May 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: cs42l73: Sync digital mixer kcontrols to allow for 0dB
by Brian Austin 11 May '12

11 May '12

Some of the Digital mixer kcontrol max values were off by 1 not allowing a max of 0dB. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 5a537b7..e0d45fd 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -565,22 +565,22 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { attn_tlv), SOC_SINGLE_TLV("SPK-IP Mono Volume", - CS42L73_SPKMIPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMIPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-XSP Mono Volume", - CS42L73_SPKMXSPA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMXSPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-ASP Mono Volume", - CS42L73_SPKMASPA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMASPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("SPK-VSP Mono Volume", - CS42L73_SPKMVSPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_SPKMVSPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-IP Mono Volume", - CS42L73_ESLMIPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMIPMA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-XSP Mono Volume", - CS42L73_ESLMXSPA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMXSPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-ASP Mono Volume", - CS42L73_ESLMASPA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMASPA, 0, 0x3F, 1, attn_tlv), SOC_SINGLE_TLV("ESL-VSP Mono Volume", - CS42L73_ESLMVSPMA, 0, 0x3E, 1, attn_tlv), + CS42L73_ESLMVSPMA, 0, 0x3F, 1, attn_tlv), SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), -- 1.7.10

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[alsa-devel] [PATCH] usb-audio: Fix comment
by Mark Hills 11 May '12

11 May '12

Explained by Takashi in <s5hfwbtmz0q.wl%tiwai(a)suse.de> > The reason is because get_min_max*() isn't called in the place you > created these controls, and get_min_max() would be called only for > integer volumes later even if uninitialized. A short cut for booleans. Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- sound/usb/mixer_quirks.c | 12 ++---------- 1 files changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 12125ca..41f4b69 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -79,16 +79,8 @@ static int snd_create_std_mono_ctl(struct usb_mixer_interface *mixer, cval->control = control; cval->cmask = cmask; - /* FIXME: Do we need this? - * The following values are for compatibility with - * Ebox-44 mixer. - * But the corresponding ebox-44 function says: - * "Volume controls will override these values" - * - * These values don't have any effect at all for - * M-Audio FTUs. - * So I think, we can safely omit the range settings here. - */ + /* get_min_max() is called only for integer volumes later, + * so provide a short-cut for booleans */ cval->min = 0; cval->max = 1; cval->res = 0; -- 1.7.4.4

2 1

[alsa-devel] ALSA calling pcm_pointer excessively?
by Russell King - ARM Linux 11 May '12

11 May '12

I'm wondering what is causing ALSA to behave as follows - I have a 64KiB buffer, which is divided up into 4 16KiB periods. The log below is from my soc-dmaengine.c code which supports audio on the Assabet (as we don't have cyclic DMA support there yet.) I'm just trying to sort out the DMA engine tx_status residue code, and am using this as proof that its working. However, it seems that ALSA for whatever reason likes to hammer away at the pcm_pointer callback from time to time: soc_dmaengine_pointer: index 3 stat 1 residue 0018 cookie 3cfe pos 10000 bytes ffe8 soc_dmaengine_pointer: index 3 stat 1 residue 0008 cookie 3cfe pos 10000 bytes fff8 soc_dmaengine_callback: complete soc_dmaengine_pointer: index 0 stat 1 residue 3ff0 cookie 3cff pos 4000 bytes 0010 soc_dmaengine_submit: submitting index 3 size 4000 cookie 3d02 soc_dmaengine_pointer: index 0 stat 1 residue 3fd0 cookie 3cff pos 4000 bytes 0030 soc_dmaengine_pointer: index 0 stat 1 residue 17b8 cookie 3cff pos 4000 bytes 2848 soc_dmaengine_pointer: index 0 stat 1 residue 1770 cookie 3cff pos 4000 bytes 2890 soc_dmaengine_pointer: index 0 stat 1 residue 1768 cookie 3cff pos 4000 bytes 2898 soc_dmaengine_pointer: index 0 stat 1 residue 1758 cookie 3cff pos 4000 bytes 28a8 soc_dmaengine_pointer: index 0 stat 1 residue 1748 cookie 3cff pos 4000 bytes 28b8 soc_dmaengine_pointer: index 0 stat 1 residue 1740 cookie 3cff pos 4000 bytes 28c0 soc_dmaengine_pointer: index 0 stat 1 residue 1730 cookie 3cff pos 4000 bytes 28d0 soc_dmaengine_pointer: index 0 stat 1 residue 1720 cookie 3cff pos 4000 bytes 28e0 soc_dmaengine_pointer: index 0 stat 1 residue 1718 cookie 3cff pos 4000 bytes 28e8 ... soc_dmaengine_pointer: index 0 stat 1 residue 0018 cookie 3cff pos 4000 bytes 3fe8 soc_dmaengine_pointer: index 0 stat 1 residue 0008 cookie 3cff pos 4000 bytes 3ff8 soc_dmaengine_callback: complete soc_dmaengine_pointer: index 1 stat 1 residue 3ff0 cookie 3d00 pos 8000 bytes 4010 soc_dmaengine_submit: submitting index 0 size 4000 cookie 3d03 soc_dmaengine_pointer: index 1 stat 1 residue 3fd0 cookie 3d00 pos 8000 bytes 4030 soc_dmaengine_pointer: index 1 stat 1 residue 3b28 cookie 3d00 pos 8000 bytes 44d8 soc_dmaengine_pointer: index 1 stat 1 residue 3b10 cookie 3d00 pos 8000 bytes 44f0 soc_dmaengine_pointer: index 1 stat 1 residue 3b00 cookie 3d00 pos 8000 bytes 4500 'pos' is the end of the current DMA buffer ((index + 1) * period_byte_size) 'residue' is the value returned from tx_status 'bytes' is the value passed into bytes_to_frames() to return from my pcm_pointer function. The question is - what would cause ALSA to hammer away at the buffer like that - checking it every 16 or 8 DMA bytes transferred? I'm also seeing around 75% system CPU time, which really isn't good. The above was captured while aplay was running, with no apparant audio corruption. Any ideas?

