", "Multimedia Uplink Switch", "HSDAC Left"}, {"Mixer", "Multimedia Uplink Switch", "HSDAC Right"}, {"Mixer", "Multimedia Switch", "HSDAC Left"}, {"Mixer", "Multimedia Switch", "HSDAC Right"}, But this does not seem to work. Can anyone give me some tips? Any comment would be appreciated. Thanks, Jorge

5 6

[alsa-devel] [PATCH] ASoC: Fix passing platform_data to ac97 bus users and fix a leak.
by Graham Gower 25 Mar '10

25 Mar '10

snd_soc_new_ac97_codec() allocates for codec->ac97, snd_soc_new_pcms() then sets the platform_data. However, snd_ac97_mixer() overwrites codec->ac97 with its own allocated struct snd_ac97. Signed-off-by: Graham Gower <graham.gower(a)gmail.com> --- sound/soc/codecs/ac97.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a1bbe16..64ec797 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -80,9 +80,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; struct snd_soc_codec *codec; struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; + int i; int ret = 0; printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); @@ -118,11 +120,21 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; + /* free the ac97 here so that we don't leak it in snd_ac97_mixer */ + snd_soc_free_ac97_codec(codec); + memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97); if (ret < 0) goto bus_err; + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].codec_dai->ac97_control) { + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } + } + return 0; bus_err: -- 1.6.4

3 5

[alsa-devel] [PATCH - ALSA: usbaudio: Add basic support for M-AudioAdd basic support for M-Audio Fast Track Ultra series
by Felix Homann 25 Mar '10

25 Mar '10

This adds basic support for M-Audio's Fast Track Ultra series of USB audio interfaces. It is a refactored version of the patch Clemens Ladisch posted some time ago. Neither playback nor capturing work properly at 44100 Hz (don't know why). The other sampling rates work properly. There's no support for the DSP mixer, yet. Kind regards, Felix Signed-off-by: Felix Homann <fexpop(a)web.de> diff --git a/usb/endpoint.c b/usb/endpoint.c index b1309cd..7b84b61 100644 --- a/usb/endpoint.c +++ b/usb/endpoint.c @@ -317,6 +317,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ + case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; diff --git a/usb/quirks-table.h b/usb/quirks-table.h index 81c5f8a..91ddef3 100644 --- a/usb/quirks-table.h +++ b/usb/quirks-table.h @@ -1826,6 +1826,60 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0763, 0x2080), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "M-Audio", */ + /* .product_name = "Fast Track Ultra 8", */ + .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_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* interface 3 (MIDI) is standard compliant */ + { + .ifnum = -1 + } + } + } +}, +{ + USB_DEVICE(0x0763, 0x2081), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "M-Audio", */ + /* .product_name = "Fast Track Ultra 8R", */ + .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_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* interface 3 (MIDI) is standard compliant */ + { + .ifnum = -1 + } + } + } +}, /* Casio devices */ { diff --git a/usb/urb.c b/usb/urb.c index ad50d43..e2b7c47 100644 --- a/usb/urb.c +++ b/usb/urb.c @@ -918,6 +918,11 @@ void snd_usb_init_substream(struct snd_usb_stream *as, case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; break; + case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ + case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ + subs->ops.prepare_sync = prepare_playback_sync_urb; + subs->ops.retire_sync = retire_playback_sync_urb; + break; } } -- 1.7.0.3

2 1

[alsa-devel] aconnect patch
by lieven moors 25 Mar '10

25 Mar '10

Hi everybody, I'm a musician working intensively with Linux, and I like writing programs. Now I am working on a little command line utility to store and restore all ALSA (midi) and JACK connections to/from an xml file, and I've been using aconnect as an example for writing the ALSA part. I also needed a function to remove all connections, similar to the remove_connections function in aconnect. After using aconnect -x for some time, I noticed that it didn't work as it should. Very often a number of connections were left untouched after issuing this command. After digging a little bit deeper, I think I found the reason for the bug, and wrote a fix for it. This is what I think happens in remove_connections: Every time the for loop is executed, the index of the snd_seq_query_subscribe_t is updated (+ 1). If the connection has the right capabilities, it is removed. When the connection is removed, I think the index gets reassigned, and the next connection gets the position at index 0. The result is that when one connection is removed, the removal of the next connection is skipped (because the for loop updates the index every time). In the patch, I only update the index when unsubscribing fails, or if it is unwanted because the capabilities or wrong. This seems to solve the problems I had with aconnect -x, and now it disconnects all connections as it should. In the original function, connections are removed both ways. First all ports with SND_SEQ_PORT_CAP_SUBS_READ are checked, and associated connections removed, and then all ports with SND_SEQ_PORT_CAP_SUBS_WRITE. In the patch, I removed the second part, because I think it is doing the same thing twice. Please let me know what you think about this patch, and if you think my reasoning is sane... Greetings, Lieven Moors diff --git a/aconnect.c b/aconnect.c index 1a50666..6092748 100644 --- a/aconnect.c +++ b/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); + } } }

3 7

[alsa-devel] [rfc patch] wm8994: range checking issue
by Dan Carpenter 25 Mar '10

25 Mar '10

Smatch complained about BUG_ON(reg > WM8994_MAX_REGISTER) because the actual number of elements in the array was WM8994_REG_CACHE_SIZE + 1. I changed the BUG_ON() to return -EINVAL. I was confused why WM8994_REG_CACHE_SIZE was different from the actual size of ->reg_cache and I was concerned because some places used ARRAY_SIZE() to find the end of the array and other places used WM8994_REG_CACHE_SIZE. In my patch, I made them the same. I don't have the hardware to test this and some of this patch is just guess work. For example, I left it in, but why is there a -1 here? 3711 /* Fill the cache with physical values we inherited; don't reset */ 3712 ret = wm8994_bulk_read(codec->control_data, 0, 3713 ARRAY_SIZE(wm8994->reg_cache) - 1, 3714 codec->reg_cache); I didn't sign this off because I figured I'd probably need to send a second version after I hear the feed back. regards, dan carpenter diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771..d9c179a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -59,13 +59,13 @@ static int wm8994_retune_mobile_base[] = { WM8994_AIF2_EQ_GAINS_1, }; -#define WM8994_REG_CACHE_SIZE 0x621 +#define WM8994_REG_CACHE_SIZE 0x622 /* codec private data */ struct wm8994_priv { struct wm_hubs_data hubs; struct snd_soc_codec codec; - u16 reg_cache[WM8994_REG_CACHE_SIZE + 1]; + u16 reg_cache[WM8994_REG_CACHE_SIZE]; int sysclk[2]; int sysclk_rate[2]; int mclk[2]; @@ -1697,7 +1697,8 @@ static int wm8994_write(struct snd_soc_codec *codec, unsigned int reg, { struct wm8994_priv *wm8994 = codec->private_data; - BUG_ON(reg > WM8994_MAX_REGISTER); + if (reg >= WM8994_REG_CACHE_SIZE) + return -EINVAL; if (!wm8994_volatile(reg)) wm8994->reg_cache[reg] = value; @@ -1710,7 +1711,8 @@ static unsigned int wm8994_read(struct snd_soc_codec *codec, { u16 *reg_cache = codec->reg_cache; - BUG_ON(reg > WM8994_MAX_REGISTER); + if (reg >= WM8994_REG_CACHE_SIZE) + return -EINVAL; if (wm8994_volatile(reg)) return wm8994_reg_read(codec->control_data, reg); @@ -3700,7 +3702,7 @@ static int wm8994_codec_probe(struct platform_device *pdev) codec->set_bias_level = wm8994_set_bias_level; codec->dai = &wm8994_dai[0]; codec->num_dai = 3; - codec->reg_cache_size = WM8994_MAX_REGISTER; + codec->reg_cache_size = WM8994_REG_CACHE_SIZE; codec->reg_cache = &wm8994->reg_cache; codec->dev = &pdev->dev;

2 4

[alsa-devel] [PATCH 4/4 v2] ASoC: fsi: Add FSI2 device support
by Kuninori Morimoto 25 Mar '10

25 Mar '10

ARM-SHMOBILE series have FIFO-buffered serial interface 2 (FSI2) device which is advanced version of FSI. This patch add simple support for it. Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o based on new [PATCH 2/4 v2] ASoC: fsi: Add FIFO size calculate sound/soc/sh/Kconfig | 3 +- sound/soc/sh/fsi.c | 56 +++++++++++++++++++++++++++++++++++++++++-------- 2 files changed, 48 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index f07f6d8..a1d14bc 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,5 +1,5 @@ menu "SoC Audio support for SuperH" - depends on SUPERH + depends on SUPERH || ARCH_SHMOBILE config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" @@ -22,7 +22,6 @@ config SND_SOC_SH4_SSI config SND_SOC_SH4_FSI tristate "SH4 FSI support" - depends on CPU_SUBTYPE_SH7724 help This option enables FSI sound support diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index ae88865..f14bbb0 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -40,6 +40,10 @@ #define MUTE_ST 0x0028 #define REG_END MUTE_ST + +#define CPU_INT_ST 0x01F4 +#define CPU_IEMSK 0x01F8 +#define CPU_IMSK 0x01FC #define INT_ST 0x0200 #define IEMSK 0x0204 #define IMSK 0x0208 @@ -47,7 +51,7 @@ #define CLK_RST 0x0210 #define SOFT_RST 0x0214 #define FIFO_SZ 0x0218 -#define MREG_START INT_ST +#define MREG_START CPU_INT_ST #define MREG_END FIFO_SZ /* DO_FMT */ @@ -116,11 +120,18 @@ struct fsi_priv { int periods; }; +struct fsi_regs { + u32 int_st; + u32 iemsk; + u32 imsk; +}; + struct fsi_master { void __iomem *base; int irq; struct fsi_priv fsia; struct fsi_priv fsib; + struct fsi_regs *regs; struct sh_fsi_platform_info *info; spinlock_t lock; }; @@ -337,8 +348,8 @@ static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) u32 data = fsi_port_ab_io_bit(fsi, is_play); struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(master, IMSK, data, data); - fsi_master_mask_set(master, IEMSK, data, data); + fsi_master_mask_set(master, master->regs->imsk, data, data); + fsi_master_mask_set(master, master->regs->iemsk, data, data); } static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) @@ -346,18 +357,18 @@ static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) u32 data = fsi_port_ab_io_bit(fsi, is_play); struct fsi_master *master = fsi_get_master(fsi); - fsi_master_mask_set(master, IMSK, data, 0); - fsi_master_mask_set(master, IEMSK, data, 0); + fsi_master_mask_set(master, master->regs->imsk, data, 0); + fsi_master_mask_set(master, master->regs->iemsk, data, 0); } static u32 fsi_irq_get_status(struct fsi_master *master) { - return fsi_master_read(master, INT_ST); + return fsi_master_read(master, master->regs->int_st); } static void fsi_irq_clear_all_status(struct fsi_master *master) { - fsi_master_write(master, INT_ST, 0x0000000); + fsi_master_write(master, master->regs->int_st, 0x0000000); } static void fsi_irq_clear_status(struct fsi_priv *fsi) @@ -369,7 +380,7 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi) data |= fsi_port_ab_io_bit(fsi, 1); /* clear interrupt factor */ - fsi_master_mask_set(master, INT_ST, data, 0); + fsi_master_mask_set(master, master->regs->int_st, data, 0); } /************************************************************************ @@ -953,6 +964,7 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; + const struct platform_device_id *id_entry; struct resource *res; unsigned int irq; int ret; @@ -962,6 +974,12 @@ static int fsi_probe(struct platform_device *pdev) return -ENODEV; } + id_entry = pdev->id_entry; + if (!id_entry) { + dev_err(&pdev->dev, "unknown fsi device\n"); + return -ENODEV; + } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); if (!res || (int)irq <= 0) { @@ -990,6 +1008,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsib.base = master->base + 0x40; master->fsib.master = master; + master->regs = (struct fsi_regs *)id_entry->driver_data; spin_lock_init(&master->lock); pm_runtime_enable(&pdev->dev); @@ -1002,7 +1021,8 @@ static int fsi_probe(struct platform_device *pdev) fsi_soft_all_reset(master); - ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); + ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, + id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); goto exit_iounmap; @@ -1069,6 +1089,23 @@ static struct dev_pm_ops fsi_pm_ops = { .runtime_resume = fsi_runtime_nop, }; +static struct fsi_regs fsi_regs = { + .int_st = INT_ST, + .iemsk = IEMSK, + .imsk = IMSK, +}; + +static struct fsi_regs fsi2_regs = { + .int_st = CPU_INT_ST, + .iemsk = CPU_IEMSK, + .imsk = CPU_IMSK, +}; + +static struct platform_device_id fsi_id_table[] = { + { "sh_fsi", (kernel_ulong_t)&fsi_regs }, + { "sh_fsi2", (kernel_ulong_t)&fsi2_regs }, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", @@ -1076,6 +1113,7 @@ static struct platform_driver fsi_driver = { }, .probe = fsi_probe, .remove = fsi_remove, + .id_table = fsi_id_table, }; static int __init fsi_mobile_init(void) -- 1.6.3.3

1 0

[alsa-devel] [PATCH 0/4] ASoC: fsi: FSI2 device support
by Kuninori Morimoto 25 Mar '10

25 Mar '10

Dear Mark These patches are FSI2 support Kuninori Morimoto (4): ASoC: fsi: ensures process inside master lock ASoC: fsi: Add FIFO size calculate ASoC: fsi: IRQ related process had be united ASoC: fsi: Add FSI2 device support ARM SH-MOBILE series have FSI2/FMSI device which are advance version of FSI. Above patches add simple FSI2 support. These patches are tested on AP4 board Best regards -- Kuninori Morimoto

3 12

[alsa-devel] [PATCH 0/7] snd-asihpi playback improvement + others
by linux＠audioscience.com 24 Mar '10

24 Mar '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Important: alsa-firmware/asihpi should also be updated with contents of http://www.audioscience.com/internet/download/firmware/dspbins40313.zip asihpi.c + hpi6205.c fix choppy audio playback on busmastering adapters the rest are minor fixes or cosmetic Eliot Blennerhassett (7): Remove obsolete comment - const correct pointer parameters. - add a new stream state wait and function Reinit response size for every msg/response transaction. Minor fix const ptr Bump lib version due to added and removed APIs Use adapter properties for stream buffer constraints. Default to less logging, remove VPRINTK3 Log buffer info in decimal. Keep HPI buffer pointers in sync with ALSA after prewrite. Support mic control caching. Move an enum out of public api. pci/asihpi/asihpi.c | 102 +++++---- pci/asihpi/hpi.h | 513 +++++++++++++++++++++++---------------------- pci/asihpi/hpi6205.c | 31 ++-- pci/asihpi/hpi_internal.h | 33 ++- pci/asihpi/hpicmn.c | 10 + pci/asihpi/hpidebug.h | 6 +- pci/asihpi/hpifunc.c | 469 ++++++++++++++++++++++------------------- pci/asihpi/hpimsginit.c | 2 +- pci/asihpi/hpimsgx.c | 26 ++- pci/asihpi/hpioctl.c | 3 - 10 files changed, 641 insertions(+), 554 deletions(-)

3 11

Re: [alsa-devel] 2.6.34-rc1: regression: ^Z no longer stops sound
by Pavel Machek 24 Mar '10

24 Mar '10

On Sun 2010-03-14 17:20:13, Bill Davidsen wrote: > Pavel Machek wrote: >> 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 > That's not new, I have seen it back as far as 2.6.27.25 doing a suspend > in a VM (KVM, no libvirt). But it only happens a few times a year, so I > never chased it. I hate to report an occasional problem in an obsolete > kernel, and I can't reproduce it. > > Does it happen repeatably with the new kernel? It happened reliably so far, yes. -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

3 2

[alsa-devel] Would it be possible to Alias current versions??
by Wolf Halton 24 Mar '10

24 Mar '10

I am working on fixing a script that was supposed to update alsa drivers on ubuntu. I am trying to figure out how to get the script to wget the most recent versions of everything without having to rewrite the script every time. How difficult would it be to alias the most recent versions? for instance ftp://ftp.alsa-project.org/pub/driver/alsa-driver-current.tar.bz2 I could work up a script for you that would check the current stable versions and update the aliases. -Wolf Halton -- This Apt Has Super Cow Powers - http://sourcefreedom.com

2 1

[alsa-devel] [patch] wm8994: playback => capture
by Dan Carpenter 24 Mar '10

24 Mar '10

Sparse caught that initialize "playback" two times instead of initializing "capture". Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771..d10d651 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3401,7 +3401,7 @@ struct snd_soc_dai wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, - .playback = { + .capture = { .stream_name = "AIF3 Capture", .channels_min = 2, .channels_max = 2,

3 2

Re: [alsa-devel] cmipci and sound skipping ?
by Clemens Ladisch 24 Mar '10

24 Mar '10

Matija Nalis wrote: > On Tue, Mar 23, 2010 at 09:36:01AM +0100, Clemens Ladisch wrote: > > Are you okay with your email address (which one?) to be published with > > the patch: > > Reported-and-tested-by: Matija Nalis <...> > > Sure, mnalis-alsadev @ voyager.hr would be ok for e-mail. Applied. Thanks! Clemens

1 0

[alsa-devel] [PATCH] ALSA: aureon - Patch for suspend/resume for Terratec Aureon cards.
by Bernhard Urban 23 Mar '10

23 Mar '10

Add proper suspend/resume code for Terratec Aureon cards. Based on ice1724 suspend/resume work of Igor Chernyshev. Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4944 Tested on linux-2.6.32.9 Signed-off-by: Bernhard Urban <lewurm(a)gmail.com> --- sound/pci/ice1712/aureon.c | 90 +++++++++++++++++++++++++++++++------------ 1 files changed, 65 insertions(+), 25 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 110d16e..1772e1a 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -1971,11 +1971,10 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) return 0; } - /* - * initialize the chip + * reset the chip */ -static int __devinit aureon_init(struct snd_ice1712 *ice) +static int aureon_reset(struct snd_ice1712 *ice) { static const unsigned short wm_inits_aureon[] = { /* These come first to reduce init pop noise */ @@ -2062,30 +2061,10 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) 0x0605, /* slave, 24bit, MSB on second OSCLK, SDOUT for right channel when OLRCK is high */ (unsigned short)-1 }; - struct aureon_spec *spec; unsigned int tmp; const unsigned short *p; - int err, i; - - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (!spec) - return -ENOMEM; - ice->spec = spec; - - if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { - ice->num_total_dacs = 6; - ice->num_total_adcs = 2; - } else { - /* aureon 7.1 and prodigy 7.1 */ - ice->num_total_dacs = 8; - ice->num_total_adcs = 2; - } - - /* to remeber the register values of CS8415 */ - ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); - if (!ice->akm) - return -ENOMEM; - ice->akm_codecs = 1; + int err; + struct aureon_spec *spec = ice->spec; err = aureon_ac97_init(ice); if (err != 0) @@ -2133,6 +2112,62 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) /* initialize PCA9554 pin directions & set default input */ aureon_pca9554_write(ice, PCA9554_DIR, 0x00); aureon_pca9554_write(ice, PCA9554_OUT, 0x00); /* internal AUX */ + return 0; +} + +/* + * suspend/resume + */ +#ifdef CONFIG_PM +static int aureon_resume(struct snd_ice1712 *ice) +{ + struct aureon_spec *spec = ice->spec; + int err, i; + + err = aureon_reset(ice); + if (err != 0) + return err; + + /* workaround for poking volume with alsamixer after resume: + * just set stored volume again */ + for (i = 0; i < ice->num_total_dacs; i++) { + wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); + } + return 0; +} +#endif + +/* + * initialize the chip + */ +static int __devinit aureon_init(struct snd_ice1712 *ice) +{ + struct aureon_spec *spec; + int i, err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AUREON51_SKY) { + ice->num_total_dacs = 6; + ice->num_total_adcs = 2; + } else { + /* aureon 7.1 and prodigy 7.1 */ + ice->num_total_dacs = 8; + ice->num_total_adcs = 2; + } + + /* to remeber the register values of CS8415 */ + ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); + if (!ice->akm) + return -ENOMEM; + ice->akm_codecs = 1; + + err = aureon_reset(ice); + if (err != 0) + return err; spec->master[0] = WM_VOL_MUTE; spec->master[1] = WM_VOL_MUTE; @@ -2141,6 +2176,11 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) wm_set_vol(ice, i, spec->vol[i], spec->master[i % 2]); } +#ifdef CONFIG_PM + ice->pm_resume = aureon_resume; + ice->pm_suspend_enabled = 1; +#endif + return 0; } -- 1.7.0.1

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 23 Mar '10

23 Mar '10

The following changes since commit 74511020dde10252f8b8e648690d99dba721de14: Mark Brown (1): Merge branch 'for-2.6.34' 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 Kuninori Morimoto (3): ASoC: ak4642: Add pll select support ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops ASoC: ak4642: Add enhanced sampling rate Mark Brown (1): ASoC: Implement WM8994 DAI tristate support sound/soc/codecs/ak4642.c | 175 ++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8994.c | 38 ++++++++++ sound/soc/sh/fsi-ak4642.c | 14 ++++ 3 files changed, 192 insertions(+), 35 deletions(-)

2 1

[alsa-devel] [PATCH 0/3] ASoC: ak4642: sampling rate update
by Kuninori Morimoto 23 Mar '10

23 Mar '10

Dear Mark These patches are sampling rate update for ak4642 Kuninori Morimoto (3): ASoC: ak4642: Add pll select support ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops ASoC: ak4642: Add enhanced sampling rate It are tested on ms7724se/AP4 board. Best regards -- Kuninori Morimoto

3 7

[alsa-devel] [PATCH] ASoC: TWL4030: PM fix for output amplifiers
by Peter Ujfalusi 23 Mar '10

23 Mar '10

Gain controls on outputs affect the power consumption when the gain is set to non 0 value. To prevent amps beeing enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 72 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 56 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af..bf59b8a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -135,9 +135,11 @@ struct twl4030_priv { unsigned int sysclk; - /* Headset output state handling */ - unsigned int hsl_enabled; - unsigned int hsr_enabled; + /* Output (with associated amp) states */ + u8 hsl_enabled, hsr_enabled; + u8 earpiece_enabled; + u8 predrivel_enabled, predriver_enabled; + u8 carkitl_enabled, carkitr_enabled; }; /* @@ -173,12 +175,47 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + struct twl4030_priv *twl4030 = codec->private_data; + int write_to_reg = 0; + twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, - reg); - else - return 0; + if (likely(reg < TWL4030_REG_SW_SHADOW)) { + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; + } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + } + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) @@ -525,26 +562,26 @@ static int micpath_event(struct snd_soc_dapm_widget *w, * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the * amplifier associated with the output pin) - * On mute bypass the reg_cache and mute the volume - * On unmute: restore the register content + * On mute bypass the reg_cache and write 0 to the register + * On unmute: restore the register content from the reg_cache * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R */ #define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ struct snd_kcontrol *kcontrol, int event) \ { \ - u8 reg_val; \ + struct twl4030_priv *twl4030 = w->codec->private_data; \ \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ + twl4030->pin_name##_enabled = 1; \ twl4030_write(w->codec, reg, \ twl4030_read_reg_cache(w->codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ - reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ - reg_val & (~mask), \ - reg); \ + twl4030->pin_name##_enabled = 0; \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + 0, reg); \ break; \ } \ return 0; \ @@ -664,7 +701,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + /* Actually write to the register */ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain, + TWL4030_REG_HS_GAIN_SET); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Wait ramp delay time + 1, so the VMID can settle */ -- 1.7.0.2

3 13

[alsa-devel] cmipci and sound skipping ?
by Matija Nalis 23 Mar '10

23 Mar '10

My card is "Asonic C-Media8738, 4 channel" and mostly works with snd_cmipci driver - it is just that (even on completely unloaded system) the sound is (very annoyingly) skipping. The speaker-test application throws out "Write error: -32,Broken pipe" errors on every skip (usually every 5-30 times a wav is played). mpg321 is much worse (every few seconds it clips the sample) making the listening to music unedurable. I'm using Debian Lenny with kernel 2.6.26 (also tried 2.6.32, no change) I've upgraded to the latest snapshot of driver, lib and utils as recommended, the bug is still there. When driver is compiled with debug, on every skip it throws out lines in dmesg with info as: ALSA pcm_lib.c:316: BUG: pcmC0D0p:0, pos = 4294918144, buffer size = 16384, period size = 1024 Note that the pos is *always* the same (4294918144 = 0xffff4000 = -16384) which might be important. I've reported the problem in bug #4899 at https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4899 (alsa-info.sh and more is attached there) There I got pointer to contact the alsa-devel list with XRUN_Debug info. So here it is. When I enable it with "echo 101 > /proc/asound/card0/pcm0p/xrun_debug" I get the following output : [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816904, pos=1037/1024/16384, hwptr=230413/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2052/1024/16384, hwptr=230413/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231428/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2053/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2061/1024/16384, hwptr=231429/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816909, pos=2061/1024/16384, hwptr=231437/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816914, pos=3076/1024/16384, hwptr=231437/229376 [33802.573395] ALSA pcm_lib.c:249: hwptr log: pcmC0D0p:0: j=4302816914, pos=3076/1024/16384, hwptr=232452/229376 [33802.573395] ALSA pcm_lib.c:316: BUG: pcmC0D0p:0, pos = 4294918144, buffer size = 16384, period size = 1024 [33802.573395] ALSA pcm_lib.c:415: PCM: hw_ptr skipping! (pos=0, delta=13308, period=1024, jdelta=0/69/0, hw_ptr=232452/232452) [33806.867658] ALSA pcm_lib.c:316: BUG: pcmC0D0p:0, pos = 4294918144, buffer size = 16384, period size = 1024 [33806.867666] ALSA pcm_lib.c:415: PCM: hw_ptr skipping! (pos=0, delta=16380, period=1024, jdelta=0/85/0, hw_ptr=425988/425988) Also it seems that enabling xrun_debug kludges around the bug - the annoying skipping (and "Write error: -32,Broken pipe" errors in speaker-test) disappears or at least becomes almost inaudiable. Would some other xrun_debug setting be more useful to give better insight ? Anything else I can provide or test to help debug the problem ? Please CC: me as I'm not subscribed to the list. -- Opinions above are GNU-copylefted.

2 3

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 23 Mar '10

23 Mar '10

The following changes since commit fb91e0beb95e7e253a22fc87b39b5543073eae33: Stephen Rothwell (1): Add linux-next specific files for 20100322 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 Pull request against -next due to the merge up of v2.6.34-rc2 (log only, diffstat got confused by the use of -next and tried to include the entire contents of the branch not just the additions). Daniel Mack (1): ASoC: pxa-pcm-lib: initialize DMA channel to -1 Mark Brown (8): Merge commit 'v2.6.34-rc2' into for-2.6.34 ASoC: Remove BROKEN from i.MX audio after dependencies merged ASoC: Add a notifier for jack status changes ASoC: Allow force enabled pins to be disabled ASoC: Move WM8350 microphone detection bias managment out of driver ASoC: Allow disabling of WM835x jack detection ASoC: Allow WM8903 mic detect disable and don't force bias on Merge branch 'for-2.6.34' into for-2.6.35 Peter Ujfalusi (1): ASoC: TWL4030: PM fix for output amplifiers

2 1

[alsa-devel] [PATCH v2] ASoC: TWL4030: PM fix for output amplifiers
by Peter Ujfalusi 22 Mar '10

22 Mar '10

Gain controls on outputs affect the power consumption when the gain is set to non 0 value. Outputs with amps have one register to configure the routing and the gain: PREDL_CTL (0x25): bit 0: Voice enable bit 1: Audio L1 enable bit 2: Audio L2 enable bit 3: Audio R2 enable bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB) bit 0 - 3: is handled in DAPM domain (DAPM_MIXER) bit 4 - 5: has simple volume control If there is no audio activity (BIAS_STANDBY), and user changes the volume, than the output amplifier will be enabled. If the user changes the routing (but the codec remains in BIAS_STANDBY), than the cached gain value also be written to the register, which enables the amplifier. The existing workaround for this is to have virtual PGAs associated with the outputs, and whit DAPM PMD the gain on the output will be forced to 0 (off) by bypassing the regcache. This failed to disable the amplifiers in several scenario (as mentioned above). Also if the codec is in BIAS_ON state, and user modifies a volume control, which path is actually not enabled, than that amplifier will be enabled as well, but it will be not turned off, since there is no DAPM path, which would make mute it. To prevent amps being enabled, when they are not needed, introduce the following workaround: Track the state of each of this type of output. In twl4030_write only allow actual write, when the given output is enabled, otherwise only update the reg_cache. The PGA event handlers on power up will write the cached value to the chip (restoring gain, routing selection). On power down 0 is written to the register (disabling the amp, and also just in case clearing the routing). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/codecs/twl4030.c | 72 ++++++++++++++++++++++++++++++++++---------- 1 files changed, 56 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af..bf59b8a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -135,9 +135,11 @@ struct twl4030_priv { unsigned int sysclk; - /* Headset output state handling */ - unsigned int hsl_enabled; - unsigned int hsr_enabled; + /* Output (with associated amp) states */ + u8 hsl_enabled, hsr_enabled; + u8 earpiece_enabled; + u8 predrivel_enabled, predriver_enabled; + u8 carkitl_enabled, carkitr_enabled; }; /* @@ -173,12 +175,47 @@ static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, static int twl4030_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + struct twl4030_priv *twl4030 = codec->private_data; + int write_to_reg = 0; + twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, - reg); - else - return 0; + if (likely(reg < TWL4030_REG_SW_SHADOW)) { + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; + } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + } + return 0; } static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) @@ -525,26 +562,26 @@ static int micpath_event(struct snd_soc_dapm_widget *w, * Output PGA builder: * Handle the muting and unmuting of the given output (turning off the * amplifier associated with the output pin) - * On mute bypass the reg_cache and mute the volume - * On unmute: restore the register content + * On mute bypass the reg_cache and write 0 to the register + * On unmute: restore the register content from the reg_cache * Outputs handled in this way: Earpiece, PreDrivL/R, CarkitL/R */ #define TWL4030_OUTPUT_PGA(pin_name, reg, mask) \ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ struct snd_kcontrol *kcontrol, int event) \ { \ - u8 reg_val; \ + struct twl4030_priv *twl4030 = w->codec->private_data; \ \ switch (event) { \ case SND_SOC_DAPM_POST_PMU: \ + twl4030->pin_name##_enabled = 1; \ twl4030_write(w->codec, reg, \ twl4030_read_reg_cache(w->codec, reg)); \ break; \ case SND_SOC_DAPM_POST_PMD: \ - reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ - reg_val & (~mask), \ - reg); \ + twl4030->pin_name##_enabled = 0; \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + 0, reg); \ break; \ } \ return 0; \ @@ -664,7 +701,10 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) /* Headset ramp-up according to the TRM */ hs_pop |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain); + /* Actually write to the register */ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + hs_gain, + TWL4030_REG_HS_GAIN_SET); hs_pop |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop); /* Wait ramp delay time + 1, so the VMID can settle */ -- 1.7.0.2

