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

[alsa-devel] [PATCH] ASoC: wm_adsp: Move DSP Rate controls into the codec
by Richard Fitzgerald 19 Jun '15

19 Jun '15

The rate controls are codec-specific, it's not possible to generically say what the range or the meaning of each control is (or even if they exist at all) - that depends on the particular codec. This is currently being handled for Arizona codecs by putting an Arizona-specific table of controls inside the wm_adsp driver. This creates a dependency between wm_adsp and arizona.c, and is an awkward solution if the ADSP is used in another family of codecs Fix this by moving the Arizona-specific rate controls into the Arizona codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 27 ++++++++++++++++++++++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 5 ++++ sound/soc/codecs/wm5110.c | 6 +++++ sound/soc/codecs/wm_adsp.c | 52 +++++-------------------------------------- sound/soc/codecs/wm_adsp.h | 2 +- 7 files changed, 48 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 88f6df2..802e05e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2286,6 +2286,33 @@ int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) } EXPORT_SYMBOL_GPL(arizona_set_output_mode); +static const struct soc_enum arizona_adsp2_rate_enum[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; + +const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { + SOC_ENUM("DSP1 Rate", arizona_adsp2_rate_enum[0]), + SOC_ENUM("DSP2 Rate", arizona_adsp2_rate_enum[1]), + SOC_ENUM("DSP3 Rate", arizona_adsp2_rate_enum[2]), + SOC_ENUM("DSP4 Rate", arizona_adsp2_rate_enum[3]), +}; +EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 8824f5b..9881b6f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -217,6 +217,8 @@ extern const struct soc_enum arizona_ng_hold; extern const struct soc_enum arizona_in_hpf_cut_enum; extern const struct soc_enum arizona_in_dmic_osr[]; +extern const struct snd_kcontrol_new arizona_adsp2_rate_controls[]; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 5a9da28..c830832 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1555,7 +1555,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) wm2200->codec = codec; - ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c5ec519..d097f09e5 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1880,6 +1880,11 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = snd_soc_add_codec_controls(codec, + arizona_adsp2_rate_controls, 1); + if (ret) + return ret; + arizona_init_spk(codec); arizona_init_gpio(codec); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 5f032a3..709fcc6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1614,6 +1614,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) return ret; } + ret = snd_soc_add_codec_controls(codec, + arizona_adsp2_rate_controls, + WM5110_NUM_ADSP); + if (ret) + return ret; + snd_soc_dapm_disable_pin(dapm, "HAPTICS"); return 0; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9f90b0..0bb415a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -458,57 +458,17 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; -const struct snd_kcontrol_new wm_adsp1_fw_controls[] = { +const struct snd_kcontrol_new wm_adsp_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], + wm_adsp_fw_get, wm_adsp_fw_put), }; -EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls); - -static const struct soc_enum wm_adsp2_rate_enum[] = { - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), -}; - -static const struct snd_kcontrol_new wm_adsp2_fw_controls[4][2] = { - { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]), - }, - { - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]), - }, - { - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]), - }, - { - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]), - }, -}; +EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -2125,8 +2085,8 @@ int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) wm_adsp2_init_debugfs(dsp, codec); return snd_soc_add_codec_controls(codec, - wm_adsp2_fw_controls[dsp->num - 1], - ARRAY_SIZE(wm_adsp2_fw_controls[0])); + &wm_adsp_fw_controls[dsp->num - 1], + 1); } EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 5042cbd..579a635 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -83,7 +83,7 @@ struct wm_adsp { #define WM_ADSP2(wname, num) \ WM_ADSP2_E(wname, num, wm_adsp2_early_event) -extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: Intel Kconfig: add SND_SOC_INTEL_BOARDS option
by Michele Curti 19 Jun '15

19 Jun '15

Machine drivers under boards folder must be available for both SND_SST_MFLD_PLATFORM and SND_SOC_INTEL_SST. This patch adds a new kernel option SND_SOC_INTEL_BOARD to permit this. Signed-off-by: Michele Curti <michele.curti(a)gmail.com> --- sound/soc/intel/Kconfig | 5 +++++ sound/soc/intel/Makefile | 2 +- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f3060a4..c442cd7 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -10,7 +10,11 @@ config SND_MFLD_MACHINE Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BOARDS + tristate + config SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_BOARDS tristate config SND_SST_IPC @@ -28,6 +32,7 @@ config SND_SST_IPC_ACPI config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI + select SND_SOC_INTEL_BOARDS depends on (X86 || COMPILE_TEST) depends on DW_DMAC_CORE help diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 3853ec2..4c4bf74 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -7,4 +7,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += baytrail/ obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += atom/ # Machine support -obj-$(CONFIG_SND_SOC_INTEL_SST) += boards/ +obj-$(CONFIG_SND_SOC_INTEL_BOARDS) += boards/ -- 2.4.3