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Re: [alsa-devel] Realtek ACL892: audio gap ~ each 10 seconds? [SOLVED]
by Takashi Iwai 11 May '12

11 May '12

[Re-added Cc to alsa-devel to let others following] At Fri, 11 May 2012 10:37:44 -0300, Dâniel Fraga wrote: > > On Fri, 11 May 2012 09:43:48 +0200 > Takashi Iwai <tiwai(a)suse.de> wrote: > > > Does it mean that you have no headphone, or does the machine have no > > headphone jack? > > The machine has a headphone jack, but I never use a headphone. > > > That's why I'm asking to test. Usually when such a noise occurs due > > to the auto-mute feature, it's because the bogus unsolicited events > > are fired up too much although no jack is plugged actually. Thus > > usually the auto-mute feature itself doesn't work in such a case > > (either no hardware implementation or the hardware has its own > > switching mechanism.) > > I understand. Ok... so you want me to plug the headphone and > test if the auto-mute works right? Ok, I did that and yes, it works > fine as expected (the audio from speakers is automatically muted when a > headphone is inserted). Hm, so it's strange that the auto-mute causes a problem. Could you check which unsolicited events are triggered on your machine? When you build your kernel with the tracing support (CONFIG_TRACEPOINTS=y), you can enable the tracing via % echo 1 > /sys/kernel/debug/tracing/events/hda/hda_unsol_event/enable then after some time, gather the result by checking /sys/kernel/debug/tracing/trace file. The output is like below. % cat /sys/kernel/debug/tracing/trace # tracer: nop # # entries-in-buffer/entries-written: 4/4 #P:4 # # _-----=> irqs-off # / _----=> need-resched # | / _---=> hardirq/softirq # || / _--=> preempt-depth # ||| / delay # TASK-PID CPU# |||| TIMESTAMP FUNCTION # | | | |||| | | <idle>-0 [001] d.h3 10157.942108: hda_unsol_event: [0] res=10010020, res_ex=10 <idle>-0 [001] d.h3 10158.848271: hda_unsol_event: [0] res=10010000, res_ex=10 ..... Check whether the events come up frequently even without plugging/unplugging any jacks. Usually this shouldn't happen. But in the case of problematic hardware with the auto-mute, many bogus unsol events are triggered. thanks, Takashi

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[alsa-devel] Realtek ACL892: audio gap ~ each 10 seconds?
by Dâniel Fraga 11 May '12

11 May '12

I changed my motherboard to Asus P8Z68V-Pro/gen3 and it uses the Realtek ACL892. Everything works, except for the fact that the sound has a "gap" each 10 seconds... it's annoying... it's just like a small pop or click... What can I do to debug that? Is there a workaround I can try? Thanks. -- Linux 3.3.0: Saber-toothed Squirrel http://www.youtube.com/DanielFragaBR

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[alsa-devel] [PATCH 1/1] aplay: print vu-meter to stderr, not stdout
by Jerome Carretero 11 May '12