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 22 Mar '10

22 Mar '10

The following changes since commit 220bf991b0366cc50a94feede3d7341fa5710ee4: Linus Torvalds (1): Linux 2.6.34-rc2 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 Plus a merge up of 2.6.34-rc2 which means that the ARM build issues are pretty much gone. Only the two new patches for ASoC are listed below. Daniel Mack (1): ASoC: pxa-pcm-lib: initialize DMA channel to -1 Mark Brown (1): ASoC: Remove BROKEN from i.MX audio after dependencies merged sound/arm/pxa2xx-pcm-lib.c | 1 + sound/soc/imx/Kconfig | 2 +-

2 1

[alsa-devel] Compiling older alsa version (1.0.15) on newer kernel (2.6.32.10) - errors ?
by Christian Steffen 22 Mar '10

22 Mar '10

Hi there, I want to compile some new alsa drivers for a MID device running ubuntu, because actual sound drivers on the device make some cracking noise all the time. The device (o2 Joggler) comes with an intel Poulsbo chipset (Sigmatel 9202). The original software on the device (a small linux with no big functionality) runs the alsa driver 1.0.15. That's the version, we have existig patches for. See those links: (Patches 0010 and 0014) http://software.intel.com/en-us/articles/patches-for-moblin-kernel-on-menlow / http://www.soulinfo.com/~hugang/tmp/ubuntu-hardy%2Btuxonice/debian/binary-cu stom.d/lpia/patchset/ It looks like these patches aren't still concluded in the newest alsa-drivers, therefore I'd like to use and compile version 1.0.15. But this seems not working with newer kernels. Here are the error messages I'm getting: make -C /root/Desktop/linux-2.6.32.10 SUBDIRS=/root/Downloads/alsa-driver-1.0.15 CPP="gcc -E" CC="gcc" modules make[1]: Betrete Verzeichnis '/root/Desktop/linux-2.6.32.10' scripts/Makefile.build:49: *** CFLAGS was changed in "/root/Downloads/alsa-driver-1.0.15/acore/Makefile". Fix it to use EXTRA_CFLAGS. Schluss. make[2]: *** [/root/Downloads/alsa-driver-1.0.15/acore] Fehler 2 make[1]: *** [_module_/root/Downloads/alsa-driver-1.0.15] Fehler 2 make[1]: Verlasse Verzeichnis '/root/Desktop/linux-2.6.32.10' make: *** [compile] Fehler 2 and still 1.0.16 which has corrected the EXTRA_CFLAGS, gives me this list of errors: make[1]: Betrete Verzeichnis '/root/Desktop/linux-2.6.32.10' CC [M] /root/Downloads/alsa-driver-1.0.16/acore/hwdep.o In file included from /root/Downloads/alsa-driver-1.0.16/acore/hwdep.c:1: /root/Downloads/alsa-driver-1.0.16/include/adriver.h:823: error: static declaration of 'jiffies_to_msecs' follows non-static declaration include/linux/jiffies.h:296: note: previous declaration of 'jiffies_to_msecs' was here /root/Downloads/alsa-driver-1.0.16/include/adriver.h:842: error: static declaration of 'msecs_to_jiffies' follows non-static declaration include/linux/jiffies.h:298: note: previous declaration of 'msecs_to_jiffies' was here In file included from /root/Downloads/alsa-driver-1.0.16/include/adriver.h:940, from /root/Downloads/alsa-driver-1.0.16/acore/hwdep.c:1: include/linux/pci.h:714: error: expected identifier or '(' before numeric constant In file included from /root/Desktop/linux-2.6.32.10/arch/x86/include/asm/pci.h:4, from include/linux/pci.h:1126, from /root/Downloads/alsa-driver-1.0.16/include/adriver.h:940, from /root/Downloads/alsa-driver-1.0.16/acore/hwdep.c:1: include/linux/mm.h:269: error: conflicting types for 'snd_compat_vmalloc_to_page' /root/Downloads/alsa-driver-1.0.16/include/adriver.h:744: note: previous declaration of 'snd_compat_vmalloc_to_page' was here In file included from /root/Downloads/alsa-driver-1.0.16/acore/hwdep.c:1: /root/Downloads/alsa-driver-1.0.16/include/adriver.h: In function 'snd_pci_orig_save_state': /root/Downloads/alsa-driver-1.0.16/include/adriver.h:1182: error: too many arguments to function 'pci_save_state' /root/Downloads/alsa-driver-1.0.16/include/adriver.h: In function 'snd_pci_orig_restore_state': /root/Downloads/alsa-driver-1.0.16/include/adriver.h:1186: error: too many arguments to function 'pci_restore_state' make[3]: *** [/root/Downloads/alsa-driver-1.0.16/acore/hwdep.o] Fehler 1 make[2]: *** [/root/Downloads/alsa-driver-1.0.16/acore] Fehler 2 make[1]: *** [_module_/root/Downloads/alsa-driver-1.0.16] Fehler 2 make[1]: Verlasse Verzeichnis '/root/Desktop/linux-2.6.32.10' make: *** [compile] Fehler 2 -------------------------- Is there any possibitliy to compile version 1.0.15 with a new kernel? Or the second way - is it difficult to apply those patches on a newer alsa driver? I tried to apply them "manually" on a new alsa driver, but without success. I thank you for any help on this! Best, Christian

1 0

[alsa-devel] [PATCH] ALSA: usb/caiaq: Add support for Traktor Kontrol X1
by Daniel Mack 22 Mar '10

22 Mar '10

This device does not have audio controllers and backlit buttons only. Input data is handled over a dedicated USB endpoint. All functions are supported by the driver now. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Dmitry Torokhov <dtor(a)mail.ru> --- sound/usb/Kconfig | 1 + sound/usb/caiaq/control.c | 99 ++++++++++++++++++++++------ sound/usb/caiaq/device.c | 8 ++- sound/usb/caiaq/device.h | 24 ++++--- sound/usb/caiaq/input.c | 162 +++++++++++++++++++++++++++++++++++++++++---- 5 files changed, 249 insertions(+), 45 deletions(-) diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index c570ae3..c4dcbad 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -65,6 +65,7 @@ config SND_USB_CAIAQ * Native Instruments Audio 8 DJ * Native Instruments Guitar Rig Session I/O * Native Instruments Guitar Rig mobile + * Native Instruments Traktor Kontrol X1 To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index 537102b..36ed703 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -35,33 +35,41 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; - unsigned int id = dev->chip.usb_id; + int maxval = 63; uinfo->count = 1; pos &= ~CNT_INTVAL; - if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) - && (pos == 0)) { - /* current input mode of A8DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 2; - return 0; - } + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): + if (pos == 0) { + /* current input mode of A8DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 2; + return 0; + } + break; - if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ) - && (pos == 0)) { - /* current input mode of A4DJ */ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + if (pos == 0) { + /* current input mode of A4DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + maxval = 127; + break; } if (is_intval) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 64; + uinfo->value.integer.max = maxval; } else { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->value.integer.min = 0; @@ -102,9 +110,10 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_audio *chip = snd_kcontrol_chip(kcontrol); struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + unsigned char cmd = EP1_CMD_WRITE_IO; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): { /* A4DJ has only one control */ /* do not expose hardware input mode 0 */ dev->control_state[0] = ucontrol->value.integer.value[0] + 1; @@ -113,10 +122,15 @@ static int control_put(struct snd_kcontrol *kcontrol, return 1; } + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + cmd = EP1_CMD_DIMM_LEDS; + break; + } + if (pos & CNT_INTVAL) { dev->control_state[pos & ~CNT_INTVAL] = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + snd_usb_caiaq_send_command(dev, cmd, dev->control_state, sizeof(dev->control_state)); } else { if (ucontrol->value.integer.value[0]) @@ -124,7 +138,7 @@ static int control_put(struct snd_kcontrol *kcontrol, else dev->control_state[pos / 8] &= ~(1 << (pos % 8)); - snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + snd_usb_caiaq_send_command(dev, cmd, dev->control_state, sizeof(dev->control_state)); } @@ -273,6 +287,43 @@ static struct caiaq_controller a4dj_controller[] = { { "Current input mode", 0 | CNT_INTVAL } }; +static struct caiaq_controller kontrolx1_controller[] = { + { "LED FX A: ON", 7 | CNT_INTVAL }, + { "LED FX A: 1", 6 | CNT_INTVAL }, + { "LED FX A: 2", 5 | CNT_INTVAL }, + { "LED FX A: 3", 4 | CNT_INTVAL }, + { "LED FX B: ON", 3 | CNT_INTVAL }, + { "LED FX B: 1", 2 | CNT_INTVAL }, + { "LED FX B: 2", 1 | CNT_INTVAL }, + { "LED FX B: 3", 0 | CNT_INTVAL }, + + { "LED Hotcue", 28 | CNT_INTVAL }, + { "LED Shift (white)", 29 | CNT_INTVAL }, + { "LED Shift (green)", 30 | CNT_INTVAL }, + + { "LED Deck A: FX1", 24 | CNT_INTVAL }, + { "LED Deck A: FX2", 25 | CNT_INTVAL }, + { "LED Deck A: IN", 17 | CNT_INTVAL }, + { "LED Deck A: OUT", 16 | CNT_INTVAL }, + { "LED Deck A: < BEAT", 19 | CNT_INTVAL }, + { "LED Deck A: BEAT >", 18 | CNT_INTVAL }, + { "LED Deck A: CUE/ABS", 21 | CNT_INTVAL }, + { "LED Deck A: CUP/REL", 20 | CNT_INTVAL }, + { "LED Deck A: PLAY", 23 | CNT_INTVAL }, + { "LED Deck A: SYNC", 22 | CNT_INTVAL }, + + { "LED Deck B: FX1", 26 | CNT_INTVAL }, + { "LED Deck B: FX2", 27 | CNT_INTVAL }, + { "LED Deck B: IN", 15 | CNT_INTVAL }, + { "LED Deck B: OUT", 14 | CNT_INTVAL }, + { "LED Deck B: < BEAT", 13 | CNT_INTVAL }, + { "LED Deck B: BEAT >", 12 | CNT_INTVAL }, + { "LED Deck B: CUE/ABS", 11 | CNT_INTVAL }, + { "LED Deck B: CUP/REL", 10 | CNT_INTVAL }, + { "LED Deck B: PLAY", 9 | CNT_INTVAL }, + { "LED Deck B: SYNC", 8 | CNT_INTVAL }, +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -321,10 +372,16 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(a8dj_controller, ARRAY_SIZE(a8dj_controller), dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): ret = add_controls(a4dj_controller, ARRAY_SIZE(a4dj_controller), dev); break; + + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + ret = add_controls(kontrolx1_controller, + ARRAY_SIZE(kontrolx1_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index a3f02dd..08ee254 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -46,7 +46,8 @@ MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," "{Native Instruments, Session I/O}," - "{Native Instruments, GuitarRig mobile}"); + "{Native Instruments, GuitarRig mobile}" + "{Native Instruments, Traktor Kontrol X1}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -127,6 +128,11 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_AUDIO2DJ }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_TRAKTORKONTROLX1 + }, { /* terminator */ } }; diff --git a/sound/usb/caiaq/device.h b/sound/usb/caiaq/device.h index 44e3edf..f1117ec 100644 --- a/sound/usb/caiaq/device.h +++ b/sound/usb/caiaq/device.h @@ -5,18 +5,20 @@ #define USB_VID_NATIVEINSTRUMENTS 0x17cc -#define USB_PID_RIGKONTROL2 0x1969 -#define USB_PID_RIGKONTROL3 0x1940 -#define USB_PID_KORECONTROLLER 0x4711 -#define USB_PID_KORECONTROLLER2 0x4712 -#define USB_PID_AK1 0x0815 -#define USB_PID_AUDIO2DJ 0x041c -#define USB_PID_AUDIO4DJ 0x0839 -#define USB_PID_AUDIO8DJ 0x1978 -#define USB_PID_SESSIONIO 0x1915 -#define USB_PID_GUITARRIGMOBILE 0x0d8d +#define USB_PID_RIGKONTROL2 0x1969 +#define USB_PID_RIGKONTROL3 0x1940 +#define USB_PID_KORECONTROLLER 0x4711 +#define USB_PID_KORECONTROLLER2 0x4712 +#define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO2DJ 0x041c +#define USB_PID_AUDIO4DJ 0x0839 +#define USB_PID_AUDIO8DJ 0x1978 +#define USB_PID_SESSIONIO 0x1915 +#define USB_PID_GUITARRIGMOBILE 0x0d8d +#define USB_PID_TRAKTORKONTROLX1 0x2305 #define EP1_BUFSIZE 64 +#define EP4_BUFSIZE 512 #define CAIAQ_USB_STR_LEN 0xff #define MAX_STREAMS 32 @@ -104,6 +106,8 @@ struct snd_usb_caiaqdev { struct input_dev *input_dev; char phys[64]; /* physical device path */ unsigned short keycode[64]; + struct urb *ep4_in_urb; + unsigned char ep4_in_buf[EP4_BUFSIZE]; #endif /* ALSA */ diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index a48d309..27ed0bc 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -19,6 +19,7 @@ #include <linux/init.h> #include <linux/usb.h> #include <linux/usb/input.h> +#include <sound/core.h> #include <sound/pcm.h> #include "device.h" @@ -65,6 +66,8 @@ static unsigned short keycode_kore[] = { KEY_BRL_DOT5 }; +#define KONTROLX1_INPUTS 40 + #define DEG90 (range / 2) #define DEG180 (range) #define DEG270 (DEG90 + DEG180) @@ -162,6 +165,17 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev, input_report_abs(input_dev, ABS_Z, (buf[4] << 8) | buf[5]); input_sync(input_dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + input_report_abs(input_dev, ABS_HAT0X, (buf[8] << 8) | buf[9]); + input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8) | buf[5]); + input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]); + input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8) | buf[3]); + input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]); + input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8) | buf[1]); + input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]); + input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8) | buf[7]); + input_sync(input_dev); + break; } } @@ -201,7 +215,7 @@ static void snd_caiaq_input_read_erp(struct snd_usb_caiaqdev *dev, } static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, - char *buf, unsigned int len) + unsigned char *buf, unsigned int len) { struct input_dev *input_dev = dev->input_dev; unsigned short *keycode = input_dev->keycode; @@ -218,15 +232,84 @@ static void snd_caiaq_input_read_io(struct snd_usb_caiaqdev *dev, input_report_key(input_dev, keycode[i], buf[i / 8] & (1 << (i % 8))); - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER) || - dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2)) + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): input_report_abs(dev->input_dev, ABS_MISC, 255 - buf[4]); + break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + /* rotary encoders */ + input_report_abs(dev->input_dev, ABS_X, buf[5] & 0xf); + input_report_abs(dev->input_dev, ABS_Y, buf[5] >> 4); + input_report_abs(dev->input_dev, ABS_Z, buf[6] & 0xf); + input_report_abs(dev->input_dev, ABS_MISC, buf[6] >> 4); + break; + } input_sync(input_dev); } +static void snd_usb_caiaq_ep4_reply_dispatch(struct urb *urb) +{ + struct snd_usb_caiaqdev *dev = urb->context; + unsigned char *buf = urb->transfer_buffer; + int ret; + + if (urb->status || !dev || urb != dev->ep4_in_urb) + return; + + if (urb->actual_length < 24) + goto requeue; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (buf[0] & 0x3) + snd_caiaq_input_read_io(dev, buf + 1, 7); + + if (buf[0] & 0x4) + snd_caiaq_input_read_analog(dev, buf + 8, 16); + + break; + } + +requeue: + dev->ep4_in_urb->actual_length = 0; + ret = usb_submit_urb(dev->ep4_in_urb, GFP_ATOMIC); + if (ret < 0) + log("unable to submit urb. OOM!?\n"); +} + +static int snd_usb_caiaq_input_open(struct input_dev *idev) +{ + struct snd_usb_caiaqdev *dev = input_get_drvdata(idev); + + if (!dev) + return -EINVAL; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + if (usb_submit_urb(dev->ep4_in_urb, GFP_KERNEL) != 0) + return -EIO; + break; + } + + return 0; +} + +static void snd_usb_caiaq_input_close(struct input_dev *idev) +{ + struct snd_usb_caiaqdev *dev = input_get_drvdata(idev); + + if (!dev) + return; + + switch (dev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + usb_kill_urb(dev->ep4_in_urb); + break; + } +} + void snd_usb_caiaq_input_dispatch(struct snd_usb_caiaqdev *dev, char *buf, unsigned int len) @@ -251,7 +334,7 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) { struct usb_device *usb_dev = dev->chip.dev; struct input_dev *input; - int i, ret; + int i, ret = 0; input = input_allocate_device(); if (!input) @@ -265,7 +348,9 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) usb_to_input_id(usb_dev, &input->id); input->dev.parent = &usb_dev->dev; - switch (dev->chip.usb_id) { + input_set_drvdata(input, dev); + + switch (dev->chip.usb_id) { case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL2): input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); input->absbit[0] = BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | @@ -326,25 +411,72 @@ int snd_usb_caiaq_input_init(struct snd_usb_caiaqdev *dev) input_set_abs_params(input, ABS_MISC, 0, 255, 0, 1); snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + input->evbit[0] = BIT_MASK(EV_KEY) | BIT_MASK(EV_ABS); + input->absbit[0] = BIT_MASK(ABS_HAT0X) | BIT_MASK(ABS_HAT0Y) | + BIT_MASK(ABS_HAT1X) | BIT_MASK(ABS_HAT1Y) | + BIT_MASK(ABS_HAT2X) | BIT_MASK(ABS_HAT2Y) | + BIT_MASK(ABS_HAT3X) | BIT_MASK(ABS_HAT3Y) | + BIT_MASK(ABS_X) | BIT_MASK(ABS_Y) | + BIT_MASK(ABS_Z); + input->absbit[BIT_WORD(ABS_MISC)] |= BIT_MASK(ABS_MISC); + BUILD_BUG_ON(sizeof(dev->keycode) < KONTROLX1_INPUTS); + for (i = 0; i < KONTROLX1_INPUTS; i++) + dev->keycode[i] = BTN_MISC + i; + input->keycodemax = KONTROLX1_INPUTS; + + /* analog potentiometers */ + input_set_abs_params(input, ABS_HAT0X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT0Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT1X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT1Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT2X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT2Y, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT3X, 0, 4096, 0, 10); + input_set_abs_params(input, ABS_HAT3Y, 0, 4096, 0, 10); + + /* rotary encoders */ + input_set_abs_params(input, ABS_X, 0, 0xf, 0, 1); + input_set_abs_params(input, ABS_Y, 0, 0xf, 0, 1); + input_set_abs_params(input, ABS_Z, 0, 0xf, 0, 1); + input_set_abs_params(input, ABS_MISC, 0, 0xf, 0, 1); + + dev->ep4_in_urb = usb_alloc_urb(0, GFP_KERNEL); + if (!dev->ep4_in_urb) { + ret = -ENOMEM; + goto exit_free_idev; + } + + usb_fill_bulk_urb(dev->ep4_in_urb, usb_dev, + usb_rcvbulkpipe(usb_dev, 0x4), + dev->ep4_in_buf, EP4_BUFSIZE, + snd_usb_caiaq_ep4_reply_dispatch, dev); + + snd_usb_caiaq_set_auto_msg(dev, 1, 10, 5); + + break; default: /* no input methods supported on this device */ - input_free_device(input); - return 0; + goto exit_free_idev; } + input->open = snd_usb_caiaq_input_open; + input->close = snd_usb_caiaq_input_close; input->keycode = dev->keycode; input->keycodesize = sizeof(unsigned short); for (i = 0; i < input->keycodemax; i++) __set_bit(dev->keycode[i], input->keybit); ret = input_register_device(input); - if (ret < 0) { - input_free_device(input); - return ret; - } + if (ret < 0) + goto exit_free_idev; dev->input_dev = input; return 0; + +exit_free_idev: + input_free_device(input); + return ret; } void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) @@ -352,6 +484,10 @@ void snd_usb_caiaq_input_free(struct snd_usb_caiaqdev *dev) if (!dev || !dev->input_dev) return; + usb_kill_urb(dev->ep4_in_urb); + usb_free_urb(dev->ep4_in_urb); + dev->ep4_in_urb = NULL; + input_unregister_device(dev->input_dev); dev->input_dev = NULL; } -- 1.7.0

2 2

[alsa-devel] Gentoo alsa output
by cdstealer 22 Mar '10

22 Mar '10

2 4

[alsa-devel] [PATCH 0/4] extend ad1938 codec/machine driver to ad193x supporting ad1936/7/8/9
by Barry Song 22 Mar '10

22 Mar '10

Hi Mark, The following patch series extends ad1938 codec/machine driver to support ad193x(including ad1936/ad1937/ad1938/ad1939). Some codecs use i2c bus and others use spi bus. patch 1: extend ad1938 codec driver to ad193x supporting ad1936/7/8/9 patch 2: change bf5xx-ad1938 machine driver to bf5xx-ad193x machine driver patch 3: soc-cache: add i2c read entry for 8_8 mode patch 4: soc-cache: let reg be AND'ed by 0xff instead of data buffer for 8_8 mode Thanks Barry

4 23

[alsa-devel] [PATCH] ALSA: ASoC: pxa-pcm-lib: initialize DMA channel to -1
by Daniel Mack 22 Mar '10

22 Mar '10

This fixes a warning ("pxa_free_dma: trying to free channel 0 which is already freed") when a device was opened but the hw_params() call failed. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Nicolas Pitre <nico(a)marvell.com> --- sound/arm/pxa2xx-pcm-lib.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 743ac6a..fd51fa8 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -205,6 +205,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (!rtd->dma_desc_array) goto err1; + rtd->dma_ch = -1; runtime->private_data = rtd; return 0; -- 1.6.3.3

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[alsa-devel] Question about dmix?
by cee1 22 Mar '10

22 Mar '10

Hi all, I know alsa support userspace software multi-playback by dmix plugin. How does it realize? I expected a daemon accepting multi-streams which then mixes these streams to a single one -- it seems not the case. Regards,

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[alsa-devel] [PATCH] ALSA: hda: Fix 0 dB offset for HP laptops using CX20551 (Waikiki)
by Daniel Chen 22 Mar '10

22 Mar '10

Hi, Attached is a patch to fix distortion at max PCM setting for HP laptops using CX20551 (Waikiki). Best, -Dan

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[alsa-devel] [RFC PATCH 09/19] pcmcia: dev_node removal (write-only drivers)
by Dominik Brodowski 22 Mar '10

