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

[alsa-devel] [sound:topic/hda-ca0132-dsp 4/11] sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)'
by Dan Carpenter 17 Dec '12

17 Dec '12

Hi Ian, FYI, there are new smatch warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/hda-ca0132-dsp head: cdc83c59e59d57b250be46c4b9d31e3b2b5ae589 commit: 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 [4/11] ALSA: hda - Update CA0132 codec to load DSP firmware binary + sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true condition '(res >= 0) => (0-u32max >= 0)' sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not propagate 'status' from dsp_dma_stop() instead of (-5)? sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential null dereference 'dma_engine'. (kzalloc returns null) sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we previously assumed 'dma_engine' could be null (see line 1857) git remote add sound git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git remote update sound git checkout 01ef7dbffb411d9d78d1150b268d9c757f9f2f93 vim +714 sound/pci/hda/patch_ca0132.c 01ef7dbf Ian Minett 2012-09-20 698 VENDOR_CHIPIO_PLL_PMU_WRITE, 0xff); 01ef7dbf Ian Minett 2012-09-20 699 mutex_unlock(&spec->chipio_mutex); 01ef7dbf Ian Minett 2012-09-20 700 } 01ef7dbf Ian Minett 2012-09-20 701 01ef7dbf Ian Minett 2012-09-20 702 /* 01ef7dbf Ian Minett 2012-09-20 703 * CA0132 DSP IO stuffs 01ef7dbf Ian Minett 2012-09-20 704 */ 01ef7dbf Ian Minett 2012-09-20 705 static int dspio_send(struct hda_codec *codec, unsigned int reg, 01ef7dbf Ian Minett 2012-09-20 706 unsigned int data) 01ef7dbf Ian Minett 2012-09-20 707 { 01ef7dbf Ian Minett 2012-09-20 708 unsigned int res; 01ef7dbf Ian Minett 2012-09-20 709 int retry = 50; 01ef7dbf Ian Minett 2012-09-20 710 01ef7dbf Ian Minett 2012-09-20 711 /* send bits of data specified by reg to dsp */ 01ef7dbf Ian Minett 2012-09-20 712 do { 01ef7dbf Ian Minett 2012-09-20 713 res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); 01ef7dbf Ian Minett 2012-09-20 @714 if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) 01ef7dbf Ian Minett 2012-09-20 715 return res; 01ef7dbf Ian Minett 2012-09-20 716 } while (--retry); 01ef7dbf Ian Minett 2012-09-20 717 01ef7dbf Ian Minett 2012-09-20 718 return -EIO; 01ef7dbf Ian Minett 2012-09-20 719 } 01ef7dbf Ian Minett 2012-09-20 720 01ef7dbf Ian Minett 2012-09-20 721 static void dspio_write_wait(struct hda_codec *codec) 01ef7dbf Ian Minett 2012-09-20 722 { --- 0-DAY kernel build testing backend Open Source Technology Center Fengguang Wu, Yuanhan Liu Intel Corporation

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[alsa-devel] The new alsa-driver repository
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Hello all, the ALSA server has new alsa-driver repository which contains several branches for now: master - just empty one with README build - the previous master branch (but all was moved to the alsa/ subtree) mirror - mirrored code from upstream (ALSA parts only) release - merging branch for mirror/build trees to create a relationship between them All branches are ophan. It means that they have no common parent. I put some handy configuration hints to the ALSA WIKI: http://www.alsa-project.org/main/index.php/GIT_Server#Syncing_the_branch_re… The reason to move the old master branch to the alsa/ subtree is quite simple - you can merge/pull the branch to the full linux-2.6 kernel tree now and do your development there. Also, cherry picks between mirror and the linux-2.6 tree should work. Another handy extension - in the current repo - might be to create the linux-2.6 branch with the whole upstream and create another branch for merging the build branch with the whole tree like 'release26' or so. In this way, we can manage another releases of the out-of-kernel build code for the ALSA tree in stable linux kernel versions. Comments, other ideas are welcome. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ALSA: hda - Fix clicking noise on Thinkpad T61/R61i
by David Henningsson 17 Dec '12

17 Dec '12

The bug reporter reports that setting the speaker pin to D3 before turning off its pinctl fixes the clicking noise on powersave for Thinkpad T61. Thanks to c4pp4 for doing most of the work with this bug, including reporting and testing, and writing preliminary patches. BugLink: https://bugs.launchpad.net/bugs/886975 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 36 ++++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) Note: c4pp4 pointed out that in his case, only the speaker needed to go to D3 before pinctl, but noticed no regressions from the below implementation, so I made the fixup more generic (easier to reuse later). diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a3a2263..27e6ca7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -144,6 +144,7 @@ struct conexant_spec { unsigned int asus:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; + unsigned int d3_before_shutup:1; /* Needs speaker to go to D3 before pinctl */ unsigned int adc_switching:1; @@ -558,13 +559,22 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM static int conexant_suspend(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; + + if (spec->d3_before_shutup) + snd_hda_codec_set_power_to_all(codec, codec->afg, AC_PWRST_D3, + false); snd_hda_shutup_pins(codec); return 0; } -#endif + +static void conexant_reboot_notify(struct hda_codec *codec) +{ + conexant_suspend(codec); +} + static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, @@ -575,7 +585,7 @@ static const struct hda_codec_ops conexant_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4408,7 +4418,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = { #ifdef CONFIG_PM .suspend = conexant_suspend, #endif - .reboot_notify = snd_hda_shutup_pins, + .reboot_notify = conexant_reboot_notify, }; /* @@ -4421,6 +4431,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_D3_BEFORE_SHUTUP, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4430,6 +4441,13 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt_fixup_d3_before_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + spec->d3_before_shutup = 1; +} + static void cxt5066_increase_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -4499,6 +4517,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_D3_BEFORE_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_d3_before_shutup, + }, +}; + +static const struct snd_pci_quirk cxt5045_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20ac, "Lenovo T61/R61i", CXT_FIXUP_D3_BEFORE_SHUTUP), + {} }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4553,6 +4580,7 @@ static int patch_conexant_auto(struct hda_codec *codec) switch (codec->vendor_id) { case 0x14f15045: codec->single_adc_amp = 1; + snd_hda_pick_fixup(codec, NULL, cxt5045_fixups, cxt_fixups); break; case 0x14f15051: add_cx5051_fake_mutes(codec); -- 1.7.9.5

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[alsa-devel] ASoC updates for v3.8
by Mark Brown 17 Dec '12

17 Dec '12

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[alsa-devel] [PATCH 0/3] at91/ssc: fixes on ASoC tree for 3.8
by Nicolas Ferre 17 Dec '12

17 Dec '12

Hi Mark, This series goes on top of current Linus' git tree and fixes an error that we have while compiling DTBs for AT91: ERROR (phandle_references): Reference to non-existent node or label "pinctrl_ssc0_tx" ERROR: Input tree has errors, aborting (use -f to force output) make[3]: *** [arch/arm/boot/dts/at91sam9g20ek.dtb] Error 2 I have included pending material that is dealing with SSC and pinctrl. The pinctrl part can be merged more easily now that the big pinctrl update for AT91 has been merged upstream. I have also included a typo patch that you have already taken as 315656b (ASoC: atmel-ssc: change disable to disable in dts node) in your topic/atmel branch but that was not described properly. If you find it easier, I can provide you a git tree for this material: just tell me... Thanks for your help, best regards, Bo Shen (3): ARM: at91/dts: fix typo in SSC status property ARM: at91/dts: add pinctrl support for SSC peripheral ASoC: atmel-ssc: add pinctrl selection to driver arch/arm/boot/dts/at91sam9260.dtsi | 18 +++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9g45.dtsi | 40 ++++++++++++++++++++++++++++++++++++-- arch/arm/boot/dts/at91sam9n12.dtsi | 26 +++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 153 insertions(+), 11 deletions(-) -- 1.8.0

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Re: [alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Nicolas Ferre 17 Dec '12

17 Dec '12

On 12/17/2012 10:39 AM, Jean-Christophe PLAGNIOL-VILLARD : > On 11:24 Wed 21 Nov , Mark Brown wrote: >> On Wed, Nov 21, 2012 at 10:17:09AM +0800, Bo Shen wrote: >>> Add pinctrl support for atmel ssc peripheral >>> >>> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> >> >> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > you handle this one with scc Hold on Jean-Christophe, I made a little experiment this morning, and this patch is not applying correctly on: - linus' tree - linux-next - neither Mark's for-next branch So, *I* will manage this patch and try to make sure that Mark is able to seamlessly include it in his tree on top of a known source for a future ASoC "fixes" pull-request. Mark, what about rebasing this one on top of current Linus' tree (toguether with the one that you have just applied on the topic/atmel branch: (ASoC: atmel-ssc: change disable to disable in dts node)? I hope that once reworked, they will be able to reach Linus' tree soon because the lack of this patch is causing DTB compilation errors on all AT91... Best regards, -- Nicolas Ferre

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[alsa-devel] [PATCH] ASoC: atmel-ssc: change disable to disable in dts node
by Bo Shen 17 Dec '12

17 Dec '12

Change the value of status to disabled to keep the consistent Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- arch/arm/boot/dts/at91sam9260.dtsi | 2 +- arch/arm/boot/dts/at91sam9263.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9g45.dtsi | 4 ++-- arch/arm/boot/dts/at91sam9x5.dtsi | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index aaa42d8..03c56cb 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,7 +217,7 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffe0000 { diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 3b721ee..15f12dd 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -179,14 +179,14 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fff9c000 { compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; macb0: ethernet@fffbc000 { diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index acfa207..44a38d0 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -232,14 +232,14 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; - status = "disable"; + status = "disabled"; }; ssc1: ssc@fffa0000 { compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; - status = "disable"; + status = "disabled"; }; adc0: adc@fffb0000 { diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 69667d0..0beff72 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -92,7 +92,7 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xf0010000 0x4000>; interrupts = <28 4 5>; - status = "disable"; + status = "disabled"; }; tcb0: timer@f8008000 { -- 1.7.9.5

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[alsa-devel] "hda-intel: azx_get_response timeout"
by Mantas M. 17 Dec '12

17 Dec '12

After upgrading to Linux 3.7 (on Arch Linux, x86_64), I've been getting the following messages in dmesg: > [Dec11 21:52] hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x001f0500 > [ +1.006709] hda-intel: No response from codec, disabling MSI: last cmd=0x001f0500 > [ +1.006773] hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f0500 This seems to happen when resuming from suspend-to-memory, and is new to v3.7 – I haven't seen any such messages on any kernels I've used earlier (between v3.0 and v3.6.9). The device is: > # lspci -v > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) > Subsystem: ASUSTeK Computer Inc. Device 1643 > Flags: bus master, fast devsel, latency 0, IRQ 22 > Memory at d5200000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ > Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 > Capabilities: [100] Virtual Channel > Capabilities: [130] Root Complex Link > Kernel driver in use: snd_hda_intel Is there anything I can/should do about this? -- Mantas Mikulėnas <grawity(a)gmail.com>

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Re: [alsa-devel] Could you help me about the ALSA?
by Zhang wei 17 Dec '12

17 Dec '12

> Yes,when buffer_size wraped the 'pos' is smallar than last one.If we > want to confirm the hw_base need to adjust there must be consider > another confidition: > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) > > how to connect this code with checking unnormal interrupt(lost or delayed)? This condition is for the double interrupt acknowledge problems (see comment). It makes sure, that the DMA buffer position was wrapped using another external timing source. Indeed, it's mostly a lowlevel driver fault to call this pointer update routine with the in_interrupt flag when the period was not really elapsed. -------------------------------------- I am sorry. I forget signed my info and this mail sent by me. Yes,I saw the comment and known the target of the pices of code and known alsa use jiffies as another timing source to detect the wrap situation,but I do not understand how to detect or check that case.In other word,why does alsa use the value runtime->hw_ptr_buffer_jiffies/2 not runtime->hw_ptr_buffer_jiffies/3 or runtime->hw_ptr_buffer_jiffies/4 ...? Best Regards zw

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Re: [alsa-devel] Could you help me about the ALSA?
by Jaroslav Kysela 17 Dec '12

17 Dec '12

Date 14.12.2012 09:07, Zhang wei wrote: > Good afternoon . > > Excuse me if I disturb you, I found your contact information from the > alsa-devel > http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030605.html > . > > Recently I am studying the alsa-driver and the code blocked me which > from the function snd_pcm_update_hw_ptr0inthe pcm_lib.c. > > =====================start==============================è > > if (in_interrupt) { > > /* we know that one period was processed */ > > /* delta = "expected next hw_ptr" for in_interrupt != > 0 */ > > delta = runtime->hw_ptr_interrupt + runtime->period_size; > > if (delta > new_hw_ptr) { > > /* check for double acknowledged interrupts */ > > hdelta = jiffies - runtime->hw_ptr_jiffies; > > if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { > > hw_base += runtime->buffer_size; > > if (hw_base >= runtime->boundary) > > hw_base = 0; > > new_hw_ptr = hw_base + pos; > > goto __delta; > > } > > } > > } > > ç================end================================= > > I’ve already read your mail but I still do not understand your intend. I > know the target of the pices of code is prevent the variable hw_ptr and > hw_base delayed update from some kind of case .so my question is > > (1) how could happen on delta>new_hw_ptr? And why? > > (2) Why we can conclue the hw_ptr crossed the boundary of > buffer_size when hdelta>runitme->hw_ptr_jiffies/2? What principle we > based on? The interrupts can be lost or delayed in some cases (small period sizes etc.). So you need to check for these situations. The driver returns only position in the ring buffer without count of "cross" points, so we need to trace these situations using another clock source (like system clock). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: cs42l52: Fix off-by-one boundary checking for cs42l52_get_clk()
by Axel Lin 16 Dec '12

16 Dec '12

The caller uses the return value of cs42l52_get_clk as array index, the valid array index is 0 .. ARRAY_SIZE(clk_map_table) - 1. Thus we need to ensure ret is less than ARRAY_SIZE(clk_map_table). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c6..66817f6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -749,7 +749,7 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) + if (ret >= ARRAY_SIZE(clk_map_table)) return -EINVAL; return ret; } -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: pcm: allow backend hardware to be freed in pause state
by Patrick Lai 15 Dec '12

15 Dec '12

When front-end PCM session is in paused state, back-end PCM session will be put in paused state as well if given front-end PCM session is the only client of given back-end. Then, application closes front-end PCM session, DPCM framework will not allow back-end enters HW_FREE state so back-end will never get shutdown completely. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- sound/soc/soc-pcm.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ef22d0b..d340644 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1240,7 +1240,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; -- The Qualcomm Innovation Center, Inc. is a member of Code Aurora Forum, hosted by The Linux Foundation 1.7.8.3

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[alsa-devel] [PATCH] ASoC: Prevent pop_wait overwrite
by Misael Lopez Cruz 15 Dec '12

15 Dec '12

pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> --- include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 12 ++++++------ 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 628db7b..3953cea 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,7 +242,6 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; struct snd_pcm_runtime *runtime; unsigned int active; - unsigned char pop_wait:1; unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; diff --git a/include/sound/soc.h b/include/sound/soc.h index 91244a0..769e27c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1039,6 +1039,7 @@ struct snd_soc_pcm_runtime { struct snd_soc_dpcm_runtime dpcm[2]; long pmdown_time; + unsigned char pop_wait:1; /* runtime devices */ struct snd_pcm *pcm; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e1..5fbfb06 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } else - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } else { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5c3ca2a..d7711fc 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work) dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } @@ -408,7 +408,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } @@ -480,8 +480,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; + rtd->pop_wait) { + rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); } -- 1.7.9.5