11 May '12

It allows showing the vu-meter while recording to stdout, eg: aplay -C -V stereo | oggenc - > out.ogg Signed-off-by: Jerome Carretero <cJ(a)zougloub.eu> diff --git a/aplay/aplay.c b/aplay/aplay.c index 1ce34c4..8462484 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1453,9 +1453,9 @@ static void print_vu_meter_mono(int perc, int maxperc) sprintf(line + val, "| MAX"); else sprintf(line + val, "| %02i%%", maxperc); - fputs(line, stdout); + fputs(line, stderr); if (perc > 100) - printf(_(" !clip ")); + fprintf(stderr, _(" !clip ")); } static void print_vu_meter_stereo(int *perc, int *maxperc) @@ -1493,7 +1493,7 @@ static void print_vu_meter_stereo(int *perc, int *maxperc) memcpy(line + bar_length, tmp, 3); } line[bar_length * 2 + 6 + 2] = 0; - fputs(line, stdout); + fputs(line, stderr); } static void print_vu_meter(signed int *perc, signed int *maxperc) @@ -1631,9 +1631,9 @@ static void compute_max_peak(u_char *data, size_t count) if (perc[c] > maxperc[c]) maxperc[c] = perc[c]; - putchar('\r'); + putc('\r', stderr); print_vu_meter(perc, maxperc); - fflush(stdout); + fflush(stderr); } else if(verbose==3) { printf(_("Max peak (%li samples): 0x%08x "), (long)ocount, max_peak[0]); @@ -1642,8 +1642,8 @@ static void compute_max_peak(u_char *data, size_t count) putchar('#'); else putchar(' '); - printf(" %i%%\n", perc[0]); - fflush(stdout); + fprintf(stderr, " %i%%\n", perc[0]); + fflush(stderr); } } -- 1.7.10

2 2

[alsa-devel] [PATCH 0/6] Add device tree support for mxs audio
by Shawn Guo 11 May '12

11 May '12

This series adds device tree probe for mxs-saif driver and mxs-sgtl5000 ASoC machine driver, and is tested on imx28-evk board. It's based on sound, clkdev, mxs pinctrl and DT branches, and probably easier to have it go through arm-soc with sound/for-3.5 pulled in as dependency. Shawn Guo (6): ASoC: mxs: add __devinit for mxs_saif_probe ASoC: mxs: use devm_clk_get for mxs_saif_probe ASoC: mxs: mxs-pcm does not need to be a plaform_driver ASoC: mxs: add device tree support for mxs-saif ASoC: mxs: add device tree support for mxs-sgtl5000 ARM: dts: enable audio support for imx28-evk .../bindings/sound/mxs-audio-sgtl5000.txt | 17 +++ .../devicetree/bindings/sound/mxs-saif.txt | 36 ++++++ arch/arm/boot/dts/imx28-evk.dts | 41 +++++++ arch/arm/boot/dts/imx28.dtsi | 23 ++++ sound/soc/mxs/mxs-pcm.c | 24 +--- sound/soc/mxs/mxs-pcm.h | 3 + sound/soc/mxs/mxs-saif.c | 119 +++++++++++--------- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 61 ++++++++++- 9 files changed, 253 insertions(+), 72 deletions(-)

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Re: [alsa-devel] Getting alsa on Caanoo
by Clemens Ladisch 11 May '12

11 May '12

Marc Nostromo [M-.-n] wrote: > > The alsa-driver package has the snddevices script, but it works only if > > the kernel was compiled without CONFIG_SND_DYNAMIC_MINORS. > > So I should be able to use it to create the devices ? Yes. Regards, Clemens

1 0

[alsa-devel] Getting alsa on Caanoo
by Marc Nostromo [M-.-n] 11 May '12

11 May '12

Hi, I'm trying to get alsa to work on an open source korean game console called the caanoo http://en.wikipedia.org/wiki/CAANOO It's open source and pretty cheap with respect to the horse power it delivers, so using it as base for an embedded synth / sequencer makes a lot of sense to me. I've successfully compiled and loaded the modules for audio & midi support but I'm still stuck with the last step so I wondered if I could get any help over here, I hope it's the right place to ask. Right now, when I connect a midi device through USB, I get the following console message root@wiz:/mnt/sd/ko_s# dmesg > usb 1-1: new full speed USB device using UBI9032 Test HCD and address 2 > usb 1-1: configuration #1 chosen from 1 choice > MIDIStreaming version 01.00 > EP 82: 1 jack(s) > EP 02: 1 jack(s) > created 1 output and 1 input ports And querying /proc/asound gives me root@wiz:/mnt/sd/ko_s# cat /proc/asound/* > 0 [nanoKONTROL ]: USB-Audio =96 nanoKONTROL > KORG INC. nanoKONTROL at usb-ubisys-usb-1, full speed > 0: [ 0] : control > 1: : sequencer > 8: [ 0- 0]: raw midi > 33: : timer 0 snd_usb_audio > G0: system timer : 10000.000us (10000000 ticks) > Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Nov 20 > 19:16:42 2007 UTC). So I'm thinking I'm on the right track. But I don't see anything in /dev to connect to so I can't really do anything with my device. _From what I read here http://www.gp32x.com/board/index.php?/topic/60064-usbrs232-device-with-the-= caanoo/page__p__954641&#entry954641 it could be due to the fact /dev is mounted in read-only (and that there's no /dev/midiXX present) and that I need to create the device file with mknod. However I'm not sure of what I should do in the specific context of midi/audio with alsa. Right now all I'm looking for is raw midi & pcm out. Could anyone give me a clue as what to try next ? Thanks a lot Marc --- http://www.marc-nostromo.com

