March 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sébastien Bourdeauducq 30 Mar '15

30 Mar '15

Hi, On 02/23/2015 08:58 PM, Jarkko Nikula wrote: >> Is there a way I can try sound/soc/codecs/rt5670.c instead? I edited it >> so that it attaches to my ACPI ID (and rt5640 doesn't), but that did not >> work. I noticed that the kernel has some Baytrail-specific modules for >> RT564x, and I guess those would need to be changed as well - but it's >> not straightforward. >> > Basically three changes are needed: > > 1. Hack add "10EC5640" to rt5670.c as you have done > 2. Add a new machine driver sound/soc/intel/sst-acpi.c/byt-rt5670 > I guess cht_bsw_rt5672.c can be used as a starting point for audio > routes and byt-rt5640.c for dai link definitions.mess > 3. Hack change sound/soc/intel/sst-acpi.c so that "10EC5640" loads > the new machine driver instead of byt-rt5640. Thanks. I did that and I get the following in the kernel log: [ 6.831422] byt-rt5670 byt-rt5670: ASoC: CPU DAI baytrail-pcm-audio not regis tered [ 6.831443] byt-rt5670 byt-rt5670: snd_soc_register_card failed -517 [ 6.831453] platform byt-rt5670: Driver byt-rt5670 requests probe deferral [ 6.884013] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 6.884023] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 Note: it seems to me that all Baytrail machines will have this "error: invalid DMA engine" message. AFAICT it is emitted from sst-firmware.c because sst->pdata->dma_engine is 0 and in sst-acpi.c the function sst_acpi_probe does not set it for Baytrail because resindex_dma_base is -1 in sst_acpi_baytrail_desc. [ 6.924094] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 6.924106] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 6.924109] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 6.931503] byt-rt5670 byt-rt5670: rt5670-aif1 <-> baytrail-pcm-audio mapping ok [ 6.931624] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Tx [ 6.931629] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Tx -> direct -> AIF1 Playback [ 6.931636] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out0 [ 6.931640] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out0 -> direct -> ssp2 Tx [ 6.931647] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out1 [ 6.931650] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out1 -> direct -> ssp2 Tx [ 6.931656] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931660] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in0 [ 6.931666] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931670] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in1 [ 6.931677] byt-rt5670 byt-rt5670: ASoC: no sink widget found for ssp2 Rx [ 6.931680] byt-rt5670 byt-rt5670: ASoC: Failed to add route AIF1 Capture -> direct -> ssp2 Rx My byt-rt5670 is there: https://gist.github.com/sbourdeauducq/8c5da8ac0da13e21ee31 I have to say that I am unfamiliar with Intel SST and I'm don't quite know what I am doing there. I wish sound devices were straightforward DMA-to-DAC devices instead of this complicated DSP architecture which AFAICT boils down to a glorified memcpy/resampler under Linux (this seems to be a pattern with this SoC though; I had a look at how to get the MIPI cameras to work on this machine and it's an even worse mess). > However, above changes may not make audio working since there has been > couple other new Baytrail based machines with correct codec but audio > seems to not move between the DSP and codec chip. Despite the error messages, the audio device is created and userspace applications can send data into it, but no sound is coming out. I also have to say that the settings in alsamixer are a terrible mess which seems not to originate from my hacks. > One assumption is if they use another SSP port than SSP2. Unfortunately > it's difficult to test since Linux DSP FW is hard-coded for SSP2. If only there were no DSP in the first place... Sébastien

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[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 30 Mar '15

30 Mar '15

Hi Mark These are v2 of remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 35 ++++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 31 +++--- sound/soc/sh/rcar/gen.c | 6 -- sound/soc/sh/rcar/rsnd.h | 19 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 93 +++++++++++------- sound/soc/sh/rcar/ssi.c | 36 ++++--- 12 files changed, 729 insertions(+), 86 deletions(-) _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] Atomic operations not multi-thread aware
by Joël Krähemann 30 Mar '15