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[alsa-devel] [PATCH] configure: do not detect incorrect cross-compiler
by Olivier Blin 15 Dec '12

15 Dec '12

On Ubuntu 11.04, configuring with --build=x86_64-unknown-linux-gnu --host=x86_64-linux-gnu finds a wrong cross-compiler: checking for cross-compiler... x86_64-x86_64-pc-linux-gnu-gcc This happens because of a dash vs underscore inconsistency in configure.in: host=x86_64-pc-linux-gnu host_cpu=x86_64 host_os=linux-gnu which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 && echo ${host_cpu}-${host-os}-gcc This bug has been introduced in the initial --with-host support from 2002, commit eb267ade29c9a49c07b1c33dc9bf7a6790217400 This configure command is about "cross-compiling for i586", where the system compiler is used, which just -m32 additional options. The --build value comes from config.guess. --- configure.in | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.in b/configure.in index e397dad..4af917b 100644 --- a/configure.in +++ b/configure.in @@ -31,7 +31,7 @@ then which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host-os}-gcc + && CC=${host_cpu}-${host_os}-gcc which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ && CC=${host_cpu}-${host_vendor}-${host_os}-gcc -- 1.8.0.2

2 1

[alsa-devel] [PATCH] usb-audio: ignore-quirk for HP Wireless Audio
by Eldad Zack 15 Dec '12

15 Dec '12

As Joe Cooper <swelljoe(a)gmail.com> reported, "On most HP Envy laptops the snd-usb-audio module causes the system to become unresponsive and Gnome Shell 3 to crash.". See also: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.h… Add a quirk to ignore this device (for now) to solve the instability issue and allow other USB audio devices to be used. Reported-by: Joe Cooper <swelljoe(a)gmail.com> Tested-by: Isaac Smith <hunternet93(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/quirks-table.h | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) Applies against mainline 3.7.0 and sound-3.8. diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 49f9af9..e3c59f0 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -99,6 +99,42 @@ }, /* + * HP Wireless Audio + * When not ignored, causes instability issues for some users, forcing them to + * blacklist the entire module. + */ +{ + USB_DEVICE(0x0424, 0xb832), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Standard Microsystems Corp.", + .product_name = "HP Wireless Audio", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + /* Mixer */ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* HID Device, .ifnum = 3 */ + { + .ifnum = -1, + } + } + } +}, + +/* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. */ -- 1.7.8.6

2 1

[alsa-devel] [PATCH v2 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..42fdf14 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,54 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,6 +1746,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); -- 1.7.9.5

2 2

[alsa-devel] [PATCH 1/1] ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pin
by mengdong.lin＠intel.com 14 Dec '12

14 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI. So this patch - allow generic hdmi patch to pick up and apply fixups for different codecs. - add fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override invalid connection list entries reported by hardware. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 71555cc..2bf1758 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -38,6 +38,7 @@ #include <sound/tlv.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_auto_parser.h" #include "hda_jack.h" static bool static_hdmi_pcm; @@ -79,6 +80,7 @@ struct hdmi_spec_per_pin { }; struct hdmi_spec { + struct hda_gen_spec gen; int num_cvts; struct hdmi_spec_per_cvt cvts[MAX_HDMI_CVTS]; @@ -1687,6 +1689,69 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_enable_dp12_mode(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); +} + +static void intel_haswell_fixup_connect_list(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + unsigned int vendor_param; + + if (action != HDA_FIXUP_ACT_PROBE) + return; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + snd_hda_override_conn_list(codec, 0x06, 3, list); + hdmi_read_pin_conn(codec, 1); +} + + +/* available models for fixup */ +enum { + INTEL_HASWELL, + INTEL_HASWELL_CONNECT_LIST, +}; + +static const struct hda_model_fixup hdmi_models[] = { + {.id = INTEL_HASWELL, .name = "Haswell"}, + {} +}; + +static const struct snd_pci_quirk hdmi_fixup_tbl[] = { + SND_PCI_QUIRK(0x8086, 0x2010, "Haswell", INTEL_HASWELL), + {} /* terminator */ +}; + +static const struct hda_fixup hdmi_fixups[] = { + [INTEL_HASWELL] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_enable_dp12_mode, + .chained = true, + .chain_id = INTEL_HASWELL_CONNECT_LIST, + }, + [INTEL_HASWELL_CONNECT_LIST] = { + .type = HDA_FIXUP_FUNC, + .v.func = intel_haswell_fixup_connect_list, + }, +}; + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1696,11 +1761,19 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + snd_hda_gen_init(&spec->gen); + + snd_hda_pick_fixup(codec, hdmi_models, hdmi_fixup_tbl, hdmi_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); return -EINVAL; } + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + codec->patch_ops = generic_hdmi_patch_ops; generic_hdmi_init_per_pins(codec); -- 1.7.9.5

3 4

Re: [alsa-devel] ALSA:aloop -- inconsistent {in-softirq-W} -> {softirq-on-W} usage
by Takashi Iwai 14 Dec '12

14 Dec '12

At Thu, 13 Dec 2012 21:36:05 +0100, Andreas Fenkart wrote: > > > Hi, > > Seen this error when stopping playback to loopback. > > # aplay -Dhw:0 -fcd /dev/urandom > Playing raw data '/dev/urandom' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo > ^CAborted by signa[ 65.562992] > [ 65.565493] ================================= > [ 65.570044] [ INFO: inconsistent lock state ] > [ 65.574601] 3.7.0-rc8+ #303 Not tainted > [ 65.578605] --------------------------------- > [ 65.583156] inconsistent {IN-SOFTIRQ-W} -> {SOFTIRQ-ON-W} usage. > [ 65.589436] aplay/156 [HC0[0]:SC0[0]:HE1:SE1] takes: > [ 65.594620] (&(&dpcm->timer_lock)->rlock){+.?...}, at: [<c02e90cd>] loopback_close+0x29/0x98 > [ 65.603604] {IN-SOFTIRQ-W} state was registered at: > [ 65.608699] [<c005d8ef>] mark_irqflags+0xa3/0x11c > [ 65.613820] [<c005dca1>] __lock_acquire+0x339/0x77c > [ 65.619112] [<c005e575>] lock_acquire+0x65/0xac > [ 65.624038] [<c03e0c17>] _raw_spin_lock+0x1f/0x2c > [ 65.629155] [<c02e8cdd>] loopback_timer_start+0x3d/0x108 > [ 65.634899] [<c02e9455>] loopback_timer_function+0x21/0x3c > [ 65.640821] [<c0035f3f>] call_timer_fn+0x63/0xfc > [ 65.645849] [<c0036127>] run_timer_softirq+0x14f/0x190 > [ 65.651410] [<c00311d1>] __do_softirq+0xb1/0x174 > [ 65.656426] [<c0031529>] irq_exit+0x6d/0x70 > [ 65.660989] [<c000dde9>] handle_IRQ+0x2d/0x60 > [ 65.665743] [<c000848d>] omap3_intc_handle_irq+0x59/0x60 > [ 65.671484] [<c000cfff>] __irq_svc+0x3f/0x64 > [ 65.676138] [<c000de86>] default_idle+0x22/0x24 > [ 65.681063] [<c000df89>] cpu_idle+0x89/0x94 > [ 65.685626] [<c05cf5bb>] start_kernel+0x25b/0x264 > [ 65.690743] irq event stamp: 74155 > [ 65.694296] hardirqs last enabled at (74155): [<c03df9b5>] __mutex_unlock_slowpath+0x85/0xec > [ 65.703215] hardirqs last disabled at (74154): [<c03df96b>] __mutex_unlock_slowpath+0x3b/0xec > [ 65.712132] softirqs last enabled at (72724): [<c0031215>] __do_softirq+0xf5/0x174 > [ 65.720141] softirqs last disabled at (72717): [<c0031529>] irq_exit+0x6d/0x70 > [ 65.727697] > [ 65.727697] other info that might help us debug this: > [ 65.734516] Possible unsafe locking scenario: > [ 65.734516] > [ 65.740697] CPU0 > [ 65.743250] ---- > [ 65.745802] lock(&(&dpcm->timer_lock)->rlock); > [ 65.750639] <Interrupt> > [ 65.753373] lock(&(&dpcm->timer_lock)->rlock); > [ 65.758392] > [ 65.758392] *** DEADLOCK *** > [ 65.758392] > [ 65.764581] 1 lock held by aplay/156: > [ 65.768404] #0: (&pcm->open_mutex){+.+.+.}, at: [<c02e0c49>] snd_pcm_release+0x1d/0x7c > [ 65.776907] > [ 65.776907] stack backtrace: > [ 65.781487] [<c0011a15>] (unwind_backtrace+0x1/0xa0) from [<c005d2c9>] (print_usage_bug+0x149/0x184) > [ 65.791039] [<c005d2c9>] (print_usage_bug+0x149/0x184) from [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) > [ 65.800590] [<c005d4c5>] (mark_lock_irq+0x1c1/0x1e0) from [<c005d5f7>] (mark_lock+0x113/0x368) > [ 65.809597] [<c005d5f7>] (mark_lock+0x113/0x368) from [<c005d915>] (mark_irqflags+0xc9/0x11c) > [ 65.818514] [<c005d915>] (mark_irqflags+0xc9/0x11c) from [<c005dca1>] (__lock_acquire+0x339/0x77c) > [ 65.827884] [<c005dca1>] (__lock_acquire+0x339/0x77c) from [<c005e575>] (lock_acquire+0x65/0xac) > [ 65.846083] [<c03e0c17>] (_raw_spin_lock+0x1f/0x2c) from [<c02e90cd>] (loopback_close+0x29/0x98) > [ 65.855273] [<c02e90cd>] (loopback_close+0x29/0x98) from [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) > [ 65.865458] [<c02e0bf1>] (snd_pcm_release_substream+0x39/0x74) from [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) > [ 65.875748] [<c02e0c4f>] (snd_pcm_release+0x23/0x7c) from [<c00b4dad>] (__fput+0x61/0x170) > [ 65.884409] [<c00b4dad>] (__fput+0x61/0x170) from [<c0040485>] (task_work_run+0x5d/0x8c) > [ 65.892879] [<c0040485>] (task_work_run+0x5d/0x8c) from [<c001017f>] (do_work_pending+0x67/0x7c) > [ 65.902069] [<c001017f>] (do_work_pending+0x67/0x7c) from [<c000d569>] (work_pending+0x9/0x18) > > This seems to be a real deadlock, since loopback_timer_stop is > triggered from ioctl, while loopback_timer_start can be called > from softirq. Could you check the latest Linus tree? The spinlocks of aloop have been already fixed there. Takashi

1 0

[alsa-devel] Polling issues with the PCM multi plugin
by Devin Anderson 14 Dec '12

14 Dec '12

Hi all, On the jack-devel list, we've been having a discussion about the ALSA PCM multi plugin: http://thread.gmane.org/gmane.comp.audio.jackit/26571/focus=26571 I've been having some issues with setting up my 2 Echo Layla 3Gs as a PCM multi. In particular, I've found that there are times when polling the file descriptors of the multi when poll() returns, but there isn't actually a period of data available for capture/playback, even though the poll descriptors indicate that the data should be available. After some debugging and learning about the PCM multi driver: 1.) The PCM multi driver only returns poll descriptors for one of the devices that makes up the multi 2.) When this happens, the device that's actually being polled *does* have data available 3.) When this happens, the device that's not being polled does not yet have a period's worth of data available After making sure that I had word clock enabled and that my word clock cable was working properly, I started delving into the problem some more. From what I can gather, the PCM multi device uses the function `snd_pcm_link()` to sync its devices, and, for start, stop, drain, pause, suspend, resume, reset, and prepare operations, this appears to work fine; however, I can't find any evidence that a PCM device checks to make sure that all of its linked slave devices have their poll descriptors ready with events before indicating that events are available. Questions: 1.) Should `snd_pcm_link` make I/O availability indication dependent on I/O availability of its linked slave devices? 2.) If not, how should this operation be handled? I'm sure it *can* be handled in userspace by bypassing the multi and polling the file descriptors of all sound cards directly, but duplicating this code for each application that wants to poll multiple sound cards seems wrong. Thanks, -- Devin Anderson surfacepatterns (at) gmail (dot) com blog - http://surfacepatterns.blogspot.com/ psinsights - http://psinsights.googlecode.com/ synthclone - http://synthclone.googlecode.com/

2 5

[alsa-devel] [PATCH] ASoC: wm5110: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 77 +++++++++++++++++++++++++++------------------ 1 file changed, 46 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 794f050..55a1348 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -55,37 +55,52 @@ SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN4R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_IN4L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN4R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4R, + ARIZONA_IN4R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm2000: Use clock API integration to configure MCLK divisor
by Mark Brown 14 Dec '12

14 Dec '12

Since we are now using the clock API integration to manage MCLK we can now use clk_get_rate() to determine if we need to divide MCLK without relying on platform data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm2000.h | 3 --- sound/soc/codecs/wm2000.c | 6 +++--- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h index aa388ca..4de81f4 100644 --- a/include/sound/wm2000.h +++ b/include/sound/wm2000.h @@ -15,9 +15,6 @@ struct wm2000_platform_data { /** Filename for system-specific image to download to device. */ const char *download_file; - /** Divide MCLK by 2 for system clock? */ - unsigned int mclkdiv2:1; - /** Disable speech clarity enhancement, for use when an * external algorithm is used. */ unsigned int speech_enh_disable:1; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 0aba8ce..85550dc 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -73,7 +73,6 @@ struct wm2000_priv { unsigned int anc_eng_ena:1; unsigned int spk_ena:1; - unsigned int mclk_div:1; unsigned int speech_clarity:1; int anc_download_size; @@ -133,6 +132,7 @@ static int wm2000_poll_bit(struct i2c_client *i2c, static int wm2000_power_up(struct i2c_client *i2c, int analogue) { struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); + unsigned long rate; int ret; BUG_ON(wm2000->anc_mode != ANC_OFF); @@ -145,7 +145,8 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) return ret; } - if (!wm2000->mclk_div) { + rate = clk_get_rate(wm2000->mclk); + if (rate <= 13500000) { dev_dbg(&i2c->dev, "Disabling MCLK divider\n"); wm2000_write(i2c, WM2000_REG_SYS_CTL2, WM2000_MCLK_DIV2_ENA_CLR); @@ -847,7 +848,6 @@ static int wm2000_i2c_probe(struct i2c_client *i2c, filename = "wm2000_anc.bin"; pdata = dev_get_platdata(&i2c->dev); if (pdata) { - wm2000->mclk_div = pdata->mclkdiv2; wm2000->speech_clarity = !pdata->speech_enh_disable; if (pdata->download_file) -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Split input PGA controls
by Mark Brown 14 Dec '12

14 Dec '12

Though the controls are named as stereo controls in the part the main use case for the analogue inputs to the WM5102 is mono. Reflect this in the controls exposed to userspace, providing a series of mono controls rather than stereo ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 64 +++++++++++++++++++++++++++------------------ 1 file changed, 38 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b20feb2..3831645 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -611,32 +611,44 @@ SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1R_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2R_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3R_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), - -SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), -SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), - -SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, - 0xbf, 0, digital_tlv), +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE("IN1L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN1R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN2R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3L Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("IN3R Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN3R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_IN3R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), -- 1.7.10.4

1 0

[alsa-devel] Presonus 44VSL AudioBox drivers
by Nicholas Hadaway 14 Dec '12

14 Dec '12

I own a Presonus AudioBox 44VSL and it does not work with alsa currently. I am an active linux user that runs git kernels and keeps an active development environment. I would love to get this hardware supported. what do I need to do to help you guys with this? Thanks! -Nick

1 0

[alsa-devel] Request for HW auto-recognition: HP Pavilion dv7 with STAC92xx
by Tomas Pospisek 13 Dec '12