3 2

[alsa-devel] [PATCH] atmel/ac97c: correct the unexpected behavior when using uninitial value for reset pin
by Bo Shen 11 May '12

11 May '12

When pdata->reset_pin is passed with a negative value (means gpio is invalid), then chip->reset_pin will not be assigned to a vaule, it will use default value 0. This will cause unexpected behavior. So, add this patch to correct. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/atmel/ac97c.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 115313e..f5ded64 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -991,6 +991,8 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) gpio_direction_output(pdata->reset_pin, 1); chip->reset_pin = pdata->reset_pin; } + } else { + chip->reset_pin = -EINVAL; } snd_card_set_dev(card, &pdev->dev); -- 1.7.10

2 1

[alsa-devel] Play a sine wave with alsa
by Luca Longhi 11 May '12

11 May '12

I've written this code to play a sine wave but I don't think it's right. Could you tell me where I am wrong? Especially I don't know what period_time and buffer_time are. What I have to do to extend the playback time? Code: #include <alsa/asoundlib.h> #include <math.h> int main() { int err; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int Fc, val; int channels=2; int size; int dir=0; unsigned int period_time= 400000; unsigned int buffer_time= 800000; snd_pcm_uframes_t frames; snd_pcm_uframes_t bufferSize; char *buffer; /* Open PCM device for playback. */ err = snd_pcm_open(&handle, "default",SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr,"unable to open pcm device: %s\n",snd_strerror(err)); exit(1); } snd_pcm_hw_params_alloca(&params); err=snd_pcm_hw_params_any(handle, params); if (err < 0) { printf("Broken configuration for playback: no configurations available: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { printf("Access type not available for playback: %s\n", snd_strerror(err)); return err; } err=snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); if (err < 0) { printf("Sample format not available for playback: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err)); return err; } Fc = 5000; dir=1; err=snd_pcm_hw_params_set_rate_near(handle, params, &Fc, &dir); if (err < 0) { printf("Rate %iHz not available for playback: %s\n", Fc, snd_strerror(err)); return err; } dir=0; err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir); if (err < 0) { printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params(handle, params); if (err < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(err)); return err; } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, 0); size = frames * 4; short buf[size]; unsigned short ampiezza; double frequenza = 500; double omega; /* Pulsazione */ int x; omega = 2 * M_PI * frequenza / 180; ampiezza = 20000; for (x = 0; x < size ; x++) { buf[x] = (short)(ampiezza * (sin (omega*x))); } err = snd_pcm_writei(handle, buf, frames); if (err == -EPIPE) { fprintf(stderr, "underrun occurred\n"); snd_pcm_prepare(handle); } else if (err < 0) { fprintf(stderr, "error from writei: %s\n", snd_strerror(err)); } else if (err != (int)frames) { fprintf(stderr, "short write, write %d frames\n", err); } snd_pcm_drain(handle); snd_pcm_close(handle); return 0; }

2 1

[alsa-devel] [PATCH] ASoC: pcm: Staticise non-exported functions
by Mark Brown 10 May '12

10 May '12

They pollute the global namespace and cause sparse to complain. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a3ea8a7..82766af 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -632,7 +632,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, + int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; @@ -1087,7 +1088,7 @@ unwind: return err; } -void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) +static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -1247,7 +1248,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) return 0; } -int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) +static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *fe = substream->private_data; int err, stream = substream->stream; @@ -1354,8 +1355,8 @@ unwind: return ret; } -int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *fe = substream->private_data; int ret, stream = substream->stream; @@ -1404,7 +1405,8 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) +static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, + int cmd) { struct snd_soc_dpcm *dpcm; int ret = 0; @@ -1497,7 +1499,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); -int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int dpcm_fe_dai_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret; @@ -1605,7 +1607,7 @@ static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) +static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *fe = substream->private_data; int stream = substream->stream, ret = 0; @@ -1935,7 +1937,7 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) return 0; } -int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) +static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_soc_dpcm *dpcm; @@ -1972,7 +1974,7 @@ int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) return ret; } -int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) +static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) { struct snd_soc_pcm_runtime *fe = fe_substream->private_data; struct snd_soc_dpcm *dpcm; -- 1.7.10