22 Mar '10

dev_node_t was only used to transport some minor/major numbers from the PCMCIA device drivers to deprecated userspace helpers. However, only a few drivers made use of it, and the userspace helpers are deprecated anyways. Therefore, get rid of dev_node_t . As a first step, remove any usage of dev_node_t from drivers which only wrote to this typedef/struct, but did not make use of it. CC: linux-bluetooth(a)vger.kernel.org CC: Harald Welte <laforge(a)gnumonks.org> CC: linux-mtd(a)lists.infradead.org CC: linux-wireless(a)vger.kernel.org CC: netdev(a)vger.kernel.org CC: linux-serial(a)vger.kernel.org CC: alsa-devel(a)alsa-project.org Signed-off-by: Dominik Brodowski <linux(a)dominikbrodowski.net> --- drivers/bluetooth/bluecard_cs.c | 4 --- drivers/bluetooth/bt3c_cs.c | 4 --- drivers/bluetooth/btuart_cs.c | 4 --- drivers/bluetooth/dtl1_cs.c | 4 --- drivers/char/pcmcia/cm4040_cs.c | 5 --- drivers/mtd/maps/pcmciamtd.c | 3 -- drivers/net/wireless/atmel_cs.c | 42 ------------------------------- drivers/net/wireless/hostap/hostap_cs.c | 9 +----- drivers/parport/parport_cs.c | 5 --- drivers/scsi/pcmcia/aha152x_stub.c | 3 -- drivers/scsi/pcmcia/fdomain_stub.c | 3 -- drivers/scsi/pcmcia/nsp_cs.c | 3 -- drivers/scsi/pcmcia/nsp_cs.h | 1 - drivers/scsi/pcmcia/qlogic_stub.c | 3 -- drivers/scsi/pcmcia/sym53c500_cs.c | 3 -- drivers/serial/serial_cs.c | 9 ------ drivers/telephony/ixj_pcmcia.c | 3 -- sound/pcmcia/pdaudiocf/pdaudiocf.c | 1 - sound/pcmcia/pdaudiocf/pdaudiocf.h | 1 - sound/pcmcia/vx/vxpocket.c | 2 - sound/pcmcia/vx/vxpocket.h | 1 - 21 files changed, 2 insertions(+), 111 deletions(-) diff --git a/drivers/bluetooth/bluecard_cs.c b/drivers/bluetooth/bluecard_cs.c index a34653d..6f907eb 100644 --- a/drivers/bluetooth/bluecard_cs.c +++ b/drivers/bluetooth/bluecard_cs.c @@ -65,7 +65,6 @@ MODULE_LICENSE("GPL"); typedef struct bluecard_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct hci_dev *hdev; @@ -916,9 +915,6 @@ static int bluecard_config(struct pcmcia_device *link) if (bluecard_open(info) != 0) goto failed; - strcpy(info->node.dev_name, info->hdev->name); - link->dev_node = &info->node; - return 0; failed: diff --git a/drivers/bluetooth/bt3c_cs.c b/drivers/bluetooth/bt3c_cs.c index 1ad9694..21e05fd 100644 --- a/drivers/bluetooth/bt3c_cs.c +++ b/drivers/bluetooth/bt3c_cs.c @@ -72,7 +72,6 @@ MODULE_FIRMWARE("BT3CPCC.bin"); typedef struct bt3c_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct hci_dev *hdev; @@ -751,9 +750,6 @@ found_port: if (bt3c_open(info) != 0) goto failed; - strcpy(info->node.dev_name, info->hdev->name); - link->dev_node = &info->node; - return 0; failed: diff --git a/drivers/bluetooth/btuart_cs.c b/drivers/bluetooth/btuart_cs.c index 1073d66..4ed7288 100644 --- a/drivers/bluetooth/btuart_cs.c +++ b/drivers/bluetooth/btuart_cs.c @@ -67,7 +67,6 @@ MODULE_LICENSE("GPL"); typedef struct btuart_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct hci_dev *hdev; @@ -680,9 +679,6 @@ found_port: if (btuart_open(info) != 0) goto failed; - strcpy(info->node.dev_name, info->hdev->name); - link->dev_node = &info->node; - return 0; failed: diff --git a/drivers/bluetooth/dtl1_cs.c b/drivers/bluetooth/dtl1_cs.c index 3d72afd..ef044d5 100644 --- a/drivers/bluetooth/dtl1_cs.c +++ b/drivers/bluetooth/dtl1_cs.c @@ -67,7 +67,6 @@ MODULE_LICENSE("GPL"); typedef struct dtl1_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct hci_dev *hdev; @@ -629,9 +628,6 @@ static int dtl1_config(struct pcmcia_device *link) if (dtl1_open(info) != 0) goto failed; - strcpy(info->node.dev_name, info->hdev->name); - link->dev_node = &info->node; - return 0; failed: diff --git a/drivers/char/pcmcia/cm4040_cs.c b/drivers/char/pcmcia/cm4040_cs.c index a6a70e4..c0775c8 100644 --- a/drivers/char/pcmcia/cm4040_cs.c +++ b/drivers/char/pcmcia/cm4040_cs.c @@ -72,7 +72,6 @@ static struct class *cmx_class; struct reader_dev { struct pcmcia_device *p_dev; - dev_node_t node; wait_queue_head_t devq; wait_queue_head_t poll_wait; wait_queue_head_t read_wait; @@ -568,10 +567,6 @@ static int reader_config(struct pcmcia_device *link, int devno) } dev = link->priv; - sprintf(dev->node.dev_name, DEVICE_NAME "%d", devno); - dev->node.major = major; - dev->node.minor = devno; - dev->node.next = &dev->node; DEBUGP(2, dev, "device " DEVICE_NAME "%d at 0x%.4x-0x%.4x\n", devno, link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1); diff --git a/drivers/mtd/maps/pcmciamtd.c b/drivers/mtd/maps/pcmciamtd.c index 689d6a7..87b2b8f 100644 --- a/drivers/mtd/maps/pcmciamtd.c +++ b/drivers/mtd/maps/pcmciamtd.c @@ -52,7 +52,6 @@ static const int debug = 0; struct pcmciamtd_dev { struct pcmcia_device *p_dev; - dev_node_t node; /* device node */ caddr_t win_base; /* ioremapped address of PCMCIA window */ unsigned int win_size; /* size of window */ unsigned int offset; /* offset into card the window currently points at */ @@ -647,9 +646,7 @@ static int pcmciamtd_config(struct pcmcia_device *link) pcmciamtd_release(link); return -ENODEV; } - snprintf(dev->node.dev_name, sizeof(dev->node.dev_name), "mtd%d", mtd->index); info("mtd%d: %s", mtd->index, mtd->name); - link->dev_node = &dev->node; return 0; failed: diff --git a/drivers/net/wireless/atmel_cs.c b/drivers/net/wireless/atmel_cs.c index 759cdc4..c2746fc 100644 --- a/drivers/net/wireless/atmel_cs.c +++ b/drivers/net/wireless/atmel_cs.c @@ -85,41 +85,7 @@ static void atmel_release(struct pcmcia_device *link); static void atmel_detach(struct pcmcia_device *p_dev); -/* - You'll also need to prototype all the functions that will actually - be used to talk to your device. See 'pcmem_cs' for a good example - of a fully self-sufficient driver; the other drivers rely more or - less on other parts of the kernel. -*/ - -/* - A linked list of "instances" of the atmelnet device. Each actual - PCMCIA card corresponds to one device instance, and is described - by one struct pcmcia_device structure (defined in ds.h). - - You may not want to use a linked list for this -- for example, the - memory card driver uses an array of struct pcmcia_device pointers, where minor - device numbers are used to derive the corresponding array index. -*/ - -/* - A driver needs to provide a dev_node_t structure for each device - on a card. In some cases, there is only one device per card (for - example, ethernet cards, modems). In other cases, there may be - many actual or logical devices (SCSI adapters, memory cards with - multiple partitions). The dev_node_t structures need to be kept - in a linked list starting at the 'dev' field of a struct pcmcia_device - structure. We allocate them in the card's private data structure, - because they generally shouldn't be allocated dynamically. - - In this case, we also provide a flag to indicate if a device is - "stopped" due to a power management event, or card ejection. The - device IO routines can use a flag like this to throttle IO to a - card that is not ready to accept it. -*/ - typedef struct local_info_t { - dev_node_t node; struct net_device *eth_dev; } local_info_t; @@ -297,14 +263,6 @@ static int atmel_config(struct pcmcia_device *link) goto failed; - /* - At this point, the dev_node_t structure(s) need to be - initialized and arranged in a linked list at link->dev_node. - */ - strcpy(dev->node.dev_name, ((local_info_t*)link->priv)->eth_dev->name ); - dev->node.major = dev->node.minor = 0; - link->dev_node = &dev->node; - return 0; failed: diff --git a/drivers/net/wireless/hostap/hostap_cs.c b/drivers/net/wireless/hostap/hostap_cs.c index 7e5f4c0..56d7d90 100644 --- a/drivers/net/wireless/hostap/hostap_cs.c +++ b/drivers/net/wireless/hostap/hostap_cs.c @@ -38,7 +38,6 @@ MODULE_PARM_DESC(ignore_cis_vcc, "Ignore broken CIS VCC entry"); /* struct local_info::hw_priv */ struct hostap_cs_priv { - dev_node_t node; struct pcmcia_device *link; int sandisk_connectplus; }; @@ -624,8 +623,6 @@ static int prism2_config(struct pcmcia_device *link) local = iface->local; local->hw_priv = hw_priv; hw_priv->link = link; - strcpy(hw_priv->node.dev_name, dev->name); - link->dev_node = &hw_priv->node; ret = pcmcia_request_irq(link, prism2_interrupt); if (ret) @@ -664,11 +661,9 @@ static int prism2_config(struct pcmcia_device *link) sandisk_enable_wireless(dev); ret = prism2_hw_config(dev, 1); - if (!ret) { + if (!ret) ret = hostap_hw_ready(dev); - if (ret == 0 && local->ddev) - strcpy(hw_priv->node.dev_name, local->ddev->name); - } + return ret; failed: diff --git a/drivers/parport/parport_cs.c b/drivers/parport/parport_cs.c index 80c9052..fd8cfe9 100644 --- a/drivers/parport/parport_cs.c +++ b/drivers/parport/parport_cs.c @@ -75,7 +75,6 @@ INT_MODULE_PARM(epp_mode, 1); typedef struct parport_info_t { struct pcmcia_device *p_dev; int ndev; - dev_node_t node; struct parport *port; } parport_info_t; @@ -193,11 +192,7 @@ static int parport_config(struct pcmcia_device *link) if (epp_mode) p->modes |= PARPORT_MODE_TRISTATE | PARPORT_MODE_EPP; info->ndev = 1; - info->node.major = LP_MAJOR; - info->node.minor = p->number; info->port = p; - strcpy(info->node.dev_name, p->name); - link->dev_node = &info->node; return 0; diff --git a/drivers/scsi/pcmcia/aha152x_stub.c b/drivers/scsi/pcmcia/aha152x_stub.c index c1de4ba..9d70aef 100644 --- a/drivers/scsi/pcmcia/aha152x_stub.c +++ b/drivers/scsi/pcmcia/aha152x_stub.c @@ -80,7 +80,6 @@ MODULE_LICENSE("Dual MPL/GPL"); typedef struct scsi_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct Scsi_Host *host; } scsi_info_t; @@ -185,8 +184,6 @@ static int aha152x_config_cs(struct pcmcia_device *link) goto failed; } - sprintf(info->node.dev_name, "scsi%d", host->host_no); - link->dev_node = &info->node; info->host = host; return 0; diff --git a/drivers/scsi/pcmcia/fdomain_stub.c b/drivers/scsi/pcmcia/fdomain_stub.c index 8cb6d71..21b1411 100644 --- a/drivers/scsi/pcmcia/fdomain_stub.c +++ b/drivers/scsi/pcmcia/fdomain_stub.c @@ -63,7 +63,6 @@ MODULE_LICENSE("Dual MPL/GPL"); typedef struct scsi_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct Scsi_Host *host; } scsi_info_t; @@ -155,8 +154,6 @@ static int fdomain_config(struct pcmcia_device *link) goto failed; scsi_scan_host(host); - sprintf(info->node.dev_name, "scsi%d", host->host_no); - link->dev_node = &info->node; info->host = host; return 0; diff --git a/drivers/scsi/pcmcia/nsp_cs.c b/drivers/scsi/pcmcia/nsp_cs.c index 24c78ed..0f0e112 100644 --- a/drivers/scsi/pcmcia/nsp_cs.c +++ b/drivers/scsi/pcmcia/nsp_cs.c @@ -1754,8 +1754,6 @@ static int nsp_cs_config(struct pcmcia_device *link) scsi_scan_host(host); - snprintf(info->node.dev_name, sizeof(info->node.dev_name), "scsi%d", host->host_no); - link->dev_node = &info->node; info->host = host; /* Finally, report what we've done */ @@ -1813,7 +1811,6 @@ static void nsp_cs_release(struct pcmcia_device *link) if (info->host != NULL) { scsi_remove_host(info->host); } - link->dev_node = NULL; if (link->win) { if (data != NULL) { diff --git a/drivers/scsi/pcmcia/nsp_cs.h b/drivers/scsi/pcmcia/nsp_cs.h index 8c61a4f..d68c9f2 100644 --- a/drivers/scsi/pcmcia/nsp_cs.h +++ b/drivers/scsi/pcmcia/nsp_cs.h @@ -224,7 +224,6 @@ typedef struct scsi_info_t { struct pcmcia_device *p_dev; struct Scsi_Host *host; - dev_node_t node; int stop; } scsi_info_t; diff --git a/drivers/scsi/pcmcia/qlogic_stub.c b/drivers/scsi/pcmcia/qlogic_stub.c index c3a4428..f0fc6ba 100644 --- a/drivers/scsi/pcmcia/qlogic_stub.c +++ b/drivers/scsi/pcmcia/qlogic_stub.c @@ -82,7 +82,6 @@ static struct scsi_host_template qlogicfas_driver_template = { typedef struct scsi_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct Scsi_Host *host; unsigned short manf_id; } scsi_info_t; @@ -235,8 +234,6 @@ static int qlogic_config(struct pcmcia_device * link) goto failed; } - sprintf(info->node.dev_name, "scsi%d", host->host_no); - link->dev_node = &info->node; info->host = host; return 0; diff --git a/drivers/scsi/pcmcia/sym53c500_cs.c b/drivers/scsi/pcmcia/sym53c500_cs.c index 6eacf4c..a511641 100644 --- a/drivers/scsi/pcmcia/sym53c500_cs.c +++ b/drivers/scsi/pcmcia/sym53c500_cs.c @@ -191,7 +191,6 @@ struct scsi_info_t { struct pcmcia_device *p_dev; - dev_node_t node; struct Scsi_Host *host; unsigned short manf_id; }; @@ -792,8 +791,6 @@ SYM53C500_config(struct pcmcia_device *link) */ data->fast_pio = USE_FAST_PIO; - sprintf(info->node.dev_name, "scsi%d", host->host_no); - link->dev_node = &info->node; info->host = host; if (scsi_add_host(host, NULL)) diff --git a/drivers/serial/serial_cs.c b/drivers/serial/serial_cs.c index 7519d27..048bd89 100644 --- a/drivers/serial/serial_cs.c +++ b/drivers/serial/serial_cs.c @@ -89,7 +89,6 @@ struct serial_info { int manfid; int prodid; int c950ctrl; - dev_node_t node[4]; int line[4]; const struct serial_quirk *quirk; }; @@ -280,8 +279,6 @@ static void serial_remove(struct pcmcia_device *link) for (i = 0; i < info->ndev; i++) serial8250_unregister_port(info->line[i]); - info->p_dev->dev_node = NULL; - if (!info->slave) pcmcia_disable_device(link); } @@ -401,11 +398,6 @@ static int setup_serial(struct pcmcia_device *handle, struct serial_info * info, } info->line[info->ndev] = line; - sprintf(info->node[info->ndev].dev_name, "ttyS%d", line); - info->node[info->ndev].major = TTY_MAJOR; - info->node[info->ndev].minor = 0x40 + line; - if (info->ndev > 0) - info->node[info->ndev - 1].next = &info->node[info->ndev]; info->ndev++; return 0; @@ -702,7 +694,6 @@ static int serial_config(struct pcmcia_device * link) if (info->quirk->post(link)) goto failed; - link->dev_node = &info->node[0]; return 0; failed: diff --git a/drivers/telephony/ixj_pcmcia.c b/drivers/telephony/ixj_pcmcia.c index d442fd3..99cb224 100644 --- a/drivers/telephony/ixj_pcmcia.c +++ b/drivers/telephony/ixj_pcmcia.c @@ -22,7 +22,6 @@ typedef struct ixj_info_t { int ndev; - dev_node_t node; struct ixj *port; } ixj_info_t; @@ -155,8 +154,6 @@ static int ixj_config(struct pcmcia_device * link) j = ixj_pcmcia_probe(link->io.BasePort1, link->io.BasePort1 + 0x10); info->ndev = 1; - info->node.major = PHONE_MAJOR; - link->dev_node = &info->node; ixj_get_serial(link, j); return 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 60fddf3..df110df 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -234,7 +234,6 @@ static int pdacf_config(struct pcmcia_device *link) if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq) < 0) goto failed; - link->dev_node = &pdacf->node; return 0; failed: diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h index b060183..a0a7ec6 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.h +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h @@ -117,7 +117,6 @@ struct snd_pdacf { /* pcmcia stuff */ struct pcmcia_device *p_dev; - dev_node_t node; }; static inline void pdacf_reg_write(struct snd_pdacf *chip, unsigned char reg, unsigned short val) diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 4198395..5f21d10 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -210,7 +210,6 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq static int vxpocket_config(struct pcmcia_device *link) { struct vx_core *chip = link->priv; - struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; int ret; snd_printdd(KERN_DEBUG "vxpocket_config called\n"); @@ -244,7 +243,6 @@ static int vxpocket_config(struct pcmcia_device *link) if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq) < 0) goto failed; - link->dev_node = &vxp->node; return 0; failed: diff --git a/sound/pcmcia/vx/vxpocket.h b/sound/pcmcia/vx/vxpocket.h index 27ea002..ea4df16 100644 --- a/sound/pcmcia/vx/vxpocket.h +++ b/sound/pcmcia/vx/vxpocket.h @@ -43,7 +43,6 @@ struct snd_vxpocket { /* pcmcia stuff */ struct pcmcia_device *p_dev; - dev_node_t node; }; extern struct snd_vx_ops snd_vxpocket_ops; -- 1.6.3.3

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[alsa-devel] [RFC PATCH 07/19] pcmcia: re-work pcmcia_request_irq()
by Dominik Brodowski 22 Mar '10