13 Dec '12

Hello, I have a HP Pavilion dv7 laptop (LS066EA#UUZ), with: # cat /proc/asound/card0/codec* | grep Codec Codec: IDT 92HD81B1X5 Codec: Intel CougarPoint HDMI when play sound without any tweaks, I get a very weak sound output. If I want to have "beats audio", that is subwoofer and all speakers, I need to set: # cat /etc/modprobe.d/alsa-base.conf options snd-hda-intel model=hp-dv7-4000 Could the snd_hda_intel alsa driver be updated to recognize and setup the driver accordingly? alsa-info.sh output here: http://www.alsa-project.org/db/?f=54ea874f43f12c1572aff3ee3113dc4b0ad0b398 Thanks, *t

2 3

[alsa-devel] [PATCH v2 1/1] ALSA: hda - use usleep_range in link reset and change timeout check
by mengdong.lin＠intel.com 13 Dec '12

13 Dec '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Reducing the time on HDA link reset can help to reduce the driver loading time. So we replace msleep with usleep_range to get more accurate time control and change the value to a smaller one. And a 100ms timeout is set for both entering and exiting the link reset. Signed-off-by: Xingchao Wang <xingchao.wang(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 72b085a..ed3a290 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1041,7 +1041,7 @@ static void azx_power_notify(struct hda_bus *bus, bool power_up); /* reset codec link */ static int azx_reset(struct azx *chip, int full_reset) { - int count; + unsigned long timeout; if (!full_reset) goto __skip; @@ -1052,24 +1052,26 @@ static int azx_reset(struct azx *chip, int full_reset) /* reset controller */ azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET); - count = 50; - while (azx_readb(chip, GCTL) && --count) - msleep(1); + timeout = jiffies + msecs_to_jiffies(100); + while (azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - msleep(1); + usleep_range(500, 1000); /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); - count = 50; - while (!azx_readb(chip, GCTL) && --count) - msleep(1); + timeout = jiffies + msecs_to_jiffies(100); + while (!azx_readb(chip, GCTL) && + time_before(jiffies, timeout)) + usleep_range(500, 1000); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ - msleep(1); + usleep_range(1000, 1200); __skip: /* check to see if controller is ready */ -- 1.7.9.5

5 12

[alsa-devel] [PATCH 0/2] Make ASUS Xonar U3 S/PDIF channel accessible
by Denis Washington 13 Dec '12

13 Dec '12

Hello, This is my first ALSA patch ever (in fact, also my first kernel patch and my first e-mail to this mailing list); as such, I'm sorry when I did something wrong and/or didn't follow the procedure properly. I hope you can forgive me. I recently bought an ASUS Xonar U3 USB audio card. It general it works well with Linux, but unfortunately, it's PCM output channel is unusable for headphones because it allows virtually no volume control (maximum loudness is already reach at a volume level of 2 or 3). The card does also have a digital (S/PDIF) output channel, which however isn't switched on by the Linux USB audio driver: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg28970.html When I boot into Windows (where the ASUS driver seemingly enables the channel) and then reboot into Linux, I can use this channel and control volume as expected. It's just this turning on of S/PDIF which is missing, as suggested by the alsa-user thread linked to above. The patchset that follows enables S/PDIF on the Xonar U3 by using the (seemingly compatible) mixer quirks code as for the U1, just as done and tested by the initiator of the alsa-user thread, together with a very minor cleanup (renaming internal xonar_u1_* identifiers to just xonar_u_*). The change seems to work well (tested with the 3.7.0 Ubuntu kernel as found in today' daily version). Regard, Denis Washington

2 4

Re: [alsa-devel] [PATCH 1/2] ASoC: tlv320aic326x: Support for TI TLV320AIC3262 audio driver
by Mark Brown 13 Dec '12

13 Dec '12

On Wed, Dec 12, 2012 at 04:23:48PM +0530, Mehar Bajwa wrote: > TI TLV320AIC3262 is a flexible, low-power, low-voltage stereo audio > codec with digital microphone inputs and programmable outputs. It includes > powertune capabilities, enhanced fully-programmable miniDSP, fixed predefined > and parameterizable signal processing blocks, integrated PLL and flexible digital > interfaces. Is this in some way different to the patch set you sent yesterday?

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: Resubmit device driver
by Mark Brown 13 Dec '12

13 Dec '12

On Wed, Dec 12, 2012 at 04:36:10PM -0800, Jerry Wong wrote: > I have made the changes you suggested below, except: I've not read the patch yet but I see that once more I need to ask you to look at and follow SubmittingPatches - you're getting closer but the at least the changelog is nothing like it should be. Please look at both SubmittingPatches and other people's patches and try to follow the standard form.

1 0

[alsa-devel] [PATCH 1/1] ASoC: tpa6130a2: Use devm_* APIs
by Sachin Kamat 13 Dec '12

13 Dec '12

Converted to use devm_gpio_request and devm_regulator_get APIs. These are device managed and make error handling and cleanup a bit simpler. Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- Compile tested on for-next branch of following tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git --- sound/soc/codecs/tpa6130a2.c | 23 ++++++----------------- 1 files changed, 6 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 565ff39..ec78073 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -398,7 +398,8 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, TPA6130A2_MUTE_L; if (data->power_gpio >= 0) { - ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + ret = devm_gpio_request(dev, data->power_gpio, + "tpa6130a2 enable"); if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); @@ -419,16 +420,16 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, break; } - data->supply = regulator_get(dev, regulator); + data->supply = devm_regulator_get(dev, regulator); if (IS_ERR(data->supply)) { ret = PTR_ERR(data->supply); dev_err(dev, "Failed to request supply: %d\n", ret); - goto err_regulator; + goto err_gpio; } ret = tpa6130a2_power(1); if (ret != 0) - goto err_power; + goto err_gpio; /* Read version */ @@ -440,15 +441,10 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, /* Disable the chip */ ret = tpa6130a2_power(0); if (ret != 0) - goto err_power; + goto err_gpio; return 0; -err_power: - regulator_put(data->supply); -err_regulator: - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); err_gpio: tpa6130a2_client = NULL; @@ -457,14 +453,7 @@ err_gpio: static int __devexit tpa6130a2_remove(struct i2c_client *client) { - struct tpa6130a2_data *data = i2c_get_clientdata(client); - tpa6130a2_power(0); - - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); - - regulator_put(data->supply); tpa6130a2_client = NULL; return 0; -- 1.7.4.1

2 3

[alsa-devel] [Resend PATCH v6 1/4] pinctrl: atmel-ssc: add pinctrl support
by Bo Shen 13 Dec '12

13 Dec '12

Add pinctrl support for atmel ssc peripheral Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Resend - based on linux-next-20121115 - commit id: 3805711eb31d3b980c4a55a6994bbc8b22a9ec65 Change since v5 - correct the pull setting, without pull up Change since v4 - This is initial version --- arch/arm/boot/dts/at91sam9260.dtsi | 18 ++++++++++++++++++ arch/arm/boot/dts/at91sam9263.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9g45.dtsi | 36 ++++++++++++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9n12.dtsi | 26 ++++++++++++++++++++++++++ arch/arm/boot/dts/at91sam9x5.dtsi | 32 +++++++++++++++++++++++++------- drivers/misc/atmel-ssc.c | 8 ++++++++ 6 files changed, 149 insertions(+), 7 deletions(-) diff --git a/arch/arm/boot/dts/at91sam9260.dtsi b/arch/arm/boot/dts/at91sam9260.dtsi index 3164dcd..f9dc7db 100644 --- a/arch/arm/boot/dts/at91sam9260.dtsi +++ b/arch/arm/boot/dts/at91sam9260.dtsi @@ -217,6 +217,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <1 16 0x1 0x0 /* PB16 periph A */ + 1 17 0x1 0x0 /* PB17 periph A */ + 1 18 0x1 0x0>; /* PB18 periph A */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <1 19 0x1 0x0 /* PB19 periph A */ + 1 20 0x1 0x0 /* PB20 periph A */ + 1 21 0x1 0x0>; /* PB21 periph A */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -350,6 +366,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfffbc000 0x4000>; interrupts = <14 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9263.dtsi b/arch/arm/boot/dts/at91sam9263.dtsi index 96e0bc9..6e9a4ff 100644 --- a/arch/arm/boot/dts/at91sam9263.dtsi +++ b/arch/arm/boot/dts/at91sam9263.dtsi @@ -165,6 +165,38 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <1 0 0x2 0x0 /* PB0 periph B */ + 1 1 0x2 0x0 /* PB1 periph B */ + 1 2 0x2 0x0>; /* PB2 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <1 3 0x2 0x0 /* PB3 periph B */ + 1 4 0x2 0x0 /* PB4 periph B */ + 1 5 0x2 0x0>; /* PB5 periph B */ + }; + }; + + ssc1 { + pinctrl_ssc1_tx: ssc1_tx-0 { + atmel,pins = + <1 6 0x1 0x0 /* PB6 periph A */ + 1 7 0x1 0x0 /* PB7 periph A */ + 1 8 0x1 0x0>; /* PB8 periph A */ + }; + + pinctrl_ssc1_rx: ssc1_rx-0 { + atmel,pins = + <1 9 0x1 0x0 /* PB9 periph A */ + 1 10 0x1 0x0 /* PB10 periph A */ + 1 11 0x1 0x0>; /* PB11 periph A */ + }; + }; + pioA: gpio@fffff200 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff200 0x200>; @@ -262,6 +294,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff98000 0x4000>; interrupts = <16 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; @@ -269,6 +303,8 @@ compatible = "atmel,at91rm9200-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <17 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc1_tx &pinctrl_ssc1_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9g45.dtsi b/arch/arm/boot/dts/at91sam9g45.dtsi index c40c3eb..63cf206 100644 --- a/arch/arm/boot/dts/at91sam9g45.dtsi +++ b/arch/arm/boot/dts/at91sam9g45.dtsi @@ -198,6 +198,38 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <3 0 0x1 0x0 /* PD0 periph A */ + 3 1 0x1 0x0 /* PD1 periph A */ + 3 2 0x1 0x0>; /* PD2 periph A */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <3 3 0x1 0x0 /* PD3 periph A */ + 3 4 0x1 0x0 /* PD4 periph A */ + 3 5 0x1 0x0>; /* PD5 periph A */ + }; + }; + + ssc1 { + pinctrl_ssc1_tx: ssc1_tx-0 { + atmel,pins = + <3 10 0x1 0x0 /* PD10 periph A */ + 3 11 0x1 0x0 /* PD11 periph A */ + 3 12 0x1 0x0>; /* PD12 periph A */ + }; + + pinctrl_ssc1_rx: ssc1_rx-0 { + atmel,pins = + <3 13 0x1 0x0 /* PD13 periph A */ + 3 14 0x1 0x0 /* PD14 periph A */ + 3 15 0x1 0x0>; /* PD15 periph A */ + }; + }; + pioA: gpio@fffff200 { compatible = "atmel,at91rm9200-gpio"; reg = <0xfffff200 0x200>; @@ -331,6 +363,8 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfff9c000 0x4000>; interrupts = <16 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; status = "disable"; }; @@ -338,6 +372,8 @@ compatible = "atmel,at91sam9g45-ssc"; reg = <0xfffa0000 0x4000>; interrupts = <17 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc1_tx &pinctrl_ssc1_rx>; status = "disable"; }; diff --git a/arch/arm/boot/dts/at91sam9n12.dtsi b/arch/arm/boot/dts/at91sam9n12.dtsi index 7b644c5..b211da6 100644 --- a/arch/arm/boot/dts/at91sam9n12.dtsi +++ b/arch/arm/boot/dts/at91sam9n12.dtsi @@ -28,6 +28,7 @@ tcb1 = &tcb1; i2c0 = &i2c0; i2c1 = &i2c1; + ssc0 = &ssc0; }; cpus { cpu@0 { @@ -199,6 +200,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <0 24 0x2 0x0 /* PA24 periph B */ + 0 25 0x2 0x0 /* PA25 periph B */ + 0 26 0x2 0x0>; /* PA26 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <0 27 0x2 0x0 /* PA27 periph B */ + 0 28 0x2 0x0 /* PA28 periph B */ + 0 29 0x2 0x0>; /* PA29 periph B */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91sam9x5-gpio", "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -249,6 +266,15 @@ status = "disabled"; }; + ssc0: ssc@f0010000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0xf0010000 0x4000>; + interrupts = <28 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; + status = "disable"; + }; + usart0: serial@f801c000 { compatible = "atmel,at91sam9260-usart"; reg = <0xf801c000 0x4000>; diff --git a/arch/arm/boot/dts/at91sam9x5.dtsi b/arch/arm/boot/dts/at91sam9x5.dtsi index 1cb26c28..df84f39 100644 --- a/arch/arm/boot/dts/at91sam9x5.dtsi +++ b/arch/arm/boot/dts/at91sam9x5.dtsi @@ -88,13 +88,6 @@ interrupts = <1 4 7>; }; - ssc0: ssc@f0010000 { - compatible = "atmel,at91sam9g45-ssc"; - reg = <0xf0010000 0x4000>; - interrupts = <28 4 5>; - status = "disable"; - }; - tcb0: timer@f8008000 { compatible = "atmel,at91sam9x5-tcb"; reg = <0xf8008000 0x100>; @@ -214,6 +207,22 @@ }; }; + ssc0 { + pinctrl_ssc0_tx: ssc0_tx-0 { + atmel,pins = + <0 24 0x2 0x0 /* PA24 periph B */ + 0 25 0x2 0x0 /* PA25 periph B */ + 0 26 0x2 0x0>; /* PA26 periph B */ + }; + + pinctrl_ssc0_rx: ssc0_rx-0 { + atmel,pins = + <0 27 0x2 0x0 /* PA27 periph B */ + 0 28 0x2 0x0 /* PA28 periph B */ + 0 29 0x2 0x0>; /* PA29 periph B */ + }; + }; + pioA: gpio@fffff400 { compatible = "atmel,at91sam9x5-gpio", "atmel,at91rm9200-gpio"; reg = <0xfffff400 0x200>; @@ -266,6 +275,15 @@ status = "disabled"; }; + ssc0: ssc@f0010000 { + compatible = "atmel,at91sam9g45-ssc"; + reg = <0xf0010000 0x4000>; + interrupts = <28 4 5>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_ssc0_tx &pinctrl_ssc0_rx>; + status = "disable"; + }; + usart0: serial@f801c000 { compatible = "atmel,at91sam9260-usart"; reg = <0xf801c000 0x200>; diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index a769719..3b3d0e0 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/of.h> +#include <linux/pinctrl/consumer.h> /* Serialize access to ssc_list and user count */ static DEFINE_SPINLOCK(user_lock); @@ -131,6 +132,13 @@ static int ssc_probe(struct platform_device *pdev) struct resource *regs; struct ssc_device *ssc; const struct atmel_ssc_platform_data *plat_dat; + struct pinctrl *pinctrl; + + pinctrl = devm_pinctrl_get_select_default(&pdev->dev); + if (IS_ERR(pinctrl)) { + dev_err(&pdev->dev, "Failed to request pinctrl\n"); + return PTR_ERR(pinctrl); + } ssc = devm_kzalloc(&pdev->dev, sizeof(struct ssc_device), GFP_KERNEL); if (!ssc) { -- 1.7.9.5

2 3

[alsa-devel] "disable" or "disabled" in DT?
by Takashi Iwai 13 Dec '12

13 Dec '12

Hi, in the recent commit for atmel-ssc to add DT support, you added the lines status = "disable"; Now looking through all commits for merging 3.8-rc1, I noticed that other codes have status = "disabled"; I think this should be kept consistent. Takashi