2 1

[alsa-devel] [PATCH] ASoC: ux500: Don't make the CPU I2S driver user selectable
by Mark Brown 10 May '12

10 May '12

It can only be used with a machine driver so the idiomatic thing for ASoC is to select this driver from the machine driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/ux500/Kconfig | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 1be1f50..44cf434 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -10,7 +10,5 @@ menuconfig SND_SOC_UX500 any of the Ux500 platforms (e.g. U8500). config SND_SOC_UX500_PLAT_MSP_I2S - tristate "Platform - MSP (I2S)" + tristate depends on SND_SOC_UX500 - help - Say Y if you want to enable the Ux500 MSP I2S-driver. -- 1.7.10

1 0

[alsa-devel] Sizeof int in Alsa
by Alan Horstmann 10 May '12

10 May '12

Hi, I have often wondered whether Alsa had an intrinsic requirement that an int would always be at least 32-bit, but been unable to find a clear statement with web searches. Looking recently at snd_pcm_hw_params_set_rate(), for example, the rate value parameter is unsigned int, which suggests to me that it would be broken with 16-bit ints; is it correct then to deduce that Alsa does not support 16-bit int platforms? Regards, Alan

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[alsa-devel] [PATCH v3] OMAPDSS: HDMI: Implement DSS driver interface for audio
by Ricardo Neri 10 May '12

10 May '12

Hi, This patch is the third version of PATCH 14 of this series: http://www.spinics.net/lists/linux-omap/msg69466.html I send this patch separately to ease review, as the rest of the patches in the series were regarded as OK by the DSS maintainer. Changes with respect to v2: *An improved locking strategy. The new strategy maintains an audio state machine. The audio state may only be transitioned if the panel is active. Audio can only be started if the previous state is AUDIO_ENABLED. Such state may only be reached if the panel is enabled. If the panel is disabled, it will also disable audio. As the panel and audio states are protected by a mutex and a spinlock, a video thread and audio thread cannot concurrently execute audio functions. BR, Ricardo Ricardo Neri (1): OMAPDSS: HDMI: Implement DSS driver interface for audio drivers/video/omap2/dss/dss.h | 8 ++ drivers/video/omap2/dss/hdmi.c | 42 ++++++++++ drivers/video/omap2/dss/hdmi_panel.c | 140 ++++++++++++++++++++++++++++++++++ 3 files changed, 190 insertions(+), 0 deletions(-) -- 1.7.5.4

1 1

[alsa-devel] [PATCH v3] Provide interface for audio support in DSS
by Ricardo Neri 10 May '12

10 May '12

Hi, This patch is proposed as a result of a previous discussion related to include audio support in the DSS device driver (http://www.spinics.net/lists/linux-omap/msg64748.html) At the moment, this is intended to cover display technologies with audio based on CEA-861 and IEC-60958, such as HDMI and DisplayPort. However, provisions are made to make it extensible (see below). These are the changes with respect to v2: * A new enum and field is added to DSS to help implementations of the interface to keep track of the audio state. These are the changes with respect to v1: * The audio_enable and audio_start functions were split into enable/disable and start/stop respectively to improve readability. * Comments were added on whether the DSS audio functions may sleep. * In order to make the audio configuration parameters extensible, a struct omap_dss_audio is defined. This helps to easily add more config parameters in the future if/when required (e.g., speaker order configuration). BR, Ricardo Ricardo Neri (1): OMAPDSS: Provide an interface for audio support Documentation/arm/OMAP/DSS | 45 ++++++++++++++++++++++++++++++++++++++++++++ include/video/omapdss.h | 34 +++++++++++++++++++++++++++++++++ 2 files changed, 79 insertions(+), 0 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: cs42l52: Staticise non-exported symbols
by Mark Brown 09 May '12

09 May '12

Makes sparse happy and avoids polluting the global namespace. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs42l52.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 703060d..a710941 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -919,7 +919,7 @@ static struct snd_soc_dai_ops cs42l52_ops = { .set_sysclk = cs42l52_set_sysclk, }; -struct snd_soc_dai_driver cs42l52_dai = { +static struct snd_soc_dai_driver cs42l52_dai = { .name = "cs42l52", .playback = { .stream_name = "Playback", @@ -1163,7 +1163,7 @@ static int cs42l52_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { +static struct snd_soc_codec_driver soc_codec_dev_cs42l52 = { .probe = cs42l52_probe, .remove = cs42l52_remove, .suspend = cs42l52_suspend, -- 1.7.10