22 Mar '10

Instead of the old pcmcia_request_irq() interface, drivers may now choose between: - calling request_irq/free_irq directly. Use the IRQ from *p_dev->irq. - use pcmcia_request_irq(p_dev, handler_t); the PCMCIA core will clean up automatically on calls to pcmcia_disable_device() or device ejection. - drivers still not capable of IRQF_SHARED (or not telling us so) may use the deprecated pcmcia_request_exclusive_irq() for the time being; they might receive a shared IRQ nonetheless. CC: linux-bluetooth(a)vger.kernel.org CC: netdev(a)vger.kernel.org CC: linux-wireless(a)vger.kernel.org CC: linux-serial(a)vger.kernel.org CC: alsa-devel(a)alsa-project.org CC: linux-usb(a)vger.kernel.org CC: linux-ide(a)vger.kernel.org Signed-off-by: Dominik Brodowski <linux(a)dominikbrodowski.net> --- Documentation/pcmcia/driver-changes.txt | 10 ++ drivers/ata/pata_pcmcia.c | 6 +- drivers/bluetooth/bluecard_cs.c | 7 +- drivers/bluetooth/bt3c_cs.c | 7 +- drivers/bluetooth/btuart_cs.c | 7 +- drivers/bluetooth/dtl1_cs.c | 7 +- drivers/char/pcmcia/ipwireless/main.c | 11 +-- drivers/char/pcmcia/synclink_cs.c | 12 +-- drivers/ide/ide-cs.c | 8 +- drivers/isdn/hardware/avm/avm_cs.c | 17 +-- drivers/isdn/hisax/avma1_cs.c | 12 +-- drivers/isdn/hisax/elsa_cs.c | 13 +-- drivers/isdn/hisax/sedlbauer_cs.c | 23 +--- drivers/isdn/hisax/teles_cs.c | 13 +-- drivers/net/pcmcia/3c574_cs.c | 6 +- drivers/net/pcmcia/3c589_cs.c | 7 +- drivers/net/pcmcia/axnet_cs.c | 13 +-- drivers/net/pcmcia/com20020_cs.c | 8 +- drivers/net/pcmcia/fmvj18x_cs.c | 10 +-- drivers/net/pcmcia/ibmtr_cs.c | 8 +- drivers/net/pcmcia/nmclan_cs.c | 6 +- drivers/net/pcmcia/pcnet_cs.c | 8 +- drivers/net/pcmcia/smc91c92_cs.c | 8 +- drivers/net/pcmcia/xirc2ps_cs.c | 7 +- drivers/net/wireless/airo_cs.c | 25 +--- drivers/net/wireless/atmel_cs.c | 28 +---- drivers/net/wireless/b43/pcmcia.c | 5 +- drivers/net/wireless/hostap/hostap_cs.c | 29 +---- drivers/net/wireless/libertas/if_cs.c | 21 +--- drivers/net/wireless/orinoco/orinoco_cs.c | 13 +-- drivers/net/wireless/orinoco/spectrum_cs.c | 13 +-- drivers/net/wireless/ray_cs.c | 8 +- drivers/net/wireless/wl3501_cs.c | 8 +- drivers/parport/parport_cs.c | 8 +- drivers/pcmcia/ds.c | 3 +- drivers/pcmcia/pcmcia_resource.c | 137 +++++++++------------- drivers/scsi/pcmcia/aha152x_stub.c | 6 +- drivers/scsi/pcmcia/fdomain_stub.c | 6 +- drivers/scsi/pcmcia/nsp_cs.c | 20 +--- drivers/scsi/pcmcia/qlogic_stub.c | 10 +- drivers/scsi/pcmcia/sym53c500_cs.c | 6 +- drivers/serial/serial_cs.c | 27 ++--- drivers/ssb/main.c | 2 +- drivers/staging/comedi/drivers/cb_das16_cs.c | 26 +--- drivers/staging/comedi/drivers/das08_cs.c | 16 +-- drivers/staging/comedi/drivers/ni_daq_700.c | 23 +--- drivers/staging/comedi/drivers/ni_daq_dio24.c | 23 +--- drivers/staging/comedi/drivers/ni_labpc_cs.c | 24 +--- drivers/staging/comedi/drivers/ni_mio_cs.c | 9 +- drivers/staging/comedi/drivers/quatech_daqp_cs.c | 22 +--- drivers/staging/netwave/netwave_cs.c | 8 +- drivers/staging/wavelan/wavelan_cs.c | 15 +-- drivers/staging/wlags49_h2/wl_cs.c | 9 +- drivers/usb/host/sl811_cs.c | 18 +-- include/pcmcia/cs.h | 7 - include/pcmcia/ds.h | 11 ++- sound/pcmcia/pdaudiocf/pdaudiocf.c | 9 +- sound/pcmcia/vx/vxpocket.c | 8 +- 58 files changed, 244 insertions(+), 593 deletions(-) diff --git a/Documentation/pcmcia/driver-changes.txt b/Documentation/pcmcia/driver-changes.txt index 446f43b..3268a9a 100644 --- a/Documentation/pcmcia/driver-changes.txt +++ b/Documentation/pcmcia/driver-changes.txt @@ -1,4 +1,14 @@ This file details changes in 2.6 which affect PCMCIA card driver authors: +* New IRQ request rules (as of 2.6.35) + Instead of the old pcmcia_request_irq() interface, drivers may now + choose between: + - calling request_irq/free_irq directly. Use the IRQ from *p_dev->irq. + - use pcmcia_request_irq(p_dev, handler_t); the PCMCIA core will + clean up automatically on calls to pcmcia_disable_device() or + device ejection. + - drivers still not capable of IRQF_SHARED (or not telling us so) may + use the deprecated pcmcia_request_exclusive_irq() for the time + being; they might receive a shared IRQ nonetheless. * no cs_error / CS_CHECK / CONFIG_PCMCIA_DEBUG (as of 2.6.33) Instead of the cs_error() callback or the CS_CHECK() macro, please use diff --git a/drivers/ata/pata_pcmcia.c b/drivers/ata/pata_pcmcia.c index 147de2f..1336216 100644 --- a/drivers/ata/pata_pcmcia.c +++ b/drivers/ata/pata_pcmcia.c @@ -267,7 +267,6 @@ static int pcmcia_init_one(struct pcmcia_device *pdev) pdev->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; pdev->io.Attributes2 = IO_DATA_PATH_WIDTH_8; pdev->io.IOAddrLines = 3; - pdev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; pdev->conf.Attributes = CONF_ENABLE_IRQ; pdev->conf.IntType = INT_MEMORY_AND_IO; @@ -292,8 +291,7 @@ static int pcmcia_init_one(struct pcmcia_device *pdev) } io_base = pdev->io.BasePort1; ctl_base = stk->ctl_base; - ret = pcmcia_request_irq(pdev, &pdev->irq); - if (ret) + if (!pdev->irq) goto failed; ret = pcmcia_request_configuration(pdev, &pdev->conf); @@ -343,7 +341,7 @@ static int pcmcia_init_one(struct pcmcia_device *pdev) } /* activate */ - ret = ata_host_activate(host, pdev->irq.AssignedIRQ, ata_sff_interrupt, + ret = ata_host_activate(host, pdev->irq, ata_sff_interrupt, IRQF_SHARED, &pcmcia_sht); if (ret) goto failed; diff --git a/drivers/bluetooth/bluecard_cs.c b/drivers/bluetooth/bluecard_cs.c index d9bf87c..a34653d 100644 --- a/drivers/bluetooth/bluecard_cs.c +++ b/drivers/bluetooth/bluecard_cs.c @@ -869,9 +869,6 @@ static int bluecard_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts1 = 8; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - - link->irq.Handler = bluecard_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -908,9 +905,9 @@ static int bluecard_config(struct pcmcia_device *link) if (i != 0) goto failed; - i = pcmcia_request_irq(link, &link->irq); + i = pcmcia_request_irq(link, bluecard_interrupt); if (i != 0) - link->irq.AssignedIRQ = 0; + goto failed; i = pcmcia_request_configuration(link, &link->conf); if (i != 0) diff --git a/drivers/bluetooth/bt3c_cs.c b/drivers/bluetooth/bt3c_cs.c index 027cb8b..1ad9694 100644 --- a/drivers/bluetooth/bt3c_cs.c +++ b/drivers/bluetooth/bt3c_cs.c @@ -661,9 +661,6 @@ static int bt3c_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts1 = 8; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - - link->irq.Handler = bt3c_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -743,9 +740,9 @@ static int bt3c_config(struct pcmcia_device *link) goto failed; found_port: - i = pcmcia_request_irq(link, &link->irq); + i = pcmcia_request_irq(link, &bt3c_interrupt); if (i != 0) - link->irq.AssignedIRQ = 0; + goto failed; i = pcmcia_request_configuration(link, &link->conf); if (i != 0) diff --git a/drivers/bluetooth/btuart_cs.c b/drivers/bluetooth/btuart_cs.c index 60c0953..1073d66 100644 --- a/drivers/bluetooth/btuart_cs.c +++ b/drivers/bluetooth/btuart_cs.c @@ -590,9 +590,6 @@ static int btuart_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts1 = 8; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - - link->irq.Handler = btuart_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -672,9 +669,9 @@ static int btuart_config(struct pcmcia_device *link) goto failed; found_port: - i = pcmcia_request_irq(link, &link->irq); + i = pcmcia_request_irq(link, btuart_interrupt); if (i != 0) - link->irq.AssignedIRQ = 0; + goto failed; i = pcmcia_request_configuration(link, &link->conf); if (i != 0) diff --git a/drivers/bluetooth/dtl1_cs.c b/drivers/bluetooth/dtl1_cs.c index 1778831..3d72afd 100644 --- a/drivers/bluetooth/dtl1_cs.c +++ b/drivers/bluetooth/dtl1_cs.c @@ -575,9 +575,6 @@ static int dtl1_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts1 = 8; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - - link->irq.Handler = dtl1_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -621,9 +618,9 @@ static int dtl1_config(struct pcmcia_device *link) if (pcmcia_loop_config(link, dtl1_confcheck, NULL) < 0) goto failed; - i = pcmcia_request_irq(link, &link->irq); + i = pcmcia_request_irq(link, dtl1_interrupt); if (i != 0) - link->irq.AssignedIRQ = 0; + goto failed; i = pcmcia_request_configuration(link, &link->conf); if (i != 0) diff --git a/drivers/char/pcmcia/ipwireless/main.c b/drivers/char/pcmcia/ipwireless/main.c index dff24da..2237890 100644 --- a/drivers/char/pcmcia/ipwireless/main.c +++ b/drivers/char/pcmcia/ipwireless/main.c @@ -195,9 +195,6 @@ static int config_ipwireless(struct ipw_dev *ipw) link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = ipwireless_interrupt; - INIT_WORK(&ipw->work_reboot, signalled_reboot_work); ipwireless_init_hardware_v1(ipw->hardware, link->io.BasePort1, @@ -205,8 +202,7 @@ static int config_ipwireless(struct ipw_dev *ipw) ipw->is_v2_card, signalled_reboot_callback, ipw); - ret = pcmcia_request_irq(link, &link->irq); - + ret = pcmcia_request_irq(link, ipwireless_interrupt); if (ret != 0) goto exit; @@ -217,7 +213,7 @@ static int config_ipwireless(struct ipw_dev *ipw) (unsigned int) link->io.BasePort1, (unsigned int) (link->io.BasePort1 + link->io.NumPorts1 - 1), - (unsigned int) link->irq.AssignedIRQ); + (unsigned int) link->irq); if (ipw->attr_memory && ipw->common_memory) printk(KERN_INFO IPWIRELESS_PCCARD_NAME ": attr memory 0x%08lx-0x%08lx, common memory 0x%08lx-0x%08lx\n", @@ -271,8 +267,6 @@ exit: static void release_ipwireless(struct ipw_dev *ipw) { - pcmcia_disable_device(ipw->link); - if (ipw->common_memory) { release_mem_region(ipw->request_common_memory.Base, ipw->request_common_memory.Size); @@ -288,7 +282,6 @@ static void release_ipwireless(struct ipw_dev *ipw) if (ipw->attr_memory) pcmcia_release_window(ipw->link, ipw->handle_attr_memory); - /* Break the link with Card Services */ pcmcia_disable_device(ipw->link); } diff --git a/drivers/char/pcmcia/synclink_cs.c b/drivers/char/pcmcia/synclink_cs.c index c31a0d9..027690b 100644 --- a/drivers/char/pcmcia/synclink_cs.c +++ b/drivers/char/pcmcia/synclink_cs.c @@ -552,10 +552,6 @@ static int mgslpc_probe(struct pcmcia_device *link) /* Initialize the struct pcmcia_device structure */ - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - link->conf.Attributes = 0; link->conf.IntType = INT_MEMORY_AND_IO; @@ -608,9 +604,7 @@ static int mgslpc_config(struct pcmcia_device *link) link->conf.ConfigIndex = 8; link->conf.Present = PRESENT_OPTION; - link->irq.Handler = mgslpc_isr; - - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, mgslpc_isr); if (ret) goto failed; ret = pcmcia_request_configuration(link, &link->conf); @@ -618,7 +612,7 @@ static int mgslpc_config(struct pcmcia_device *link) goto failed; info->io_base = link->io.BasePort1; - info->irq_level = link->irq.AssignedIRQ; + info->irq_level = link->irq; /* add to linked list of devices */ sprintf(info->node.dev_name, "mgslpc0"); @@ -628,7 +622,7 @@ static int mgslpc_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x:", info->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); diff --git a/drivers/ide/ide-cs.c b/drivers/ide/ide-cs.c index ab87e4f..afd1ab7 100644 --- a/drivers/ide/ide-cs.c +++ b/drivers/ide/ide-cs.c @@ -102,7 +102,6 @@ static int ide_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; link->io.IOAddrLines = 3; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -285,8 +284,7 @@ static int ide_config(struct pcmcia_device *link) io_base = link->io.BasePort1; ctl_base = stk->ctl_base; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); if (ret) @@ -299,11 +297,11 @@ static int ide_config(struct pcmcia_device *link) if (is_kme) outb(0x81, ctl_base+1); - host = idecs_register(io_base, ctl_base, link->irq.AssignedIRQ, link); + host = idecs_register(io_base, ctl_base, link->irq, link); if (host == NULL && link->io.NumPorts1 == 0x20) { outb(0x02, ctl_base + 0x10); host = idecs_register(io_base + 0x10, ctl_base + 0x10, - link->irq.AssignedIRQ, link); + link->irq, link); } if (host == NULL) diff --git a/drivers/isdn/hardware/avm/avm_cs.c b/drivers/isdn/hardware/avm/avm_cs.c index 94b796d..0d485f6 100644 --- a/drivers/isdn/hardware/avm/avm_cs.c +++ b/drivers/isdn/hardware/avm/avm_cs.c @@ -107,9 +107,6 @@ static int avmcs_probe(struct pcmcia_device *p_dev) p_dev->io.Attributes1 = IO_DATA_PATH_WIDTH_8; p_dev->io.NumPorts2 = 0; - /* Interrupt setup */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - /* General socket configuration */ p_dev->conf.Attributes = CONF_ENABLE_IRQ; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -172,7 +169,7 @@ static int avmcs_configcheck(struct pcmcia_device *p_dev, static int avmcs_config(struct pcmcia_device *link) { local_info_t *dev; - int i; + int i = -1; char devname[128]; int cardtype; int (*addcard)(unsigned int port, unsigned irq); @@ -190,11 +187,7 @@ static int avmcs_config(struct pcmcia_device *link) return -ENODEV; do { - /* - * allocate an interrupt line - */ - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) { + if (!link->irq) { /* undo */ pcmcia_disable_device(link); break; @@ -249,9 +242,9 @@ static int avmcs_config(struct pcmcia_device *link) default: case AVM_CARDTYPE_B1: addcard = b1pcmcia_addcard_b1; break; } - if ((i = (*addcard)(link->io.BasePort1, link->irq.AssignedIRQ)) < 0) { + if ((i = (*addcard)(link->io.BasePort1, link->irq)) < 0) { printk(KERN_ERR "avm_cs: failed to add AVM-%s-Controller at i/o %#x, irq %d\n", - dev->node.dev_name, link->io.BasePort1, link->irq.AssignedIRQ); + dev->node.dev_name, link->io.BasePort1, link->irq); avmcs_release(link); return -ENODEV; } @@ -270,7 +263,7 @@ static int avmcs_config(struct pcmcia_device *link) static void avmcs_release(struct pcmcia_device *link) { - b1pcmcia_delcard(link->io.BasePort1, link->irq.AssignedIRQ); + b1pcmcia_delcard(link->io.BasePort1, link->irq); pcmcia_disable_device(link); } /* avmcs_release */ diff --git a/drivers/isdn/hisax/avma1_cs.c b/drivers/isdn/hisax/avma1_cs.c index e5deb15..96d0b5d 100644 --- a/drivers/isdn/hisax/avma1_cs.c +++ b/drivers/isdn/hisax/avma1_cs.c @@ -119,9 +119,6 @@ static int avma1cs_probe(struct pcmcia_device *p_dev) p_dev->io.Attributes2 = IO_DATA_PATH_WIDTH_16; p_dev->io.IOAddrLines = 5; - /* Interrupt setup */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - /* General socket configuration */ p_dev->conf.Attributes = CONF_ENABLE_IRQ; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -177,7 +174,7 @@ static int avma1cs_configcheck(struct pcmcia_device *p_dev, static int avma1cs_config(struct pcmcia_device *link) { local_info_t *dev; - int i; + int i = -1; char devname[128]; IsdnCard_t icard; int busy = 0; @@ -197,8 +194,7 @@ static int avma1cs_config(struct pcmcia_device *link) /* * allocate an interrupt line */ - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) { + if (!link->irq) { /* undo */ pcmcia_disable_device(link); break; @@ -230,9 +226,9 @@ static int avma1cs_config(struct pcmcia_device *link) } printk(KERN_NOTICE "avma1_cs: checking at i/o %#x, irq %d\n", - link->io.BasePort1, link->irq.AssignedIRQ); + link->io.BasePort1, link->irq); - icard.para[0] = link->irq.AssignedIRQ; + icard.para[0] = link->irq; icard.para[1] = link->io.BasePort1; icard.protocol = isdnprot; icard.typ = ISDN_CTYPE_A1_PCMCIA; diff --git a/drivers/isdn/hisax/elsa_cs.c b/drivers/isdn/hisax/elsa_cs.c index c9a30b1..242c434 100644 --- a/drivers/isdn/hisax/elsa_cs.c +++ b/drivers/isdn/hisax/elsa_cs.c @@ -136,10 +136,6 @@ static int elsa_cs_probe(struct pcmcia_device *link) local->cardnr = -1; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -223,11 +219,8 @@ static int elsa_cs_config(struct pcmcia_device *link) if (i != 0) goto failed; - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) { - link->irq.AssignedIRQ = 0; + if (!link->irq) goto failed; - } i = pcmcia_request_configuration(link, &link->conf); if (i != 0) @@ -244,7 +237,7 @@ static int elsa_cs_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x: ", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); @@ -253,7 +246,7 @@ static int elsa_cs_config(struct pcmcia_device *link) link->io.BasePort2+link->io.NumPorts2-1); printk("\n"); - icard.para[0] = link->irq.AssignedIRQ; + icard.para[0] = link->irq; icard.para[1] = link->io.BasePort1; icard.protocol = protocol; icard.typ = ISDN_CTYPE_ELSA_PCMCIA; diff --git a/drivers/isdn/hisax/sedlbauer_cs.c b/drivers/isdn/hisax/sedlbauer_cs.c index 7836ec3..967371b 100644 --- a/drivers/isdn/hisax/sedlbauer_cs.c +++ b/drivers/isdn/hisax/sedlbauer_cs.c @@ -143,10 +143,6 @@ static int sedlbauer_probe(struct pcmcia_device *link) local->p_dev = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -227,9 +223,7 @@ static int sedlbauer_config_check(struct pcmcia_device *p_dev, else if (dflt->vpp1.present & (1<<CISTPL_POWER_VNOM)) p_dev->conf.Vpp = dflt->vpp1.param[CISTPL_POWER_VNOM]/10000; - /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -313,17 +307,6 @@ static int sedlbauer_config(struct pcmcia_device *link) goto failed; /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } - - /* This actually configures the PCMCIA socket -- setting up the I/O windows and the interrupt mapping, and putting the card and host interface into "Memory and IO" mode. @@ -346,7 +329,7 @@ static int sedlbauer_config(struct pcmcia_device *link) if (link->conf.Vpp) printk(", Vpp %d.%d", link->conf.Vpp/10, link->conf.Vpp%10); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); @@ -358,7 +341,7 @@ static int sedlbauer_config(struct pcmcia_device *link) req->Base+req->Size-1); printk("\n"); - icard.para[0] = link->irq.AssignedIRQ; + icard.para[0] = link->irq; icard.para[1] = link->io.BasePort1; icard.protocol = protocol; icard.typ = ISDN_CTYPE_SEDLBAUER_PCMCIA; diff --git a/drivers/isdn/hisax/teles_cs.c b/drivers/isdn/hisax/teles_cs.c index b0c5976..75e075c 100644 --- a/drivers/isdn/hisax/teles_cs.c +++ b/drivers/isdn/hisax/teles_cs.c @@ -126,10 +126,6 @@ static int teles_probe(struct pcmcia_device *link) local->p_dev = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -213,11 +209,8 @@ static int teles_cs_config(struct pcmcia_device *link) if (i != 0) goto cs_failed; - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) { - link->irq.AssignedIRQ = 0; + if (!link->irq) goto cs_failed; - } i = pcmcia_request_configuration(link, &link->conf); if (i != 0) @@ -234,7 +227,7 @@ static int teles_cs_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x:", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); @@ -243,7 +236,7 @@ static int teles_cs_config(struct pcmcia_device *link) link->io.BasePort2+link->io.NumPorts2-1); printk("\n"); - icard.para[0] = link->irq.AssignedIRQ; + icard.para[0] = link->irq; icard.para[1] = link->io.BasePort1; icard.protocol = protocol; icard.typ = ISDN_CTYPE_TELESPCMCIA; diff --git a/drivers/net/pcmcia/3c574_cs.c b/drivers/net/pcmcia/3c574_cs.c index 61e33cf..839a29c 100644 --- a/drivers/net/pcmcia/3c574_cs.c +++ b/drivers/net/pcmcia/3c574_cs.c @@ -282,8 +282,6 @@ static int tc574_probe(struct pcmcia_device *link) spin_lock_init(&lp->window_lock); link->io.NumPorts1 = 32; link->io.Attributes1 = IO_DATA_PATH_WIDTH_16; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = &el3_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; @@ -352,7 +350,7 @@ static int tc574_config(struct pcmcia_device *link) if (i != 0) goto failed; - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, el3_interrupt); if (ret) goto failed; @@ -360,7 +358,7 @@ static int tc574_config(struct pcmcia_device *link) if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; ioaddr = dev->base_addr; diff --git a/drivers/net/pcmcia/3c589_cs.c b/drivers/net/pcmcia/3c589_cs.c index 091e0b0..405d772 100644 --- a/drivers/net/pcmcia/3c589_cs.c +++ b/drivers/net/pcmcia/3c589_cs.c @@ -194,8 +194,7 @@ static int tc589_probe(struct pcmcia_device *link) spin_lock_init(&lp->lock); link->io.NumPorts1 = 16; link->io.Attributes1 = IO_DATA_PATH_WIDTH_16; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = &el3_interrupt; + link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; @@ -271,7 +270,7 @@ static int tc589_config(struct pcmcia_device *link) if (i != 0) goto failed; - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, el3_interrupt); if (ret) goto failed; @@ -279,7 +278,7 @@ static int tc589_config(struct pcmcia_device *link) if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; ioaddr = dev->base_addr; EL3WINDOW(0); diff --git a/drivers/net/pcmcia/axnet_cs.c b/drivers/net/pcmcia/axnet_cs.c index 09291e6..41d2def 100644 --- a/drivers/net/pcmcia/axnet_cs.c +++ b/drivers/net/pcmcia/axnet_cs.c @@ -169,7 +169,6 @@ static int axnet_probe(struct pcmcia_device *link) info = PRIV(dev); info->p_dev = link; link->priv = dev; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -266,12 +265,9 @@ static int try_io_port(struct pcmcia_device *link) int j, ret; if (link->io.NumPorts1 == 32) { link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; - if (link->io.NumPorts2 > 0) { - /* for master/slave multifunction cards */ + /* for master/slave multifunction cards */ + if (link->io.NumPorts2 > 0) link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; - link->irq.Attributes = - IRQ_TYPE_DYNAMIC_SHARING; - } } else { /* This should be two 16-port windows */ link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; @@ -337,8 +333,7 @@ static int axnet_config(struct pcmcia_device *link) if (ret != 0) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; if (link->io.NumPorts2 == 8) { @@ -350,7 +345,7 @@ static int axnet_config(struct pcmcia_device *link) if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; if (!get_prom(link)) { diff --git a/drivers/net/pcmcia/com20020_cs.c b/drivers/net/pcmcia/com20020_cs.c index 21d9c9d..278438b 100644 --- a/drivers/net/pcmcia/com20020_cs.c +++ b/drivers/net/pcmcia/com20020_cs.c @@ -163,7 +163,6 @@ static int com20020_probe(struct pcmcia_device *p_dev) p_dev->io.Attributes1 = IO_DATA_PATH_WIDTH_8; p_dev->io.NumPorts1 = 16; p_dev->io.IOAddrLines = 16; - p_dev->irq.Attributes = IRQ_TYPE_EXCLUSIVE; p_dev->conf.Attributes = CONF_ENABLE_IRQ; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -275,15 +274,14 @@ static int com20020_config(struct pcmcia_device *link) dev_dbg(&link->dev, "got ioaddr %Xh\n", ioaddr); dev_dbg(&link->dev, "request IRQ %d\n", - link->irq.AssignedIRQ); - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) + link->irq); + if (!link->irq) { dev_dbg(&link->dev, "requestIRQ failed totally!\n"); goto failed; } - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; ret = pcmcia_request_configuration(link, &link->conf); if (ret) diff --git a/drivers/net/pcmcia/fmvj18x_cs.c b/drivers/net/pcmcia/fmvj18x_cs.c index b9dc80b..31f1a03 100644 --- a/drivers/net/pcmcia/fmvj18x_cs.c +++ b/drivers/net/pcmcia/fmvj18x_cs.c @@ -254,10 +254,6 @@ static int fmvj18x_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 5; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = fjn_interrupt; - /* General socket configuration */ link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -425,8 +421,6 @@ static int fmvj18x_config(struct pcmcia_device *link) } if (link->io.NumPorts2 != 0) { - link->irq.Attributes = - IRQ_TYPE_DYNAMIC_SHARING; ret = mfc_try_io_port(link); if (ret != 0) goto failed; } else if (cardtype == UNGERMANN) { @@ -437,14 +431,14 @@ static int fmvj18x_config(struct pcmcia_device *link) if (ret) goto failed; } - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, fjn_interrupt); if (ret) goto failed; ret = pcmcia_request_configuration(link, &link->conf); if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; if (link->io.BasePort2 != 0) { diff --git a/drivers/net/pcmcia/ibmtr_cs.c b/drivers/net/pcmcia/ibmtr_cs.c index 37f4a6f..0225cdf 100644 --- a/drivers/net/pcmcia/ibmtr_cs.c +++ b/drivers/net/pcmcia/ibmtr_cs.c @@ -156,8 +156,6 @@ static int __devinit ibmtr_attach(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts1 = 4; link->io.IOAddrLines = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - link->irq.Handler = ibmtr_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.Present = PRESENT_OPTION; @@ -238,11 +236,11 @@ static int __devinit ibmtr_config(struct pcmcia_device *link) } dev->base_addr = link->io.BasePort1; - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_exclusive_irq(link, ibmtr_interrupt); if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; - ti->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; + ti->irq = link->irq; ti->global_int_enable=GLOBAL_INT_ENABLE+((dev->irq==9) ? 2 : dev->irq); /* Allocate the MMIO memory window */ diff --git a/drivers/net/pcmcia/nmclan_cs.c b/drivers/net/pcmcia/nmclan_cs.c index c717b14..b779e3a 100644 --- a/drivers/net/pcmcia/nmclan_cs.c +++ b/drivers/net/pcmcia/nmclan_cs.c @@ -463,8 +463,6 @@ static int nmclan_probe(struct pcmcia_device *link) link->io.NumPorts1 = 32; link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 5; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - link->irq.Handler = mace_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; @@ -652,14 +650,14 @@ static int nmclan_config(struct pcmcia_device *link) ret = pcmcia_request_io(link, &link->io); if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_exclusive_irq(link, mace_interrupt); if (ret) goto failed; ret = pcmcia_request_configuration(link, &link->conf); if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; ioaddr = dev->base_addr; diff --git a/drivers/net/pcmcia/pcnet_cs.c b/drivers/net/pcmcia/pcnet_cs.c index 776cad2..22a3350 100644 --- a/drivers/net/pcmcia/pcnet_cs.c +++ b/drivers/net/pcmcia/pcnet_cs.c @@ -265,7 +265,6 @@ static int pcnet_probe(struct pcmcia_device *link) info->p_dev = link; link->priv = dev; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -489,8 +488,6 @@ static int try_io_port(struct pcmcia_device *link) if (link->io.NumPorts2 > 0) { /* for master/slave multifunction cards */ link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; - link->irq.Attributes = - IRQ_TYPE_DYNAMIC_SHARING; } } else { /* This should be two 16-port windows */ @@ -560,8 +557,7 @@ static int pcnet_config(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; if (link->io.NumPorts2 == 8) { @@ -575,7 +571,7 @@ static int pcnet_config(struct pcmcia_device *link) ret = pcmcia_request_configuration(link, &link->conf); if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; if (info->flags & HAS_MISC_REG) { if ((if_port == 1) || (if_port == 2)) diff --git a/drivers/net/pcmcia/smc91c92_cs.c b/drivers/net/pcmcia/smc91c92_cs.c index 5adc662..1257d10 100644 --- a/drivers/net/pcmcia/smc91c92_cs.c +++ b/drivers/net/pcmcia/smc91c92_cs.c @@ -329,8 +329,6 @@ static int smc91c92_probe(struct pcmcia_device *link) link->io.NumPorts1 = 16; link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 4; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = &smc_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -453,7 +451,6 @@ static int mhz_mfc_config(struct pcmcia_device *link) link->conf.Attributes |= CONF_ENABLE_SPKR; link->conf.Status = CCSR_AUDIO_ENA; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->io.IOAddrLines = 16; link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts2 = 8; @@ -651,7 +648,6 @@ static int osi_config(struct pcmcia_device *link) link->conf.Attributes |= CONF_ENABLE_SPKR; link->conf.Status = CCSR_AUDIO_ENA; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->io.NumPorts1 = 64; link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts2 = 8; @@ -876,7 +872,7 @@ static int smc91c92_config(struct pcmcia_device *link) if (i) goto config_failed; - i = pcmcia_request_irq(link, &link->irq); + i = pcmcia_request_irq(link, smc_interrupt); if (i) goto config_failed; i = pcmcia_request_configuration(link, &link->conf); @@ -886,7 +882,7 @@ static int smc91c92_config(struct pcmcia_device *link) if (smc->manfid == MANFID_MOTOROLA) mot_config(link); - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; if ((if_port >= 0) && (if_port <= 2)) dev->if_port = if_port; diff --git a/drivers/net/pcmcia/xirc2ps_cs.c b/drivers/net/pcmcia/xirc2ps_cs.c index 4d1802e..c9e7d7d 100644 --- a/drivers/net/pcmcia/xirc2ps_cs.c +++ b/drivers/net/pcmcia/xirc2ps_cs.c @@ -555,7 +555,6 @@ xirc2ps_probe(struct pcmcia_device *link) link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; - link->irq.Handler = xirc2ps_interrupt; /* Fill in card specific entries */ dev->netdev_ops = &netdev_ops; @@ -841,7 +840,6 @@ xirc2ps_config(struct pcmcia_device * link) link->conf.Attributes |= CONF_ENABLE_SPKR; link->conf.Status |= CCSR_AUDIO_ENA; } - link->irq.Attributes |= IRQ_TYPE_DYNAMIC_SHARING; link->io.NumPorts2 = 8; link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; if (local->dingo) { @@ -866,7 +864,6 @@ xirc2ps_config(struct pcmcia_device * link) } printk(KNOT_XIRC "no ports available\n"); } else { - link->irq.Attributes |= IRQ_TYPE_DYNAMIC_SHARING; link->io.NumPorts1 = 16; for (ioaddr = 0x300; ioaddr < 0x400; ioaddr += 0x10) { link->io.BasePort1 = ioaddr; @@ -885,7 +882,7 @@ xirc2ps_config(struct pcmcia_device * link) * Now allocate an interrupt line. Note that this does not * actually assign a handler to the interrupt. */ - if ((err=pcmcia_request_irq(link, &link->irq))) + if ((err=pcmcia_request_irq(link, xirc2ps_interrupt))) goto config_error; /**************** @@ -982,7 +979,7 @@ xirc2ps_config(struct pcmcia_device * link) printk(KNOT_XIRC "invalid if_port requested\n"); /* we can now register the device with the net subsystem */ - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; if (local->dingo) diff --git a/drivers/net/wireless/airo_cs.c b/drivers/net/wireless/airo_cs.c index f6036fb..7867c51 100644 --- a/drivers/net/wireless/airo_cs.c +++ b/drivers/net/wireless/airo_cs.c @@ -132,10 +132,6 @@ static int airo_probe(struct pcmcia_device *p_dev) dev_dbg(&p_dev->dev, "airo_attach()\n"); - /* Interrupt setup */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - p_dev->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -212,9 +208,7 @@ static int airo_cs_config_check(struct pcmcia_device *p_dev, else if (dflt->vpp1.present & (1<<CISTPL_POWER_VNOM)) p_dev->conf.Vpp = dflt->vpp1.param[CISTPL_POWER_VNOM]/10000; - /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -300,16 +294,8 @@ static int airo_config(struct pcmcia_device *link) if (ret) goto failed; - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + if (!link->irq) + goto failed; /* This actually configures the PCMCIA socket -- setting up @@ -320,7 +306,7 @@ static int airo_config(struct pcmcia_device *link) if (ret) goto failed; ((local_info_t *)link->priv)->eth_dev = - init_airo_card(link->irq.AssignedIRQ, + init_airo_card(link->irq, link->io.BasePort1, 1, &link->dev); if (!((local_info_t *)link->priv)->eth_dev) goto failed; @@ -338,8 +324,7 @@ static int airo_config(struct pcmcia_device *link) dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Vpp) printk(", Vpp %d.%d", link->conf.Vpp/10, link->conf.Vpp%10); - if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); diff --git a/drivers/net/wireless/atmel_cs.c b/drivers/net/wireless/atmel_cs.c index 3240791..759cdc4 100644 --- a/drivers/net/wireless/atmel_cs.c +++ b/drivers/net/wireless/atmel_cs.c @@ -141,10 +141,6 @@ static int atmel_probe(struct pcmcia_device *p_dev) dev_dbg(&p_dev->dev, "atmel_attach()\n"); - /* Interrupt setup */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - p_dev->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -226,9 +222,7 @@ static int atmel_config_check(struct pcmcia_device *p_dev, else if (dflt->vpp1.present & (1<<CISTPL_POWER_VNOM)) p_dev->conf.Vpp = dflt->vpp1.param[CISTPL_POWER_VNOM]/10000; - /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -278,15 +272,9 @@ static int atmel_config(struct pcmcia_device *link) if (pcmcia_loop_config(link, atmel_config_check, NULL)) goto failed; - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; + if (!link->irq) { + dev_err(&link->dev, "atmel: cannot assign IRQ: check that CONFIG_ISA is set in kernel config."); + goto failed; } /* @@ -298,14 +286,8 @@ static int atmel_config(struct pcmcia_device *link) if (ret) goto failed; - if (link->irq.AssignedIRQ == 0) { - printk(KERN_ALERT - "atmel: cannot assign IRQ: check that CONFIG_ISA is set in kernel config."); - goto failed; - } - ((local_info_t*)link->priv)->eth_dev = - init_atmel_card(link->irq.AssignedIRQ, + init_atmel_card(link->irq, link->io.BasePort1, did ? did->driver_info : ATMEL_FW_TYPE_NONE, &link->dev, diff --git a/drivers/net/wireless/b43/pcmcia.c b/drivers/net/wireless/b43/pcmcia.c index 984174b..b9ea107 100644 --- a/drivers/net/wireless/b43/pcmcia.c +++ b/drivers/net/wireless/b43/pcmcia.c @@ -97,10 +97,7 @@ static int __devinit b43_pcmcia_probe(struct pcmcia_device *dev) if (res != 0) goto err_disable; - dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - dev->irq.Handler = NULL; /* The handler is registered later. */ - res = pcmcia_request_irq(dev, &dev->irq); - if (res != 0) + if (!dev->irq) goto err_disable; res = pcmcia_request_configuration(dev, &dev->conf); diff --git a/drivers/net/wireless/hostap/hostap_cs.c b/drivers/net/wireless/hostap/hostap_cs.c index d19748d..7e5f4c0 100644 --- a/drivers/net/wireless/hostap/hostap_cs.c +++ b/drivers/net/wireless/hostap/hostap_cs.c @@ -555,15 +555,7 @@ static int prism2_config_check(struct pcmcia_device *p_dev, p_dev->conf.Vpp = dflt->vpp1.param[CISTPL_POWER_VNOM] / 10000; /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; - else if (!(p_dev->conf.Attributes & CONF_ENABLE_IRQ)) { - /* At least Compaq WL200 does not have IRQInfo1 set, - * but it does not work without interrupts.. */ - printk(KERN_WARNING "Config has no IRQ info, but trying to " - "enable IRQ anyway..\n"); - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; - } + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ PDEBUG(DEBUG_EXTRA, "IO window settings: cfg->io.nwin=%d " @@ -635,18 +627,9 @@ static int prism2_config(struct pcmcia_device *link) strcpy(hw_priv->node.dev_name, dev->name); link->dev_node = &hw_priv->node; - /* - * Allocate an interrupt line. Note that this does not assign a - * handler to the interrupt, unless the 'Handler' member of the - * irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = prism2_interrupt; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + ret = pcmcia_request_irq(link, prism2_interrupt); + if (ret) + goto failed; /* * This actually configures the PCMCIA socket -- setting up @@ -657,7 +640,7 @@ static int prism2_config(struct pcmcia_device *link) if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; /* Finally, report what we've done */ @@ -667,7 +650,7 @@ static int prism2_config(struct pcmcia_device *link) printk(", Vpp %d.