2 1

[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by isaac smith 12 Dec '12

12 Dec '12

Success! I applied the patch against kernel 3.6.9 (AMD's Catalyst driver doesn't support 3.7 yet) and I can now use the snd-usb-audio driver without it hanging up on the wireless audio. I tested a USB sound card, it worked perfectly. There's no sign of the HP Wireless card in ALSA. Thanks a lot! Now I can finally use my slightly decent speakers and my USB MIDI controller. On Tue, Dec 11, 2012 at 12:59 PM, Eldad Zack <eldad(a)fogrefinery.com> wrote: > > On Mon, 10 Dec 2012, isaac smith wrote: > >> I have the same issue with my HP Envy 15. The sound card in question >> is an HP Wireless Audio card that only works with HP-branded wireless >> speakers, so playing to the card produces no output whatsoever, but >> does hang whatever process tries to access it. I doubt that many >> people ever use this feature, so unless someone invests the time to >> get it working I'd like a way to just ignore it completely. I've >> blacklisted the snd-usb-audio module for now, but I have a USB sound >> card and a USB MIDI controller that I'd like to be able to use. > > Hi Isaac, > > Try this patch - applies against 3.7. > > Let me know if that works for you and you can successfully use other USB > audio devices. > > Cheers, > Eldad > > diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h > index 88d8ceb..be68c66 100644 > --- a/sound/usb/quirks-table.h > +++ b/sound/usb/quirks-table.h > @@ -99,6 +99,42 @@ > }, > > /* > + * Standard Microsystems Corp. > + * HP Wireless Audio > + */ > +{ > + USB_DEVICE(0x0424, 0xb832), > + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { > + .vendor_name = "Standard Microsystems Corp.", > + .product_name = "HP Wireless Audio", > + .ifnum = QUIRK_ANY_INTERFACE, > + .type = QUIRK_COMPOSITE, > + .data = (const struct snd_usb_audio_quirk[]) { > + /* Mixer */ > + { > + .ifnum = 0, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* Playback */ > + { > + .ifnum = 1, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* Capture */ > + { > + .ifnum = 2, > + .type = QUIRK_IGNORE_INTERFACE, > + }, > + /* HID Device, maybe standard compliant? */ > + { > + /* .ifnum = 3 */ > + .ifnum = -1, > + } > + } > + } > +}, > + > +/* > * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface > * class matches do not take effect without an explicit ID match. > */ > >

1 0

[alsa-devel] Bug in aplay?
by Damien Zammit 12 Dec '12

12 Dec '12

Hi, I think i found a small bug in alsa utility "aplay". If you have a sound device that supports the S24_3BE format natively, and you have a wav file of the same format, aplay will not play it because it detects the wrong format and assumes it is default format. (ie 8000Hz Mono 1 channel). I compiled my own version of aplay and it works because I hacked the default format to match this format, but it is not the ideal solution. pi@raspberrypi ~ $ aplay --version aplay: version 1.0.25 by Jaroslav Kysela <perex(a)perex.cz> pi@raspberrypi ~ $ aplay -D hw:1 48000-S24_3BE.wav Playing raw data '48000-S24_3BE.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:1081: Sample format non available Available formats: - S24_3BE My version: pi@raspberrypi ~ $ ./aplay -D hw:1 48000-S24_3BE.wav Playing raw data '48000-S24_3BE.wav' : Signed 24 bit Big Endian in 3bytes, Rate 48000 Hz, Stereo aplay: set_params:1145: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_3BE SUBFORMAT: STD SAMPLE_BITS: 24 FRAME_BITS: 48 CHANNELS: 2 RATE: 48000 PERIOD_TIME: 125000 PERIOD_SIZE: 6000 PERIOD_BYTES: 36000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 24000 BUFFER_BYTES: 144000 TICK_TIME: 0 pi@raspberrypi ~ $ This is making it tricky for me to use my sound card. For some reason it works better with jackd. The error above is also a mystery to me, but it seems to be detecting the format better. Regards, Damien

3 6

[alsa-devel] [PATCH 0/2] Fix usb-audio delay calculation
by Takashi Iwai 12 Dec '12

12 Dec '12

Hi, the following two patches are for fixing the wrong PCM delay reported from usb-audio driver. The first one is trivial, just to fix / optimize for the capture direction. The second one is a deeper fix for corner cases like playback pause. Takashi

2 7

[alsa-devel] [PATCH 2/2] ASoC: aic: Support for AIC family DSPs
by Mehar Bajwa 12 Dec '12