2 1

[alsa-devel] [PATCH 1/2] ASoC: wm8994: Add debounce to wm8994 mic detection
by Mark Brown 09 May '12

09 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 27 ++++++++++++++++++++------- sound/soc/codecs/wm8994.h | 2 ++ 2 files changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index dfc4e90..93d81a9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3227,22 +3227,20 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } EXPORT_SYMBOL_GPL(wm8994_mic_detect); -static irqreturn_t wm8994_mic_irq(int irq, void *data) +static void wm8994_mic_work(struct work_struct *work) { - struct wm8994_priv *priv = data; + struct wm8994_priv *priv = container_of(work, + struct wm8994_priv, + mic_work.work); struct snd_soc_codec *codec = priv->codec; int reg; int report; -#ifndef CONFIG_SND_SOC_WM8994_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); -#endif - reg = snd_soc_read(codec, WM8994_INTERRUPT_RAW_STATUS_2); if (reg < 0) { dev_err(codec->dev, "Failed to read microphone status: %d\n", reg); - return IRQ_HANDLED; + return; } dev_dbg(codec->dev, "Microphone status: %x\n", reg); @@ -3284,6 +3282,20 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) snd_soc_jack_report(priv->micdet[1].jack, report, SND_JACK_HEADSET | SND_JACK_BTN_0); +} + +static irqreturn_t wm8994_mic_irq(int irq, void *data) +{ + struct wm8994_priv *priv = data; + struct snd_soc_codec *codec = priv->codec; + +#ifndef CONFIG_SND_SOC_WM8994_MODULE + trace_snd_soc_jack_irq(dev_name(codec->dev)); +#endif + + pm_wakeup_event(codec->dev, 300); + + schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3656,6 +3668,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->codec = codec; mutex_init(&wm8994->accdet_lock); + INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work); for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) init_completion(&wm8994->fll_locked[i]); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 91650bb..d77e06f 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -12,6 +12,7 @@ #include <sound/soc.h> #include <linux/firmware.h> #include <linux/completion.h> +#include <linux/workqueue.h> #include "wm_hubs.h" @@ -127,6 +128,7 @@ struct wm8994_priv { struct mutex accdet_lock; struct wm8994_micdet micdet[2]; + struct delayed_work mic_work; bool mic_detecting; bool jack_mic; int btn_mask; -- 1.7.10

1 1

Re: [alsa-devel] Fast Track Ultra latency and new streaming logic.
by Takashi Iwai 09 May '12

09 May '12

At Wed, 9 May 2012 15:05:47 +0200, Felix Homann wrote: > > Am 03.05.2012 16:03 schrieb "Felix Homann" <linuxaudio(a)showlabor.de>: > > > > Is this a > > flaw in Alsa, Jack or the USB stack? > > I'm starting to think that this is a Jack issue. At least I can set rather > small period and buffer sizes in aplay without any xruns. Or a full-duplex issue? You can test the latency via alsa-lib/test/latency program. (Yes, the usage is quite undocumented. Anyone contributing the howto would be great...) Takashi

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Fix sparse warning due to use of 0 as NULL
by Mark Brown 09 May '12

09 May '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b3b2d5f..dfc4e90 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -792,7 +792,7 @@ static void vmid_reference(struct snd_soc_codec *codec) switch (wm8994->vmid_mode) { default: - WARN_ON(0 == "Invalid VMID mode"); + WARN_ON(NULL == "Invalid VMID mode"); case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, -- 1.7.10

1 0

[alsa-devel] [PATCH RESEND 3/3] ASoC: cs42l73: Use DAPM routes to hook AIF widgets to DAI's
by Brian Austin 09 May '12