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[alsa-devel] question about dapm-routes and uda1380.c
by Andrea Venturi 19 Jun '15

19 Jun '15

hello, i'm working on integrating on an Allwinner A20 based SBC an external codec based on NXP UDA1380 through I2S DAI; the SBC is Olime Olinuxino Lime2, the codec board is this one, as it's pretty cheap on usual china sites: http://www.wvshare.com/product/UDA1380-Board.htm i'm using mainline kernel, 4.0.x from here: https://github.com/jwrdegoede/linux-sunxi/tree/sunxi-wip i'm planning to just port the I2S DAI from the legacy allwinner linux-sunxi 3.4 and use a "simple-audio-card" approach. i have more or less the setup in place with my 3 modules and DTS someway hacked to work out the connection between SOC/CPU + DAI + CODEC. now i load the three modules in the kernel this way (i'm wondering if there's a way for autoloading, i'll look for when it's stable..): # insert the platform/DAI driver modprobe snd-soc-sunxi-i2s sleep 1 modprobe snd-soc-simple-card sleep 1 modprobe snd-soc-uda1380 sleep 1 then , i've a number of warning and error (it's normal as i'm in early debugging now..). the one for which i'm posting is related to "dapm routes" in the .../codecs/uda1380.c, these are errors in the log: [ 33.876179] uda1380-codec 1-0018: Control not supported for path Right PGA -> [Mic + Line R] -> Right ADC [ 33.885756] uda1380-codec 1-0018: ASoC: no dapm match for Right PGA --> Mic + Line R --> Right ADC [ 33.894719] uda1380-codec 1-0018: ASoC: Failed to add route Right PGA -> Mic + Line R -> Right ADC [ 33.903669] uda1380-codec 1-0018: Control not supported for path Right PGA -> [Line] -> Right ADC [ 33.912542] uda1380-codec 1-0018: ASoC: no dapm match for Right PGA --> Line --> Right ADC [ 33.920807] uda1380-codec 1-0018: ASoC: Failed to add route Right PGA -> Line -> Right ADC to clear them out, i had to patch the dapm_route structure of uda1380.c with: --- sound/soc/codecs/uda1380-orig.c 2015-05-26 12:41:58.650111434 +0200 +++ sound/soc/codecs/uda1380.c 2015-06-18 22:38:51.845892195 +0200 @@ -403,8 +403,11 @@ {"Input Mux", "Line", "Left PGA"}, /* right input */ - {"Right ADC", "Mic + Line R", "Right PGA"}, - {"Right ADC", "Line", "Right PGA"}, + {"Right ADC", NULL, "Input Mux"}, + {"Input Mux", "Mic", "Mic LNA"}, + {"Input Mux", "Mic + Line R", "Right PGA"}, + {"Input Mux", "Line L", "Left PGA"}, + {"Input Mux", "Line", "Right PGA"}, /* inputs */ {"Mic LNA", NULL, "VINM"}, i added the same "Input Mux" DAPM widget as it's done on the left input of the codec, that's already bound to this "dapm widget" and is managing the couple of codec register bits at 0x22, SEL_LNA and SEL_MIC, that select one of the many input path (see the PDF to better understand.. i could add a shot but i don't know if it's policy permitted ) http://www.nxp.com/documents/data_sheet/UDA1380.pdf as far as i can see, it should be an issue independent by my "development" setup, but should happening, at module load time, to all the devices having such a codec and using a fairly recent mainline kernel in an ASOC environment; is that right? could you confirm this feeling? let me sincerely tell you that there's not that much documentation about Alsa ASoC, simple-audio-card, codec, DAIs, dapm, dts, up to date, or at least is scattered here and there. so it takes a bit of time to read, understand (maybe), code, test and debug this kind of drivers, really! i can understand there's lot's ongoing here, and API are on flux, for example i had to put together stuff from different card drivers as i'm willing to use also Dmaengine and regmap and don't remember whatever else, but many drivers are still frozen on previous API (i understand it would take lot's of janitoring effort and testing on regression cases..) of course, i still can't say this patch is good for something until i've worked out all the other issues and finally listened some good music out of this combo SOC+codec! :-) hope to post more about this setup when it's working.. bests