%d", link->conf.Vpp / 10, link->conf.Vpp % 10); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); diff --git a/drivers/net/wireless/libertas/if_cs.c b/drivers/net/wireless/libertas/if_cs.c index 1f6cb58..0c99b7d 100644 --- a/drivers/net/wireless/libertas/if_cs.c +++ b/drivers/net/wireless/libertas/if_cs.c @@ -776,7 +776,7 @@ static void if_cs_release(struct pcmcia_device *p_dev) lbs_deb_enter(LBS_DEB_CS); - free_irq(p_dev->irq.AssignedIRQ, card); + free_irq(p_dev->irq, card); pcmcia_disable_device(p_dev); if (card->iobase) ioport_unmap(card->iobase); @@ -806,8 +806,7 @@ static int if_cs_ioprobe(struct pcmcia_device *p_dev, p_dev->io.NumPorts1 = cfg->io.win[0].len; /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ if (cfg->io.nwin != 1) { @@ -836,9 +835,6 @@ static int if_cs_probe(struct pcmcia_device *p_dev) card->p_dev = p_dev; p_dev->priv = card; - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - p_dev->irq.Handler = NULL; - p_dev->conf.Attributes = 0; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -853,13 +849,8 @@ static int if_cs_probe(struct pcmcia_device *p_dev) * a handler to the interrupt, unless the 'Handler' member of * the irq structure is initialized. */ - if (p_dev->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(p_dev, &p_dev->irq); - if (ret) { - lbs_pr_err("error in pcmcia_request_irq\n"); - goto out1; - } - } + if (!p_dev->irq) + goto out1; /* Initialize io access */ card->iobase = ioport_map(p_dev->io.BasePort1, p_dev->io.NumPorts1); @@ -882,7 +873,7 @@ static int if_cs_probe(struct pcmcia_device *p_dev) /* Finally, report what we've done */ lbs_deb_cs("irq %d, io 0x%04x-0x%04x\n", - p_dev->irq.AssignedIRQ, p_dev->io.BasePort1, + p_dev->irq, p_dev->io.BasePort1, p_dev->io.BasePort1 + p_dev->io.NumPorts1 - 1); /* @@ -939,7 +930,7 @@ static int if_cs_probe(struct pcmcia_device *p_dev) priv->fw_ready = 1; /* Now actually get the IRQ */ - ret = request_irq(p_dev->irq.AssignedIRQ, if_cs_interrupt, + ret = request_irq(p_dev->irq, if_cs_interrupt, IRQF_SHARED, DRV_NAME, card); if (ret) { lbs_pr_err("error in request_irq\n"); diff --git a/drivers/net/wireless/orinoco/orinoco_cs.c b/drivers/net/wireless/orinoco/orinoco_cs.c index 1d4ada1..5d29b11 100644 --- a/drivers/net/wireless/orinoco/orinoco_cs.c +++ b/drivers/net/wireless/orinoco/orinoco_cs.c @@ -119,10 +119,6 @@ orinoco_cs_probe(struct pcmcia_device *link) card->p_dev = link; link->priv = priv; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = orinoco_interrupt; - /* General socket configuration defaults can go here. In this * client, we assume very little, and rely on the CIS for * almost everything. In most clients, many details (i.e., @@ -258,12 +254,7 @@ orinoco_cs_config(struct pcmcia_device *link) goto failed; } - /* - * Allocate an interrupt line. Note that this does not assign - * a handler to the interrupt, unless the 'Handler' member of - * the irq structure is initialized. - */ - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, orinoco_interrupt); if (ret) goto failed; @@ -296,7 +287,7 @@ orinoco_cs_config(struct pcmcia_device *link) /* Register an interface with the stack */ if (orinoco_if_add(priv, link->io.BasePort1, - link->irq.AssignedIRQ) != 0) { + link->irq) != 0) { printk(KERN_ERR PFX "orinoco_if_add() failed\n"); goto failed; } diff --git a/drivers/net/wireless/orinoco/spectrum_cs.c b/drivers/net/wireless/orinoco/spectrum_cs.c index 59bda24..7a8e056 100644 --- a/drivers/net/wireless/orinoco/spectrum_cs.c +++ b/drivers/net/wireless/orinoco/spectrum_cs.c @@ -193,10 +193,6 @@ spectrum_cs_probe(struct pcmcia_device *link) card->p_dev = link; link->priv = priv; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = orinoco_interrupt; - /* General socket configuration defaults can go here. In this * client, we assume very little, and rely on the CIS for * almost everything. In most clients, many details (i.e., @@ -332,12 +328,7 @@ spectrum_cs_config(struct pcmcia_device *link) goto failed; } - /* - * Allocate an interrupt line. Note that this does not assign - * a handler to the interrupt, unless the 'Handler' member of - * the irq structure is initialized. - */ - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, orinoco_interrupt); if (ret) goto failed; @@ -374,7 +365,7 @@ spectrum_cs_config(struct pcmcia_device *link) /* Register an interface with the stack */ if (orinoco_if_add(priv, link->io.BasePort1, - link->irq.AssignedIRQ) != 0) { + link->irq) != 0) { printk(KERN_ERR PFX "orinoco_if_add() failed\n"); goto failed; } diff --git a/drivers/net/wireless/ray_cs.c b/drivers/net/wireless/ray_cs.c index 7f0f855..e7ed2a4 100644 --- a/drivers/net/wireless/ray_cs.c +++ b/drivers/net/wireless/ray_cs.c @@ -321,10 +321,6 @@ static int ray_probe(struct pcmcia_device *p_dev) p_dev->io.Attributes1 = IO_DATA_PATH_WIDTH_8; p_dev->io.IOAddrLines = 5; - /* Interrupt setup. For PCMCIA, driver takes what's given */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - p_dev->irq.Handler = &ray_interrupt; - /* General socket configuration */ p_dev->conf.Attributes = CONF_ENABLE_IRQ; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -417,10 +413,10 @@ static int ray_config(struct pcmcia_device *link) /* Now allocate an interrupt line. Note that this does not actually assign a handler to the interrupt. */ - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, ray_interrupt); if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; /* This actually configures the PCMCIA socket -- setting up the I/O windows and the interrupt mapping. diff --git a/drivers/net/wireless/wl3501_cs.c b/drivers/net/wireless/wl3501_cs.c index 7b9621d..c43f05b 100644 --- a/drivers/net/wireless/wl3501_cs.c +++ b/drivers/net/wireless/wl3501_cs.c @@ -1897,10 +1897,6 @@ static int wl3501_probe(struct pcmcia_device *p_dev) p_dev->io.Attributes1 = IO_DATA_PATH_WIDTH_8; p_dev->io.IOAddrLines = 5; - /* Interrupt setup */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - p_dev->irq.Handler = wl3501_interrupt; - /* General socket configuration */ p_dev->conf.Attributes = CONF_ENABLE_IRQ; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -1961,7 +1957,7 @@ static int wl3501_config(struct pcmcia_device *link) /* Now allocate an interrupt line. Note that this does not actually * assign a handler to the interrupt. */ - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, wl3501_interrupt); if (ret) goto failed; @@ -1972,7 +1968,7 @@ static int wl3501_config(struct pcmcia_device *link) if (ret) goto failed; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; SET_NETDEV_DEV(dev, &link->dev); if (register_netdev(dev)) { diff --git a/drivers/parport/parport_cs.c b/drivers/parport/parport_cs.c index 7dd370f..80c9052 100644 --- a/drivers/parport/parport_cs.c +++ b/drivers/parport/parport_cs.c @@ -105,7 +105,6 @@ static int parport_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.Attributes2 = IO_DATA_PATH_WIDTH_8; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -174,20 +173,19 @@ static int parport_config(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); if (ret) goto failed; p = parport_pc_probe_port(link->io.BasePort1, link->io.BasePort2, - link->irq.AssignedIRQ, PARPORT_DMA_NONE, + link->irq, PARPORT_DMA_NONE, &link->dev, IRQF_SHARED); if (p == NULL) { printk(KERN_NOTICE "parport_cs: parport_pc_probe_port() at " "0x%3x, irq %u failed\n", link->io.BasePort1, - link->irq.AssignedIRQ); + link->irq); goto failed; } diff --git a/drivers/pcmcia/ds.c b/drivers/pcmcia/ds.c index e0b1efc..74245cc 100644 --- a/drivers/pcmcia/ds.c +++ b/drivers/pcmcia/ds.c @@ -531,7 +531,6 @@ struct pcmcia_device *pcmcia_device_add(struct pcmcia_socket *s, unsigned int fu p_dev->function_config = tmp_dev->function_config; p_dev->io = tmp_dev->io; p_dev->irq = tmp_dev->irq; - p_dev->irq_v = tmp_dev->irq_v; kref_get(&p_dev->function_config->ref); } @@ -556,7 +555,7 @@ struct pcmcia_device *pcmcia_device_add(struct pcmcia_socket *s, unsigned int fu dev_printk(KERN_NOTICE, &p_dev->dev, "pcmcia: registering new device %s (IRQ: %d)\n", - p_dev->devname, p_dev->irq_v); + p_dev->devname, p_dev->irq); pcmcia_device_query(p_dev); diff --git a/drivers/pcmcia/pcmcia_resource.c b/drivers/pcmcia/pcmcia_resource.c index f629aac..c907a3f 100644 --- a/drivers/pcmcia/pcmcia_resource.c +++ b/drivers/pcmcia/pcmcia_resource.c @@ -407,41 +407,6 @@ out: } /* pcmcia_release_io */ -static int pcmcia_release_irq(struct pcmcia_device *p_dev, irq_req_t *req) -{ - struct pcmcia_socket *s = p_dev->socket; - config_t *c; - int ret = -EINVAL; - - mutex_lock(&s->ops_mutex); - - c = p_dev->function_config; - - if (!p_dev->_irq) - goto out; - - p_dev->_irq = 0; - - if (c->state & CONFIG_LOCKED) - goto out; - - if (s->pcmcia_irq != req->AssignedIRQ) { - dev_dbg(&s->dev, "IRQ must match assigned one\n"); - goto out; - } - - if (req->Handler) - free_irq(req->AssignedIRQ, p_dev->priv); - - ret = 0; - -out: - mutex_unlock(&s->ops_mutex); - - return ret; -} /* pcmcia_release_irq */ - - int pcmcia_release_window(struct pcmcia_device *p_dev, window_handle_t wh) { struct pcmcia_socket *s = p_dev->socket; @@ -679,61 +644,66 @@ out: EXPORT_SYMBOL(pcmcia_request_io); -/** pcmcia_request_irq +/** + * pcmcia_request_irq() - attempt to request a IRQ for a PCMCIA device * - * Request_irq() reserves an irq for this client. + * pcmcia_request_irq() is a wrapper around request_irq which will allow + * the PCMCIA core to clean up the registration in pcmcia_disable_device(). + * Drivers are free to use request_irq() directly, but then they need to + * call free_irq themselfves, too. Also, only IRQF_SHARED capable IRQ + * handlers are allowed. */ - -int pcmcia_request_irq(struct pcmcia_device *p_dev, irq_req_t *req) +int __must_check pcmcia_request_irq(struct pcmcia_device *p_dev, + irq_handler_t handler) { - struct pcmcia_socket *s = p_dev->socket; - config_t *c; - int ret = -EINVAL, irq = p_dev->irq_v; - int type = IRQF_SHARED; + int ret; - mutex_lock(&s->ops_mutex); + if (!p_dev->irq) + return -EINVAL; - if (!(s->state & SOCKET_PRESENT)) { - dev_dbg(&s->dev, "No card present\n"); - goto out; - } - c = p_dev->function_config; - if (c->state & CONFIG_LOCKED) { - dev_dbg(&s->dev, "Configuration is locked\n"); - goto out; - } + ret = request_irq(p_dev->irq, handler, IRQF_SHARED, + p_dev->devname, p_dev->priv); + if (!ret) + p_dev->_irq = 1; - if (!irq) { - dev_dbg(&s->dev, "no IRQ available\n"); - goto out; - } + return ret; +} +EXPORT_SYMBOL(pcmcia_request_irq); - if (!(req->Attributes & IRQ_TYPE_DYNAMIC_SHARING)) { - req->Attributes |= IRQ_TYPE_DYNAMIC_SHARING; - dev_printk(KERN_WARNING, &p_dev->dev, "pcmcia: the driver " - "needs updating to supported shared IRQ lines.\n"); - } - if (req->Handler) { - ret = request_irq(irq, req->Handler, type, - p_dev->devname, p_dev->priv); - if (ret) { - dev_printk(KERN_INFO, &s->dev, - "request_irq() failed\n"); - goto out; - } - } +/** + * pcmcia_request_exclusive_irq() - attempt to request an exclusive IRQ first + * + * pcmcia_request_exclusive_irq() is a wrapper around request_irq which + * attempts first to request an exclusive IRQ. If it fails, it also accepts + * a shared IRQ, but prints out a warning. PCMCIA drivers should allow for + * IRQ sharing and either use request_irq directly (then they need to call + * free_irq themselves, too), or the pcmcia_request_irq() function. + */ +int __must_check +pcmcia_request_exclusive_irq(struct pcmcia_device *p_dev, irq_handler_t handler) +{ + int ret; - req->AssignedIRQ = irq; + if (!p_dev->irq) + return -EINVAL; - p_dev->_irq = 1; + ret = request_irq(p_dev->irq, handler, 0, p_dev->devname, p_dev->priv); + if (ret) { + ret = pcmcia_request_irq(p_dev, handler); + dev_printk(KERN_WARNING, &p_dev->dev, "pcmcia: " + "request for exclusive IRQ could not be fulfilled.\n"); + dev_printk(KERN_WARNING, &p_dev->dev, "pcmcia: the driver " + "needs updating to supported shared IRQ lines.\n"); + } + if (ret) + dev_printk(KERN_INFO, &p_dev->dev, "request_irq() failed\n"); + else + p_dev->_irq = 1; - ret = 0; -out: - mutex_unlock(&s->ops_mutex); return ret; -} /* pcmcia_request_irq */ -EXPORT_SYMBOL(pcmcia_request_irq); +} /* pcmcia_request_exclusive_irq */ +EXPORT_SYMBOL(pcmcia_request_exclusive_irq); #ifdef CONFIG_PCMCIA_PROBE @@ -777,7 +747,7 @@ static int pcmcia_setup_isa_irq(struct pcmcia_device *p_dev, int type) p_dev); if (!ret) { free_irq(irq, p_dev); - p_dev->irq_v = s->pcmcia_irq = irq; + p_dev->irq = s->pcmcia_irq = irq; pcmcia_used_irq[irq]++; break; } @@ -818,12 +788,12 @@ int pcmcia_setup_irq(struct pcmcia_device *p_dev) { struct pcmcia_socket *s = p_dev->socket; - if (p_dev->irq_v) + if (p_dev->irq) return 0; /* already assigned? */ if (s->pcmcia_irq) { - p_dev->irq_v = s->pcmcia_irq; + p_dev->irq = s->pcmcia_irq; return 0; } @@ -837,7 +807,7 @@ int pcmcia_setup_irq(struct pcmcia_device *p_dev) /* but use the PCI irq otherwise */ if (s->pci_irq) { - p_dev->irq_v = s->pcmcia_irq = s->pci_irq; + p_dev->irq = s->pcmcia_irq = s->pci_irq; return 0; } @@ -945,7 +915,8 @@ void pcmcia_disable_device(struct pcmcia_device *p_dev) { pcmcia_release_configuration(p_dev); pcmcia_release_io(p_dev, &p_dev->io); - pcmcia_release_irq(p_dev, &p_dev->irq); + if (p_dev->_irq) + free_irq(p_dev->irq, p_dev->priv); if (p_dev->win) pcmcia_release_window(p_dev, p_dev->win); } diff --git a/drivers/scsi/pcmcia/aha152x_stub.c b/drivers/scsi/pcmcia/aha152x_stub.c index 528733b..c1de4ba 100644 --- a/drivers/scsi/pcmcia/aha152x_stub.c +++ b/drivers/scsi/pcmcia/aha152x_stub.c @@ -105,7 +105,6 @@ static int aha152x_probe(struct pcmcia_device *link) link->io.NumPorts1 = 0x20; link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 10; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.Present = PRESENT_OPTION; @@ -160,8 +159,7 @@ static int aha152x_config_cs(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); @@ -172,7 +170,7 @@ static int aha152x_config_cs(struct pcmcia_device *link) memset(&s, 0, sizeof(s)); s.conf = "PCMCIA setup"; s.io_port = link->io.BasePort1; - s.irq = link->irq.AssignedIRQ; + s.irq = link->irq; s.scsiid = host_id; s.reconnect = reconnect; s.parity = parity; diff --git a/drivers/scsi/pcmcia/fdomain_stub.c b/drivers/scsi/pcmcia/fdomain_stub.c index 9140406..8cb6d71 100644 --- a/drivers/scsi/pcmcia/fdomain_stub.c +++ b/drivers/scsi/pcmcia/fdomain_stub.c @@ -88,7 +88,6 @@ static int fdomain_probe(struct pcmcia_device *link) link->io.NumPorts1 = 0x10; link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 10; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.Present = PRESENT_OPTION; @@ -133,8 +132,7 @@ static int fdomain_config(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); if (ret) @@ -144,7 +142,7 @@ static int fdomain_config(struct pcmcia_device *link) release_region(link->io.BasePort1, link->io.NumPorts1); /* Set configuration options for the fdomain driver */ - sprintf(str, "%d,%d", link->io.BasePort1, link->irq.AssignedIRQ); + sprintf(str, "%d,%d", link->io.BasePort1, link->irq); fdomain_setup(str); host = __fdomain_16x0_detect(&fdomain_driver_template); diff --git a/drivers/scsi/pcmcia/nsp_cs.c b/drivers/scsi/pcmcia/nsp_cs.c index 0212464..24c78ed 100644 --- a/drivers/scsi/pcmcia/nsp_cs.c +++ b/drivers/scsi/pcmcia/nsp_cs.c @@ -1563,13 +1563,6 @@ static int nsp_cs_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 10; /* not used */ - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - - /* Interrupt handler */ - link->irq.Handler = &nspintr; - link->irq.Attributes |= IRQF_SHARED; - /* General socket configuration */ link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -1646,8 +1639,7 @@ static int nsp_cs_config_check(struct pcmcia_device *p_dev, } /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -1720,10 +1712,8 @@ static int nsp_cs_config(struct pcmcia_device *link) if (ret) goto cs_failed; - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - if (pcmcia_request_irq(link, &link->irq)) - goto cs_failed; - } + if (pcmcia_request_irq(link, nspintr)) + goto cs_failed; ret = pcmcia_request_configuration(link, &link->conf); if (ret) @@ -1741,7 +1731,7 @@ static int nsp_cs_config(struct pcmcia_device *link) /* Set port and IRQ */ data->BaseAddress = link->io.BasePort1; data->NumAddress = link->io.NumPorts1; - data->IrqNumber = link->irq.AssignedIRQ; + data->IrqNumber = link->irq; nsp_dbg(NSP_DEBUG_INIT, "I/O[0x%x+0x%x] IRQ %d", data->BaseAddress, data->NumAddress, data->IrqNumber); @@ -1775,7 +1765,7 @@ static int nsp_cs_config(struct pcmcia_device *link) printk(", Vpp %d.%d", link->conf.Vpp/10, link->conf.Vpp%10); } if (link->conf.Attributes & CONF_ENABLE_IRQ) { - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); } if (link->io.NumPorts1) { printk(", io 0x%04x-0x%04x", link->io.BasePort1, diff --git a/drivers/scsi/pcmcia/qlogic_stub.c b/drivers/scsi/pcmcia/qlogic_stub.c index f85f094..c3a4428 100644 --- a/drivers/scsi/pcmcia/qlogic_stub.c +++ b/drivers/scsi/pcmcia/qlogic_stub.c @@ -161,7 +161,6 @@ static int qlogic_probe(struct pcmcia_device *link) link->io.NumPorts1 = 16; link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 10; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.Present = PRESENT_OPTION; @@ -209,8 +208,7 @@ static int qlogic_config(struct pcmcia_device * link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); @@ -227,10 +225,10 @@ static int qlogic_config(struct pcmcia_device * link) /* The KXL-810AN has a bigger IO port window */ if (link->io.NumPorts1 == 32) host = qlogic_detect(&qlogicfas_driver_template, link, - link->io.BasePort1 + 16, link->irq.AssignedIRQ); + link->io.BasePort1 + 16, link->irq); else host = qlogic_detect(&qlogicfas_driver_template, link, - link->io.BasePort1, link->irq.AssignedIRQ); + link->io.BasePort1, link->irq); if (!host) { printk(KERN_INFO "%s: no SCSI devices found\n", qlogic_name); @@ -258,7 +256,7 @@ static void qlogic_release(struct pcmcia_device *link) scsi_remove_host(info->host); - free_irq(link->irq.AssignedIRQ, info->host); + free_irq(link->irq, info->host); pcmcia_disable_device(link); scsi_host_put(info->host); diff --git a/drivers/scsi/pcmcia/sym53c500_cs.c b/drivers/scsi/pcmcia/sym53c500_cs.c index e7564d8..6eacf4c 100644 --- a/drivers/scsi/pcmcia/sym53c500_cs.c +++ b/drivers/scsi/pcmcia/sym53c500_cs.c @@ -719,8 +719,7 @@ SYM53C500_config(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); - if (ret) + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); @@ -752,7 +751,7 @@ SYM53C500_config(struct pcmcia_device *link) * 0x320, 0x330, 0x340, 0x350 */ port_base = link->io.BasePort1; - irq_level = link->irq.AssignedIRQ; + irq_level = link->irq; DEB(printk("SYM53C500: port_base=0x%x, irq=%d, fast_pio=%d\n", port_base, irq_level, USE_FAST_PIO);) @@ -866,7 +865,6 @@ SYM53C500_probe(struct pcmcia_device *link) link->io.NumPorts1 = 16; link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.IOAddrLines = 10; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; diff --git a/drivers/serial/serial_cs.c b/drivers/serial/serial_cs.c index e91db4b..7519d27 100644 --- a/drivers/serial/serial_cs.c +++ b/drivers/serial/serial_cs.c @@ -334,7 +334,6 @@ static int serial_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_8; link->io.NumPorts1 = 8; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; if (do_sound) { link->conf.Attributes |= CONF_ENABLE_SPKR; @@ -477,7 +476,7 @@ static int simple_config(struct pcmcia_device *link) } if (info->slave) { return setup_serial(link, info, port, - link->irq.AssignedIRQ); + link->irq); } } @@ -498,10 +497,6 @@ static int simple_config(struct pcmcia_device *link) return -1; found_port: - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) - link->irq.AssignedIRQ = 0; - if (info->multi && (info->manfid == MANFID_3COM)) link->conf.ConfigIndex &= ~(0x08); @@ -514,7 +509,7 @@ found_port: i = pcmcia_request_configuration(link, &link->conf); if (i != 0) return -1; - return setup_serial(link, info, link->io.BasePort1, link->irq.AssignedIRQ); + return setup_serial(link, info, link->io.BasePort1, link->irq); } static int multi_config_check(struct pcmcia_device *p_dev, @@ -577,13 +572,9 @@ static int multi_config(struct pcmcia_device *link) } } - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) { - /* FIXME: comment does not fit, error handling does not fit */ - printk(KERN_NOTICE - "serial_cs: no usable port range found, giving up\n"); - link->irq.AssignedIRQ = 0; - } + if (!link->irq) + dev_warn(&link->dev, + "serial_cs: no usable IRQ found, continuing...\n"); /* * Apply any configuration quirks. @@ -606,11 +597,11 @@ static int multi_config(struct pcmcia_device *link) if (link->conf.ConfigIndex == 1 || link->conf.ConfigIndex == 3) { err = setup_serial(link, info, base2, - link->irq.AssignedIRQ); + link->irq); base2 = link->io.BasePort1; } else { err = setup_serial(link, info, link->io.BasePort1, - link->irq.AssignedIRQ); + link->irq); } info->c950ctrl = base2; @@ -624,10 +615,10 @@ static int multi_config(struct pcmcia_device *link) return 0; } - setup_serial(link, info, link->io.BasePort1, link->irq.AssignedIRQ); + setup_serial(link, info, link->io.BasePort1, link->irq); for (i = 0; i < info->multi - 1; i++) setup_serial(link, info, base2 + (8 * i), - link->irq.AssignedIRQ); + link->irq); return 0; } diff --git a/drivers/ssb/main.c b/drivers/ssb/main.c index 03dfd27..1f0bddb 100644 --- a/drivers/ssb/main.c +++ b/drivers/ssb/main.c @@ -489,7 +489,7 @@ static int ssb_devices_register(struct ssb_bus *bus) break; case SSB_BUSTYPE_PCMCIA: #ifdef CONFIG_SSB_PCMCIAHOST - sdev->irq = bus->host_pcmcia->irq.AssignedIRQ; + sdev->irq = bus->host_pcmcia->irq; dev->parent = &bus->host_pcmcia->dev; #endif break; diff --git a/drivers/staging/comedi/drivers/cb_das16_cs.c b/drivers/staging/comedi/drivers/cb_das16_cs.c index bc375e7..7711036 100644 --- a/drivers/staging/comedi/drivers/cb_das16_cs.c +++ b/drivers/staging/comedi/drivers/cb_das16_cs.c @@ -179,12 +179,12 @@ static int das16cs_attach(struct comedi_device *dev, } printk("\n"); - ret = request_irq(link->irq.AssignedIRQ, das16cs_interrupt, + ret = request_irq(link->irq, das16cs_interrupt, IRQF_SHARED, "cb_das16_cs", dev); if (ret < 0) { return ret; } - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; printk("irq=%u ", dev->irq); dev->board_ptr = das16cs_probe(dev, link); @@ -701,10 +701,6 @@ static int das16cs_pcmcia_attach(struct pcmcia_device *link) link->priv = local; /* Initialize the pcmcia_device structure */ - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - link->conf.Attributes = 0; link->conf.IntType = INT_MEMORY_AND_IO; @@ -739,8 +735,7 @@ static int das16cs_pcmcia_config_loop(struct pcmcia_device *p_dev, return -EINVAL; /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -779,16 +774,9 @@ static void das16cs_pcmcia_config(struct pcmcia_device *link) goto failed; } - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + if (!link->irq) + goto failed; + /* This actually configures the PCMCIA socket -- setting up the I/O windows and the interrupt mapping, and putting the @@ -810,7 +798,7 @@ static void das16cs_pcmcia_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %u", link->irq.AssignedIRQ); + printk(", irq %u", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1 + link->io.NumPorts1 - 1); diff --git a/drivers/staging/comedi/drivers/das08_cs.c b/drivers/staging/comedi/drivers/das08_cs.c index f12ef1c..906cc2a 100644 --- a/drivers/staging/comedi/drivers/das08_cs.c +++ b/drivers/staging/comedi/drivers/das08_cs.c @@ -171,10 +171,6 @@ static int das08_pcmcia_attach(struct pcmcia_device *link) local->link = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -228,8 +224,7 @@ static int das08_pcmcia_config_loop(struct pcmcia_device *p_dev, return -ENODEV; /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -276,11 +271,8 @@ static void das08_pcmcia_config(struct pcmcia_device *link) goto failed; } - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + if (!link->irq) + goto failed; /* This actually configures the PCMCIA socket -- setting up @@ -303,7 +295,7 @@ static void das08_pcmcia_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %u", link->irq.AssignedIRQ); + printk(", irq %u", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1 + link->io.NumPorts1 - 1); diff --git a/drivers/staging/comedi/drivers/ni_daq_700.c b/drivers/staging/comedi/drivers/ni_daq_700.c index c9b0395..04fa19a 100644 --- a/drivers/staging/comedi/drivers/ni_daq_700.c +++ b/drivers/staging/comedi/drivers/ni_daq_700.c @@ -379,7 +379,7 @@ static int dio700_attach(struct comedi_device *dev, struct comedi_devconfig *it) return -EIO; iobase = link->io.BasePort1; #ifdef incomplete - irq = link->irq.AssignedIRQ; + irq = link->irq; #endif break; default: @@ -501,10 +501,6 @@ static int dio700_cs_attach(struct pcmcia_device *link) local->link = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -576,8 +572,7 @@ static int dio700_pcmcia_config_loop(struct pcmcia_device *p_dev, } /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -638,16 +633,8 @@ static void dio700_config(struct pcmcia_device *link) goto failed; } - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + if (!link->irq) + goto failed; /* This actually configures the PCMCIA socket -- setting up @@ -670,7 +657,7 @@ static void dio700_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1 + link->io.NumPorts1 - 1); diff --git a/drivers/staging/comedi/drivers/ni_daq_dio24.c b/drivers/staging/comedi/drivers/ni_daq_dio24.c index 9017be3..4b0c608 100644 --- a/drivers/staging/comedi/drivers/ni_daq_dio24.c +++ b/drivers/staging/comedi/drivers/ni_daq_dio24.c @@ -130,7 +130,7 @@ static int dio24_attach(struct comedi_device *dev, struct comedi_devconfig *it) return -EIO; iobase = link->io.BasePort1; #ifdef incomplete - irq = link->irq.AssignedIRQ; + irq = link->irq; #endif break; default: @@ -252,10 +252,6 @@ static int dio24_cs_attach(struct pcmcia_device *link) local->link = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -327,8 +323,7 @@ static int dio24_pcmcia_config_loop(struct pcmcia_device *p_dev, } /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -389,16 +384,8 @@ static void dio24_config(struct pcmcia_device *link) goto failed; } - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + if (!link->irq) + goto failed; /* This actually configures the PCMCIA socket -- setting up @@ -421,7 +408,7 @@ static void dio24_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1 + link->io.NumPorts1 - 1); diff --git a/drivers/staging/comedi/drivers/ni_labpc_cs.c b/drivers/staging/comedi/drivers/ni_labpc_cs.c index 5a0e5b9..53caa78 100644 --- a/drivers/staging/comedi/drivers/ni_labpc_cs.c +++ b/drivers/staging/comedi/drivers/ni_labpc_cs.c @@ -143,7 +143,7 @@ static int labpc_attach(struct comedi_device *dev, struct comedi_devconfig *it) if (!link) return -EIO; iobase = link->io.BasePort1; - irq = link->irq.AssignedIRQ; + irq = link->irq; break; default: printk("bug! couldn't determine board type\n"); @@ -228,10 +228,6 @@ static int labpc_cs_attach(struct pcmcia_device *link) local->link = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = NULL; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -305,9 +301,7 @@ static int labpc_pcmcia_config_loop(struct pcmcia_device *p_dev, } /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= - (CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ); + p_dev->conf.Attributes |= CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -367,16 +361,8 @@ static void labpc_config(struct pcmcia_device *link) goto failed; } - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + if (!link->irq) + goto failed; /* This actually configures the PCMCIA socket -- setting up @@ -399,7 +385,7 @@ static void labpc_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1 + link->io.NumPorts1 - 1); diff --git a/drivers/staging/comedi/drivers/ni_mio_cs.c b/drivers/staging/comedi/drivers/ni_mio_cs.c index dc4849a..a6007ab 100644 --- a/drivers/staging/comedi/drivers/ni_mio_cs.c +++ b/drivers/staging/comedi/drivers/ni_mio_cs.c @@ -272,7 +272,6 @@ static int cs_attach(struct pcmcia_device *link) { link->io.Attributes1 = IO_DATA_PATH_WIDTH_16; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -344,10 +343,8 @@ static void mio_cs_config(struct pcmcia_device *link) return; } - ret = pcmcia_request_irq(link, &link->irq); - if (ret) { - printk("pcmcia_request_irq() returned error: %i\n", ret); - } + if (!link->irq) + dev_info(&link->dev, "no IRQ available\n"); ret = pcmcia_request_configuration(link, &link->conf); @@ -369,7 +366,7 @@ static int mio_cs_attach(struct comedi_device *dev, struct comedi_devconfig *it) dev->driver = &driver_ni_mio_cs; dev->iobase = link->io.BasePort1; - irq = link->irq.AssignedIRQ; + irq = link->irq; printk("comedi%d: %s: DAQCard: io 0x%04lx, irq %u, ", dev->minor, dev->driver->driver_name, dev->iobase, irq); diff --git a/drivers/staging/comedi/drivers/quatech_daqp_cs.c b/drivers/staging/comedi/drivers/quatech_daqp_cs.c index 3325f24..7aa17f9 100644 --- a/drivers/staging/comedi/drivers/quatech_daqp_cs.c +++ b/drivers/staging/comedi/drivers/quatech_daqp_cs.c @@ -1040,10 +1040,6 @@ static int daqp_cs_attach(struct pcmcia_device *link) local->link = link; link->priv = local; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = daqp_interrupt; - /* General socket configuration defaults can go here. In this client, we assume very little, and rely on the CIS for almost @@ -1105,8 +1101,7 @@ static int daqp_pcmcia_config_loop(struct pcmcia_device *p_dev, return -ENODEV; /* Do we need to allocate an interrupt? */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -1144,16 +1139,9 @@ static void daqp_cs_config(struct pcmcia_device *link) goto failed; } - /* - Allocate an interrupt line. Note that this does not assign a - handler to the interrupt, unless the 'Handler' member of the - irq structure is initialized. - */ - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } + ret = pcmcia_request_irq(link, daqp_interrupt); + if (ret) + goto failed; /* This actually configures the PCMCIA socket -- setting up @@ -1180,7 +1168,7 @@ static void daqp_cs_config(struct pcmcia_device *link) printk(KERN_INFO "%s: index 0x%02x", dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Attributes & CONF_ENABLE_IRQ) - printk(", irq %u", link->irq.AssignedIRQ); + printk(", irq %u", link->irq); if (link->io.NumPorts1) printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1 + link->io.NumPorts1 - 1); diff --git a/drivers/staging/netwave/netwave_cs.c b/drivers/staging/netwave/netwave_cs.c index 26477fd..1206b9c 100644 --- a/drivers/staging/netwave/netwave_cs.c +++ b/drivers/staging/netwave/netwave_cs.c @@ -382,10 +382,6 @@ static int netwave_probe(struct pcmcia_device *link) link->io.Attributes2 = IO_DATA_PATH_WIDTH_16; */ link->io.IOAddrLines = 5; - /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - link->irq.Handler = &netwave_interrupt; - /* General socket configuration */ link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; @@ -732,7 +728,7 @@ static int netwave_pcmcia_config(struct pcmcia_device *link) { * Now allocate an interrupt line. Note that this does not * actually assign a handler to the interrupt. */ - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_irq(link, netwave_interrupt); if (ret) goto failed; @@ -767,7 +763,7 @@ static int netwave_pcmcia_config(struct pcmcia_device *link) { ramBase = ioremap(req.Base, 0x8000); priv->ramBase = ramBase; - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; SET_NETDEV_DEV(dev, &link->dev); diff --git a/drivers/staging/wavelan/wavelan_cs.c b/drivers/staging/wavelan/wavelan_cs.c index 04f691d..37fa855 100644 --- a/drivers/staging/wavelan/wavelan_cs.c +++ b/drivers/staging/wavelan/wavelan_cs.c @@ -3850,12 +3850,8 @@ wv_pcmcia_config(struct pcmcia_device * link) if (i != 0) break; - /* - * Now allocate an interrupt line. Note that this does not - * actually assign a handler to the interrupt. - */ - i = pcmcia_request_irq(link, &link->irq); - if (i != 0) + i = pcmcia_request_interrupt(link, wavelan_interrupt); + if (!i) break; /* @@ -3890,7 +3886,7 @@ wv_pcmcia_config(struct pcmcia_device * link) break; /* Feed device with this info... */ - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; netif_start_queue(dev); @@ -4437,10 +4433,6 @@ wavelan_probe(struct pcmcia_device *p_dev) p_dev->io.Attributes1 = IO_DATA_PATH_WIDTH_8; p_dev->io.IOAddrLines = 3; - /* Interrupt setup */ - p_dev->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; - p_dev->irq.Handler = wavelan_interrupt; - /* General socket configuration */ p_dev->conf.Attributes = CONF_ENABLE_IRQ; p_dev->conf.IntType = INT_MEMORY_AND_IO; @@ -4487,7 +4479,6 @@ wavelan_probe(struct pcmcia_device *p_dev) ret = wv_hw_config(dev); if (ret) { - dev->irq = 0; pcmcia_disable_device(p_dev); return ret; } diff --git a/drivers/staging/wlags49_h2/wl_cs.c b/drivers/staging/wlags49_h2/wl_cs.c index 811a8da..b2eb158 100644 --- a/drivers/staging/wlags49_h2/wl_cs.c +++ b/drivers/staging/wlags49_h2/wl_cs.c @@ -157,15 +157,12 @@ static int wl_adapter_attach(struct pcmcia_device *link) link->io.NumPorts1 = HCF_NUM_IO_PORTS; link->io.Attributes1 = IO_DATA_PATH_WIDTH_16; link->io.IOAddrLines = 6; - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING | IRQ_HANDLE_PRESENT; - link->irq.IRQInfo1 = IRQ_INFO2_VALID | IRQ_LEVEL_ID; - link->irq.Handler = &wl_isr; link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 5; link->conf.Present = PRESENT_OPTION; - link->priv = link->irq.Instance = dev; + link->priv = dev; lp = wl_priv(dev); lp->link = link; @@ -319,11 +316,11 @@ void wl_adapter_insert( struct pcmcia_device *link ) link->conf.Attributes |= CONF_ENABLE_IRQ; CS_CHECK(RequestIO, pcmcia_request_io(link, &link->io)); - CS_CHECK(RequestIRQ, pcmcia_request_irq(link, &link->irq)); + CS_CHECK(RequestIRQ, pcmcia_request_irq(link, wl_isr)); CS_CHECK(RequestConfiguration, pcmcia_request_configuration(link, &link->conf)); - dev->irq = link->irq.AssignedIRQ; + dev->irq = link->irq; dev->base_addr = link->io.BasePort1; SET_NETDEV_DEV(dev, &handle_to_dev(link)); diff --git a/drivers/usb/host/sl811_cs.c b/drivers/usb/host/sl811_cs.c index 39d253e..a712788 100644 --- a/drivers/usb/host/sl811_cs.c +++ b/drivers/usb/host/sl811_cs.c @@ -163,8 +163,7 @@ static int sl811_cs_config_check(struct pcmcia_device *p_dev, dflt->vpp1.param[CISTPL_POWER_VNOM]/10000; /* we need an interrupt */ - if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= CONF_ENABLE_IRQ; /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; @@ -197,11 +196,8 @@ static int sl811_cs_config(struct pcmcia_device *link) /* require an IRQ and two registers */ if (!link->io.NumPorts1 || link->io.NumPorts1 < 2) goto failed; - if (link->conf.Attributes & CONF_ENABLE_IRQ) { - ret = pcmcia_request_irq(link, &link->irq); - if (ret) - goto failed; - } else + + if (!link->irq) goto failed; ret = pcmcia_request_configuration(link, &link->conf); @@ -216,12 +212,12 @@ static int sl811_cs_config(struct pcmcia_device *link) dev->node.dev_name, link->conf.ConfigIndex); if (link->conf.Vpp) printk(", Vpp %d.%d", link->conf.Vpp/10, link->conf.Vpp%10); - printk(", irq %d", link->irq.AssignedIRQ); + printk(", irq %d", link->irq); printk(", io 0x%04x-0x%04x", link->io.BasePort1, link->io.BasePort1+link->io.NumPorts1-1); printk("\n"); - if (sl811_hc_init(parent, link->io.BasePort1, link->irq.AssignedIRQ) + if (sl811_hc_init(parent, link->io.BasePort1, link->irq) < 0) { failed: printk(KERN_WARNING "sl811_cs_config failed\n"); @@ -241,10 +237,6 @@ static int sl811_cs_probe(struct pcmcia_device *link) local->p_dev = link; link->priv = local; - /* Initialize */ - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - link->irq.Handler = NULL; - link->conf.Attributes = 0; link->conf.IntType = INT_MEMORY_AND_IO; diff --git a/include/pcmcia/cs.h b/include/pcmcia/cs.h index af61571..a04e6ca 100644 --- a/include/pcmcia/cs.h +++ b/include/pcmcia/cs.h @@ -114,13 +114,6 @@ typedef struct io_req_t { #define IO_DATA_PATH_WIDTH_16 0x08 #define IO_DATA_PATH_WIDTH_AUTO 0x10 -/* For RequestIRQ and ReleaseIRQ */ -typedef struct irq_req_t { - u_int Attributes; - u_int AssignedIRQ; - irq_handler_t Handler; -} irq_req_t; - /* Attributes for RequestIRQ and ReleaseIRQ */ #define IRQ_TYPE 0x03 #define IRQ_TYPE_EXCLUSIVE 0x00 diff --git a/include/pcmcia/ds.h b/include/pcmcia/ds.h index f70c5c1..81b6d86 100644 --- a/include/pcmcia/ds.h +++ b/include/pcmcia/ds.h @@ -90,12 +90,11 @@ struct pcmcia_device { dev_node_t *dev_node; u_int open; io_req_t io; - irq_req_t irq; config_req_t conf; window_handle_t win; /* device setup */ - unsigned int irq_v; /* do not use directly yet */ + unsigned int irq; /* Is the device suspended, or in the process of * being removed? */ @@ -195,7 +194,13 @@ int pcmcia_access_configuration_register(struct pcmcia_device *p_dev, /* device configuration */ int pcmcia_request_io(struct pcmcia_device *p_dev, io_req_t *req); -int pcmcia_request_irq(struct pcmcia_device *p_dev, irq_req_t *req); + +int __must_check __deprecated +pcmcia_request_exclusive_irq(struct pcmcia_device *p_dev, + irq_handler_t handler); +int __must_check pcmcia_request_irq(struct pcmcia_device *p_dev, + irq_handler_t handler); + int pcmcia_request_configuration(struct pcmcia_device *p_dev, config_req_t *req); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index dcf8539..60fddf3 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -142,11 +142,6 @@ static int snd_pdacf_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - /* FIXME: This driver should be updated to allow for dynamic IRQ sharing */ - /* link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; */ - - link->irq.Handler = pdacf_interrupt; link->conf.Attributes = CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; @@ -228,7 +223,7 @@ static int pdacf_config(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_exclusive_irq(link, pdacf_interrupt); if (ret) goto failed; @@ -236,7 +231,7 @@ static int pdacf_config(struct pcmcia_device *link) if (ret) goto failed; - if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq.AssignedIRQ) < 0) + if (snd_pdacf_assign_resources(pdacf, link->io.BasePort1, link->irq) < 0) goto failed; link->dev_node = &pdacf->node; diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7be3b33..4198395 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -161,10 +161,6 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl, link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; - - link->irq.Handler = &snd_vx_irq_handler; - link->conf.Attributes = CONF_ENABLE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; @@ -234,7 +230,7 @@ static int vxpocket_config(struct pcmcia_device *link) if (ret) goto failed; - ret = pcmcia_request_irq(link, &link->irq); + ret = pcmcia_request_exclusive_irq(link, snd_vx_irq_handler); if (ret) goto failed; @@ -245,7 +241,7 @@ static int vxpocket_config(struct pcmcia_device *link) chip->dev = &link->dev; snd_card_set_dev(chip->card, chip->dev); - if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0) + if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq) < 0) goto failed; link->dev_node = &vxp->node; -- 1.6.3.3