09 May '12

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 41 ++++++++++++++++++++++++++++------------- 1 file changed, 28 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a94e1e9..0aa56aa 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -599,17 +599,17 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS42L73_PWRCTL2, 7, 1, NULL, 0), - SND_SOC_DAPM_AIF_OUT("XSPOUTL", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTL", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("XSPOUTR", "XSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("XSPOUTR", NULL, 0, CS42L73_PWRCTL2, 1, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTL", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTL", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("ASPOUTR", "ASP Capture", 0, + SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTL", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0, CS42L73_PWRCTL2, 4, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTR", "VSP Capture", 0, + SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -638,21 +638,21 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_AIF_IN("XSPINL", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINR", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINR", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("XSPINM", "XSP Playback", 0, + SND_SOC_DAPM_AIF_IN("XSPINM", NULL, 0, CS42L73_PWRCTL2, 0, 1), - SND_SOC_DAPM_AIF_IN("ASPINL", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINL", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINR", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINR", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("ASPINM", "ASP Playback", 0, + SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, + SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -776,6 +776,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"HL Left Mixer", NULL, "VSPIN"}, {"HL Right Mixer", NULL, "VSPIN"}, + {"ASPINL", NULL, "ASP Playback"}, + {"ASPINM", NULL, "ASP Playback"}, + {"ASPINR", NULL, "ASP Playback"}, + {"XSPINL", NULL, "XSP Playback"}, + {"XSPINM", NULL, "XSP Playback"}, + {"XSPINR", NULL, "XSP Playback"}, + {"VSPIN", NULL, "VSP Playback"}, + /* Capture Paths */ {"MIC1", NULL, "MIC1 Bias"}, {"PGA Left Mux", "Mic 1", "MIC1"}, @@ -822,6 +830,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"VSPOUTL", NULL, "VSPL Output Mixer"}, {"VSPOUTR", NULL, "VSPR Output Mixer"}, + + {"ASP Capture", NULL, "ASPOUTL"}, + {"ASP Capture", NULL, "ASPOUTR"}, + {"XSP Capture", NULL, "XSPOUTL"}, + {"XSP Capture", NULL, "XSPOUTR"}, + {"VSP Capture", NULL, "VSPOUTL"}, + {"VSP Capture", NULL, "VSPOUTR"}, }; struct cs42l73_mclk_div { -- 1.7.10

2 1

[alsa-devel] [PATCH 1/3] ASoC: cs42l73: Convert to module_i2c_driver()
by Brian Austin 09 May '12

09 May '12

Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l73.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a94e1e9..60b7f53 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1428,25 +1428,7 @@ static struct i2c_driver cs42l73_i2c_driver = { }; -static int __init cs42l73_modinit(void) -{ - int ret; - ret = i2c_add_driver(&cs42l73_i2c_driver); - if (ret != 0) { - pr_err("Failed to register CS42L73 I2C driver: %d\n", ret); - return ret; - } - return 0; -} - -module_init(cs42l73_modinit); - -static void __exit cs42l73_exit(void) -{ - i2c_del_driver(&cs42l73_i2c_driver); -} - -module_exit(cs42l73_exit); +module_i2c_driver(cs42l73_i2c_driver); MODULE_DESCRIPTION("ASoC CS42L73 driver"); MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <joe(a)nucleusys.com>"); -- 1.7.10

2 8

[alsa-devel] [PATCH 11/11] ASoC: Ux500: Add machine-driver
by Ola Lilja 09 May '12

09 May '12

Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 11 ++ sound/soc/ux500/Makefile | 3 +- sound/soc/ux500/mop500.c | 117 +++++++++++++++++ sound/soc/ux500/mop500_ab8500.c | 270 +++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/mop500_ab8500.h | 21 +++ 5 files changed, 421 insertions(+), 1 deletions(-) create mode 100644 sound/soc/ux500/mop500.c create mode 100644 sound/soc/ux500/mop500_ab8500.c create mode 100644 sound/soc/ux500/mop500_ab8500.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 59da6ce..c158e39 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -21,3 +21,14 @@ config SND_SOC_UX500_PLAT_DMA help Say Y if you want to enable the Ux500 platform-driver. ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 0c66a44..cce0c11 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -6,4 +6,5 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o - +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 0000000..5d632ff --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,117 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/mach-types.h> + +#include <linux/module.h> +#include <linux/io.h> +#include <linux/spi/spi.h> + +#include <mach/board-mop500-audio.h> + +#include <sound/soc.h> +#include <sound/initval.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include <mop500_ab8500.h> + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); + + diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 0000000..5b8151d --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,270 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <linux/io.h> + +#include <mach/hardware.h> +#include <mach/board-mop500-audio.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_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->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct device *dev = rtd->card->dev; + unsigned int fmt, fmt_if1; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + driver_mode = (channels == 8) ? + DRIVERMODE_CODEC_ONLY : DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + sw_codec = (driver_mode == DRIVERMODE_NORMAL) ? sw_cpu : 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dev_dbg(dev, "%s: Setup IF1 for FM-radio.\n", __func__); + fmt_if1 = SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_I2S; + ret = ab8500_audio_setup_if1(codec_dai->codec, fmt_if1, 16, 1); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_audio_platform_data *pdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + /* Setup AB8500 according to board-settings */ + pdata = (struct mop500_audio_platform_data *)dev_get_platdata(dev); + ret = ab8500_audio_setup_mics(codec, &pdata->amics); + if (ret < 0) { + pr_err("%s: Failed to setup mics (%d)!\n", __func__, ret); + return ret; + } + ret = ab8500_audio_set_ear_cmv(codec, pdata->ear_cmv); + if (ret < 0) { + pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 0000000..642b5dd --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,21 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); + +#endif -- 1.7.8.3