30 Mar '15

Hi, please take a a look at gsequencer.org especially at: libags-audio * http://gsequencer.org/api/ags/ags-AgsPort.html gsequencer * http://gsequencer.org/api/ags/ags-AgsLineMember.html I can crash my application while modifying values of the LADSPA plugin assigned dial widget what uses atomic operations and a task engine ... Where can me read about state of Atomic Operations of linux or glibc? Me doing further investigation. bests Joël

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[alsa-devel] [PATCH v2 0/2] ALSA: jack: Refactoring for jack kctls
by Jie Yang 30 Mar '15

30 Mar '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this implement, we are no longer need hda jack specific kctls, so here also remove jack kctls for hda. Any comments are welcome. Jie Yang (2): ALSA: jack: create jack kcontrols for every jack input device ALSA: hda - Remove jack kctls include/sound/jack.h | 8 ++++ sound/core/Kconfig | 3 -- sound/core/Makefile | 3 +- sound/core/jack.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 10 +---- sound/pci/hda/hda_jack.c | 45 ++-------------------- sound/pci/hda/hda_jack.h | 3 -- 7 files changed, 110 insertions(+), 61 deletions(-) -- 1.9.1

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 28 Mar '15

28 Mar '15

2015-3-28 下午5:52 於 "Christian Gleerup" <cromozon(a)swn.nu> 寫道： > > Trying again with my previous kernel since the new kernel seems broken regarding loading the sound module > > nothin plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > > Headphone plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff 0xffffffff = 0x80000000 + 0x7ffffff Try the modified version of hda_jack_sense_test.py which need hda_analyzer 's codec parser and alsa python https://bugzilla.kernel.org/show_bug.cgi?id=66621#c39 it need time for the impedance measurement to complete and become stable, You may need to increase time.sleep(0.02) if you still get 0xffffffff https://bugzilla.kernel.org/show_bug.cgi?id=66621#c41 # the hda_codec.py file comes from Jaroslav Kysela's hda_analyzer program. from hda_codec import * import time def parseoptions(): from optparse import OptionParser parser = OptionParser() parser.add_option("-c", "--card", dest="cardindex", default=0, metavar="CARD", help="card index (as can be seen in /proc/asound/cards)") parser.add_option("-i", "--codec", dest="codecindex", default=0, metavar="CODEC", help="codec index (as can be seen in /proc/asound/cardX/codecY)") (options, args) = parser.parse_args() return int(options.cardindex), int(options.codecindex) cardindex, codecindex = parseoptions() codec = HDACodec(cardindex, codecindex) codec.analyze() for nid in codec.nodes: node = codec.get_node(nid) if WIDGET_TYPE_IDS[node.wtype] == 'PIN': pd = " " impedance = " " if ((node.pincaps & 4) == 4): if (node.pincaps & 2): codec.rw(nid, VERBS['SET_PIN_SENSE'], 0) time.sleep(0.02) sense = codec.rw(nid, VERBS['GET_PIN_SENSE'], 0) if (node.pincaps & 1): impedance = "Imp=%d" % (sense & 0x7fffffff) if (sense & 0x80000000): pd = "PD=1" else: pd = "PD=0" print "Pin 0x%.2x [%s] %s %s at %s (%s) %s : %s %s" % (nid, node.jack_conn_name, node.jack_type_name, node.jack_location_name, node.jack_location2_name, node.jack_color_name, node.defcfg_misc , pd, impedance) > > Mic plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > > > > >

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[alsa-devel] getting the sound to work on my asus m6s00v
by Christian Gleerup 28 Mar '15

28 Mar '15

I had some problems getting the sound to work on my asus m6s00v (I think that is the model, it is a bit worn so I can't really read it) the symptoms was * on the integrated speakers, no sound at all * on the headphone speakers, sound for a very very short period that faded into nothing. with help from debianuser on freenode#alsa we managed to get it working. the option that i had to load the module with is the following: sudo modprobe snd-hda-intel model=3stack-digout the status now (perfect)... * sound on integrated speaker. * sound in headphones (disables integrated speaker output which is fine) * mic works, both for direct playback and recording * when recording it is the 'digital' channel that controls the volume of the recording level, all others Notice: I don't have OSS or pulseaudio installed. I am thinking it would be really nice if the driver automatically could detect that this option is required. this is the output from the alsa-info script http://www.alsa-project.org/db/?f=28b98a3bd1efc6a80dbd5c54b52c4472d8ea5496 let me know if I can do anything to help with this. Regards.