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[alsa-devel] [RFC PATCH 06/19] pcmcia: pass FORCED_PULSE parameter in pcmcia_request_configuration()
by Dominik Brodowski 22 Mar '10

22 Mar '10

As it's only used there it makes no sense relying on pcmcia_request_irq(). CC: alsa-devel(a)alsa-project.org Signed-off-by: Dominik Brodowski <linux(a)dominikbrodowski.net> --- drivers/pcmcia/pcmcia_resource.c | 10 ++-------- drivers/staging/comedi/drivers/ni_labpc_cs.c | 5 +++-- include/pcmcia/cs.h | 1 + sound/pcmcia/pdaudiocf/pdaudiocf.c | 6 +++--- 4 files changed, 9 insertions(+), 13 deletions(-) diff --git a/drivers/pcmcia/pcmcia_resource.c b/drivers/pcmcia/pcmcia_resource.c index fb08e80..f629aac 100644 --- a/drivers/pcmcia/pcmcia_resource.c +++ b/drivers/pcmcia/pcmcia_resource.c @@ -425,10 +425,6 @@ static int pcmcia_release_irq(struct pcmcia_device *p_dev, irq_req_t *req) if (c->state & CONFIG_LOCKED) goto out; - if (c->irq.Attributes != req->Attributes) { - dev_dbg(&s->dev, "IRQ attributes must match assigned ones\n"); - goto out; - } if (s->pcmcia_irq != req->AssignedIRQ) { dev_dbg(&s->dev, "IRQ must match assigned one\n"); goto out; @@ -552,9 +548,8 @@ int pcmcia_request_configuration(struct pcmcia_device *p_dev, if (req->Present & PRESENT_IOBASE_0) c->Option |= COR_ADDR_DECODE; } - if (req->Attributes & CONF_ENABLE_IRQ) - if (!(c->irq.Attributes & IRQ_FORCED_PULSE)) - c->Option |= COR_LEVEL_REQ; + if (req->Attributes & CONF_ENABLE_PULSE_IRQ) + c->Option |= COR_LEVEL_REQ; pcmcia_write_cis_mem(s, 1, (base + CISREG_COR)>>1, 1, &c->Option); mdelay(40); } @@ -729,7 +724,6 @@ int pcmcia_request_irq(struct pcmcia_device *p_dev, irq_req_t *req) } } - c->irq.Attributes = req->Attributes; req->AssignedIRQ = irq; p_dev->_irq = 1; diff --git a/drivers/staging/comedi/drivers/ni_labpc_cs.c b/drivers/staging/comedi/drivers/ni_labpc_cs.c index 0b963bb..5a0e5b9 100644 --- a/drivers/staging/comedi/drivers/ni_labpc_cs.c +++ b/drivers/staging/comedi/drivers/ni_labpc_cs.c @@ -229,7 +229,7 @@ static int labpc_cs_attach(struct pcmcia_device *link) link->priv = local; /* Interrupt setup */ - link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING | IRQ_FORCED_PULSE; + link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; link->irq.Handler = NULL; /* @@ -306,7 +306,8 @@ static int labpc_pcmcia_config_loop(struct pcmcia_device *p_dev, /* Do we need to allocate an interrupt? */ if (cfg->irq.IRQInfo1 || dflt->irq.IRQInfo1) - p_dev->conf.Attributes |= CONF_ENABLE_IRQ; + p_dev->conf.Attributes |= + (CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ); /* IO window settings */ p_dev->io.NumPorts1 = p_dev->io.NumPorts2 = 0; diff --git a/include/pcmcia/cs.h b/include/pcmcia/cs.h index 75fa353..af61571 100644 --- a/include/pcmcia/cs.h +++ b/include/pcmcia/cs.h @@ -85,6 +85,7 @@ typedef struct config_req_t { #define CONF_ENABLE_IRQ 0x01 #define CONF_ENABLE_DMA 0x02 #define CONF_ENABLE_SPKR 0x04 +#define CONF_ENABLE_PULSE_IRQ 0x08 #define CONF_VALID_CLIENT 0x100 /* IntType field */ diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index edaa729..dcf8539 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -142,12 +142,12 @@ static int snd_pdacf_probe(struct pcmcia_device *link) link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO; link->io.NumPorts1 = 16; - link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE; + link->irq.Attributes = IRQ_TYPE_EXCLUSIVE; /* FIXME: This driver should be updated to allow for dynamic IRQ sharing */ - /* link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING | IRQ_FORCED_PULSE; */ + /* link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING; */ link->irq.Handler = pdacf_interrupt; - link->conf.Attributes = CONF_ENABLE_IRQ; + link->conf.Attributes = CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ; link->conf.IntType = INT_MEMORY_AND_IO; link->conf.ConfigIndex = 1; link->conf.Present = PRESENT_OPTION; -- 1.6.3.3

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[alsa-devel] [PATCH] sound/oss/vidc.c: change the field used with DMA_ACTIVE
by Julia Lawall 21 Mar '10

21 Mar '10

From: Julia Lawall <julia(a)diku.dk> The constant DMA_ACTIVE is defined with the dma_buffparams structure rather than with the audio_operations structure. Takashi Iwai suggested that the dmap_out field of the audio_operations structure should be used instead. This is not tested. Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/oss/vidc.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index 725fef0..a4127ba 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -363,13 +363,13 @@ static void vidc_audio_trigger(int dev, int enable_bits) struct audio_operations *adev = audio_devs[dev]; if (enable_bits & PCM_ENABLE_OUTPUT) { - if (!(adev->flags & DMA_ACTIVE)) { + if (!(adev->dmap_out->flags & DMA_ACTIVE)) { unsigned long flags; local_irq_save(flags); /* prevent recusion */ - adev->flags |= DMA_ACTIVE; + adev->dmap_out->flags |= DMA_ACTIVE; dma_interrupt = vidc_audio_dma_interrupt; vidc_sound_dma_irq(0, NULL);

2 1

[alsa-devel] [PATCH] ALSA: hda - Add PCI quirk for HP dv6-1110ax.
by Kunal Gangakhedkar 21 Mar '10

21 Mar '10

Adding this PCI quirk fixes the board config detection. This also fixes jack sensing by using "hp_detect=1" via properly detected board config. Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c416bb..c4be3fa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1730,6 +1730,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- 1.6.6.60.gc2ff1

2 1

[alsa-devel] PATCH: Fix snd-hda-intel boot hang on ASUS A8N-VM
by Brian Lindholm 21 Mar '10

21 Mar '10

This patch fixes a regression in 2.6.33.x. Was working in 2.6.32.x. Symptom: Hang on boot at "HDA Intel". Boots only if on-board sound is disabled or if "snd-hda-intel.enable_msi=0" is appended to boot parameters. Machine: ASUS A8N-VM with Athlon-X2 processor. Relevant output from lspci -nn: 00:10.1 Audio device [0403]: nVidia Corporation MCP51 High Definition Audio [10de:026c] (rev a2) Relevant output from alsa-info.sh: 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) 00:10.1 0403: 10de:026c (rev a2) Subsystem: 10de:cb84 Proposed patch (which has been confirmed to allow boot, with sound active): diff -dru linux-2.6.33-orig/sound/pci/hda/hda_intel.c linux-2.6.33/sound/pci/hda/hda_intel.c --- linux-2.6.33-orig/sound/pci/hda/hda_intel.c 2010-03-16 17:33:23.487938000 -0400 +++ linux-2.6.33/sound/pci/hda/hda_intel.c 2010-03-16 17:40:36.141225010 -0400 @@ -2354,6 +2354,7 @@ static void __devinit check_probe_mask(s static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ + SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS", 0), /* nvidia */ {} }; Signed-off-by: Brian E Lindholm <brian_lindholm(a)users.sourceforge.net>

3 3

[alsa-devel] question about sound/oss/vidc.c
by Julia Lawall 20 Mar '10

20 Mar '10

In sound/oss/vidc.c, the function vidc_audio_trigger contains thefollowing code: struct audio_operations *adev = audio_devs[dev]; if (enable_bits & PCM_ENABLE_OUTPUT) { if (!(adev->flags & DMA_ACTIVE)) { unsigned long flags; local_irq_save(flags); /* prevent recusion */ adev->flags |= DMA_ACTIVE; ... } } I wonder if the use of DMA_ACTIVE is correct? This constant is defined inthe file sound/oss/dev_table.h within the definition of the structure dma_buffparms. But here it is being used with the structure audio_operations. The definition of audio_operations also contains the definitions of some constants. Indeed, one of them, DMA_DUPLEX has the same value as DMA_ACTIVE. thanks, julia

3 3

[alsa-devel] Problem about building alsa driver for wm8974 on i.mx27 platform
by jjinfo＠nudt.edu.cn 20 Mar '10

20 Mar '10

Hi, I'm trying to build alsa-1.0.22 driver for my wm8974 codec on i.Mx27 platform. I need to cross compile it on x86 platform for i.Mx27. Here is my configure options: ./configure --wit-cross=arm-linux- --with-kernel=<path-to-ARM-kernel> --with-build=<path-to-kernel> When configure process completes, a "toplevel.config" file is generated. But in the toplevel.config file, the following config options are not set by the configure process( which prevent wm8974 driver from being built): CONFIG_SND_MX1_MX2_SOC= CONFIG_ARCH_MX2= CONFIG_ARCH_MX1= CONFIG_SND_MXC_SOC_SSI= CONFIG_SND_SOC_MX27VIS_WM8974= CONFIG_MACH_MX27= CONFIG_MACH_IMX27_VISSTRIM_M10= Since the default value of "--with-cards" is "all", why these options are not set? (Event I specifiy --with-cards=soc, mx1-mx2-soc, soc-mx27vis-wm8974, these options are still not set in toplevel.config.) Any idea? Regards, Jie

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[alsa-devel] Question about support for TLV320AIC3104 codec
by jjinfo＠nudt.edu.cn 20 Mar '10

20 Mar '10

Hi all, Does the codec driver of tlv320aic3x.c/.h in the latest release of alsa-1.0.22 support TLV320AIC3204 codec? Regards, Jie

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[PATCH] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 20 Mar '10

20 Mar '10

Enable wm8753 codec in aspenite via ssp interface. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/aspenite.c | 218 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 229 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/aspenite.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 286d52a..0f10ee2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -113,6 +113,15 @@ 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_PXA2XX_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 a74e6c9..74054aa 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o obj-$(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 @@ -38,6 +39,7 @@ 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..733c95c --- /dev/null +++ b/sound/soc/pxa/aspenite.c @@ -0,0 +1,218 @@ +/* + * 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 L", NULL), + SND_SOC_DAPM_SPK("Headset Speaker R", 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 L", NULL, "LOUT2"}, + {"Headset Speaker L", NULL, "OUT4"}, + {"Headset Speaker R", NULL, "OUT3"}, + {"Headset Speaker R", 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 L"), + SOC_DAPM_PIN_SWITCH("Headset Speaker R"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Line LIN"), + SOC_DAPM_PIN_SWITCH("Line RIN"), +}; + +/* Seek the index of MCLK configuration table */ +static int pxa168_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 */ +static int pxa168_get_mclk(int i) +{ + if ((i < 0) || (i >= ARRAY_SIZE(mclk_conf))) + return -EINVAL; + return mclk_conf[i].mclk; +} + +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 = pxa168_seek_mclk_conf(rate, width, channel); + if (ret < 0) + return ret; + index = ret; + mclk = pxa168_get_mclk(ret); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + 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, 0, index, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aspenite_hifi_ops = { + .hw_params = aspenite_hifi_hw_params, +}; + +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)); + + 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"); -- 1.5.6.5

1 0

[PATCH] ASoC: enable wm8753 in aspenite
by Haojian Zhuang 20 Mar '10

20 Mar '10

Enable wm8753 codec in aspenite via ssp interface. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 9 ++ sound/soc/pxa/Makefile | 2 + sound/soc/pxa/aspenite.c | 190 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 201 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/aspenite.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 286d52a..0f10ee2 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -113,6 +113,15 @@ 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_PXA2XX_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 a74e6c9..74054aa 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -12,6 +12,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o obj-$(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 @@ -38,6 +39,7 @@ 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..f82acdc --- /dev/null +++ b/sound/soc/pxa/aspenite.c @@ -0,0 +1,190 @@ +/* + * 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 L", NULL), + SND_SOC_DAPM_SPK("Headset Speaker R", 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 L", NULL, "LOUT2"}, + {"Headset Speaker L", NULL, "OUT4"}, + {"Headset Speaker R", NULL, "OUT3"}, + {"Headset Speaker R", 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 L"), + SOC_DAPM_PIN_SWITCH("Headset Speaker R"), + 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; + int mclk, ret; + + mclk = pxa168_query_mclk(substream, params); + if (mclk < 0) + return mclk; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS); + 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, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aspenite_hifi_ops = { + .hw_params = aspenite_hifi_hw_params, +}; + +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)); + + 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"); -- 1.5.6.5

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 19 Mar '10

19 Mar '10

The following changes since commit f4bee1bb0083869b1b7d73c6eda7703a7aae0506: Barry Song (1): ASoC: soc-cache: let reg be AND'ed by 0xff instead of data buffer for 8_8 mode 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 Barry Song (2): ASoC: rename ad1938 to ad193x and add support for ad1936/7/8/9 ASoC: change bf5xx-ad1938 machine driver to bf5xx-ad193x machine driver Guennadi Liakhovetski (2): ASoC: SIU driver shall select FW_LOADER ASoC: remove a card from the list, if instantiation failed Jorge Eduardo Candelaria (1): ASoC: McPDM: Use tabs for indentation Mark Brown (9): ALSA: Rename jack switch table in preparation for button support ALSA: Add support for key reporting via the jack interface ASoC: Correct typoed Mic2 connections on 1133-EV1 board ASoC: Bail out of wm_hubs DC servo if calibration fails ASoC: Unexport AD193x bus probe/remove functions Merge branch 'topic/jack' into for-2.6.35 ASoC: Hook up microphone jack detection on 1133-EV1 board ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream Merge branch 'for-2.6.34' into for-2.6.35 Misael Lopez Cruz (1): ASoC: TWL6040: Add twl6040 codec driver Olaya, Margarita (1): OMAP4: PMIC: Rename twl6030_codec as twl6040_codec Peter Ujfalusi (3): ASoC: omap-mcbsp: Add support for Left Justified format ASoC: tlv320dac33: Fix DSP modes ASoC: tlv320dac33: Internal clocking changes drivers/mfd/twl-core.c | 4 +- include/linux/i2c/twl.h | 6 +- include/sound/jack.h | 8 + include/sound/soc-dai.h | 18 +- include/sound/soc.h | 1 + sound/core/jack.c | 71 +- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 6 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/blackfin/Makefile | 4 +- .../blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} | 66 +- sound/soc/codecs/Kconfig | 8 +- sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/ad1938.c | 521 --------- sound/soc/codecs/ad1938.h | 100 -- sound/soc/codecs/ad193x.c | 558 +++++++++ sound/soc/codecs/ad193x.h | 81 ++ sound/soc/codecs/tlv320dac33.c | 10 +- sound/soc/codecs/twl6040.c | 1227 ++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++ sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/davinci/davinci-i2s.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 3 +- sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-vcif.c | 3 +- sound/soc/imx/imx-pcm-dma-mx2.c | 8 +- sound/soc/imx/imx-ssi.c | 7 +- sound/soc/imx/wm1133-ev1.c | 19 +- sound/soc/omap/mcpdm.c | 548 +++++----- sound/soc/omap/omap-mcbsp.c | 16 +- sound/soc/omap/omap-mcpdm.c | 3 +- sound/soc/omap/omap-pcm.c | 4 +- sound/soc/pxa/pxa-ssp.c | 23 +- sound/soc/pxa/pxa2xx-ac97.c | 17 +- sound/soc/pxa/pxa2xx-i2s.c | 7 +- sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/s3c24xx/s3c-ac97.c | 21 +- sound/soc/s3c24xx/s3c-dma.c | 4 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 13 +- sound/soc/s3c24xx/s3c-pcm.c | 7 +- sound/soc/s3c24xx/s3c2412-i2s.c | 7 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 19 +- sound/soc/s6000/s6000-i2s.c | 3 +- sound/soc/s6000/s6000-pcm.c | 40 +- sound/soc/sh/Kconfig | 1 + sound/soc/soc-core.c | 27 +- 46 files changed, 2611 insertions(+), 1049 deletions(-) rename sound/soc/blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} (65%) delete mode 100644 sound/soc/codecs/ad1938.c delete mode 100644 sound/soc/codecs/ad1938.h create mode 100644 sound/soc/codecs/ad193x.c create mode 100644 sound/soc/codecs/ad193x.h create mode 100755 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 19 Mar '10

19 Mar '10

The following changes since commit 83f40d31ca3fba3627a289e262b9423e21e1f01a: Takashi Iwai (1): Merge branch 'fix/asoc' into for-next 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: Bail out of wm_hubs DC servo if calibration fails Peter Ujfalusi (2): ASoC: tlv320dac33: Fix DSP modes ASoC: tlv320dac33: Internal clocking changes sound/soc/codecs/tlv320dac33.c | 10 +++------- sound/soc/codecs/wm_hubs.c | 2 +- 2 files changed, 4 insertions(+), 8 deletions(-)

2 1

[alsa-devel] [PATCH v2] ASoC: remove a card from the list, if instantiation failed
by Guennadi Liakhovetski 19 Mar '10

19 Mar '10

If instantiation of a card failed, we still have to remove it from the card list on unregistration. This fixes an Oops on Migo-R, triggering, when after a failed firmware load attempt the driver modules are removed and re-inserted again. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Mark, is this more like what you mean? diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..f63b59a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1232,26 +1232,25 @@ static int soc_remove(struct platform_device *pdev) struct snd_soc_platform *platform = card->platform; struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (!card->instantiated) - return 0; + if (card->instantiated) { + run_delayed_work(&card->delayed_work); - run_delayed_work(&card->delayed_work); + if (platform->remove) + platform->remove(pdev); - if (platform->remove) - platform->remove(pdev); + if (codec_dev->remove) + codec_dev->remove(pdev); - if (codec_dev->remove) - codec_dev->remove(pdev); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + if (cpu_dai->remove) + cpu_dai->remove(pdev, cpu_dai); + } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + if (card->remove) + card->remove(pdev); } - if (card->remove) - card->remove(pdev); - snd_soc_unregister_card(card); return 0;

2 2

[alsa-devel] Are ALSA io plugins implemented as threads or does ALSA use fork() ?
by Stefan Schoenleitner 19 Mar '10

19 Mar '10

Hello, from what I have learned so far ALSA io plugins are implemented as shared libraries with their entry point defined with the SND_PCM_PLUGIN_DEFINE_FUNC() macro. However, each plugin instance can only work for either the SND_PCM_STREAM_PLAYBACK or SND_PCM_STREAM_CAPTURE stream. Hence this is why io plugins supporting playback and capture usually set up their callback functions depending on what type of stream is used. For example in the pulseaudio io plugin, this is done with: pcm->io.callback = stream == SND_PCM_STREAM_PLAYBACK ? &pulse_playback_callback : &pulse_capture_callback; * If such a plugin is used for capture AND playback at the same time, it seems that two instances of the plugin need to be running, right ? * Are these instances running as separate processes (i.e. fork()) or are they implemented as threads ? How is this implemented internally in the ALSA code ? In my plugin I'm planning that both the playback and the capture plugin instances talk to a common data processing thread over shared buffers. The reason for this is that the data processing threads needs to have access to the capture and playback samples *at the same time*. I'm guessing that the plugin instances run with different PIDs so that I would need to use common IPC mechanisms in order to let them communicate with each other, right ? Hence my first approach would be to set up a common shared memory together with a semaphore so that only one process can access the shared memory at a time. Any better ideas ? cheers, Stefan

2 2

[alsa-devel] [PATCH v2 2/4] change bf5xx-ad1938 machine driver to bf5xx-ad193x machine driver
by Barry Song 19 Mar '10

19 Mar '10

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 9 ++- sound/soc/blackfin/Makefile | 4 +- .../blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} | 66 ++++++++++---------- 3 files changed, 40 insertions(+), 39 deletions(-) rename sound/soc/blackfin/{bf5xx-ad1938.c => bf5xx-ad193x.c} (65%) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 97f1a25..8ef2502 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -49,13 +49,14 @@ config SND_BF5XX_SOC_AD1836 help Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. -config SND_BF5XX_SOC_AD1938 - tristate "SoC AD1938 Audio support for Blackfin" +config SND_BF5XX_SOC_AD193X + tristate "SoC AD193X Audio support for Blackfin" depends on SND_BF5XX_TDM select SND_BF5XX_SOC_TDM - select SND_SOC_AD1938 + select SND_SOC_AD193X help - Say Y if you want to add support for AD1938 codec on Blackfin. + Say Y if you want to add support for AD193X codec on Blackfin. + This driver supports AD1936, AD1937, AD1938 and AD1939. config SND_BF5XX_AC97 tristate "SoC AC97 Audio for the ADI BF5xx chip" diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 87e3042..49af3f3 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -20,10 +20,10 @@ snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o -snd-ad1938-objs := bf5xx-ad1938.o +snd-ad193x-objs := bf5xx-ad193x.o obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o -obj-$(CONFIG_SND_BF5XX_SOC_AD1938) += snd-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad193x.c similarity index 65% rename from sound/soc/blackfin/bf5xx-ad1938.c rename to sound/soc/blackfin/bf5xx-ad193x.c index 2ef1e50..b8c9060 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -1,9 +1,9 @@ /* - * File: sound/soc/blackfin/bf5xx-ad1938.c + * File: sound/soc/blackfin/bf5xx-ad193x.c * Author: Barry Song <Barry.Song(a)analog.com> * * Created: Thur June 4 2009 - * Description: Board driver for ad1938 sound chip + * Description: Board driver for ad193x sound chip * * Bugs: Enter bugs at http://blackfin.uclinux.org/ * @@ -38,15 +38,15 @@ #include <asm/dma.h> #include <asm/portmux.h> -#include "../codecs/ad1938.h" +#include "../codecs/ad193x.h" #include "bf5xx-sport.h" #include "bf5xx-tdm-pcm.h" #include "bf5xx-tdm.h" -static struct snd_soc_card bf5xx_ad1938; +static struct snd_soc_card bf5xx_ad193x; -static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) +static int bf5xx_ad193x_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -55,7 +55,7 @@ static int bf5xx_ad1938_startup(struct snd_pcm_substream *substream) return 0; } -static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, +static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -89,61 +89,61 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops bf5xx_ad1938_ops = { - .startup = bf5xx_ad1938_startup, - .hw_params = bf5xx_ad1938_hw_params, +static struct snd_soc_ops bf5xx_ad193x_ops = { + .startup = bf5xx_ad193x_startup, + .hw_params = bf5xx_ad193x_hw_params, }; -static struct snd_soc_dai_link bf5xx_ad1938_dai = { - .name = "ad1938", - .stream_name = "AD1938", +static struct snd_soc_dai_link bf5xx_ad193x_dai = { + .name = "ad193x", + .stream_name = "AD193X", .cpu_dai = &bf5xx_tdm_dai, - .codec_dai = &ad1938_dai, - .ops = &bf5xx_ad1938_ops, + .codec_dai = &ad193x_dai, + .ops = &bf5xx_ad193x_ops, }; -static struct snd_soc_card bf5xx_ad1938 = { - .name = "bf5xx_ad1938", +static struct snd_soc_card bf5xx_ad193x = { + .name = "bf5xx_ad193x", .platform = &bf5xx_tdm_soc_platform, - .dai_link = &bf5xx_ad1938_dai, + .dai_link = &bf5xx_ad193x_dai, .num_links = 1, }; -static struct snd_soc_device bf5xx_ad1938_snd_devdata = { - .card = &bf5xx_ad1938, - .codec_dev = &soc_codec_dev_ad1938, +static struct snd_soc_device bf5xx_ad193x_snd_devdata = { + .card = &bf5xx_ad193x, + .codec_dev = &soc_codec_dev_ad193x, }; -static struct platform_device *bfxx_ad1938_snd_device; +static struct platform_device *bfxx_ad193x_snd_device; -static int __init bf5xx_ad1938_init(void) +static int __init bf5xx_ad193x_init(void) { int ret; - bfxx_ad1938_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1938_snd_device) + bfxx_ad193x_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad193x_snd_device) return -ENOMEM; - platform_set_drvdata(bfxx_ad1938_snd_device, &bf5xx_ad1938_snd_devdata); - bf5xx_ad1938_snd_devdata.dev = &bfxx_ad1938_snd_device->dev; - ret = platform_device_add(bfxx_ad1938_snd_device); + platform_set_drvdata(bfxx_ad193x_snd_device, &bf5xx_ad193x_snd_devdata); + bf5xx_ad193x_snd_devdata.dev = &bfxx_ad193x_snd_device->dev; + ret = platform_device_add(bfxx_ad193x_snd_device); if (ret) - platform_device_put(bfxx_ad1938_snd_device); + platform_device_put(bfxx_ad193x_snd_device); return ret; } -static void __exit bf5xx_ad1938_exit(void) +static void __exit bf5xx_ad193x_exit(void) { - platform_device_unregister(bfxx_ad1938_snd_device); + platform_device_unregister(bfxx_ad193x_snd_device); } -module_init(bf5xx_ad1938_init); -module_exit(bf5xx_ad1938_exit); +module_init(bf5xx_ad193x_init); +module_exit(bf5xx_ad193x_exit); /* Module information */ MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD1938 board driver"); +MODULE_DESCRIPTION("ALSA SoC AD193X board driver"); MODULE_LICENSE("GPL"); -- 1.5.6.3

3 2

[alsa-devel] [PATCHv6 2/2][RESEND] ASoC: TWL6040: Add twl6040 codec driver
by Olaya, Margarita 19 Mar '10

19 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6040 codec driver. The TWL6040 codec uses a proprietary 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 TWL6040 codec supports power-up/down manual and automatic sequence. Manual sequence is done through a specific register writes sequence. Automatic sequence is done when the codec is powered-up through the external AUDPWRON line. The completion of the sequence is signaled through the audio interrupt. TWL6040 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 (for 44.1 KHz and 48 KHz respectively) - The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. 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/twl6040.c | 1227 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 4 files changed, 1374 insertions(+), 0 deletions(-) create mode 100755 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 16c47ed..398cbb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ 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_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -168,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6981777..98bd10c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ 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-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -85,6 +86,7 @@ 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-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(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/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100755 index 0000000..0e4dce7 --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1227 @@ +/* + * ALSA SoC TWL6040 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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x60, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x4A, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ + 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x00, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_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, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(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 = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + 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 + */ +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 *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_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", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* 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_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_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, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_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 twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + 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 */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + 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 twl6040_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 twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + 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); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* + * 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 twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_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 |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_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 |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_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); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +#ifdef CONFIG_PM +static int twl6040_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; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} +#else +#define twl6040_suspend NULL +#define twl6040_resume NULL +#endif + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_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, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_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 twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_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_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_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); + init_completion(&priv->ready); + + 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; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_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); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 0000000..c472070 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ -- 1.6.1.3

3 5

[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: DSP mode and clocking updates
by Peter Ujfalusi 19 Mar '10

19 Mar '10

Hello, tlv320dac33 can not support DSP_B mode since the data delay can not be set to be actually 0. Only DSP_A mode can be supported, so fix the DSP_A mode, and remove the DSP_B. The original internal clock distribution config has been found to be not optimal, so fix that as well. Peter --- Peter Ujfalusi (2): ASoC: tlv320dac33: Fix DSP modes ASoC: tlv320dac33: Internal clocking changes sound/soc/codecs/tlv320dac33.c | 10 +++------- 1 files changed, 3 insertions(+), 7 deletions(-)

3 4

[alsa-devel] [PATCH] ASoC: McPDM: Use tabs for indentation
by Jorge Eduardo Candelaria 19 Mar '10

19 Mar '10

Indentation in initial support for McPDM driver was converted to spaces. Use tabs to comply with open source coding-style. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/mcpdm.c | 548 ++++++++++++++++++++++++------------------------ 1 files changed, 274 insertions(+), 274 deletions(-) diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index ad8df6c..479d3b6 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -1,5 +1,5 @@ /* - * mcpdm.c -- McPDM interface driver + * mcpdm.c -- McPDM interface driver * * Author: Jorge Eduardo Candelaria <x0107209(a)ti.com> * Copyright (C) 2009 - Texas Instruments, Inc. @@ -38,46 +38,46 @@ static struct omap_mcpdm *mcpdm; static inline void omap_mcpdm_write(u16 reg, u32 val) { - __raw_writel(val, mcpdm->io_base + reg); + __raw_writel(val, mcpdm->io_base + reg); } static inline int omap_mcpdm_read(u16 reg) { - return __raw_readl(mcpdm->io_base + 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"); + 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"); } /* @@ -86,26 +86,26 @@ static void omap_mcpdm_reg_dump(void) */ static void omap_mcpdm_reset_capture(int reset) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (reset) - ctrl |= SW_UP_RST; - else - ctrl &= ~SW_UP_RST; + if (reset) + ctrl |= SW_UP_RST; + else + ctrl &= ~SW_UP_RST; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } static void omap_mcpdm_reset_playback(int reset) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (reset) - ctrl |= SW_DN_RST; - else - ctrl &= ~SW_DN_RST; + if (reset) + ctrl |= SW_DN_RST; + else + ctrl &= ~SW_DN_RST; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -114,14 +114,14 @@ static void omap_mcpdm_reset_playback(int reset) */ void omap_mcpdm_start(int stream) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (stream) - ctrl |= mcpdm->up_channels; - else - ctrl |= mcpdm->dn_channels; + if (stream) + ctrl |= mcpdm->up_channels; + else + ctrl |= mcpdm->dn_channels; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -130,14 +130,14 @@ void omap_mcpdm_start(int stream) */ void omap_mcpdm_stop(int stream) { - int ctrl = omap_mcpdm_read(MCPDM_CTRL); + int ctrl = omap_mcpdm_read(MCPDM_CTRL); - if (stream) - ctrl &= ~mcpdm->up_channels; - else - ctrl &= ~mcpdm->dn_channels; + if (stream) + ctrl &= ~mcpdm->up_channels; + else + ctrl &= ~mcpdm->dn_channels; - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); } /* @@ -146,38 +146,38 @@ void omap_mcpdm_stop(int stream) */ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) { - int irq_mask = 0; - int ctrl; + int irq_mask = 0; + int ctrl; - if (!uplink) - return -EINVAL; + if (!uplink) + return -EINVAL; - mcpdm->uplink = uplink; + mcpdm->uplink = uplink; - /* Enable irq request generation */ - irq_mask |= uplink->irq_mask & MCPDM_UPLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + /* 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; + /* Configure uplink threshold */ + if (uplink->threshold > UP_THRES_MAX) + uplink->threshold = UP_THRES_MAX; - omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); + omap_mcpdm_write(MCPDM_FIFO_CTRL_UP, uplink->threshold); - /* Configure DMA controller */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_UP_ENABLE); + /* 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; + /* 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); + /* Uplink channels */ + mcpdm->up_channels = uplink->channels & (PDM_UP_MASK | PDM_STATUS_MASK); - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); - return 0; + return 0; } /* @@ -186,38 +186,38 @@ int omap_mcpdm_capture_open(struct omap_mcpdm_link *uplink) */ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) { - int irq_mask = 0; - int ctrl; + int irq_mask = 0; + int ctrl; - if (!downlink) - return -EINVAL; + if (!downlink) + return -EINVAL; - mcpdm->downlink = downlink; + mcpdm->downlink = downlink; - /* Enable irq request generation */ - irq_mask |= downlink->irq_mask & MCPDM_DOWNLINK_IRQ_MASK; - omap_mcpdm_write(MCPDM_IRQENABLE_SET, irq_mask); + /* 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; + /* Configure uplink threshold */ + if (downlink->threshold > DN_THRES_MAX) + downlink->threshold = DN_THRES_MAX; - omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); + omap_mcpdm_write(MCPDM_FIFO_CTRL_DN, downlink->threshold); - /* Enable DMA request generation */ - omap_mcpdm_write(MCPDM_DMAENABLE_SET, DMA_DN_ENABLE); + /* 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; + /* 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); + /* Downlink channels */ + mcpdm->dn_channels = downlink->channels & (PDM_DN_MASK | PDM_CMD_MASK); - omap_mcpdm_write(MCPDM_CTRL, ctrl); + omap_mcpdm_write(MCPDM_CTRL, ctrl); - return 0; + return 0; } /* @@ -226,24 +226,24 @@ int omap_mcpdm_playback_open(struct omap_mcpdm_link *downlink) */ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) { - int irq_mask = 0; + int irq_mask = 0; - if (!uplink) - return -EINVAL; + 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 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); + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_UP_ENABLE); - /* Clear Downlink channels */ - mcpdm->up_channels = 0; + /* Clear Downlink channels */ + mcpdm->up_channels = 0; - mcpdm->uplink = NULL; + mcpdm->uplink = NULL; - return 0; + return 0; } /* @@ -252,124 +252,124 @@ int omap_mcpdm_capture_close(struct omap_mcpdm_link *uplink) */ int omap_mcpdm_playback_close(struct omap_mcpdm_link *downlink) { - int irq_mask = 0; + int irq_mask = 0; - if (!downlink) - return -EINVAL; + 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 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); + /* Disable DMA request generation */ + omap_mcpdm_write(MCPDM_DMAENABLE_CLR, DMA_DN_ENABLE); - /* clear Downlink channels */ - mcpdm->dn_channels = 0; + /* clear Downlink channels */ + mcpdm->dn_channels = 0; - mcpdm->downlink = NULL; + mcpdm->downlink = NULL; - return 0; + 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; + 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; + int ret; - clk_enable(mcpdm->clk); + clk_enable(mcpdm->clk); - spin_lock(&mcpdm->lock); + 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; + 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); + spin_unlock(&mcpdm->lock); - /* Disable lines while request is ongoing */ - omap_mcpdm_write(MCPDM_CTRL, 0x00); + /* 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; - } + 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; + return 0; err: - clk_disable(mcpdm->clk); - return ret; + 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); + 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 @@ -377,108 +377,108 @@ void omap_mcpdm_free(void) */ int omap_mcpdm_set_offset(int offset1, int offset2) { - int offset; + int offset; - if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) - return -EINVAL; + if ((offset1 > DN_OFST_MAX) || (offset2 > DN_OFST_MAX)) + return -EINVAL; - offset = (offset1 << DN_OFST_RX1) | (offset2 << DN_OFST_RX2); + 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 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; + /* 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); + omap_mcpdm_write(MCPDM_DN_OFFSET, offset); - return 0; + 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; + 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); + iounmap(mcpdm->io_base); err_resource: - kfree(mcpdm); + kfree(mcpdm); exit: - return ret; + return ret; } static int __devexit omap_mcpdm_remove(struct platform_device *pdev) { - struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); + struct omap_mcpdm *mcpdm_ptr = platform_get_drvdata(pdev); - platform_set_drvdata(pdev, NULL); + platform_set_drvdata(pdev, NULL); - clk_put(mcpdm_ptr->clk); + clk_put(mcpdm_ptr->clk); - iounmap(mcpdm_ptr->io_base); + iounmap(mcpdm_ptr->io_base); - mcpdm_ptr->clk = NULL; - mcpdm_ptr->free = 0; - mcpdm_ptr->dev = NULL; + mcpdm_ptr->clk = NULL; + mcpdm_ptr->free = 0; + mcpdm_ptr->dev = NULL; - kfree(mcpdm_ptr); + kfree(mcpdm_ptr); - return 0; + return 0; } static struct platform_driver omap_mcpdm_driver = { - .probe = omap_mcpdm_probe, - .remove = __devexit_p(omap_mcpdm_remove), - .driver = { - .name = "omap-mcpdm", - }, + .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); + return platform_driver_register(&omap_mcpdm_driver); } arch_initcall(omap_mcpdm_init); -- 1.6.0.4