12 Dec '12

AIC family of audio CODECs from TI features a programmable miniDSP for performing signal processing operations. Due to commonality of functions across the CODECs a common library will be used to provide support for them. Signed-off-by: Mehar Bajwa <mehar.bajwa(a)ti.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aic3xxx_cfw.h | 529 +++++++++++++++++++ sound/soc/codecs/aic3xxx_cfw_ops.c | 989 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/aic3xxx_cfw_ops.h | 98 ++++ 5 files changed, 1623 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/aic3xxx_cfw.h create mode 100644 sound/soc/codecs/aic3xxx_cfw_ops.c create mode 100644 sound/soc/codecs/aic3xxx_cfw_ops.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 73d8cea..52b3601 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -188,6 +188,11 @@ config SND_SOC_ADAV80X config SND_SOC_ADS117X tristate +config SND_SOC_AIC_CFW + tristate + default y if SND_SOC_TLV320AIC3262=y + default m if SND_SOC_TLV320AIC3262=m + config SND_SOC_AK4104 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a3f8f4f..ffda11b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -9,6 +9,7 @@ snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o +snd-soc-aic3xxx-cfw-ops-objs := aic3xxx_cfw_ops.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o @@ -132,6 +133,7 @@ obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o +obj-$(CONFIG_SND_SOC_AIC_CFW) += snd-soc-aic3xxx-cfw-ops.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o diff --git a/sound/soc/codecs/aic3xxx_cfw.h b/sound/soc/codecs/aic3xxx_cfw.h new file mode 100644 index 0000000..b88485d --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw.h @@ -0,0 +1,529 @@ +/* + * aic3xxx_cfw.h -- SoC audio for TI OMAP44XX SDP + * Codec Firmware Declarations + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef AIC_FIRMWARE_H_ +#define AIC_FIRMWARE_H_ + + +#define AIC_FW_MAGIC 0xC0D1F1ED + + +/** \defgroup pd Arbitrary Limitations */ +/* @{ */ +#ifndef AIC_MAX_ID +# define AIC_MAX_ID (64) /**<Max length of string identifies*/ +# define AIC_MAX_VARS (256) /**<Number of "variables" alive at the*/ + /**<same time in an acx file*/ +#endif + +/* @} */ + + + +/** \defgroup st Enums, Flags, Macros and Supporting Types */ +/* @{ */ + + +/** + * Device Family Identifier + * + */ +enum __attribute__ ((__packed__)) aic_dfamily { + AIC_DFM_TYPE_A, + AIC_DFM_TYPE_B, + AIC_DFM_TYPE_C +}; + +/** + * Device Identifier + * + */ +enum __attribute__ ((__packed__)) aic_device { + AIC_DEV_DAC3120, + AIC_DEV_DAC3100, + AIC_DEV_AIC3120, + AIC_DEV_AIC3100, + AIC_DEV_AIC3110, + AIC_DEV_AIC3111, + AIC_DEV_AIC36, + AIC_DEV_AIC3206, + AIC_DEV_AIC3204, + AIC_DEV_AIC3254, + AIC_DEV_AIC3256, + AIC_DEV_AIC3253, + AIC_DEV_AIC3212, + AIC_DEV_AIC3262, + AIC_DEV_AIC3017, + AIC_DEV_AIC3008, + + AIC_DEV_AIC3266, + AIC_DEV_AIC3285, +}; + +/** + * Transition Sequence Identifier + * + */ +enum aic_transition_t { + AIC_TRN_INIT, + AIC_TRN_RESUME, + AIC_TRN_NEUTRAL, + AIC_TRN_A_MUTE, + AIC_TRN_D_MUTE, + AIC_TRN_AD_MUTE, + AIC_TRN_A_UNMUTE, + AIC_TRN_D_UNMUTE, + AIC_TRN_AD_UNMUTE, + AIC_TRN_SUSPEND, + AIC_TRN_EXIT, + AIC_TRN_N +}; + +#ifndef __cplusplus +static const char *const aic_transition_id[] = { + [AIC_TRN_INIT] "INIT", + [AIC_TRN_RESUME] "RESUME", + [AIC_TRN_NEUTRAL] "NEUTRAL", + [AIC_TRN_A_MUTE] "A_MUTE", + [AIC_TRN_D_MUTE] "D_MUTE", + [AIC_TRN_AD_MUTE] "AD_MUTE", + [AIC_TRN_A_UNMUTE] "A_UNMUTE", + [AIC_TRN_D_UNMUTE] "D_UNMUTE", + [AIC_TRN_AD_UNMUTE]"AD_UNMUTE", + [AIC_TRN_SUSPEND] "SUSPEND", + [AIC_TRN_EXIT] "EXIT", +}; +#endif + +/* @} */ + +/** \defgroup ds Data Structures */ +/* @{ */ + + +/** +* AIC Command +* These commands do not appear in the register +* set of the device. +*/ +enum __attribute__ ((__packed__)) aic_cmd_id { + AIC_CMD_NOP = 0x80, + AIC_CMD_DELAY, + AIC_CMD_UPDTBITS, + AIC_CMD_WAITBITS, + AIC_CMD_LOCK, + AIC_CMD_BURST, + AIC_CMD_RBURST, + AIC_CMD_LOAD_VAR_IM, + AIC_CMD_LOAD_VAR_ID, + AIC_CMD_STORE_VAR, + AIC_CMD_COND, + AIC_CMD_BRANCH, + AIC_CMD_BRANCH_IM, + AIC_CMD_BRANCH_ID, + AIC_CMD_PRINT, + AIC_CMD_OP_ADD = 0xC0, + AIC_CMD_OP_SUB, + AIC_CMD_OP_MUL, + AIC_CMD_OP_DIV, + AIC_CMD_OP_AND, + AIC_CMD_OP_OR, + AIC_CMD_OP_SHL, + AIC_CMD_OP_SHR, + AIC_CMD_OP_RR, + AIC_CMD_OP_XOR, + AIC_CMD_OP_NOT, + AIC_CMD_OP_LNOT, +}; + +/** +* AIC Delay +* Used for the cmd command delay +* Has one parameter of delay time in ms +*/ +struct aic_cmd_delay { + u16 delay; + enum aic_cmd_id cid; + u8 delay_fine; +}; + +/** +* AIC Lock +* Take codec mutex to avoid clashing with DAPM operations +*/ +struct aic_cmd_lock { + u16 lock; + enum aic_cmd_id cid; + u8 unused; +}; + + +/** + * AIC UPDTBITS, WAITBITS, CHKBITS + * Both these cmd commands have same arguments + * cid will be used to specify which command it is + * has parameters of book, page, offset and mask + */ +struct aic_cmd_bitop { + u16 unused1; + enum aic_cmd_id cid; + u8 mask; +}; + +/** + * AIC CMD Burst header + * Burst writes inside command array + * Followed by burst address, first byte + */ +struct aic_cmd_bhdr { + u16 len; + enum aic_cmd_id cid; + u8 unused; +}; + +/** + * AIC CMD Burst + * Burst writes inside command array + * Followed by data to the extent indicated in previous len + * Can be safely cast to aic_burst + */ +struct aic_cmd_burst { + u8 book; + u8 page; + u8 offset; + u8 data[1]; +}; +#define AIC_CMD_BURST_LEN(n) (2 + ((n) - 1 + 3)/4) + +/** + * AIC CMD Scratch register + * For load + * if (svar != dvar) + * dvar = setbits(svar, mask) // Ignore reg + * else + * dvar = setbits(reg, mask) + * For store + * if (svar != dvar) + * reg = setbits(svar, dvar) + * else + * reg = setbits(svar, mask) + * + */ +struct aic_cmd_ldst { + u8 dvar; + u8 svar; + enum aic_cmd_id cid; + u8 mask; +}; + +/** + * AIC CMD Conditional + * May only precede branch. Followed by nmatch+1 jump + * instructions + * cond = svar&mask + * At each of the following nmatch+1 branch command + * if (cond == match) + * take the branch + */ +struct aic_cmd_cond { + u8 svar; + u8 nmatch; + enum aic_cmd_id cid; + u8 mask; +}; +#define AIC_CMD_COND_LEN(nm) (1 + ((nm)+1)) + +/** + * AIC CMD Goto + * For branch, break, continue and stop + */ +struct aic_cmd_branch { + u16 address; + enum aic_cmd_id cid; + u8 match; +}; + +/** + * AIC Debug print + * For diagnostics + */ +struct aic_cmd_print { + u8 fmtlen; + u8 nargs; + enum aic_cmd_id cid; + char fmt[1]; +}; + +#define AIC_CMD_PRINT_LEN(p) (1 + ((p).fmtlen/4) + (((p).nargs + 3)/4)) +#define AIC_CMD_PRINT_ARG(p) (1 + ((p).fmtlen/4)) + +/** + * AIC Arithmetic and logical operations + * Bit 5 indicates if op1 is indirect + * Bit 6 indicates if op2 is indirect + */ +struct aic_cmd_op { + u8 op1; + u8 op2; + enum aic_cmd_id cid; + u8 dst; +}; +#define AIC_CMD_OP1_ID (1u<<5) +#define AIC_CMD_OP2_ID (1u<<4) + +#define AIC_CMD_OP_START AIC_CMD_OP_ADD +#define AIC_CMD_OP_END (AIC_CMD_OP_LNOT|AIC_CMD_OP1_ID|AIC_CMD_OP2_ID) +#define AIC_CMD_OP_IS_UNARY(x) \ + (((x) == AIC_CMD_OP_NOT) || ((x) == AIC_CMD_OP_LNOT)) + + +/** + * AIC Register + * + * A single reg write + * + */ +union aic_register { + struct { + u8 book; + u8 page; + u8 offset; + u8 data; + }; + u32 bpod; +}; + + + +/** + * AIC Command + * + * Can be a either a + * -# single register write, or + * -# command + * + */ +union aic_cmd { + struct { + u16 unused1; + enum aic_cmd_id cid; + u8 unused2; + }; + union aic_register reg; + struct aic_cmd_delay delay; + struct aic_cmd_lock lock; + struct aic_cmd_bitop bitop; + struct aic_cmd_bhdr bhdr; + struct aic_cmd_burst burst; + struct aic_cmd_ldst ldst; + struct aic_cmd_cond cond; + struct aic_cmd_branch branch; + struct aic_cmd_print print; + u8 print_arg[4]; + struct aic_cmd_op op; +}; + +#define AIC_REG_IS_CMD(x) ((x).cid >= AIC_CMD_DELAY) + +/** + * AIC Block Type + * + * Block identifier + * + */ +enum __attribute__ ((__packed__)) aic_block_t { + AIC_BLOCK_SYSTEM_PRE, + AIC_BLOCK_A_INST, + AIC_BLOCK_A_A_COEF, + AIC_BLOCK_A_B_COEF, + AIC_BLOCK_A_F_COEF, + AIC_BLOCK_D_INST, + AIC_BLOCK_D_A1_COEF, + AIC_BLOCK_D_B1_COEF, + AIC_BLOCK_D_A2_COEF, + AIC_BLOCK_D_B2_COEF, + AIC_BLOCK_D_F_COEF, + AIC_BLOCK_SYSTEM_POST, + AIC_BLOCK_N, + AIC_BLOCK_INVALID, +}; +#define AIC_BLOCK_D_A_COEF AIC_BLOCK_D_A1_COEF +#define AIC_BLOCK_D_B_COEF AIC_BLOCK_D_B1_COEF + +/** + * AIC Block + * + * A block of logically grouped sequences/commands/cmd-commands + * + */ +struct aic_block { + enum aic_block_t type; + int ncmds; + union aic_cmd cmd[]; +}; +#define AIC_BLOCK_SIZE(ncmds) (sizeof(struct aic_block) + \ + ((ncmds)*sizeof(union aic_cmd))) + +/** + * AIC Image + * + * A downloadable image + */ +struct aic_image { + char name[AIC_MAX_ID]; /**< Name of the pfw/overlay/configuration*/ + char *desc; /**< User string*/ + int mute_flags; + struct aic_block *block[AIC_BLOCK_N]; +}; + + + +/** + * AIC PLL + * + * PLL configuration sequence and match critirea + */ +struct aic_pll { + char name[AIC_MAX_ID]; /**< Name of the PLL sequence*/ + char *desc; /**< User string*/ + struct aic_block *seq; +}; + +/** + * AIC Control + * + * Run-time control for a process flow + */ +struct aic_control { + char name[AIC_MAX_ID]; /**< Control identifier*/ + char *desc; /**< User string*/ + int mute_flags; + + int min; /**< Min value of control (*100)*/ + int max; /**< Max value of control (*100)*/ + int step; /**< Control step size (*100)*/ + + int imax; /**< Max index into controls array*/ + int ireset; /**< Reset control to defaults*/ + int icur; /**< Last value set*/ + struct aic_block **output; /**< Array of sequences to send*/ +}; + +/** + * Process flow + * + * Complete description of a process flow + */ +struct aic_pfw { + char name[AIC_MAX_ID]; /**< Name of the process flow*/ + char *desc; /**< User string*/ + u32 version; + u8 prb_a; + u8 prb_d; + int novly; /**< Number of overlays (1 or more)*/ + int ncfg; /**< Number of configurations (0 or more)*/ + int nctrl; /**< Number of run-time controls*/ + struct aic_image *base; /**< Base sequence*/ + struct aic_image **ovly_cfg; /**< Overlay and cfg*/ + /**< patches (if any)*/ + struct aic_control **ctrl; /**< Array of run-time controls*/ +}; + +#define AIC_OCFG_NDX(p, o, c) (((o)*(p)->ncfg)+(c)) +/** + * Process transition + * + * Sequence for specific state transisitions within the driver + * + */ +struct aic_transition { + char name[AIC_MAX_ID]; /**< Name of the transition*/ + char *desc; /**< User string*/ + struct aic_block *block; +}; + +/** + * Device audio mode + * + * Link operating modes to process flows, + * configurations and sequences + * + */ +struct aic_mode { + char name[AIC_MAX_ID]; + char *desc; /**< User string*/ + u32 flags; + u8 pfw; + u8 ovly; + u8 cfg; + u8 pll; + struct aic_block *entry; + struct aic_block *exit; +}; + +struct aic_asoc_toc_entry { + char etext[AIC_MAX_ID]; + int mode; + int cfg; +}; + +struct aic_asoc_toc { + int nentries; + struct aic_asoc_toc_entry entry[]; +}; + +/** + * AIC Project + * + * Top level structure describing the AIC project + */ +struct aic_project { + u32 magic; /**< magic number for identifying F/W file*/ + u32 if_id; /**< Interface match code */ + u32 size; /**< Total size of the firmware (including this header)*/ + u32 cksum; /**< CRC32 of the pickled firmware */ + u32 version; /**< Firmware version (from CFD file)*/ + u32 tstamp; /**< Time stamp of firmware build (epoch seconds)*/ + char name[AIC_MAX_ID]; /**< Project name*/ + char *desc; /**< User string*/ + enum aic_dfamily dfamily; /**< Device family*/ + enum aic_device device; /**< Device identifier*/ + u32 flags; /**< AIC flags*/ + + struct aic_transition **transition; /**< Transition sequences*/ + + u16 npll; /**< Number of PLL settings*/ + struct aic_pll **pll; /**< PLL settings*/ + + u16 npfw; /**< Number of process flows*/ + struct aic_pfw **pfw; /**< Process flows*/ + + u16 nmode; /**< Number of operating modes*/ + struct aic_mode **mode; /**< Modes*/ + + struct aic_asoc_toc *asoc_toc; /**< list of amixer controls*/ +}; + + +/* @} */ + +/* **AIC_INTERFACE_ID=0x3FA6D547** */ + +#endif /* AIC_FIRMWARE_H_ */ diff --git a/sound/soc/codecs/aic3xxx_cfw_ops.c b/sound/soc/codecs/aic3xxx_cfw_ops.c new file mode 100644 index 0000000..79b975b --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw_ops.c @@ -0,0 +1,989 @@ +/* + * linux/sound/soc/codecs/aic3xxx/aic3xxx_cfw_ops.c + * + * Copyright (C) 2011 Texas Instruments Inc., + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <linux/slab.h> +#include <sound/tlv.h> + +#include "aic3xxx_cfw.h" +#include "aic3xxx_cfw_ops.h" + +#define aic_warn(_cfw, fmt, ...)\ + dev_warn(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) +#define aic_error(_cfw, fmt, ...)\ + dev_err(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) +#define aic_dbg(_cfw, fmt, ...)\ + dev_dbg(_cfw->dev, "AIC: " fmt, ##__VA_ARGS__) + +#if defined(CONFIG_MFD_AIC3111) || defined(CONFIG_AIC3111_CORE) + +# define AIC3XXX_AIC_DEVICE "aic3111_cfw" +#elif defined(CONFIG_MFD_AIC3256) || defined(CONFIG_AIC3256_CORE) + +# define AIC3XXX_AIC_DEVICE "aic3256_cfw" +#elif defined(CONFIG_MFD_AIC262) || defined(CONFIG_AIC3262_CORE) +# define AIC3XXX_AIC_DEVICE "aic3262_cfw" +#else +# define AIC3XXX_AIC_DEVICE "aic3xxx_cfw" +#endif + +/* **Code beyond this point is compilable on host** */ + +/* + * Firmware version numbers are used to make sure that the + * host and target code stay in sync. It is _not_ recommended + * to provide this number from the outside (E.g., from a makefile) + * Instead, a set of automated tools are relied upon to keep the numbers + * in sync at the time of host testing. + */ +#undef AIC_FW_IF_ID +#define AIC_FW_IF_ID 0x3FA6D547 +static int aic_dlimage(struct aic_state *ps, struct aic_image *pim); +static int aic_dlcfg(struct aic_state *ps, struct aic_image *pim); +static int aic_dlctl(struct aic_state *ps, struct aic_block *pb, + u32 mute_flags); + +static void aic_dlcmds(struct aic_state *ps, struct aic_block *pb); +static int aic_set_mode_id(struct aic_state *ps); +static int aic_mute(struct aic_state *ps, int mute, u32 flags); +static int aic_setmode_cfg_u(struct aic_state *ps, int mode, int cfg); +static int aic_setcfg_u(struct aic_state *ps, int cfg); +static int aic_transition_u(struct aic_state *ps, char *ttype); +static int aic_set_pll_u(struct aic_state *ps, int asi); +static int aic_control_u(struct aic_state *ps, char *cname, int param); +static struct aic_project *aic_unpickle(void *pcfw, int n); + +static void aic_wait(struct aic_state *ps, unsigned int reg, u8 mask, + u8 data); +static void aic_set_bits(u8 *data, u8 mask, u8 val); + +int aic_init(struct aic_state *ps, const struct aic3xxx_codec_ops *ops, + struct snd_soc_codec *codec) +{ + ps->ops = ops; + ps->codec = codec; + ps->pjt = NULL; + mutex_init(&ps->mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(aic_init); + +int aic_lock(struct aic_state *ps, int lock) +{ + if (lock) + mutex_lock(&ps->mutex); + else + mutex_unlock(&ps->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(aic_lock); + +int aic_reload(struct aic_state *ps, void *pcfw, int n) +{ + ps->pjt = aic_unpickle(pcfw, n); + ps->cur_mode_id = + ps->cur_mode = ps->cur_pll = ps->cur_pfw = + ps->cur_ovly = ps->cur_cfg = -1; + if (ps->pjt == NULL) { + aic_error(ps, "version mismatch\n"); + return -EINVAL; + } + return 0; +} +EXPORT_SYMBOL_GPL(aic_reload); + +int aic_setmode(struct aic_state *ps, int mode) +{ + struct aic_project *pjt; + int ret; + + aic_lock(ps, 1); + pjt = ps->pjt; + if (pjt == NULL) { + aic_lock(ps, 0); + return -EINVAL; + } + ret = aic_setmode_cfg_u(ps, mode, pjt->mode[mode]->cfg); + aic_lock(ps, 0); + return ret; +} + +int aic_setcfg(struct aic_state *ps, int cfg) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_setcfg_u(ps, cfg); + aic_lock(ps, 0); + return ret; +} + +static int aic_setcfg_u(struct aic_state *ps, int cfg) +{ + struct aic_project *pjt = ps->pjt; + struct aic_pfw *pfw; + struct aic_image *patch; + + if (pjt == NULL) + return -EINVAL; + if (ps->cur_pfw < 0 || ps->cur_pfw >= pjt->npfw) + return -EINVAL; /* Non miniDSP */ + if (ps->cur_cfg == cfg) + return 0; + pfw = pjt->pfw[ps->cur_pfw]; + if (pfw->ncfg == 0 && cfg != 0) + return -EINVAL; + if (cfg > 0 && cfg >= pfw->ncfg) + return -EINVAL; + ps->cur_cfg = cfg; + aic_set_mode_id(ps); + patch = + pfw->ovly_cfg[AIC_OCFG_NDX(pfw, ps->cur_ovly, ps->cur_cfg)]; + if (pfw->ncfg != 0) + return aic_dlcfg(ps, patch); + return 0; +} + +int aic_setmode_cfg(struct aic_state *ps, int mode, int cfg) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_setmode_cfg_u(ps, mode, cfg); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_setmode_cfg); + +static int aic_setmode_cfg_u(struct aic_state *ps, int mode, int cfg) +{ + struct aic_project *pjt = ps->pjt; + struct aic_mode *pmode; + int which = 0, ocndx; + + if (pjt == NULL) + goto err; + if ((mode < 0) || (mode >= pjt->nmode)) + goto err; + if (cfg < 0) + goto err; + if (mode == ps->cur_mode) + return aic_setcfg_u(ps, cfg); + + /* Apply exit sequence for previous mode if present */ + if (ps->cur_mode >= 0) + aic_dlcmds(ps, pjt->mode[ps->cur_mode]->exit); + pmode = pjt->mode[mode]; + if (pjt->mode[mode]->pfw < pjt->npfw) { /* New mode uses miniDSP */ + struct aic_image *im; + struct aic_pfw *pfw = pjt->pfw[pmode->pfw]; + + /* Make sure cfg is valid and supported in this mode */ + if (pfw->ncfg == 0 && cfg != 0) + goto err; + if (cfg > 0 && cfg >= pfw->ncfg) + goto err; + + /* + * Decisions about which miniDSP to stop/restart are taken + * on the basis of sections present in the _base_ image + * This allows for correct sync mode operation even in cases + * where the base PFW uses both miniDSPs where a particular + * overlay applies only to one + */ + im = pfw->base; + if (im->block[AIC_BLOCK_A_INST]) + which |= AIC3XXX_COPS_MDSP_A; + if (im->block[AIC_BLOCK_D_INST]) + which |= AIC3XXX_COPS_MDSP_D; + + if (pmode->pfw != ps->cur_pfw) { + + /* New mode requires different PFW */ + ps->cur_pfw = pmode->pfw; + ps->cur_ovly = 0; + ps->cur_cfg = 0; + + which = ps->ops->stop(ps->codec, which); + aic_dlimage(ps, im); + if (pmode->ovly && pmode->ovly < pfw->novly) { + + /* New mode uses ovly */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + aic_dlimage(ps, + pfw->ovly_cfg[ocndx]); + } else if (pfw->ncfg > 0) { + + /* new mode needs only a cfg change */ + ocndx = AIC_OCFG_NDX(pfw, 0, cfg); + aic_dlimage(ps, + pfw->ovly_cfg[ocndx]); + } + ps->ops->restore(ps->codec, which); + + } else if (pmode->ovly != ps->cur_ovly) { + + /* New mode requires only an ovly change */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + which = ps->ops->stop(ps->codec, which); + aic_dlimage(ps, pfw->ovly_cfg[ocndx]); + ps->ops->restore(ps->codec, which); + } else if (pfw->ncfg > 0 && cfg != ps->cur_cfg) { + + /* New mode requires only a cfg change */ + ocndx = AIC_OCFG_NDX(pfw, pmode->ovly, cfg); + aic_dlcfg(ps, pfw->ovly_cfg[ocndx]); + } + ps->cur_ovly = pmode->ovly; + ps->cur_cfg = cfg; + + ps->cur_mode = mode; + aic_set_pll_u(ps, 0); + + } else if (pjt->mode[mode]->pfw != 0xFF) { + + /* Not bypass mode */ + aic_warn(ps, "Bad pfw setting detected (%d). Max pfw=%d", + pmode->pfw, pjt->npfw); + } + ps->cur_mode = mode; + aic_set_mode_id(ps); + + /* Transition to netural mode */ + aic_transition_u(ps, "NEUTRAL"); + + /* Apply entry sequence if present */ + aic_dlcmds(ps, pmode->entry); + + aic_dbg(ps, "setmode_cfg: DONE (mode=%d pfw=%d ovly=%d cfg=%d)", + ps->cur_mode, ps->cur_pfw, ps->cur_ovly, ps->cur_cfg); + return 0; + +err: + aic_dbg(ps, "Failed to set firmware mode"); + return -EINVAL; +} + +int aic_transition(struct aic_state *ps, char *ttype) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_transition_u(ps, ttype); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_transition); + +static int