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[alsa-devel] How to build alsa-driver ?
by Happy 18 Jun '15

18 Jun '15

Dear ALSA developers, I am trying to get into ALSA development. I need some help. I checked out to the head of the alsa-driver mirror branch (latest update 9 days ago). I made some changes to the usb/mixer_scarlett.c file. (just to test, change the text of the output channels only for the 18i20) Questions: How can I build this? How can I and update (Ubuntu 15.04) with this new build to this change? Thank you very much! Martin

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[alsa-devel] About kernel tlv320aic32x4 driver
by Kimmo Saarela 18 Jun '15

18 Jun '15

I have some patches to that driver and recently I ported those changes to reasonably fresh kernel (4.1-rc7). There are some changes that are bit distractible to current workings of the driver. (mixer settings related) Current mainline driver does nothing to connect unused inputs to common mode (datasheet states that that shall be done) and in our testing that had big impact on recorded audio quality. I've added OF bindings that check which inputs to connect and where and then connect unused inputs to common. Should I just post my patches here or should I first test them against some other kernel version? - Kimmo S.

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Re: [alsa-devel] Intel HDA & Realtek issues on ASUS G751JM
by Raymond Yau 18 Jun '15

18 Jun '15

> > > > How do the other user know the mapping even if you implemented 5.1 ? > I don't know (?) https://rog.asus.com/forum/showthread.php?54685-Asus-G751JT-supports-5-1-an… Is the support of 5.1 done by realtek hda manager or asus audio wizard ?

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[alsa-devel] Default volume for all soundcards (-20dB by default?)
by Jo 17 Jun '15

17 Jun '15

Hi, It seems that the default volume for any new soundcard is set by default to -20dB. Do you know where is it hardcoded ? In which source file (I used package "alsa-utils" in Debian) or which configuration file? Thanks, Jo PS: here's the context : I woud like to set up the default sound volume once for all, for all ALSA devices that will be connected ever. Of course, I could do "amixer ..." or even "alsamixer" to modify the volume of _currently available_ soundcards. But I really want to modify the default volume even for future soundcards that will be added later. I've seen /var/lib/alsa/asound.state but the content is specific to _currently connected_ soundcards. What I want is a solution that will apply to any soundcard that will be connected in the future. Why do I want this? I'm providing a ready-to-use Debian image for my project http://www.samplerbox.org. User #1 might use the computer's built-in-soundcard, User #2 might have a USB DAC, User #3 might have another soundcard... I would like to provide a default -3dB volume that will work for any ALSA soundcard people could have...

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[alsa-devel] [PATCH] ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
by Axel Lin 17 Jun '15

17 Jun '15

Fix trivial typo. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Charles and Dimitris, This looks like a typo. Can you confirm? Thanks, Axel. sound/soc/codecs/wm8995.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 687c4dd..505b65f 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1930,7 +1930,7 @@ static int wm8995_set_dai_sysclk(struct snd_soc_dai *dai, dai->id + 1, freq); break; case WM8995_SYSCLK_MCLK2: - wm8995->sysclk[dai->id] = WM8995_SYSCLK_MCLK1; + wm8995->sysclk[dai->id] = WM8995_SYSCLK_MCLK2; wm8995->mclk[1] = freq; dev_dbg(dai->dev, "AIF%d using MCLK2 at %uHz\n", dai->id + 1, freq); -- 2.1.0

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[alsa-devel] [PATCH v2] ASoC: wm8960: add dapm kcontrols to select left/right ADC input source
by Zidan Wang 17 Jun '15