3 2

[alsa-devel] [PATCH v2] ASoC: omap-mcbsp: Add support for Left Justified format
by Peter Ujfalusi 19 Mar '10

19 Mar '10

Basic support for Left Justified coding for OMAP McBSP. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 12 +++++++++++- 1 files changed, 11 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2952fb0..0199e95 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -331,7 +331,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || + format == SND_SOC_DAIFMT_LEFT_J)) { /* Use dual-phase frames */ regs->rcr2 |= RPHASE; regs->xcr2 |= XPHASE; @@ -376,6 +377,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID((framesize >> 1) - 1); break; @@ -427,6 +429,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_LEFT_J: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + regs->spcr1 |= RJUST(2); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); -- 1.7.0.2

4 3

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Add support for Left Justified format
by Peter Ujfalusi 19 Mar '10

19 Mar '10

Basic support for Left Justified coding for OMAP McBSP. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 10 +++++++++- 1 files changed, 9 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2952fb0..d315cc4 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -331,7 +331,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || + format == SND_SOC_DAIFMT_LEFT_J)) { /* Use dual-phase frames */ regs->rcr2 |= RPHASE; regs->xcr2 |= XPHASE; @@ -376,6 +377,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID((framesize >> 1) - 1); break; @@ -427,6 +429,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_LEFT_J: + /* 0-bit data delay */ + regs->rcr2 |= RDATDLY(0); + regs->xcr2 |= XDATDLY(0); + regs->spcr1 |= RJUST(2); + break; case SND_SOC_DAIFMT_DSP_A: /* 1-bit data delay */ regs->rcr2 |= RDATDLY(1); -- 1.7.0.2

2 4

[alsa-devel] HDMI output with headphone jack sense possible on Conexant CX20561?
by Andrew Udvare 18 Mar '10

18 Mar '10

My audio codec is Conexant CX20561 (Hermosa) (haven't been able to find anything about 'Hermosa' on the Wiki). I'd like to be able to do a few things which I have not been able to do with tutorials I found on the web: - Whenever I plug in headphones, HDMI output is muted and headphones are enabled - Right now, the way I do this is by manually editing /etc/asound.conf to have the following (commented out makes headphones work, not commented makes HDMI work without any headphone/laptop speaker output) /etc/asound.conf: defaults.pcm.card 0 defaults.pcm.device 3 where 3 is my HDMI (IEC958 2; no idea what IEC958 1 is for) output. So I have a switcher script that does the commenting and uncommenting. Would be nice if this could be automatic (even if alsa had to run such a script on detecting headphones). - Route audio output (PCM) to input (for mixer recording) (Intel HDA cannot do this apparently by default (thanks DRM), needs some hacking around?) Information from alsa-info.sh: http://www.alsa- project.org/db/?f=107f3f6012e68f04d3562d5e11aeb052a1454884 Are my model options correct? And if it's important to know, I build ALSA in-kernel. Is this not recommended? I know at the very least it makes it such that I have to reboot to make any new options to take effect. Thanks all -- Andrew Udvare audvare(a)gmail.com 978-870-5908 www.andyudvare.com

1 0

[alsa-devel] [PATCHv6 2/2] ASoC: TWL6040: Add twl6040 codec driver
by Olaya, Margarita 18 Mar '10

18 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6040 codec driver. The TWL6040 codec uses a proprietary 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 TWL6040 codec supports power-up/down manual and automatic sequence. Manual sequence is done through a specific register writes sequence. Automatic sequence is done when the codec is powered-up through the external AUDPWRON line. The completion of the sequence is signaled through the audio interrupt. TWL6040 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 (for 44.1 KHz and 48 KHz respectively) - The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. 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/twl6040.c | 1227 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 4 files changed, 1374 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 16c47ed..398cbb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ 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_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -168,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6981777..98bd10c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ 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-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -85,6 +86,7 @@ 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-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(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/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100644 index 0000000..b90bc60 --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1227 @@ +/* + * ALSA SoC TWL6040 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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x60, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x4A, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C - No input selected on Left Mic */ + 0x18, /* TWL6040_MICRCTL 0x0D - No input selected on Right Mic */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x00, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x00, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x00, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_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, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(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 = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + 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 + */ +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 *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_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", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* 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_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_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, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_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 twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + 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 */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + 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 twl6040_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 twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + 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); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* + * 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 twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_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 |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_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 |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_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); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +#ifdef CONFIG_PM +static int twl6040_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; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} +#else +#define twl6040_suspend NULL +#define twl6040_resume NULL +#endif + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_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, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_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 twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_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_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_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); + init_completion(&priv->ready); + + 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; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_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); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 0000000..c472070 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ -- 1.6.1.3

3 2

[alsa-devel] [PATCHv2 1/2] OMAP4: PMIC: Rename twl6030_codec as twl6040_codec
by Olaya, Margarita 18 Mar '10

18 Mar '10

Correction for chips: twl6030 is Phoenix Power chip twl6040 is Phoenix Audio chip Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- drivers/mfd/twl-core.c | 4 ++-- include/linux/i2c/twl.h | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 562cd49..720e099 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -109,7 +109,7 @@ #endif #if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ - defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) + defined(CONFIG_SND_SOC_TWL6040) || defined(CONFIG_SND_SOC_TWL6040_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -708,7 +708,7 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) /* 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", + child = add_child(sub_chip_id, "twl6040_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 fb6784e..ebd90ce 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -569,9 +569,9 @@ struct twl4030_codec_data { 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 */ + /* twl6040 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { -- 1.6.1.3

4 4

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 18 Mar '10

18 Mar '10

The following changes since commit dd76769dd53bdb804a4b02a3eb256f4740f78720: Mark Brown (1): ASoC: Refresh WM8750 bias management 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 Barry Song (1): ASoC: soc-cache: let reg be AND'ed by 0xff instead of data buffer for 8_8 mode Cliff Cai (1): ASoC: soc-cache: add i2c read entry for 8_8 mode sound/soc/soc-cache.c | 39 ++++++++++++++++++++++++++++++++++++++- 1 files changed, 38 insertions(+), 1 deletions(-)

2 1

[alsa-devel] ASoC update for 2.6.34
by Mark Brown 18 Mar '10

18 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-2.6.34 Guennadi Liakhovetski (1): ASoC: SIU driver shall select FW_LOADER sound/soc/sh/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] Support for button reporting on jacks
by Mark Brown 18 Mar '10

18 Mar '10

This patch series adds support for key reporting to the generic ALSA jack API. Currently a series of switches are supported, with the jack implementation assuming that only switches will be used. This isn't ideal for things like the simple buttons implemented in headsets which are used in the same way as keyboard keys and physically resemble them more closely than switches. If this is applied I'd like if possible to merge the topic into the ASoC tree so that I can use it in machine drivers there. 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 topic/jack Mark Brown (2): ALSA: Rename jack switch table in preparation for button support ALSA: Add support for key reporting via the jack interface include/sound/jack.h | 8 +++++ sound/core/jack.c | 71 +++++++++++++++++++++++++++++++++++++++++++++---- 2 files changed, 73 insertions(+), 6 deletions(-)

2 3

[alsa-devel] HDMI output with headphone jack sense possible on Conexant CX20561?
by Andrew Udvare 18 Mar '10

18 Mar '10

My audio codec is Conexant CX20561 (Hermosa) (haven't been able to find anything about 'Hermosa' on the Wiki). I'd like to be able to do a few things which I have not been able to do with tutorials I found on the web: - Whenever I plug in headphones, HDMI output is muted and headphones are enabled - Right now, the way I do this is by manually editing /etc/asound.conf to have the following (commented out makes headphones work, not commented makes HDMI work without any headphone/laptop speaker output) /etc/asound.conf: defaults.pcm.card 0 defaults.pcm.device 3 where 3 is my HDMI (IEC958 2; no idea what IEC958 1 is for) output. So I have a switcher script that does the commenting and uncommenting. Would be nice if this could be automatic (even if alsa had to run such a script on detecting headphones). - Route audio output (PCM) to input (for mixer recording) (Intel HDA cannot do this apparently by default (thanks DRM), needs some hacking around?) Information from alsa-info.sh: http://www.alsa- project.org/db/?f=107f3f6012e68f04d3562d5e11aeb052a1454884 Are my model options correct? And if it's important to know, I build ALSA in-kernel. Is this not recommended? I know at the very least it makes it such that I have to reboot to make any new options to take effect. Thanks all -- Andrew Udvare audvare(a)gmail.com 978-870-5908 www.andyudvare.com

1 0

[alsa-devel] [PATCH] ASoC: SIU driver shall select FW_LOADER
by Guennadi Liakhovetski 18 Mar '10

18 Mar '10

The SIU ASoC driver must load firmware to program the DSP, therefore it has to select FW_LOADER in its Kconfig entry. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 1066749..f07f6d8 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_SH4_SIU select DMA_ENGINE select DMADEVICES select SH_DMAE + select FW_LOADER ## ## Boards

3 2

[alsa-devel] [PATCH] ASoC: remove a card from the list, if instantiation failed
by Guennadi Liakhovetski 18 Mar '10

18 Mar '10

If instantiation of a card failed, we have to remove it from the card list. This fixes an Oops on Migo-R, triggering, when after a failed firmware load attempt the driver modules are removed and re-inserted again. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski(a)gmx.de> --- Should also be good for stable. diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..307eea0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2361,6 +2361,8 @@ static int snd_soc_register_card(struct snd_soc_card *card) mutex_lock(&client_mutex); list_add(&card->list, &card_list); snd_soc_instantiate_cards(); + if (!card->instantiated) + list_del(&card->list); mutex_unlock(&client_mutex); dev_dbg(card->dev, "Registered card '%s'\n", card->name);

2 3

[alsa-devel] [PATCHv6 0/2] TWL6040 audio codec initial support
by Olaya, Margarita 18 Mar '10

18 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 v5: - Add CONFIG_PM for twl6040_suspend and twl6030_resume functions - Add description for patch 1 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 | 1227 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 6 files changed, 1379 insertions(+), 5 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h

1 0

[PATCH] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 17 Mar '10

17 Mar '10

Support pxa168 ssp in ASoC. The SSP bit clock mechanism is totally different from pxa2xx series. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++++ sound/soc/pxa/Kconfig | 6 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa168-ssp.c | 166 ++++++++++++++++++++++++++++ sound/soc/pxa/pxa168-ssp.h | 64 +++++++++++ 5 files changed, 285 insertions(+), 1 deletions(-) create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h create mode 100644 sound/soc/pxa/pxa168-ssp.c create mode 100644 sound/soc/pxa/pxa168-ssp.h 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/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 7be1d5f..286d52a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA + depends on ARCH_PXA || ARCH_MMP select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to @@ -25,6 +25,10 @@ config SND_PXA2XX_SOC_SSP tristate select PXA_SSP +config SND_PXA168_SOC_SSP + tristate + select PXA_SSP + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 33c1579..a74e6c9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,13 @@ 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-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-ssp.o +snd-soc-pxa168-ssp-objs := pxa-ssp.o 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_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o +obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/pxa168-ssp.c b/sound/soc/pxa/pxa168-ssp.c new file mode 100644 index 0000000..d28d18a --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.c @@ -0,0 +1,166 @@ +/* + * pxa168-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.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" +#include "pxa-ssp.h" + +/* + * 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); + + /* 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 ASYSCLK is supplied by pxa168, ASSPDR should be configured. */ + if (dir == SND_SOC_CLOCK_OUT) + asspdr = (mclk_conf[freq].bclk_num << 16) + | mclk_conf[freq].bclk_denom; + + pxa_ssp_disable(ssp); + clk_disable(ssp->clk); /* SSP port internal clock */ + + /* clear ECS, NCS, MOD, ACS */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + data = sscr0 & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (sscr0 != data) + pxa_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 */ + pxa_ssp_enable(ssp); + return 0; +} + +static int pxa168_ssp_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 ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + + pxa_ssp_disable(ssp); + /* update divider register in MPMU */ + __raw_writel(0, MPMU_ASYSDR); + __raw_writel(0, MPMU_ASSPDR); + return 0; +} + +static struct snd_soc_dai_ops pxa168_ssp_dai_ops = { + .hw_free = pxa168_ssp_hw_free, + .set_sysclk = pxa168_ssp_set_dai_sysclk, +}; + +#define PXA168_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA168_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA168_SSP_DAI(_id) \ +{ \ + .name = "pxa168-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .ops = &pxa168_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa168_ssp_dai[] = { + PXA168_SSP_DAI(PXA168_DAI_SSP1), + PXA168_SSP_DAI(PXA168_DAI_SSP2), + PXA168_SSP_DAI(PXA168_DAI_SSP3), + PXA168_SSP_DAI(PXA168_DAI_SSP4), + PXA168_SSP_DAI(PXA168_DAI_SSP5), +}; +EXPORT_SYMBOL_GPL(pxa168_ssp_dai); + +static int __init pxa168_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa168_ssp_init); + +static void __exit pxa168_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + snd_soc_unregister_dai(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..bf73137 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.h @@ -0,0 +1,64 @@ +/* + * 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 /* ASYSCLK master -- pxa168 */ +#define PXA168_ASYSCLK_SLAVE 1 /* ASYSCLK slave -- pxa168 */ + +struct pxa168_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 / ASYSCLK_NUM + * BCLK = MCLK * SSPSCLK_DENOM / SSPSCLK_NUM + */ +static const struct pxa168_ssp_mclk mclk_conf[] = { + /* rate, fmt, chn, mclk, den, num, bclk, den, num */ + {96000, 16, 2, 12288000, 64, 1625, 6144000, 1, 2}, + {96000, 16, 1, 12288000, 64, 1625, 1536000, 1, 8}, + {88200, 16, 2, 11289600, 294, 8125, 5644800, 1, 2}, + {88200, 16, 1, 11289600, 294, 8125, 1411200, 1, 8}, + {48000, 16, 2, 12288000, 64, 1625, 3072000, 1, 4}, + {48000, 16, 1, 12288000, 64, 1625, 768000, 1, 16}, + {44100, 16, 2, 11289600, 294, 8125, 2822400, 1, 4}, + {44100, 16, 1, 11289600, 294, 8125, 705600, 1, 16}, + {32000, 16, 2, 12288000, 64, 1625, 2048000, 1, 6}, + {32000, 16, 1, 12288000, 64, 1625, 512000, 1, 24}, + {22050, 16, 2, 11289600, 294, 8125, 1411200, 1, 8}, + {22050, 16, 1, 11289600, 294, 8125, 352800, 1, 32}, + {16000, 16, 2, 12288000, 64, 1625, 1024000, 1, 12}, + {16000, 16, 1, 12288000, 64, 1625, 256000, 1, 48}, + {11025, 16, 2, 11289600, 294, 8125, 705600, 1, 16}, + {11025, 16, 1, 11289600, 294, 8125, 176400, 1, 64}, + { 8000, 16, 2, 12288000, 64, 1625, 512000, 1, 24}, + { 8000, 16, 1, 12288000, 64, 1625, 128000, 1, 96}, +}; + +extern struct snd_soc_dai pxa168_ssp_dai[]; + +#endif -- 1.5.6.5

1 0

[PATCH] ASoC: support pxa168 ssp in ASoC
by Haojian Zhuang 17 Mar '10

17 Mar '10

Support pxa168 ssp in ASoC. The SSP bit clock mechanism is totally different from pxa2xx series. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- arch/arm/mach-mmp/include/mach/regs-mpmu.h | 48 ++++++ sound/soc/pxa/Kconfig | 6 +- sound/soc/pxa/Makefile | 2 + sound/soc/pxa/pxa-ssp.h | 1 + sound/soc/pxa/pxa168-ssp.c | 233 ++++++++++++++++++++++++++++ sound/soc/pxa/pxa168-ssp.h | 27 ++++ 6 files changed, 316 insertions(+), 1 deletions(-) create mode 100644 arch/arm/mach-mmp/include/mach/regs-mpmu.h create mode 100644 sound/soc/pxa/pxa168-ssp.c create mode 100644 sound/soc/pxa/pxa168-ssp.h 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/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 7be1d5f..286d52a 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,6 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" - depends on ARCH_PXA + depends on ARCH_PXA || ARCH_MMP select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to @@ -25,6 +25,10 @@ config SND_PXA2XX_SOC_SSP tristate select PXA_SSP +config SND_PXA168_SOC_SSP + tristate + select PXA_SSP + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 33c1579..a74e6c9 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,13 @@ 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-pxa2xx-ssp-objs := pxa-ssp.o pxa2xx-ssp.o +snd-soc-pxa168-ssp-objs := pxa-ssp.o 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_PXA2XX_SOC_SSP) += snd-soc-pxa2xx-ssp.o +obj-$(CONFIG_SND_PXA168_SOC_SSP) += snd-soc-pxa168-ssp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index af4a09b..852660a 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -15,6 +15,7 @@ struct ssp_priv { struct ssp_device *ssp; unsigned int sysclk; + unsigned int sysclk_idx; int dai_fmt; #ifdef CONFIG_PM uint32_t cr0; diff --git a/sound/soc/pxa/pxa168-ssp.c b/sound/soc/pxa/pxa168-ssp.c new file mode 100644 index 0000000..487e899 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.c @@ -0,0 +1,233 @@ +/* + * pxa168-ssp.c -- ALSA Soc Audio Layer + * + * Copyright 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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * TODO: + * o Test network mode + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/pxa2xx-lib.h> + +#include <mach/hardware.h> +#include <mach/dma.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" +#include "pxa-ssp.h" + +struct pxa168_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 / ASYSCLK_NUM + * BCLK = MCLK * SSPSCLK_DENOM / SSPSCLK_NUM + */ +static const struct pxa168_ssp_mclk mclk_conf[] = { + /* rate, fmt, chn, mclk, den, num, bclk, den, num */ + {96000, 16, 2, 12288000, 64, 1625, 6144000, 1, 2}, + {96000, 16, 1, 12288000, 64, 1625, 1536000, 1, 8}, + {88200, 16, 2, 11289600, 294, 8125, 5644800, 1, 2}, + {88200, 16, 1, 11289600, 294, 8125, 1411200, 1, 8}, + {48000, 16, 2, 12288000, 64, 1625, 3072000, 1, 4}, + {48000, 16, 1, 12288000, 64, 1625, 768000, 1, 16}, + {44100, 16, 2, 11289600, 294, 8125, 2822400, 1, 4}, + {44100, 16, 1, 11289600, 294, 8125, 705600, 1, 16}, + {32000, 16, 2, 12288000, 64, 1625, 2048000, 1, 6}, + {32000, 16, 1, 12288000, 64, 1625, 512000, 1, 24}, + {22050, 16, 2, 11289600, 294, 8125, 1411200, 1, 8}, + {22050, 16, 1, 11289600, 294, 8125, 352800, 1, 32}, + {16000, 16, 2, 12288000, 64, 1625, 1024000, 1, 12}, + {16000, 16, 1, 12288000, 64, 1625, 256000, 1, 48}, + {11025, 16, 2, 11289600, 294, 8125, 705600, 1, 16}, + {11025, 16, 1, 11289600, 294, 8125, 176400, 1, 64}, + { 8000, 16, 2, 12288000, 64, 1625, 512000, 1, 24}, + { 8000, 16, 1, 12288000, 64, 1625, 128000, 1, 96}, +}; + +int pxa168_query_mclk(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; + struct ssp_priv *priv = cpu_dai->private_data; + unsigned int rate, width, channel; + int i, ret = -EINVAL; + + rate = params_rate(params); + width = snd_pcm_format_physical_width(params_format(params)); + channel = params_channels(params); + + for (i = 0; i < ARRAY_SIZE(mclk_conf); i++) { + if ((mclk_conf[i].rate == rate) + && (mclk_conf[i].format == width) + && (mclk_conf[i].channel == channel)) { + /* save both mclk and mclk index into ssp_priv */ + priv->sysclk = mclk_conf[i].mclk; + priv->sysclk_idx = i; + ret = priv->sysclk; + break; + } + } + return ret; +} +EXPORT_SYMBOL_GPL(pxa168_query_mclk); + +/* + * 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); + + /* freq is the index of mclk_conf table */ + if ((freq < 0) || (priv->sysclk_idx >= ARRAY_SIZE(mclk_conf))) { + dev_warn(&ssp->pdev->dev, "Wrong frequency on SYSCLK\n"); + return -EINVAL; + } + asysdr = (mclk_conf[priv->sysclk_idx].mclk_num << 16) + | mclk_conf[priv->sysclk_idx].mclk_denom; + asspdr = 0; + /* If ASYSCLK is supplied by pxa168, ASSPDR should be configured. */ + if (dir == SND_SOC_CLOCK_OUT) + asspdr = (mclk_conf[priv->sysclk_idx].bclk_num << 16) + | mclk_conf[priv->sysclk_idx].bclk_denom; + + pxa_ssp_disable(ssp); + clk_disable(ssp->clk); /* SSP port internal clock */ + + /* clear ECS, NCS, MOD, ACS */ + sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + data = sscr0 & ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); + if (sscr0 != data) + pxa_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 */ + pxa_ssp_enable(ssp); + return 0; +} + +static int pxa168_ssp_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 ssp_priv *priv = cpu_dai->private_data; + struct ssp_device *ssp = priv->ssp; + + pxa_ssp_disable(ssp); + /* update divider register in MPMU */ + __raw_writel(0, MPMU_ASYSDR); + __raw_writel(0, MPMU_ASSPDR); + return 0; +} + +static struct snd_soc_dai_ops pxa168_ssp_dai_ops = { + .hw_free = pxa168_ssp_hw_free, + .set_sysclk = pxa168_ssp_set_dai_sysclk, +}; + +#define PXA168_SSP_RATES SNDRV_PCM_RATE_8000_96000 +#define PXA168_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +#define PXA168_SSP_DAI(_id) \ +{ \ + .name = "pxa168-ssp", \ + .id = _id, \ + .playback = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .capture = { \ + .channels_min = 1, \ + .channels_max = 2, \ + .rates = PXA168_SSP_RATES, \ + .formats = PXA168_SSP_FORMATS, \ + }, \ + .ops = &pxa168_ssp_dai_ops, \ +} + +struct snd_soc_dai pxa168_ssp_dai[] = { + PXA168_SSP_DAI(PXA168_DAI_SSP1), + PXA168_SSP_DAI(PXA168_DAI_SSP2), + PXA168_SSP_DAI(PXA168_DAI_SSP3), + PXA168_SSP_DAI(PXA168_DAI_SSP4), + PXA168_SSP_DAI(PXA168_DAI_SSP5), +}; +EXPORT_SYMBOL_GPL(pxa168_ssp_dai); + +static int __init pxa168_ssp_init(void) +{ + struct snd_soc_dai *dai; + int i, ret; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + ret = pxa_ssp_register_dai(dai); + if (ret) + return ret; + } + return ret; +} +module_init(pxa168_ssp_init); + +static void __exit pxa168_ssp_exit(void) +{ + struct snd_soc_dai *dai = NULL; + int i; + + for (i = 0; i < ARRAY_SIZE(pxa168_ssp_dai); i++) { + dai = &pxa168_ssp_dai[i]; + snd_soc_unregister_dai(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..c7e23c7 --- /dev/null +++ b/sound/soc/pxa/pxa168-ssp.h @@ -0,0 +1,27 @@ +/* + * 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 /* ASYSCLK master -- pxa168 */ +#define PXA168_ASYSCLK_SLAVE 1 /* ASYSCLK slave -- pxa168 */ + +extern struct snd_soc_dai pxa168_ssp_dai[]; + +extern int pxa168_query_mclk(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +#endif -- 1.5.6.5

1 0

[alsa-devel] How to create an Alsa mixer for a USB device w/o mixer descriptor?
by Felix Homann 17 Mar '10

17 Mar '10

Hi, as posted earlier, I've got a USB device (Fast Track Ultra 8R) wich *has* a mixer unit but doesn't have a descriptor for it. It's operated by standard control messages. Although I'm specifically interested in my device, I guess there are many more USB interfaces out there with similar issues. So in general: What's the best way to build a Alsa mixer for such devices. Should one bypass all the descriptor parsing and build a mixer specifically for the device in interest? Or should one think about passing a "fake" descriptor to the mixer build process? Any ideas? Kind regards, Felix

2 3

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 17 Mar '10

17 Mar '10

The following changes since commit fbc2dae854da5fc44e2ce2f97446c8bd283c7e17: Mark Brown (1): ASoC: Support GPIO based microphone detection for WM8904 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 Mark Brown (3): ASoC: Implement WM835x microphone jack detection support ASoC: Remove current PGA control handling ASoC: Refresh WM8750 bias management The PGA control handling is the build issue fix. include/linux/mfd/wm8350/audio.h | 2 + sound/soc/codecs/wm8350.c | 58 +++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 3 + sound/soc/codecs/wm8750.c | 50 ++++----------------- sound/soc/soc-dapm.c | 93 +------------------------------------ 5 files changed, 75 insertions(+), 131 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 17 Mar '10

17 Mar '10

The following changes since commit 75581d2459d3e6e5e796a42b362e4abb612bc958: Peter Ujfalusi (1): ASoC: OMAP3: Report delay caused by the internal FIFO 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 Pretty much all of which is the first bit of some improvements to ASoC jack detection, including implementation in some drivers. Mark Brown (9): ASoC: Remove version display from WM8750 ASoC: Remove unused 'muted' flag from DAPM widgets ASoC: Allow pins to be force enabled ASoC: Add GPIO configuration support for WM8903 ASoC: Initial WM8903 microphone bias and short detection ASoC: Add WM8903 interrupt support ASoC: Implement interrupt driven microphone detection for WM8903 ASoC: Allow configuration of WM8904 GPIO pin functions ASoC: Support GPIO based microphone detection for WM8904 include/sound/soc-dapm.h | 4 +- include/sound/wm8903.h | 249 +++++++++++++++++++++++++++++++++++++++++++++ include/sound/wm8904.h | 110 +++++++++++++++++++- sound/soc/codecs/wm8750.c | 3 - sound/soc/codecs/wm8903.c | 201 ++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm8903.h | 221 +--------------------------------------- sound/soc/codecs/wm8904.c | 17 +++ sound/soc/codecs/wm8904.h | 97 ------------------ sound/soc/soc-dapm.c | 35 ++++++- 9 files changed, 603 insertions(+), 334 deletions(-) create mode 100644 include/sound/wm8903.h

2 4

[alsa-devel] snd-hda-intel: no data when recording from mic, "IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj" logged in kern.log
by Aleksander Adamowski 17 Mar '10