aic_transition_u(struct aic_state *ps, char *ttype) +{ + int i; + + if (ps->pjt == NULL) + return -EINVAL; + for (i = 0; i < AIC_TRN_N; ++i) { + if (!strcasecmp(ttype, aic_transition_id[i])) { + struct aic_transition *pt = ps->pjt->transition[i]; + aic_dbg(ps, "Sending transition %s[%d]", ttype, i); + if (pt) + aic_dlcmds(ps, pt->block); + return 0; + } + } + aic_warn(ps, "Transition %s not present or invalid", ttype); + return 0; +} + +int aic_set_pll(struct aic_state *ps, int asi) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_set_pll_u(ps, asi); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic_set_pll); + +static int aic_set_pll_u(struct aic_state *ps, int asi) +{ + struct aic_project *pjt = ps->pjt; + int pll_id; + + if (pjt == NULL) + return -EINVAL; + if (ps->cur_mode < 0) + return -EINVAL; + pll_id = pjt->mode[ps->cur_mode]->pll; + if (ps->cur_pll != pll_id) { + aic_dbg(ps, "Re-configuring PLL: %s==>%d", + pjt->pll[pll_id]->name, + pll_id); + aic_dlcmds(ps, pjt->pll[pll_id]->seq); + ps->cur_pll = pll_id; + } + return 0; +} + +int aic_control(struct aic_state *ps, char *cname, int param) +{ + int ret; + + aic_lock(ps, 1); + ret = aic_control_u(ps, cname, param); + aic_lock(ps, 0); + return ret; +} + +static int aic_control_u(struct aic_state *ps, char *cname, int param) +{ + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw < 0 || ps->cur_pfw >= ps->pjt->npfw) { + aic_warn(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + struct aic_control *pc = pfw->ctrl[i]; + if (strcasecmp(cname, pfw->ctrl[i]->name)) + continue; + if (param < 0 || param > pc->imax) { + aic_warn(ps, "Parameter out of range\n"); + return -EINVAL; + } + aic_dbg(ps, "Sending control %s[%d]", cname, param); + pc->icur = param; + aic_dlctl(ps, pc->output[param], pc->mute_flags); + return 0; + } + aic_warn(ps, "Control named %s not found in pfw %s", cname, pfw->name); + + return -EINVAL; +} + +static void aic_op(struct aic_state *ps, unsigned char *var, + struct aic_cmd_op cmd) +{ + u32 op1, op2; + u32 cid = cmd.cid; + + op1 = cmd.op1; + op2 = cmd.op2; + if (cid & AIC_CMD_OP1_ID) + op1 = var[op1]; + if (cid & AIC_CMD_OP2_ID) + op2 = var[op2]; + cid &= ~(AIC_CMD_OP1_ID | AIC_CMD_OP2_ID); + + switch (cid) { + case AIC_CMD_OP_ADD: + var[cmd.dst] = op1 + op2; + break; + case AIC_CMD_OP_SUB: + var[cmd.dst] = op1 - op2; + break; + case AIC_CMD_OP_MUL: + var[cmd.dst] = op1 * op2; + break; + case AIC_CMD_OP_DIV: + var[cmd.dst] = op1 / op2; + break; + case AIC_CMD_OP_AND: + var[cmd.dst] = op1 & op2; + break; + case AIC_CMD_OP_OR: + var[cmd.dst] = op1 | op2; + break; + case AIC_CMD_OP_SHL: + var[cmd.dst] = (op1 << op2); + break; + case AIC_CMD_OP_SHR: + var[cmd.dst] = (op1 >> op2); + break; + case AIC_CMD_OP_RR: + while (op2--) + var[cmd.dst] = (op1 >> 1) | ((op1 & 1) << 7); + break; + case AIC_CMD_OP_XOR: + var[cmd.dst] = op1 ^ op2; + break; + case AIC_CMD_OP_NOT: + var[cmd.dst] = ~op1; + break; + case AIC_CMD_OP_LNOT: + var[cmd.dst] = !op1; + break; + default: + break; + } +} + +static void aic_dlcmds(struct aic_state *ps, struct aic_block *pb) +{ + int pc = 0, cond = 0; + unsigned char var[256]; + + if (!pb) + return; + while (pc < pb->ncmds) { + union aic_cmd *c = &(pb->cmd[pc]); + if (c->cid != AIC_CMD_BRANCH_IM && + c->cid != AIC_CMD_BRANCH_ID && c->cid != AIC_CMD_NOP) + cond = 0; + switch (c->cid) { + case 0 ... (AIC_CMD_NOP - 1): + ps->ops->reg_write(ps->codec, c->reg.bpod, + c->reg.data); + pc += 1; + break; + case AIC_CMD_NOP: + pc += 1; + break; + case AIC_CMD_DELAY: + mdelay(c->delay.delay); + pc += 1; + break; + case AIC_CMD_UPDTBITS: + ps->ops->set_bits(ps->codec, c[1].reg.bpod, + c->bitop.mask, c[1].reg.data); + pc += 2; + break; + case AIC_CMD_WAITBITS: + aic_wait(ps, c[1].reg.bpod, c->bitop.mask, + c[1].reg.data); + pc += 2; + break; + case AIC_CMD_LOCK: + if (c->delay.delay) + ps->ops->lock(ps->codec); + else + ps->ops->unlock(ps->codec); + pc += 1; + break; + case AIC_CMD_BURST: + ps->ops->bulk_write(ps->codec, c[1].reg.bpod, + c->bhdr.len, c[1].burst.data); + pc += AIC_CMD_BURST_LEN(c->bhdr.len); + break; + case AIC_CMD_RBURST: + ps->ops->bulk_read(ps->codec, c[1].reg.bpod, + c->bhdr.len, c[1].burst.data); + pc += AIC_CMD_BURST_LEN(c->bhdr.len); + break; + case AIC_CMD_LOAD_VAR_IM: + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, c->ldst.svar); + pc += 1; + break; + case AIC_CMD_LOAD_VAR_ID: + if (c->ldst.svar != c->ldst.dvar) { + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, + var[c->ldst.svar]); + pc += 1; + } else { + u8 data; + data = ps->ops->reg_read(ps->codec, + c[1].reg.bpod); + aic_set_bits(&var[c->ldst.dvar], + c->ldst.mask, data); + pc += 2; + } + break; + case AIC_CMD_STORE_VAR: + if (c->ldst.svar != c->ldst.dvar) + ps->ops->set_bits(ps->codec, + c[1].reg.bpod, + var[c->ldst.dvar], + var[c->ldst.svar]); + else + ps->ops->set_bits(ps->codec, + c[1].reg.bpod, + c->ldst.mask, + var[c->ldst.svar]); + pc += 2; + break; + case AIC_CMD_COND: + cond = var[c->cond.svar] & c->cond.mask; + pc += 1; + break; + case AIC_CMD_BRANCH: + pc = c->branch.address; + break; + case AIC_CMD_BRANCH_IM: + if (c->branch.match == cond) + pc = c->branch.address; + else + pc += 1; + break; + case AIC_CMD_BRANCH_ID: + if (var[c->branch.match] == cond) + pc = c->branch.address; + else + pc += 1; + break; + case AIC_CMD_PRINT: + { + union aic_cmd *parglist = + c + AIC_CMD_PRINT_ARG(c->print); + printk(c->print.fmt, + var[parglist->print_arg[0]], + var[parglist->print_arg[1]], + var[parglist->print_arg[2]], + var[parglist->print_arg[3]]); + pc += AIC_CMD_PRINT_LEN(c->print); + } + break; + case AIC_CMD_OP_START ... AIC_CMD_OP_END: + aic_op(ps, var, c->op); + pc += 1; + break; + default: + aic_warn(ps, "Unknown cmd command %x. Skipped", c->cid); + pc += 1; + break; + } + } +} + +static void aic_wait(struct aic_state *ps, unsigned int reg, u8 mask, + u8 data) +{ + while ((ps->ops->reg_read(ps->codec, reg) & mask) != data) + usleep_range(500, 1000); +} + +static inline void aic_set_bits(u8 *data, u8 mask, u8 val) +{ + *data = (*data & (~mask)) | (val & mask); +} + +static const struct { + u32 mdsp; + int buf_a, buf_b; + u32 swap; +} csecs[] = { + { + .mdsp = AIC3XXX_COPS_MDSP_A, + .swap = AIC3XXX_ABUF_MDSP_A, + .buf_a = AIC_BLOCK_A_A_COEF, + .buf_b = AIC_BLOCK_A_B_COEF + }, + { + .mdsp = AIC3XXX_COPS_MDSP_D, + .swap = AIC3XXX_ABUF_MDSP_D1, + .buf_a = AIC_BLOCK_D_A1_COEF, + .buf_b = AIC_BLOCK_D_B1_COEF + }, + { + .mdsp = AIC3XXX_COPS_MDSP_D, + .swap = AIC3XXX_ABUF_MDSP_D2, + .buf_a = AIC_BLOCK_D_A2_COEF, + .buf_b = AIC_BLOCK_D_B2_COEF + }, +}; +static int aic_dlctl(struct aic_state *ps, struct aic_block *pb, + u32 mute_flags) +{ + int i, btype = pb->type; + int run_state = ps->ops->lock(ps->codec); + + aic_dbg(ps, "Download CTL"); + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (csecs[i].buf_a != btype && csecs[i].buf_b != btype) + continue; + aic_dbg(ps, "\tDownload once to %d\n", btype); + aic_dlcmds(ps, pb); + if (run_state & csecs[i].mdsp) { + aic_dbg(ps, "Download again to make sure it reaches B"); + aic_mute(ps, 1, run_state & mute_flags); + ps->ops->bswap(ps->codec, csecs[i].swap); + aic_mute(ps, 0, run_state & mute_flags); + aic_dlcmds(ps, pb); + } + break; + } + ps->ops->unlock(ps->codec); + return 0; +} + +static int aic_dlcfg(struct aic_state *ps, struct aic_image *pim) +{ + int i, run_state, swap; + + aic_dbg(ps, "Download CFG %s", pim->name); + run_state = ps->ops->lock(ps->codec); + swap = 0; + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (!pim->block[csecs[i].buf_a]) + continue; + aic_dlcmds(ps, pim->block[csecs[i].buf_a]); + aic_dlcmds(ps, pim->block[csecs[i].buf_b]); + if (run_state & csecs[i].mdsp) + swap |= csecs[i].swap; + } + if (swap) { + aic_mute(ps, 1, run_state & pim->mute_flags); + ps->ops->bswap(ps->codec, swap); + aic_mute(ps, 0, run_state & pim->mute_flags); + for (i = 0; i < sizeof(csecs) / sizeof(csecs[0]); ++i) { + if (!pim->block[csecs[i].buf_a]) + continue; + if (!(run_state & csecs[i].mdsp)) + continue; + aic_dlcmds(ps, pim->block[csecs[i].buf_a]); + aic_dlcmds(ps, pim->block[csecs[i].buf_b]); + } + } + ps->ops->unlock(ps->codec); + return 0; +} + +static int aic_dlimage(struct aic_state *ps, struct aic_image *pim) +{ + int i; + + if (!pim) + return 0; + aic_dbg(ps, "Download IMAGE %s", pim->name); + for (i = 0; i < AIC_BLOCK_N; ++i) + aic_dlcmds(ps, pim->block[i]); + return 0; +} + +static int aic_mute(struct aic_state *ps, int mute, u32 flags) +{ + if ((flags & AIC3XXX_COPS_MDSP_D) && (flags & AIC3XXX_COPS_MDSP_A)) + aic_transition_u(ps, + mute ? "AD_MUTE" : "AD_UNMUTE"); + else if (flags & AIC3XXX_COPS_MDSP_D) + aic_transition_u(ps, mute ? "D_MUTE" : "D_UNMUTE"); + else if (flags & AIC3XXX_COPS_MDSP_A) + aic_transition_u(ps, mute ? "A_MUTE" : "A_UNMUTE"); + return 0; +} + +static inline void *aic_ndx2ptr(void *p, u8 *base) +{ + return &base[(int)p]; +} +static inline char *aic_desc(void *p, u8 *base) +{ + if (p) + return aic_ndx2ptr(p, base); + return NULL; +} + +static void aic_unpickle_image(struct aic_image *im, void *p) +{ + int i; + + im->desc = aic_desc(im->desc, p); + for (i = 0; i < AIC_BLOCK_N; ++i) + if (im->block[i]) + im->block[i] = aic_ndx2ptr(im->block[i], p); +} + +static void aic_unpickle_control(struct aic_control *ct, void *p) +{ + int i; + + ct->output = aic_ndx2ptr(ct->output, p); + ct->desc = aic_desc(ct->desc, p); + for (i = 0; i <= ct->imax; ++i) + ct->output[i] = aic_ndx2ptr(ct->output[i], p); +} + +static struct aic_project *aic_unpickle(void *p, int n) +{ + struct aic_project *pjt = p; + int i, j; + + if (pjt->magic != AIC_FW_MAGIC || pjt->size != n || + pjt->if_id != AIC_FW_IF_ID) { + return NULL; + } + + pjt->desc = aic_desc(pjt->desc, p); + pjt->transition = aic_ndx2ptr(pjt->transition, p); + for (i = 0; i < AIC_TRN_N; i++) { + if (!pjt->transition[i]) + continue; + pjt->transition[i] = aic_ndx2ptr(pjt->transition[i], p); + pjt->transition[i]->desc = aic_desc( + pjt->transition[i]->desc, p); + pjt->transition[i]->block = aic_ndx2ptr( + pjt->transition[i]->block, p); + } + pjt->pll = aic_ndx2ptr(pjt->pll, p); + for (i = 0; i < pjt->npll; i++) { + pjt->pll[i] = aic_ndx2ptr(pjt->pll[i], p); + pjt->pll[i]->desc = aic_desc(pjt->pll[i]->desc, p); + pjt->pll[i]->seq = aic_ndx2ptr(pjt->pll[i]->seq, p); + } + + pjt->pfw = aic_ndx2ptr(pjt->pfw, p); + for (i = 0; i < pjt->npfw; i++) { + pjt->pfw[i] = aic_ndx2ptr(pjt->pfw[i], p); + pjt->pfw[i]->desc = aic_desc(pjt->pfw[i]->desc, p); + if (pjt->pfw[i]->base) { + pjt->pfw[i]->base = aic_ndx2ptr( + pjt->pfw[i]->base, p); + aic_unpickle_image(pjt->pfw[i]->base, p); + } + pjt->pfw[i]->ovly_cfg = aic_ndx2ptr( + pjt->pfw[i]->ovly_cfg, p); + for (j = 0; j < pjt->pfw[i]->novly * pjt->pfw[i]->ncfg; ++j) { + pjt->pfw[i]->ovly_cfg[j] = aic_ndx2ptr( + pjt->pfw[i]->ovly_cfg[j], p); + aic_unpickle_image(pjt->pfw[i]->ovly_cfg[j], p); + } + if (pjt->pfw[i]->nctrl) + pjt->pfw[i]->ctrl = aic_ndx2ptr( + pjt->pfw[i]->ctrl, p); + for (j = 0; j < pjt->pfw[i]->nctrl; ++j) { + pjt->pfw[i]->ctrl[j] = aic_ndx2ptr( + pjt->pfw[i]->ctrl[j], p); + aic_unpickle_control(pjt->pfw[i]->ctrl[j], p); + } + } + + pjt->mode = aic_ndx2ptr(pjt->mode, p); + for (i = 0; i < pjt->nmode; i++) { + pjt->mode[i] = aic_ndx2ptr(pjt->mode[i], p); + pjt->mode[i]->desc = aic_desc(pjt->mode[i]->desc, p); + if (pjt->mode[i]->entry) + pjt->mode[i]->entry = aic_ndx2ptr( + pjt->mode[i]->entry, p); + if (pjt->mode[i]->exit) + pjt->mode[i]->exit = aic_ndx2ptr( + pjt->mode[i]->exit, p); + } + if (pjt->asoc_toc) + pjt->asoc_toc = aic_ndx2ptr(pjt->asoc_toc, p); + else + return NULL; + + return pjt; +} +static int aic_set_mode_id(struct aic_state *ps) +{ + struct aic_asoc_toc *toc = ps->pjt->asoc_toc; + int i; + + for (i = 0; i < toc->nentries; ++i) { + if (toc->entry[i].cfg == ps->cur_cfg && + toc->entry[i].mode == ps->cur_mode) { + ps->cur_mode_id = i; + return 0; + } + } + aic_dbg(ps, "Unknown mode,cfg combination [%d,%d]", ps->cur_mode, + ps->cur_cfg); + return -EINVAL; +} + +/* **Code beyond this point is not compilable on host** */ + +static int aic3xxx_get_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw >= ps->pjt->npfw) { + aic_dbg(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + if (!strcasecmp(kcontrol->id.name, pfw->ctrl[i]->name)) { + struct aic_control *pc = pfw->ctrl[i]; + ucontrol->value.integer.value[0] = pc->icur; + return 0; + } + } + return 0; +} + +static int aic3xxx_put_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + + aic_control(ps, kcontrol->id.name, + ucontrol->value.integer.value[0]); + return 0; +} + +static int aic3xxx_info_control(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol) +{ + struct aic_state *ps = (struct aic_state *)kcontrol->private_value; + struct aic_pfw *pfw; + int i; + + if (ps->cur_pfw >= ps->pjt->npfw) { + aic_dbg(ps, "Not in MiniDSP mode"); + return 0; + } + pfw = ps->pjt->pfw[ps->cur_pfw]; + for (i = 0; i < pfw->nctrl; ++i) { + if (!strcasecmp(kcontrol->id.name, pfw->ctrl[i]->name)) { + struct aic_control *pc = pfw->ctrl[i]; + ucontrol->value.integer.min = 0; + ucontrol->value.integer.max = pc->imax; + if (pc->imax == 1) + ucontrol->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + ucontrol->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + } + } + + ucontrol->count = 1; + return 0; +} +int aic_add_controls(struct snd_soc_codec *codec, struct aic_state *ps) +{ + int i, j; + struct aic_pfw *pfw; + + for (j = 0; j < ps->pjt->npfw; ++j) { + pfw = ps->pjt->pfw[j]; + + for (i = 0; i < pfw->nctrl; ++i) { + struct aic_control *pc = pfw->ctrl[i]; + struct snd_kcontrol_new *generic_control = + devm_kzalloc(codec->dev, + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + unsigned int *tlv_array = + devm_kzalloc(codec->dev, + 4 * sizeof(unsigned int), GFP_KERNEL); + + if (generic_control == NULL) + return -ENOMEM; + generic_control->access = + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE; + tlv_array[0] = SNDRV_CTL_TLVT_DB_SCALE; + tlv_array[1] = 2 * sizeof(unsigned int); + tlv_array[2] = pc->min; + tlv_array[3] = ((pc->step) & TLV_DB_SCALE_MASK); + if (pc->step > 0) + generic_control->tlv.p = tlv_array; + generic_control->name = pc->name; + generic_control->private_value = (unsigned long) ps; + generic_control->get = aic3xxx_get_control; + generic_control->put = aic3xxx_put_control; + generic_control->info = aic3xxx_info_control; + generic_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + snd_soc_add_codec_controls(codec, generic_control, 1); + aic_dbg(ps, "Added control %s", pc->name); + } + } + return 0; + +} +EXPORT_SYMBOL_GPL(aic_add_controls); + +static int aic3xxx_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *) kcontrol->private_value; + struct aic_state *ps = (struct aic_state *) e->mask; + + ucontrol->value.enumerated.item[0] = ps->cur_mode_id; + + return 0; +} + +static int aic3xxx_put_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_enum *e = (struct soc_enum *) kcontrol->private_value; + struct aic_state *ps = (struct aic_state *) e->mask; + struct aic_asoc_toc *toc; + int index, ret; + + aic_lock(ps, 1); + toc = ps->pjt->asoc_toc; + + index = ucontrol->value.enumerated.item[0]; + if (index < 0 || index >= toc->nentries) { + aic_lock(ps, 0); + return -EINVAL; + } + ret = aic_setmode_cfg_u(ps, toc->entry[index].mode, + toc->entry[index].cfg); + aic_lock(ps, 0); + return ret; +} +EXPORT_SYMBOL_GPL(aic3xxx_put_mode); + +int aic_add_modes(struct snd_soc_codec *codec, struct aic_state *ps) +{ + int j; + struct aic_asoc_toc *toc = ps->pjt->asoc_toc; + struct soc_enum *mode_cfg_enum = + devm_kzalloc(codec->dev, sizeof(struct soc_enum), GFP_KERNEL); + struct snd_kcontrol_new *mode_cfg_control = + devm_kzalloc(codec->dev, sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + + if (mode_cfg_enum == NULL) + goto mem_err; + if (mode_cfg_control == NULL) + goto mem_err; + + mode_cfg_enum->dtexts = devm_kzalloc(codec->dev, + toc->nentries * sizeof(char *), + GFP_KERNEL); + if (mode_cfg_enum->dtexts == NULL) + goto mem_err; + + for (j = 0; j < toc->nentries; j++) + mode_cfg_enum->dtexts[j] = toc->entry[j].etext; + + mode_cfg_enum->reg = j; + mode_cfg_enum->max = toc->nentries; + mode_cfg_enum->mask = (unsigned int) ps; + mode_cfg_control->name = "Codec Firmware Setmode"; + mode_cfg_control->get = aic3xxx_get_mode; + mode_cfg_control->put = aic3xxx_put_mode; + mode_cfg_control->info = snd_soc_info_enum_ext; + mode_cfg_control->private_value = (unsigned long) mode_cfg_enum; + mode_cfg_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + snd_soc_add_codec_controls(codec, mode_cfg_control, 1); + return 0; +mem_err: + kfree(mode_cfg_control); + kfree(mode_cfg_enum); + kfree(mode_cfg_enum->texts); + return -ENOMEM; + +} +EXPORT_SYMBOL_GPL(aic_add_modes); + +MODULE_AUTHOR("Hari Rajagopala <harik(a)ti.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/aic3xxx_cfw_ops.h b/sound/soc/codecs/aic3xxx_cfw_ops.h new file mode 100644 index 0000000..6a600fb --- /dev/null +++ b/sound/soc/codecs/aic3xxx_cfw_ops.h @@ -0,0 +1,98 @@ +/* + * aic3xxx_cfw_ops.h -- SoC audio for TI OMAP44XX SDP + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef AIC3XXX_CFW_OPS_H_ +#define AIC3XXX_CFW_OPS_H_ + +#include <linux/mutex.h> +#include <sound/soc.h> +#include <linux/cdev.h> + +struct aic_project; +struct aic3xxx_codec_ops; + +struct aic_state { + const char *dev_name; + int version; + struct aic_project *pjt; + const struct aic3xxx_codec_ops *ops; + struct snd_soc_codec *codec; + struct device *dev; + struct mutex mutex; + int cur_mode_id; + int cur_pll; + int cur_mode; + int cur_pfw; + int cur_ovly; + int cur_cfg; + struct cdev cdev; + int is_open; +}; + +int aic_init(struct aic_state *ps, const struct aic3xxx_codec_ops *ops, + struct snd_soc_codec *codec); +int aic_lock(struct aic_state *ps, int lock); +int aic_reload(struct aic_state *ps, void *pcfw, int n); +int aic_setmode(struct aic_state *ps, int mode); +int aic_setmode_cfg(struct aic_state *ps, int mode, int cfg); +int aic_setcfg(struct aic_state *ps, int cfg); +int aic_transition(struct aic_state *ps, char *ttype); +int aic_set_pll(struct aic_state *ps, int asi); +int aic_control(struct aic_state *ps, char *cname, int param); +int aic_add_controls(struct snd_soc_codec *codec, struct aic_state *ps); +int aic_add_modes(struct snd_soc_codec *codec, struct aic_state *ps); + + +#define AIC3XXX_COPS_MDSP_D_L (0x00000002u) +#define AIC3XXX_COPS_MDSP_D_R (0x00000001u) +#define AIC3XXX_COPS_MDSP_D (AIC3XXX_COPS_MDSP_D_L|AIC3XXX_COPS_MDSP_D_R) + +#define AIC3XXX_COPS_MDSP_A_L (0x00000020u) +#define AIC3XXX_COPS_MDSP_A_R (0x00000010u) +#define AIC3XXX_COPS_MDSP_A (AIC3XXX_COPS_MDSP_A_L|AIC3XXX_COPS_MDSP_A_R) + +#define AIC3XXX_COPS_MDSP_ALL (AIC3XXX_COPS_MDSP_D|AIC3XXX_COPS_MDSP_A) + +#define AIC3XXX_ABUF_MDSP_D1 (0x00000001u) +#define AIC3XXX_ABUF_MDSP_D2 (0x00000002u) +#define AIC3XXX_ABUF_MDSP_A (0x00000010u) +#define AIC3XXX_ABUF_MDSP_ALL \ + (AIC3XXX_ABUF_MDSP_D1|AIC3XXX_ABUF_MDSP_D2|AIC3XXX_ABUF_MDSP_A) + + +struct aic3xxx_codec_ops { + int (*reg_read) (struct snd_soc_codec *codec, unsigned int reg); + int (*reg_write) (struct snd_soc_codec *codec, unsigned int reg, + unsigned char val); + int (*set_bits) (struct snd_soc_codec *codec, unsigned int reg, + unsigned char mask, unsigned char val); + int (*bulk_read) (struct snd_soc_codec *codec, unsigned int reg, + int count, u8 *buf); + int (*bulk_write) (struct snd_soc_codec *codec, unsigned int reg, + int count, const u8 *buf); + int (*lock) (struct snd_soc_codec *codec); + int (*unlock) (struct snd_soc_codec *codec); + int (*stop) (struct snd_soc_codec *codec, int mask); + int (*restore) (struct snd_soc_codec *codec, int runstate); + int (*bswap) (struct snd_soc_codec *codec, int mask); +}; + +MODULE_LICENSE("GPL"); + +#endif -- 1.7.0.4