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[alsa-devel] [PATCH RFT] ASoC: max98925: Fix bit-width 24 settings in max98925_dai_hw_params
by Axel Lin 27 Mar '15

27 Mar '15

Trivial typo fix. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Anish, This looks like a typo, can you confirm if this patch is correct. Thanks. sound/soc/codecs/max98925.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 0796dfa..9b5a17d 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -442,8 +442,8 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, case 24: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); - max98925->ch_size = 32; + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_24); + max98925->ch_size = 24; break; case 32: regmap_update_bits(max98925->regmap, -- 1.9.1

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Re: [alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Raymond Yau 27 Mar '15

27 Mar '15

>> >> > >> >> > >> >> numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB >> >> >> >> You need workaround since pulseaudio don't like volume control with more than two channels >> >> >> >> https://bugs.freedesktop.org/show_bug.cgi?id=84983#c35 >> >> >> >> Can you post output of your usb audio device ? >> >> >> >> lsusb -vvvv >> > >> > >> > >> > Certainly! >> > >> > Here it is: >> > >> > Bus 001 Device 003: ID 1038:1225 Ideazon, Inc. >> > Device Descriptor: >> > bLength 18 >> > bDescriptorType 1 >> > bcdUSB 2.00 >> > bDeviceClass 0 (Defined at Interface level) >> > bDeviceSubClass 0 >> > bDeviceProtocol 0 >> > bMaxPacketSize0 64 >> > idVendor 0x1038 Ideazon, Inc. >> > idProduct 0x1225 >> > bcdDevice 0.03 >> > iManufacturer 1 SteelSeries >> > iProduct 2 SteelSeries Siberia Elite >> > iSerial 3 000000000000 >> > bNumConfigurations 1 >> >> > >> > AudioControl Interface Descriptor: >> > bLength 13 >> > bDescriptorType 36 >> > bDescriptorSubtype 4 (MIXER_UNIT) >> > Warning: Descriptor too short >> >> https://github.com/gregkh/usbutils/blob/master/lsusb.c >> >> Seem no warning for descriptor too short for uac2 and can dump those uac2 specific variables >> >> > bUnitID 10 >> > bNrInPins 2 >> > baSourceID( 0) 9 >> > baSourceID( 1) 1 >> > bNrChannels 8 >> > wChannelConfig 0x063f >> > Left Front (L) >> > Right Front (R) >> > Center Front (C) >> > Low Freqency Enhancement (LFE) >> > Left Surround (LS) >> > Right Surround (RS) >> > Side Left (SL) >> > Side Right (SR) >> > iChannelNames 0 >> > bmControls 0x00 >> > bmControls 0x00 >> > iMixer 33 (error) >> > > > > Maybe it's the language barrier, but I am not sure if you wanted me to do something. I, out of curiosity, tried to compile that lsusb but I'm missing headers such as config.h. > > Should you need any more feedback from me about this device, please do not hesitate asking through mailing list or directly, just consider I'm not dev! :) > Do you mean there is no sound using alsa hw device since pulseaudio does not support the multichannel speaker playback volume comtrol ? speaker-test -c8 -t wav -D hw:CARD=Elite speaker-test -c2 -t wav -D hw:CARD=Elite You need to know how to compile alsa driver for you running kernel if you want to fix the problem >> >> > > > amixer: Control hw:2 element read error: Invalid argument If alsamixer or amixer cannot set the speaker playback volume Try skip the problematic PCM Volume controls by adding if (state->chip->usb_id == USB_ID(0x1038,0x1225)) if(unitid == 10) ich_has_control = 0; If the driver really need this mixer unit for the playback path, you need to fix those controls bUnitID 10 bNrInPins 2 baSourceID( 0) 9 microphone baSourceID( 1) 1 usb streaming This look like the first four problematic controls related to microphone and the last control related to streaming numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument

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[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

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