17 Jun '15

Add dapm kcontrols to select left/right ADC input source, one to select the left ADC input source and one for the right ADC input source. In default, the left ADC will select the left input, and the right ADC will select the right input. When the left(right) ADC select the right(left) input, the left(right) input path will be powered down. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 023e898..3ea12b7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -141,6 +141,8 @@ static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; +static const char *wm8960_adcl_src_text[] = {"Left", "Right"}; +static const char *wm8960_adcr_src_text[] = {"Right", "Left"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -150,6 +152,10 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), }; +static SOC_ENUM_SINGLE_DECL(wm8960_adcl_src_enum, + WM8960_ADDCTL1, 3, wm8960_adcl_src_text); +static SOC_ENUM_SINGLE_DECL(wm8960_adcr_src_enum, + WM8960_ADDCTL1, 2, wm8960_adcr_src_text); static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -281,6 +287,11 @@ SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), }; +static const struct snd_kcontrol_new wm8960_adcl_mux = + SOC_DAPM_ENUM("ADCL Source", wm8960_adcl_src_enum); +static const struct snd_kcontrol_new wm8960_adcr_mux = + SOC_DAPM_ENUM("ADCR Source", wm8960_adcr_src_enum); + static const struct snd_kcontrol_new wm8960_lin_boost[] = { SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0), SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0), @@ -344,6 +355,9 @@ SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0), SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), +SND_SOC_DAPM_MUX("ADCL Source", SND_SOC_NOPM, 0, 0, &wm8960_adcl_mux), +SND_SOC_DAPM_MUX("ADCR Source", SND_SOC_NOPM, 0, 0, &wm8960_adcr_mux), + SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0, &wm8960_loutput_mixer[0], ARRAY_SIZE(wm8960_loutput_mixer)), @@ -399,8 +413,14 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "Right Input Mixer", NULL, "RINPUT2" }, { "Right Input Mixer", NULL, "RINPUT3" }, - { "Left ADC", NULL, "Left Input Mixer" }, - { "Right ADC", NULL, "Right Input Mixer" }, + { "ADCL Source", "Left", "Left Input Mixer" }, + { "ADCL Source", "Right", "Right Input Mixer" }, + + { "ADCR Source", "Left", "Left Input Mixer" }, + { "ADCR Source", "Right", "Right Input Mixer" }, + + { "Left ADC", NULL, "ADCL Source" }, + { "Right ADC", NULL, "ADCR Source" }, { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" }, { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} , -- 1.9.1

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[alsa-devel] [PATCH 1/3] ALSA: hda: add hdac_ext stream creation and cleanup routines
by Vinod Koul 17 Jun '15

17 Jun '15

HDAC extended core should create streams for an extended bus and also free up those on cleanup. So introduce snd_hdac_ext_stream_init_all and snd_hdac_stream_free_all routines Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/hdaudio_ext.h | 3 +++ sound/hda/ext/hdac_ext_stream.c | 50 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 53 insertions(+) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 202350a8eddb..877661629322 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -87,6 +87,9 @@ struct hdac_ext_stream { void snd_hdac_ext_stream_init(struct hdac_ext_bus *bus, struct hdac_ext_stream *stream, int idx, int direction, int tag); +int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, + int num_stream, int dir); +void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus); struct hdac_ext_stream *snd_hdac_ext_stream_assign(struct hdac_ext_bus *bus, struct snd_pcm_substream *substream, int type); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 0677bb65a494..f8ffbdbb450d 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -18,6 +18,7 @@ */ #include <linux/delay.h> +#include <linux/slab.h> #include <sound/pcm.h> #include <sound/hda_register.h> #include <sound/hdaudio_ext.h> @@ -54,6 +55,55 @@ void snd_hdac_ext_stream_init(struct hdac_ext_bus *ebus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init); /** + * snd_hdac_ext_stream_init_all - create and initialize the stream objects + * for an extended hda bus + * @ebus: HD-audio ext core bus + * @start_idx: start index for streams + * @num_stream: number of streams to initialize + * @dir: direction of streams + */ +int snd_hdac_ext_stream_init_all(struct hdac_ext_bus *ebus, int start_idx, + int num_stream, int dir) +{ + int stream_tag = 0; + int i, tag, idx = start_idx; + + for (i = 0; i < num_stream; i++) { + struct hdac_ext_stream *stream = + kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + tag = ++stream_tag; + snd_hdac_ext_stream_init(ebus, stream, idx, dir, tag); + idx++; + } + + return 0; + +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); + +/** + * snd_hdac_stream_free_all - free hdac extended stream objects + * + * @ebus: HD-audio ext core bus + */ +void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus) +{ + struct hdac_stream *s; + struct hdac_ext_stream *stream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + while (!list_empty(&bus->stream_list)) { + s = list_first_entry(&bus->stream_list, struct hdac_stream, list); + stream = stream_to_hdac_ext_stream(s); + list_del(&s->list); + kfree(stream); + } +} +EXPORT_SYMBOL_GPL(snd_hdac_stream_free_all); + +/** * snd_hdac_ext_stream_decouple - decouple the hdac stream * @ebus: HD-audio ext core bus * @stream: HD-audio ext core stream object to initialize -- 1.9.1

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