1 0

[alsa-devel] how to update the hw_ptr in the interrupt?
by Zhang wei 12 Dec '12

12 Dec '12

Hello every one.i am in a trouble and could anyone help me figure out ? I have spent couple days trying to understand ,but I do not know the pricinple clearly which decides how to adjust the hw_ptr in the interrupt routine. Below is the reference sourcecode in the snd_pcm_update_hw_ptr0(). When the DMA interrupt ocurrs the process will enter the snd_pcm_update_hw_ptr0() with the in_interrupt=1. ========================================================== if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; goto __delta; } } } ========================================================== The question is: what can we get from the result of the compare between delta and new_hw_ptr.? Why we can conclue it? The delta represent the expected location in the range of *boundary* , and it producted based the hw_ptr_interrupt(*not fixed*) and perod_size(fixed)? The new_hw_ptr is producted by runtime->hw_ptr_base plus pos. What Is the relationship between delta and new_hw_ptr? Why we can compare them with each other ? Any tips will be ok. Thanks a lot. Best Regards zw

1 0

[alsa-devel] [PATCH] ASoC: arizona: Implement tristate support
by Mark Brown 12 Dec '12

12 Dec '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 231fae0..a85a68f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -859,11 +859,27 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return snd_soc_dapm_sync(&codec->dapm); } +static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + unsigned int reg; + + if (tristate) + reg = ARIZONA_AIF1_TRI; + else + reg = 0; + + return snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_TRI, reg); +} + const struct snd_soc_dai_ops arizona_dai_ops = { .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, + .set_tristate = arizona_set_tristate, }; EXPORT_SYMBOL_GPL(arizona_dai_ops); -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
by Fabio Estevam 12 Dec '12

12 Dec '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Refactor set_pll code to avoid the following warnings: sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized] Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid GCC warnings). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm8983.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 9fe1e04..e7c73ff 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -851,30 +851,33 @@ static int wm8983_set_pll(struct snd_soc_dai *dai, int pll_id, struct pll_div pll_div; codec = dai->codec; - if (freq_in && freq_out) { + if (!freq_in || !freq_out) { + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + return 0; + } else { ret = pll_factors(&pll_div, freq_out * 4 * 2, freq_in); if (ret) return ret; - } - - /* disable the PLL before re-programming it */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, 0); - if (!freq_in || !freq_out) - return 0; + /* disable the PLL before re-programming it */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, 0); + + /* set PLLN and PRESCALE */ + snd_soc_write(codec, WM8983_PLL_N, + (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) + | pll_div.n); + /* set PLLK */ + snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); + snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); + /* enable the PLL */ + snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, + WM8983_PLLEN_MASK, WM8983_PLLEN); + } - /* set PLLN and PRESCALE */ - snd_soc_write(codec, WM8983_PLL_N, - (pll_div.div2 << WM8983_PLL_PRESCALE_SHIFT) - | pll_div.n); - /* set PLLK */ - snd_soc_write(codec, WM8983_PLL_K_3, pll_div.k & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_2, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8983_PLL_K_1, (pll_div.k >> 18)); - /* enable the PLL */ - snd_soc_update_bits(codec, WM8983_POWER_MANAGEMENT_1, - WM8983_PLLEN_MASK, WM8983_PLLEN); return 0; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Unconditionally wait for the FLL to lock
by Mark Brown 12 Dec '12

12 Dec '12

If the FLL is being shut down we will exit early so there is no need to check here and in fact we're checking the wrong thing anyway. Reported-by: Graeme Gregory <gg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 26 ++++++++++++-------------- 1 file changed, 12 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 09816e8..50b0983 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2875,22 +2875,20 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, ret = 0; - if (fll1 & WM8962_FLL_ENA) { - /* This should be a massive overestimate but go even - * higher if we'll error out - */ - if (wm8962->irq) - timeout = msecs_to_jiffies(5); - else - timeout = msecs_to_jiffies(1); + /* This should be a massive overestimate but go even + * higher if we'll error out + */ + if (wm8962->irq) + timeout = msecs_to_jiffies(5); + else + timeout = msecs_to_jiffies(1); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); + timeout = wait_for_completion_timeout(&wm8962->fll_lock, + timeout); - if (timeout == 0 && wm8962->irq) { - dev_err(codec->dev, "FLL lock timed out"); - ret = -ETIMEDOUT; - } + if (timeout == 0 && wm8962->irq) { + dev_err(codec->dev, "FLL lock timed out"); + ret = -ETIMEDOUT; } wm8962->fll_fref = Fref; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ALSA - Replace msleep() with usleep_range() to save module loading time
by Wang Xingchao 12 Dec '12

12 Dec '12

msleep(1) is not so accurate and may cause almost 15ms delay in azx_reset(). usleep_range() can reduce ~30ms loading time for Intel Haswell which has two HDA controllers. Signed-off-by: Wang Xingchao <wangxingchao2011(a)gmail.com> --- sound/pci/hda/hda_intel.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index eb48109..865df90 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1075,22 +1075,22 @@ static int azx_reset(struct azx *chip, int full_reset) count = 50; while (azx_readb(chip, GCTL) && --count) - msleep(1); + usleep_range(1000, 1500); /* delay for >= 100us for codec PLL to settle per spec * Rev 0.9 section 5.5.1 */ - msleep(1); + usleep_range(1000, 1500); /* Bring controller out of reset */ azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET); count = 50; while (!azx_readb(chip, GCTL) && --count) - msleep(1); + usleep_range(1000, 1500); /* Brent Chartrand said to wait >= 540us for codecs to initialize */ - msleep(1); + usleep_range(1000, 1500); __skip: /* check to see if controller is ready */ -- 1.7.9.5

3 2

[alsa-devel] ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err = -22
by isaac smith 11 Dec '12

11 Dec '12

Hello, I have the same issue with my HP Envy 15. The sound card in question is an HP Wireless Audio card that only works with HP-branded wireless speakers, so playing to the card produces no output whatsoever, but does hang whatever process tries to access it. I doubt that many people ever use this feature, so unless someone invests the time to get it working I'd like a way to just ignore it completely. I've blacklisted the snd-usb-audio module for now, but I have a USB sound card and a USB MIDI controller that I'd like to be able to use. Here's the output of "lsusb -vd 0424:b832" (apologies for length): Bus 004 Device 004: ID 0424:b832 Standard Microsystems Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x0424 Standard Microsystems Corp. idProduct 0xb832 bcdDevice 75.28 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 225 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 3 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 72 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0402 Headset bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 11 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 12 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 1 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 0 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 48000 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 1 Milliseconds wLockDelay 0 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.10 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 74 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x8a EP 10 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 10 Device Status: 0x060a (Bus Powered) Remote Wakeup Enabled > > On Thu, 6 Dec 2012, Joe Cooper wrote: > > On most HP Envy laptops the snd-usb-audio module causes the system to > > become unresponsive and Gnome Shell 3 to crash. This effects every kernel > > I've tried from 3.3.4 on up to 3.7rc8. It also produces these errors on the > > console: > > > > ALSA sound/usb/mixer.c:415 cannot get current value for control 2 ch 1: err > > = -22 > > > > ALSA sound/usb/clock.c:228 3:1:1: cannot set freq 44100 to ep 0x1 > > > > Blacklisting this module allows the system to function, but with most of > > its speakers not working. It also prevents use of other USB audio devices, > > making multitrack recording impossible. > > > > I have a ticket about it in the Fedora bug tracker, but was referred to > > alsa-devel. That ticket links to other discussions of this bug, as well. > > Here's that ticket: https://bugzilla.redhat.com/show_bug.cgi?id=884368 > > > > alsa-info is here: https://bugzilla.redhat.com/attachment.cgi?id=658989 > > > > Please let me know if you need further details, or if there's anything else > > I can do to help troubleshoot this. > > Hi Joe, > > I can try to help. > > Do you get any sound output at all (even if it stutters or sounds > wrong)? > > Can you please send the output of "lsusb -v -d 0424:b832"? > I think I can work out a patch for you to skip this device, so > you can use other audio usb devices until this gets proper support. > > Cheers, > Eldad