17 Mar '10

Hi! I think I have some interesting problems to analyse for Takashi. Maybe my sound card model is a non typical one, but I have a continuous stream of capture-related problems with it. It's an integrated sound card on my Gigabyte GA-MA74GM-S2H Rev. 1.1 motherboard. "grep ^Codec /proc/asound/card0/codec#0" shows: Codec: Realtek ALC888 I'm attaching the full contents of /proc/asound/card0/codec#0 for identification. But first, some background. With this particular card, I've been continuously hit by the "azx_get_response timeout, switching to single_cmd mode" problem, described here: http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… However, in my case setting probe_mask=1 (experimentally determined) didn't make the problem go away, although it seemed to reduce its frequency (might be just a coincidence). Whenever the dreaded "azx_get_response timeout, switching to single_cmd mode" message appeared in the kernel log, recording stopped working. The card got into the bad state where it would not record sound, and arecord would be giving a stream of "overrun" error messages: arecord -f cd output.wav Recording WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo overrun!!! (at least 27.876 ms long) overrun!!! (at least 26.965 ms long) overrun!!! (at least 16.169 ms long) overrun!!! (at least 11.749 ms long) overrun!!! (at least 11.298 ms long) overrun!!! (at least 11.814 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.190 ms long) overrun!!! (at least 11.747 ms long) overrun!!! (at least 11.819 ms long) overrun!!! (at least 11.149 ms long) overrun!!! (at least 11.257 ms long) overrun!!! (at least 11.315 ms long) overrun!!! (at least 11.651 ms long) overrun!!! (at least 11.745 ms long) overrun!!! (at least 11.235 ms long) overrun!!! (at least 11.169 ms long) overrun!!! (at least 11.776 ms long) overrun!!! (at least 11.391 ms long) overrun!!! (at least 11.669 ms long) overrun!!! (at least 11.714 ms long) overrun!!! (at least 11.671 ms long) overrun!!! (at least 11.190 ms long) What's notable, sound playback would continue to work OK. After being fed up with interrupted Skype conversations (where I weren't immediately aware that the other side stopped hearing me, since I was hearing them), I've decided to upgrade do kernel 2.6.29 (citing the http://www.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.html#_cod… document: "Since 2.6.29 kernel, the driver has a more robust probing method, so this error might happen rarely, though"). After upgrading to 2.6.29 I've removed any extra snd-hda-intel options from modprobe config files. Now the azx_get_response timeout seems to be gone, but I'm experiencing another problem: after some powered up time, recording starts giving empty streams - there's no error, but audio data has zero lengh. e.g. when I arecord, I get a .wav file that is only 44 bytes long (which seems to be the length of the WAV header). When I make a Skype test call to echo123, and it starts playing back my message, it's zero length too - the playback end signal can be heard immediately after the playback start signal. This problem seems to concide with the following error message in the kernel log: hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. Do you have any idea what could be the cause and how to debug this issue further? -- Best Regards, Aleksander Adamowski http://olo.org.pl

2 10

[alsa-devel] [PATCHv5 2/2] ASoC: TWL6040: Add twl6040 codec driver
by Olaya, Margarita 16 Mar '10

16 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Initial version of TWL6040 codec driver. The TWL6040 codec uses a proprietary 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 TWL6040 codec supports power-up/down manual and automatic sequence. Manual sequence is done through a specific register writes sequence. Automatic sequence is done when the codec is powered-up through the external AUDPWRON line. The completion of the sequence is signaled through the audio interrupt. TWL6040 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 (for 44.1 KHz and 48 KHz respectively) - The high-performance PLL generates an exact 19.2 MHz clock signal from high-frequency input at 12/19.2/26/38.4 MHz. Low-power playback mode is a special scenario where only headset path (headset DAC and driver) is active. For the particular case of headset path, PLL being used defines the headset power mode: low-power, high-performance. 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/twl6040.c | 1223 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 141 +++++ 4 files changed, 1370 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/twl6040.c create mode 100644 sound/soc/codecs/twl6040.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 16c47ed..398cbb0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,6 +35,7 @@ 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_TWL6040 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM2000 if I2C @@ -168,6 +169,9 @@ config SND_SOC_TWL4030 select TWL4030_CODEC tristate +config SND_SOC_TWL6040 + tristate + config SND_SOC_UDA134X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6981777..98bd10c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ 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-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8350-objs := wm8350.o @@ -85,6 +86,7 @@ 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-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(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/twl6040.c b/sound/soc/codecs/twl6040.c new file mode 100644 index 0000000..54835cc --- /dev/null +++ b/sound/soc/codecs/twl6040.c @@ -0,0 +1,1223 @@ +/* + * ALSA SoC TWL6040 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 "twl6040.h" + +#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct twl6040_data { + struct snd_soc_codec codec; + int audpwron; + int naudint; + int codec_powered; + int pll; + int non_lp; + unsigned int sysclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; +}; + +/* + * twl6040 register cache & default register settings + */ +static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { + 0x00, /* not used 0x00 */ + 0x4B, /* TWL6040_ASICID (ro) 0x01 */ + 0x00, /* TWL6040_ASICREV (ro) 0x02 */ + 0x00, /* TWL6040_INTID 0x03 */ + 0x00, /* TWL6040_INTMR 0x04 */ + 0x00, /* TWL6040_NCPCTRL 0x05 */ + 0x00, /* TWL6040_LDOCTL 0x06 */ + 0x00, /* TWL6040_HPPLLCTL 0x07 */ + 0x00, /* TWL6040_LPPLLCTL 0x08 */ + 0x00, /* TWL6040_LPPLLDIV 0x09 */ + 0x00, /* TWL6040_AMICBCTL 0x0A */ + 0x00, /* TWL6040_DMICBCTL 0x0B */ + 0x18, /* TWL6040_MICLCTL 0x0C */ + 0x18, /* TWL6040_MICRCTL 0x0D */ + 0x00, /* TWL6040_MICGAIN 0x0E */ + 0x1B, /* TWL6040_LINEGAIN 0x0F */ + 0x00, /* TWL6040_HSLCTL 0x10 */ + 0x00, /* TWL6040_HSRCTL 0x11 */ + 0x00, /* TWL6040_HSGAIN 0x12 */ + 0x06, /* TWL6040_EARCTL 0x13 */ + 0x00, /* TWL6040_HFLCTL 0x14 */ + 0x03, /* TWL6040_HFLGAIN 0x15 */ + 0x00, /* TWL6040_HFRCTL 0x16 */ + 0x03, /* TWL6040_HFRGAIN 0x17 */ + 0x00, /* TWL6040_VIBCTLL 0x18 */ + 0x00, /* TWL6040_VIBDATL 0x19 */ + 0x00, /* TWL6040_VIBCTLR 0x1A */ + 0x00, /* TWL6040_VIBDATR 0x1B */ + 0x00, /* TWL6040_HKCTL1 0x1C */ + 0x00, /* TWL6040_HKCTL2 0x1D */ + 0x00, /* TWL6040_GPOCTL 0x1E */ + 0x00, /* TWL6040_ALB 0x1F */ + 0x00, /* TWL6040_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, /* TWL6040_TRIM1 0x28 */ + 0x00, /* TWL6040_TRIM2 0x29 */ + 0x00, /* TWL6040_TRIM3 0x2A */ + 0x00, /* TWL6040_HSOTRIM 0x2B */ + 0x00, /* TWL6040_HFOTRIM 0x2C */ + 0x09, /* TWL6040_ACCCTL 0x2D */ + 0x00, /* TWL6040_STATUS (ro) 0x2E */ +}; + +/* + * twl6040 vio/gnd registers: + * registers under vio/gnd supply can be accessed + * before the power-up sequence, after NRESPWRON goes high + */ +static const int twl6040_vio_reg[TWL6040_VIOREGNUM] = { + TWL6040_REG_ASICID, + TWL6040_REG_ASICREV, + TWL6040_REG_INTID, + TWL6040_REG_INTMR, + TWL6040_REG_NCPCTL, + TWL6040_REG_LDOCTL, + TWL6040_REG_AMICBCTL, + TWL6040_REG_DMICBCTL, + TWL6040_REG_HKCTL1, + TWL6040_REG_HKCTL2, + TWL6040_REG_GPOCTL, + TWL6040_REG_TRIM1, + TWL6040_REG_TRIM2, + TWL6040_REG_TRIM3, + TWL6040_REG_HSOTRIM, + TWL6040_REG_HFOTRIM, + TWL6040_REG_ACCCTL, + TWL6040_REG_STATUS, +}; + +/* + * twl6040 vdd/vss registers: + * registers under vdd/vss supplies can only be accessed + * after the power-up sequence + */ +static const int twl6040_vdd_reg[TWL6040_VDDREGNUM] = { + TWL6040_REG_HPPLLCTL, + TWL6040_REG_LPPLLCTL, + TWL6040_REG_LPPLLDIV, + TWL6040_REG_MICLCTL, + TWL6040_REG_MICRCTL, + TWL6040_REG_MICGAIN, + TWL6040_REG_LINEGAIN, + TWL6040_REG_HSLCTL, + TWL6040_REG_HSRCTL, + TWL6040_REG_HSGAIN, + TWL6040_REG_EARCTL, + TWL6040_REG_HFLCTL, + TWL6040_REG_HFLGAIN, + TWL6040_REG_HFRCTL, + TWL6040_REG_HFRGAIN, + TWL6040_REG_VIBCTLL, + TWL6040_REG_VIBDATL, + TWL6040_REG_VIBCTLR, + TWL6040_REG_VIBDATR, + TWL6040_REG_ALB, + TWL6040_REG_DLB, +}; + +/* + * read twl6040 register cache + */ +static inline unsigned int twl6040_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + return cache[reg]; +} + +/* + * write twl6040 register cache + */ +static inline void twl6040_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL6040_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * read from twl6040 hardware register + */ +static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl6040_write_reg_cache(codec, reg, value); + + return value; +} + +/* + * write to the twl6040 register space + */ +static int twl6040_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + if (reg >= TWL6040_CACHEREGNUM) + return -EIO; + + twl6040_write_reg_cache(codec, reg, value); + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl6040_init_vio_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + /* allow registers to be accessed by i2c */ + twl6040_write(codec, TWL6040_REG_ACCCTL, cache[TWL6040_REG_ACCCTL]); + + for (i = 0; i < TWL6040_VIOREGNUM; i++) { + reg = twl6040_vio_reg[i]; + /* skip read-only registers (ASICID, ASICREV, STATUS) */ + switch (reg) { + case TWL6040_REG_ASICID: + case TWL6040_REG_ASICREV: + case TWL6040_REG_STATUS: + continue; + default: + break; + } + twl6040_write(codec, reg, cache[reg]); + } +} + +static void twl6040_init_vdd_regs(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + int reg, i; + + for (i = 0; i < TWL6040_VDDREGNUM; i++) { + reg = twl6040_vdd_reg[i]; + twl6040_write(codec, reg, cache[reg]); + } +} + +/* twl6040 codec manual power-up sequence */ +static void twl6040_power_up(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable reference system */ + ldoctl |= TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(10); + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* enable high-side ldo */ + ldoctl |= TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable negative charge pump */ + ncpctl |= TWL6040_NCPENA | TWL6040_NCPOPEN; + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* enable low-side ldo */ + ldoctl |= TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* enable low-power pll */ + lppllctl |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* reset state machine */ + accctl |= TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + mdelay(5); + accctl &= ~TWL6040_RESETSPLIT; + twl6040_write(codec, TWL6040_REG_ACCCTL, accctl); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); +} + +/* twl6040 codec manual power-down sequence */ +static void twl6040_power_down(struct snd_soc_codec *codec) +{ + u8 ncpctl, ldoctl, lppllctl, accctl; + + ncpctl = twl6040_read_reg_cache(codec, TWL6040_REG_NCPCTL); + ldoctl = twl6040_read_reg_cache(codec, TWL6040_REG_LDOCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + accctl = twl6040_read_reg_cache(codec, TWL6040_REG_ACCCTL); + + /* enable internal oscillator */ + ldoctl |= TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(10); + /* disable low-power pll */ + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + /* disable low-side ldo */ + ldoctl &= ~TWL6040_LSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable negative charge pump */ + ncpctl &= ~(TWL6040_NCPENA | TWL6040_NCPOPEN); + twl6040_write(codec, TWL6040_REG_NCPCTL, ncpctl); + udelay(488); + /* disable high-side ldo */ + ldoctl &= ~TWL6040_HSLDOENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + udelay(244); + /* disable internal oscillator */ + ldoctl &= ~TWL6040_OSCENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + /* disable reference system */ + ldoctl &= ~TWL6040_REFENA; + twl6040_write(codec, TWL6040_REG_LDOCTL, ldoctl); + msleep(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 = TWL6040_HSDRVMODEL | TWL6040_HSDACMODEL; + + hslctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSLCTL); + hsrctl = twl6040_read_reg_cache(codec, TWL6040_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6040_write(codec, TWL6040_REG_HSLCTL, hslctl); + twl6040_write(codec, TWL6040_REG_HSRCTL, hsrctl); + + return 0; +} + +static int twl6040_power_mode_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct twl6040_data *priv = codec->private_data; + + if (SND_SOC_DAPM_EVENT_ON(event)) + priv->non_lp++; + else + priv->non_lp--; + + return 0; +} + +/* audio interrupt handler */ +static irqreturn_t twl6040_naudint_handler(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + struct twl6040_data *priv = codec->private_data; + u8 intid; + + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + + switch (intid) { + case TWL6040_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6040_PLUGINT: + case TWL6040_UNPLUGINT: + case TWL6040_HOOKINT: + break; + case TWL6040_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6040_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6040_READYINT: + complete(&priv->ready); + 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 + */ +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 *twl6040_amicl_texts[] = + {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; + +/* Right analog microphone selection */ +static const char *twl6040_amicr_texts[] = + {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; + +static const struct soc_enum twl6040_enum[] = { + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), +}; + +static const struct snd_kcontrol_new amicl_control = + SOC_DAPM_ENUM("Route", twl6040_enum[0]); + +static const struct snd_kcontrol_new amicr_control = + SOC_DAPM_ENUM("Route", twl6040_enum[1]); + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new hsdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 5, 1, 0); + +static const struct snd_kcontrol_new hsdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 5, 1, 0); + +/* Handsfree DAC playback switches */ +static const struct snd_kcontrol_new hfdacl_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hfdacr_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 2, 1, 0); + +/* Headset driver switches */ +static const struct snd_kcontrol_new hsl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSLCTL, 2, 1, 0); + +static const struct snd_kcontrol_new hsr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HSRCTL, 2, 1, 0); + +/* Handsfree driver switches */ +static const struct snd_kcontrol_new hfl_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 4, 1, 0); + +static const struct snd_kcontrol_new hfr_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); + +static const struct snd_kcontrol_new twl6040_snd_controls[] = { + /* Capture gains */ + SOC_DOUBLE_TLV("Capture Preamplifier Volume", + TWL6040_REG_MICGAIN, 6, 7, 1, 1, mic_preamp_tlv), + SOC_DOUBLE_TLV("Capture Volume", + TWL6040_REG_MICGAIN, 0, 3, 4, 0, mic_amp_tlv), + + /* Playback gains */ + SOC_DOUBLE_TLV("Headset Playback Volume", + TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), + +}; + +static const struct snd_soc_dapm_widget twl6040_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", + TWL6040_REG_MICLCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmpR", + TWL6040_REG_MICRCTL, 0, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC Left", "Left Front Capture", + TWL6040_REG_MICLCTL, 2, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right Front Capture", + TWL6040_REG_MICRCTL, 2, 0), + + /* Microphone bias */ + SND_SOC_DAPM_MICBIAS("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0), + SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0), + + /* DACs */ + SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", + TWL6040_REG_HSLCTL, 0, 0), + SND_SOC_DAPM_DAC("HSDAC Right", "Headset Playback", + TWL6040_REG_HSRCTL, 0, 0), + SND_SOC_DAPM_DAC_E("HFDAC Left", "Handsfree Playback", + TWL6040_REG_HFLCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("HFDAC Right", "Handsfree Playback", + TWL6040_REG_HFRCTL, 0, 0, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* 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_E("Handsfree Left Driver", + SND_SOC_NOPM, 0, 0, &hfl_driver_switch_controls, + twl6040_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, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Analog playback PGAs */ + SND_SOC_DAPM_PGA("HFDAC Left PGA", + TWL6040_REG_HFLCTL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HFDAC Right PGA", + TWL6040_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 twl6040_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl6040_dapm_widgets, + ARRAY_SIZE(twl6040_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int twl6040_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6040_data *priv = codec->private_data; + int time_left; + u8 intid; + + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + + if (!time_left) { + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6040_REG_INTID); + if (!(intid & TWL6040_READYINT)) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + } + + priv->codec_powered = 1; + + return 0; +} + +static int twl6040_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct twl6040_data *priv = codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (priv->codec_powered) + break; + + if (gpio_is_valid(audpwron)) { + /* use AUDPWRON line */ + gpio_set_value(audpwron, 1); + + /* wait for power-up completion */ + ret = twl6040_power_up_completion(codec, naudint); + if (ret) + return ret; + + /* sync registers updated during power-up sequence */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LPPLLCTL); + } else { + /* use manual power-up sequence */ + twl6040_power_up(codec); + priv->codec_powered = 1; + } + + /* initialize vdd/vss registers with reg_cache */ + twl6040_init_vdd_regs(codec); + 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 */ + twl6040_read_reg_volatile(codec, TWL6040_REG_NCPCTL); + twl6040_read_reg_volatile(codec, TWL6040_REG_LDOCTL); + twl6040_write_reg_cache(codec, TWL6040_REG_LPPLLCTL, + 0x00); + } else { + /* use manual power-down sequence */ + twl6040_power_down(codec); + } + + priv->codec_powered = 0; + break; + } + + codec->bias_level = level; + + 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 twl6040_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 twl6040_data *priv = codec->private_data; + + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + 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); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + u8 lppllctl; + int rate; + + /* nothing to do for high-perf pll, it supports only 48 kHz */ + if (priv->pll == TWL6040_HPPLL_ID) + return 0; + + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + rate = params_rate(params); + switch (rate) { + case 88200: + lppllctl |= TWL6040_LPLLFIN; + priv->sysclk = 17640000; + break; + case 96000: + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", rate); + return -EINVAL; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + return 0; +} + +static int twl6040_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 twl6040_data *priv = codec->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * 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 twl6040_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 twl6040_data *priv = codec->private_data; + u8 hppllctl, lppllctl; + + hppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_HPPLLCTL); + lppllctl = twl6040_read_reg_cache(codec, TWL6040_REG_LPPLLCTL); + + switch (clk_id) { + case TWL6040_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 |= TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + mdelay(5); + lppllctl &= ~TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + hppllctl &= ~TWL6040_HPLLENA; + twl6040_write(codec, TWL6040_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 |= TWL6040_LPLLFIN; + break; + case 19200000: + lppllctl &= ~TWL6040_LPLLFIN; + break; + default: + /* sysclk not yet configured */ + lppllctl &= ~TWL6040_LPLLFIN; + priv->sysclk = 19200000; + break; + } + + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + priv->pll = TWL6040_LPPLL_ID; + priv->sysclk_constraints = &lp_constraints; + break; + case TWL6040_SYSCLK_SEL_HPPLL: + hppllctl &= ~TWL6040_MCLK_MSK; + + switch (freq) { + case 12000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_12000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 19200000: + /* mclk input, pll disabled */ + hppllctl |= TWL6040_MCLK_19200KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLBP; + break; + case 26000000: + /* mclk input, pll enabled */ + hppllctl |= TWL6040_MCLK_26000KHZ | + TWL6040_HPLLSQRBP | + TWL6040_HPLLENA; + break; + case 38400000: + /* clk slicer, pll disabled */ + hppllctl |= TWL6040_MCLK_38400KHZ | + TWL6040_HPLLSQRENA | + TWL6040_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); + + twl6040_write(codec, TWL6040_REG_HPPLLCTL, hppllctl); + udelay(500); + lppllctl |= TWL6040_HPLLSEL; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + lppllctl &= ~TWL6040_LPLLENA; + twl6040_write(codec, TWL6040_REG_LPPLLCTL, lppllctl); + + /* high-performance pll can provide only 19.2 MHz */ + priv->pll = TWL6040_HPPLL_ID; + priv->sysclk = 19200000; + priv->sysclk_constraints = &hp_constraints; + break; + default: + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops twl6040_dai_ops = { + .startup = twl6040_startup, + .hw_params = twl6040_hw_params, + .trigger = twl6040_trigger, + .set_sysclk = twl6040_set_dai_sysclk, +}; + +struct snd_soc_dai twl6040_dai = { + .name = "twl6040", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}; +EXPORT_SYMBOL_GPL(twl6040_dai); + +static int twl6040_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; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl6040_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +static struct snd_soc_codec *twl6040_codec; + +static int twl6040_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + BUG_ON(!twl6040_codec); + + codec = twl6040_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, twl6040_snd_controls, + ARRAY_SIZE(twl6040_snd_controls)); + twl6040_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 twl6040_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + twl6040_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_twl6040 = { + .probe = twl6040_probe, + .remove = twl6040_remove, + .suspend = twl6040_suspend, + .resume = twl6040_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl6040); + +static int __devinit twl6040_codec_probe(struct platform_device *pdev) +{ + struct twl4030_codec_data *twl_codec = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct twl6040_data *priv; + int audpwron, naudint; + int ret = 0; + + priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + if (twl_codec) { + audpwron = twl_codec->audpwron_gpio; + naudint = twl_codec->naudint_irq; + } else { + audpwron = -EINVAL; + naudint = 0; + } + + priv->audpwron = audpwron; + priv->naudint = naudint; + + codec = &priv->codec; + codec->dev = &pdev->dev; + twl6040_dai.dev = &pdev->dev; + + codec->name = "twl6040"; + codec->owner = THIS_MODULE; + codec->read = twl6040_read_reg_cache; + codec->write = twl6040_write; + codec->set_bias_level = twl6040_set_bias_level; + codec->private_data = priv; + codec->dai = &twl6040_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(twl6040_reg); + codec->reg_cache = kmemdup(twl6040_reg, sizeof(twl6040_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); + init_completion(&priv->ready); + + 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; + } + + if (naudint) { + /* audio interrupt */ + ret = request_threaded_irq(naudint, NULL, + twl6040_naudint_handler, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "twl6040_codec", codec); + if (ret) + goto gpio2_err; + } else { + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + ~TWL6040_READYMSK & TWL6040_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6040_write_reg_cache(codec, TWL6040_REG_INTMR, + TWL6040_ALLINT_MSK); + } + } + + /* init vio registers */ + twl6040_init_vio_regs(codec); + + /* power on device */ + ret = twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto irq_err; + + ret = snd_soc_register_codec(codec); + if (ret) + goto reg_err; + + twl6040_codec = codec; + + ret = snd_soc_register_dai(&twl6040_dai); + if (ret) + goto dai_err; + + return 0; + +dai_err: + snd_soc_unregister_codec(codec); + twl6040_codec = NULL; +reg_err: + twl6040_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); +gpio1_err: + kfree(codec->reg_cache); +cache_err: + kfree(priv); + return ret; +} + +static int __devexit twl6040_codec_remove(struct platform_device *pdev) +{ + struct twl6040_data *priv = twl6040_codec->private_data; + int audpwron = priv->audpwron; + int naudint = priv->naudint; + + if (gpio_is_valid(audpwron)) + gpio_free(audpwron); + + if (naudint) + free_irq(naudint, twl6040_codec); + + snd_soc_unregister_dai(&twl6040_dai); + snd_soc_unregister_codec(twl6040_codec); + + kfree(twl6040_codec); + twl6040_codec = NULL; + + return 0; +} + +static struct platform_driver twl6040_codec_driver = { + .driver = { + .name = "twl6040_codec", + .owner = THIS_MODULE, + }, + .probe = twl6040_codec_probe, + .remove = __devexit_p(twl6040_codec_remove), +}; + +static int __init twl6040_codec_init(void) +{ + return platform_driver_register(&twl6040_codec_driver); +} +module_init(twl6040_codec_init); + +static void __exit twl6040_codec_exit(void) +{ + platform_driver_unregister(&twl6040_codec_driver); +} +module_exit(twl6040_codec_exit); + +MODULE_DESCRIPTION("ASoC TWL6040 codec driver"); +MODULE_AUTHOR("Misael Lopez Cruz"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h new file mode 100644 index 0000000..c472070 --- /dev/null +++ b/sound/soc/codecs/twl6040.h @@ -0,0 +1,141 @@ +/* + * ALSA SoC TWL6040 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 __TWL6040_H__ +#define __TWL6040_H__ + +#define TWL6040_REG_ASICID 0x01 +#define TWL6040_REG_ASICREV 0x02 +#define TWL6040_REG_INTID 0x03 +#define TWL6040_REG_INTMR 0x04 +#define TWL6040_REG_NCPCTL 0x05 +#define TWL6040_REG_LDOCTL 0x06 +#define TWL6040_REG_HPPLLCTL 0x07 +#define TWL6040_REG_LPPLLCTL 0x08 +#define TWL6040_REG_LPPLLDIV 0x09 +#define TWL6040_REG_AMICBCTL 0x0A +#define TWL6040_REG_DMICBCTL 0x0B +#define TWL6040_REG_MICLCTL 0x0C +#define TWL6040_REG_MICRCTL 0x0D +#define TWL6040_REG_MICGAIN 0x0E +#define TWL6040_REG_LINEGAIN 0x0F +#define TWL6040_REG_HSLCTL 0x10 +#define TWL6040_REG_HSRCTL 0x11 +#define TWL6040_REG_HSGAIN 0x12 +#define TWL6040_REG_EARCTL 0x13 +#define TWL6040_REG_HFLCTL 0x14 +#define TWL6040_REG_HFLGAIN 0x15 +#define TWL6040_REG_HFRCTL 0x16 +#define TWL6040_REG_HFRGAIN 0x17 +#define TWL6040_REG_VIBCTLL 0x18 +#define TWL6040_REG_VIBDATL 0x19 +#define TWL6040_REG_VIBCTLR 0x1A +#define TWL6040_REG_VIBDATR 0x1B +#define TWL6040_REG_HKCTL1 0x1C +#define TWL6040_REG_HKCTL2 0x1D +#define TWL6040_REG_GPOCTL 0x1E +#define TWL6040_REG_ALB 0x1F +#define TWL6040_REG_DLB 0x20 +#define TWL6040_REG_TRIM1 0x28 +#define TWL6040_REG_TRIM2 0x29 +#define TWL6040_REG_TRIM3 0x2A +#define TWL6040_REG_HSOTRIM 0x2B +#define TWL6040_REG_HFOTRIM 0x2C +#define TWL6040_REG_ACCCTL 0x2D +#define TWL6040_REG_STATUS 0x2E + +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) + +#define TWL6040_VIOREGNUM 18 +#define TWL6040_VDDREGNUM 21 + +/* INTID (0x03) fields */ + +#define TWL6040_THINT 0x01 +#define TWL6040_PLUGINT 0x02 +#define TWL6040_UNPLUGINT 0x04 +#define TWL6040_HOOKINT 0x08 +#define TWL6040_HFINT 0x10 +#define TWL6040_VIBINT 0x20 +#define TWL6040_READYINT 0x40 + +/* INTMR (0x04) fields */ + +#define TWL6040_READYMSK 0x40 +#define TWL6040_ALLINT_MSK 0x7B + +/* NCPCTL (0x05) fields */ + +#define TWL6040_NCPENA 0x01 +#define TWL6040_NCPOPEN 0x40 + +/* LDOCTL (0x06) fields */ + +#define TWL6040_LSLDOENA 0x01 +#define TWL6040_HSLDOENA 0x04 +#define TWL6040_REFENA 0x40 +#define TWL6040_OSCENA 0x80 + +/* HPPLLCTL (0x07) fields */ + +#define TWL6040_HPLLENA 0x01 +#define TWL6040_HPLLRST 0x02 +#define TWL6040_HPLLBP 0x04 +#define TWL6040_HPLLSQRENA 0x08 +#define TWL6040_HPLLSQRBP 0x10 +#define TWL6040_MCLK_12000KHZ (0 << 5) +#define TWL6040_MCLK_19200KHZ (1 << 5) +#define TWL6040_MCLK_26000KHZ (2 << 5) +#define TWL6040_MCLK_38400KHZ (3 << 5) +#define TWL6040_MCLK_MSK 0x60 + +/* LPPLLCTL (0x08) fields */ + +#define TWL6040_LPLLENA 0x01 +#define TWL6040_LPLLRST 0x02 +#define TWL6040_LPLLSEL 0x04 +#define TWL6040_LPLLFIN 0x08 +#define TWL6040_HPLLSEL 0x10 + +/* HSLCTL (0x10) fields */ + +#define TWL6040_HSDACMODEL 0x02 +#define TWL6040_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6040_HSDACMODER 0x02 +#define TWL6040_HSDRVMODER 0x08 + +/* ACCCTL (0x2D) fields */ + +#define TWL6040_RESETSPLIT 0x04 + +#define TWL6040_SYSCLK_SEL_LPPLL 1 +#define TWL6040_SYSCLK_SEL_HPPLL 2 + +#define TWL6040_HPPLL_ID 1 +#define TWL6040_LPPLL_ID 2 + +extern struct snd_soc_dai twl6040_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl6040; + +#endif /* End of __TWL6040_H__ */ -- 1.6.1.3

3 3

[alsa-devel] [PATCH 1/2] OMAP4: PMIC: Rename twl6030_codec as twl6040_codec
by Olaya, Margarita 16 Mar '10

16 Mar '10

Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- drivers/mfd/twl-core.c | 4 ++-- include/linux/i2c/twl.h | 6 +++--- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index 562cd49..720e099 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -109,7 +109,7 @@ #endif #if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\ - defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) + defined(CONFIG_SND_SOC_TWL6040) || defined(CONFIG_SND_SOC_TWL6040_MODULE) #define twl_has_codec() true #else #define twl_has_codec() false @@ -708,7 +708,7 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features) /* 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", + child = add_child(sub_chip_id, "twl6040_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 fb6784e..ebd90ce 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -569,9 +569,9 @@ struct twl4030_codec_data { 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 */ + /* twl6040 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ }; struct twl4030_platform_data { -- 1.6.1.3

2 2

Re: [alsa-devel] Configuration help please?
by Alex Austin 16 Mar '10

16 Mar '10

That is a very complex thing you're asking for. 5.1 sound works by having mythtv pass the raw datastream out the spdif port. Stereo analog is decoded by mythtv, so to handle both, you would need mythtv to output the raw stream to spdif at the same time as it decodes to stereo on analog. You could have the stream split to two outputs, but the spdif would always be PCM stereo. On Mar 16, 2010 9:32 AM, "Dan Armbrust" <daniel.armbrust.list(a)gmail.com> wrote: On Mon, Mar 15, 2010 at 9:11 AM, Dan Armbrust <daniel.armbrust.list(a)gmail.com> wrote: > Hi, > > I'm ... Hello? Is this really a hard thing that I'm asking to do? No one can help me configure alsa? Thanks, Dan _______________________________________________ Alsa-devel mailing list Alsa-devel@als...

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[alsa-devel] Configuration help please?
by Dan Armbrust 16 Mar '10

16 Mar '10

Hi, I'm trying to configure a mythtv box to send sound to my spdif port AND my analog stereo speakers. In order to make the spdif port work, I found I needed this in /etc/asound.conf: #---- pcm.!default { type plug slave.pcm "spdif" } #---- I thought maybe this would work to send the sound to both.... #---- pcm.!default { type plug slave.pcm "front" slave.pcm "spdif" } #---- But that was a little naive... apparently. The sound just goes to whichever item I put last. I know right next to nothing about writing alsa config files... and every example I find is very large, very complicated looking, and doesn't do what I want. Can someone tell me how to configure alsa for this? What I want - exactly - Sound to always go to both the "front" (my 2 channel stereo speakers), and my spdif output, with anywhere between 1 and 5.1 channels... depending on what my source audio has. I also want mythtv to pass the full 5.1 channels from HD to the spdif port, while sending the stereo sound to the analog. Some info: # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 0/1 Subdevice #0: subdevice #0 #aplay -L front:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC888 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Intel,DEV=0 HDA Intel, ALC888 Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) Thanks! Dan

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