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[alsa-devel] [PATCH 09/11] ASoC: Ux500: Add platform-driver
by Ola Lilja 09 May '12

09 May '12

Add platform-driver handling all DMA-activities. Signed-off-by: Ola Lilja <ola.o.lilja(a)stericsson.com> --- sound/soc/ux500/Kconfig | 7 + sound/soc/ux500/Makefile | 3 + sound/soc/ux500/ux500_pcm.c | 320 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/ux500/ux500_pcm.h | 35 +++++ 4 files changed, 365 insertions(+), 0 deletions(-) create mode 100644 sound/soc/ux500/ux500_pcm.c create mode 100644 sound/soc/ux500/ux500_pcm.h diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 9b0ae4e..59da6ce 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -14,3 +14,10 @@ config SND_SOC_UX500_PLAT_MSP_I2S help Say Y if you want to enable the Ux500 MSP I2S-driver. +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. + diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index d8e9dd2..0c66a44 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -3,4 +3,7 @@ snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..0d944d5 --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,320 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 <asm/page.h> + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/slab.h> + +#include <plat/ste_dma40.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + pcm->info_flags = 0; + strcpy(pcm->name, "UX500_PCM"); + dev_dbg(rtd->platform->dev, "%s: PCM-name: %s\n", __func__, pcm->name); + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPLv2"); + diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..77ed44d --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, + * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> + * for ST-Ericsson. + * + * License terms: + * + * 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 UX500_PCM_H +#define UX500_PCM_H + +#include <asm/page.h> + +#include <linux/workqueue.h> + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif -- 1.7.8.3

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[alsa-devel] [PATCH 0/4] omap-abe-twl6040: Device tree support
by Peter Ujfalusi 09 May '12

09 May '12

Hello, The following series adds device tree support for OMAP4+ based boards with twl6040 codec. The first three patch is for preparing the driver to be used with DT based probing. The dts structure used for this machine driver uses some of the features already available in the core (thanks to the Tegra DT support). The DAPM routing information is coming from dts. The DAI link configuration also comes via DTS. In order to handle the different number of PCMs (dai links) the dts file has to tell the number of links (abe-twl6040,number-of-links) and has to contain the same number of dai link phandle configuration in the form of: dai-linkX,codec = <phandle to codec>; dai-linkX,dai = <phandle to CPU dai>; Where X is a number from 1... In this way the driver can enumerate and fix up the dai link structure. As an example of the dts section (SDP4430): sound { /* ASoC */ compatible = "ti,abe-twl6040"; abe-twl6040,model = "SDP4430"; abe-twl6040,jack_detection = <1>; abe-twl6040,mclk_freq = <38400000>; /* Number of DAI link connections */ abe-twl6040,number-of-links = <2>; dai-link1,codec = <&twl6040_codec>; dai-link1,dai = <&mcpdm>; dai-link2,codec = <&dmic_codec>; dai-link2,dai = <&dmic>; /* Audio routing */ abe-twl6040,audio-routing = "Headset Stereophone", "HSOL", "Headset Stereophone", "HSOR", "Earphone Spk", "EP", "Ext Spk", "HFL", "Ext Spk", "HFR", "Line Out", "AUXL", "Line Out", "AUXR", "Vibrator", "VIBRAL", "Vibrator", "VIBRAR", "HSMIC", "Headset Mic", "Headset Mic", "Headset Mic Bias", "MAINMIC", "Main Handset Mic", "Main Handset Mic", "Main Mic Bias", "SUBMIC", "Sub Handset Mic", "Sub Handset Mic", "Main Mic Bias", "AFML", "Line In", "AFMR", "Line In", "DMic", "Digital Mic", "Digital Mic", "Digital Mic1 Bias"; }; Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters ASoC: omap-abe-twl6040: Add device tree support .../devicetree/bindings/sound/omap-abe-twl6040.txt | 57 ++++++ sound/soc/omap/omap-abe-twl6040.c | 199 ++++++++++++++----- 2 files changed, 203 insertions(+), 53 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-abe-twl6040.txt -- 1.7.8.6

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