2 1

[alsa-devel] Denon DN-X 1600 Support
by Florian Beginn 11 Dec '12

11 Dec '12

Hello, i just made my first steps fiddling with the alsa-driver and got my DN-X 1600 working. It's a dj-mixer connected via usb, offering 8 channel input/output and midi. Samplerate can be set to either 44.1, 48 or 96khz. It seems to not answer properly on samplerate requests from snd_usb_ctl_msg() so i hardcoded it to 96khz. Regrettably, this is a pretty ugly workaround and somehow broke midi capability, although audio is running fine. I'd be happy if one of you could help me work out how to do this right/better. Find attached the patches and all information i could gather. P.S. I'm not used to git (well, any VCS) and hope the diff format is ok for you. --- a/sound/usb/clock.c 2012-01-25 08:56:49.000000000 +0100 +++ b/sound/usb/clock.c 2012-12-11 17:33:47.685196544 +0100 @@ -255,6 +255,14 @@ int err, crate; int clock = snd_usb_clock_find_source(chip, fmt->clock); + /* + * Denon DN-X 1600 won't respond when asked for samplerate. + * Currently hardcoded to only work with 96khz + */ + + if (chip->usb_id == USB_ID(0x154e, 0x500e) && rate == 96000) + return 0; + if (clock < 0) return clock; --- a/sound/usb/format.c 2012-12-07 17:11:00.000000000 +0100 +++ b/sound/usb/format.c 2012-12-11 17:42:14.069181583 +0100 @@ -281,6 +281,21 @@ int nr_triplets, data_size, ret = 0; int clock = snd_usb_clock_find_source(chip, fp->clock); + /* + * Denon DN-X 1600 won't respond when asked for samplerate. + * Currently hardcoded to only work with 96khz + */ + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + { + fp->nr_rates = 1; + fp->rate_min = 96000; + fp->rate_max = 96000; + fp->rates = SNDRV_PCM_RATE_96000; + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + fp->rate_table[0] = 96000; + return 0; + } + if (clock < 0) { snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n", __func__, clock); --- a/sound/usb/pcm.c 2012-01-25 08:56:49.000000000 +0100 +++ b/sound/usb/pcm.c 2012-12-11 17:57:11.579301294 +0100 @@ -198,6 +198,13 @@ if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL)) return 0; + /* + * Denon DN-X 1600 doesn't properly handle this -> bail out, too + */ + + if (chip->usb_id == USB_ID(0x154e, 0x500e)) + return 0; + switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: default: cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.25. lsusb (...) Bus 002 Device 003: ID 154e:500e D&M Holdings, Inc. (Denon/Marantz) lsusb -t (...) /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 3, If 0, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 2, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 3, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 4, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 3, If 5, Class=HID, Driver=usbhid, 480M (...) lsusb -v (...) Bus 002 Device 003: ID 154e:500e D&M Holdings, Inc. (Denon/Marantz) Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 ? bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x154e D&M Holdings, Inc. (Denon/Marantz) idProduct 0x500e bcdDevice 0.01 iManufacturer 1 iProduct 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 385 bNumInterfaces 6 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 0 bInterfaceCount 3 bFunctionClass 255 Vendor Specific Class bFunctionSubClass 0 bFunctionProtocol 32 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 32 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 2.00 bCategory 0 wTotalLength 92 bmControl 0x01 Latency control Control (read-only) AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bCSourceID 7 bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) bmControls 0x0555 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Cluster Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 1 bCSourceID 7 bmControls 0x0155 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0201 Microphone bAssocTerminal 0 bCSourceID 7 bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) bmControls 0x0555 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Cluster Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 bCSourceID 7 bmControls 0x0155 Copy Protect Control (read-only) Connector Control (read-only) Overload Control (read-only) Underflow Control (read-only) Overflow Control (read-only) iTerminal 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 5 bmAttributes 0x02 Internal variable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 6 bmAttributes 0x02 Internal variable Clock bmControls 0x03 Clock Frequency Control (read/write) bAssocTerminal 0 iClockSource 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 11 (CLOCK_SELECTOR) bUnitID 7 bNrInPins 2 baCSourceID( 0) 5 baCSourceID( 1) 6 bmControls 0x01 Clock Selector Control (read-only) iClockSelector 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bmControls 0x05 Active Alternate Setting Control (read-only) Valid Alternate Setting Control (read-only) bFormatType 1 bmFormats 1 PCM bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x15 Pitch Control (read-only) Data Overrun Control (read-only) Data Underrun Control (read-only) bLockDelayUnits 1 Milliseconds wLockDelay 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 32 iInterface 0 AudioStreaming Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bmControls 0x05 Active Alternate Setting Control (read-only) Valid Alternate Setting Control (read-only) bFormatType 1 bmFormats 1 PCM bNrChannels 8 bmChannelConfig 0x000000ff Front Left (FL) Front Right (FR) Front Center (FC) Low Frequency Effects (LFE) Back Left (BL) Back Right (BR) Front Left of Center (FLC) Front Right of Center (FRC) iChannelNames 0 AudioStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bSubslotSize 3 bBitResolution 24 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0138 1x 312 bytes bInterval 1 AudioControl Endpoint Descriptor: bLength 8 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bmControls 0x15 Pitch Control (read-only) Data Overrun Control (read-only) Data Underrun Control (read-only) bLockDelayUnits 1 Milliseconds wLockDelay 0 Interface Association: bLength 8 bDescriptorType 11 bFirstInterface 3 bInterfaceCount 2 bFunctionClass 1 Audio bFunctionSubClass 0 bFunctionProtocol 0 iFunction 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 4 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 97 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 17 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 18 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 19 bNrInputPins 1 baSourceID( 0) 18 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 20 bNrInputPins 1 baSourceID( 0) 17 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 1 baAssocJackID( 1) 17 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 6 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 2 baAssocJackID( 0) 3 baAssocJackID( 1) 19 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 34 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0020 1x 32 bytes bInterval 1

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[alsa-devel] [PATCH 2/2] Rename internal xonar_u1_* names to just xonar_u_*
by Denis Washington 11 Dec '12

11 Dec '12

Now that the code applies both to the U1 and the U3, xonar_u1 is not a good name anymore. Signed-off-by: Denis Washington <denisw(a)online.de> --- sound/usb/mixer.h | 2 +- sound/usb/mixer_quirks.c | 24 ++++++++++++------------ 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index a7f3d45..9e7ec0e 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -22,7 +22,7 @@ struct usb_mixer_interface { u8 rc_buffer[6]; u8 audigy2nx_leds[3]; - u8 xonar_u1_status; + u8 xonar_u_status; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2223089..de6db70 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -416,23 +416,23 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } -static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + ucontrol->value.integer.value[0] = !!(mixer->xonar_u_status & 0x02); return 0; } -static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); u8 old_status, new_status; int err, changed; - old_status = mixer->xonar_u1_status; + old_status = mixer->xonar_u_status; if (ucontrol->value.integer.value[0]) new_status = old_status | 0x02; else @@ -449,27 +449,27 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; - mixer->xonar_u1_status = new_status; + mixer->xonar_u_status = new_status; return changed; } -static struct snd_kcontrol_new snd_xonar_u1_output_switch = { +static struct snd_kcontrol_new snd_xonar_u_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Playback Switch", .info = snd_ctl_boolean_mono_info, - .get = snd_xonar_u1_switch_get, - .put = snd_xonar_u1_switch_put, + .get = snd_xonar_u_switch_get, + .put = snd_xonar_u_switch_put, }; -static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +static int snd_xonar_u_controls_create(struct usb_mixer_interface *mixer) { int err; err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + snd_ctl_new1(&snd_xonar_u_output_switch, mixer)); if (err < 0) return err; - mixer->xonar_u1_status = 0x05; + mixer->xonar_u_status = 0x05; return 0; } @@ -1110,7 +1110,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ - err = snd_xonar_u1_controls_create(mixer); + err = snd_xonar_u_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 1.8.0

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[alsa-devel] [PATCH 1/2] Enable S/PDIF on the ASUS Xonar U3
by Denis Washington 11 Dec '12

11 Dec '12

The only required change is to extend the existing Xonar U1 mixer quirks to the U3, which seems to be controlled the same way. Signed-off-by: Denis Washington <denisw(a)online.de> --- sound/usb/mixer_quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ae2b714..2223089 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1107,8 +1107,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_ftu_create_mixer(mixer); break; - case USB_ID(0x0b05, 0x1739): - case USB_ID(0x0b05, 0x1743): + case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ + case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ + case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ err = snd_xonar_u1_controls_create(mixer); break; -- 1.8.0

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Re: [alsa-devel] is it a memory bug?
by l9jj＠163.com 11 Dec '12

11 Dec '12

thanks a lot. 在 2012-12-10 19:41:58, Clemens Ladisch <clemens(a)ladisch.de>写道: l9jj(a)163.com wrote: > just call alloca() without free()? <http://linux.die.net/man/3/alloca> Regards, Clemens _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel l9jj(a)163.com ——网易邮箱手机客户端——

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[alsa-devel] how this will happen about hw_ptr_buffer_jiffies?
by l9jj 11 Dec '12

11 Dec '12

recently,i am tracing the process into the snd_pcm_update_hw_ptr0() and find the statement. i do not understand when the if (hdelta > runtime->hw_ptr_buffer_jiffies/2) (1)will happen? (2)and the reason? (3)the other question is if the dma interrupt will can enter the snd_pcm_update_hw_ptr0() every time when it ocurrs? thanks for any suggestion. Best Regards L9jj =====================code start==============================> if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { /* check for double acknowledged interrupts */ hdelta = jiffies - runtime->hw_ptr_jiffies; if (hdelta > runtime->hw_ptr_buffer_jiffies/2) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; goto __delta; } } <=====================code end============================ -- ------------------------------------ l9jj Regards

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[alsa-devel] [PATCH 2/2] Rename internal xonar_u1_* names to just xonar_u_*
by Denis Washington 11 Dec '12

11 Dec '12

Now that the code applies both to the U1 and the U3, xonar_u1 is not a good name anymore. --- sound/usb/mixer.h | 2 +- sound/usb/mixer_quirks.c | 24 ++++++++++++------------ 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index a7f3d45..9e7ec0e 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -22,7 +22,7 @@ struct usb_mixer_interface { u8 rc_buffer[6]; u8 audigy2nx_leds[3]; - u8 xonar_u1_status; + u8 xonar_u_status; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2223089..de6db70 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -416,23 +416,23 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } -static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = !!(mixer->xonar_u1_status & 0x02); + ucontrol->value.integer.value[0] = !!(mixer->xonar_u_status & 0x02); return 0; } -static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, +static int snd_xonar_u_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); u8 old_status, new_status; int err, changed; - old_status = mixer->xonar_u1_status; + old_status = mixer->xonar_u_status; if (ucontrol->value.integer.value[0]) new_status = old_status | 0x02; else @@ -449,27 +449,27 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, up_read(&mixer->chip->shutdown_rwsem); if (err < 0) return err; - mixer->xonar_u1_status = new_status; + mixer->xonar_u_status = new_status; return changed; } -static struct snd_kcontrol_new snd_xonar_u1_output_switch = { +static struct snd_kcontrol_new snd_xonar_u_output_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Playback Switch", .info = snd_ctl_boolean_mono_info, - .get = snd_xonar_u1_switch_get, - .put = snd_xonar_u1_switch_put, + .get = snd_xonar_u_switch_get, + .put = snd_xonar_u_switch_put, }; -static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) +static int snd_xonar_u_controls_create(struct usb_mixer_interface *mixer) { int err; err = snd_ctl_add(mixer->chip->card, - snd_ctl_new1(&snd_xonar_u1_output_switch, mixer)); + snd_ctl_new1(&snd_xonar_u_output_switch, mixer)); if (err < 0) return err; - mixer->xonar_u1_status = 0x05; + mixer->xonar_u_status = 0x05; return 0; } @@ -1110,7 +1110,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ - err = snd_xonar_u1_controls_create(mixer); + err = snd_xonar_u_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 1.8.0

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[alsa-devel] [PATCH 1/2] Enable S/PDIF on the ASUS Xonar U3
by Denis Washington 11 Dec '12

11 Dec '12

The only required change is to extend the existing Xonar U1 mixer quirks to the U3, which seems to be controlled the same way. --- sound/usb/mixer_quirks.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ae2b714..2223089 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1107,8 +1107,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_ftu_create_mixer(mixer); break; - case USB_ID(0x0b05, 0x1739): - case USB_ID(0x0b05, 0x1743): + case USB_ID(0x0b05, 0x1739): /* ASUS Xonar U1 */ + case USB_ID(0x0b05, 0x1743): /* ASUS Xonar U1 (2) */ + case USB_ID(0x0b05, 0x17a0): /* ASUS Xonar U3 */ err = snd_xonar_u1_controls_create(mixer); break; -- 1.8.0

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Re: [alsa-devel] [PATCH V4 0/2] Add DT support for i2s
by Padma Venkat 10 Dec '12

10 Dec '12

Hi Mark, On Wed, Nov 28, 2012 at 4:17 PM, Padmavathi Venna <padma.v(a)samsung.com> wrote: > V4 patches are based on Mark Brown's for-next branch of > "git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git" > > Changes since V3: > - Added syntex of gpio specifier as suggested by Mark Brown in > DT document file > - Added the requirement for having I2S alias ids > - Explained about gpio lines > - Made a seperate patch to get the I2S rclk src clk from alias ID > > Changes since V2: > - Rebased on 3.7-rc3 > - Custom DT bindings are prefixed with samsung > - As generic device tree DMA helpers not yet mainlined > I am still using custom dma bindings. So added a > priliminary statement regarding the same. I will rework > on my patch once generic DMA helpers are mainlined. > > Chnages since V1: > - Rebased on 3.6-rc6 > > Padmavathi Venna (2): > ASoC: Samsung: Get I2S src_clk from clock alias id. > ASoC: SAMSUNG: Add DT support for i2s Any comments on DT support for i2s patch? If not when this patch can be pulled? > > .../devicetree/bindings/sound/samsung-i2s.txt | 78 +++++++ > include/linux/platform_data/asoc-s3c.h | 6 - > sound/soc/samsung/dma.c | 1 + > sound/soc/samsung/dma.h | 1 + > sound/soc/samsung/i2s.c | 238 +++++++++++++++---- > 5 files changed, 268 insertions(+), 56 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt > > -- > 1.7.4.4 > Thanks Padma

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[alsa-devel] is it a memory bug?
by l9jj＠163.com 10 Dec '12

10 Dec '12

project: alsa-lib-1.0.25 file: dlmisc.c function: snd_dlsym_verify() description: the pointer vname hold a memory until exit the function without release.just call alloca() without free()？ l9jj(a)163.com ——网易邮箱手机客户端——

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