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

[alsa-devel] [PATCH v5 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 09 Feb '15

09 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. v5: no functional change, only addressed Takashi comments on patch 2,3,4 fixed used of bool with true/false and ! in comparisons move trigger_tstamp_pending_update flag to snd_usb_substream, will be used by ALSA core when other devices need it Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 66 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/card.h | 2 + sound/usb/pcm.c | 9 ++ 9 files changed, 479 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

2 17

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 09 Feb '15

09 Feb '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to within 500ppm of the SSC master clock, possibly divided by 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 113 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 110 insertions(+), 4 deletions(-) Changes since v1: - I have checked all Atmel datasheets I could get my hands on (and that seemed relevant), and in every instance where they have described the SSC, the description have the exact rate limitations on the RK and TK pin that I have implemented here. - Improved the comments. - Rebased on top of the latest patches from Bo Chen. One thing remains a bit unclear, and that is the 500ppm deduction. Is that really warranted? The number was just pulled out of my hat... Cheers, Peter diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..767f65bab25d 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,96 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + /* Take away 500ppm, just to be on the safe side. */ + t.max -= t.max / 2000; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +298,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk) * 2; /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +311,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +886,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +905,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

4 14

[alsa-devel] [patch] ALSA: riptide: off by one in snd_riptide_joystick_probe()
by Dan Carpenter 09 Feb '15

09 Feb '15

There is an off by one bug in snd_riptide_joystick_probe() because we do the "dev++" toward the start of the function instead of waiting until the end. The impact of this bug is: 1) A static checker warning. 2) If you connect 32 joysticks to your computer they will fail to load with a different error message than intended. 3) If you pass a module option to specify a non-default joystick port then the parameter array has to be shifted one element. The first two are not serious. For the third one, it's probably too late to change it since the bug was introduced in 2009. Either no one noticed and we can leave it alone, or if they did notice, they probably have implemented work arounds so we can't change it now without confusing them. So instead of fixing it the "theoretically correct way", I have decided to just work around it by making the joystick_port[] one element larger. This silences the static checker warning and doesn't affect user space. Fixes: db1005ec6ff8 ('ALSA: riptide - Fix joystick resource handling') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 29f2827..324e34c 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -125,7 +125,11 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; #ifdef SUPPORT_JOYSTICK -static int joystick_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x200 }; +/* + * The joystick_port accounting is off by one but I'm afraid to change it for + * fear of breaking userspace. + */ +static int joystick_port[SNDRV_CARDS + 1] = { [0 ... (SNDRV_CARDS)] = 0x200 }; #endif static int mpu_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x330 }; static int opl3_port[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS - 1)] = 0x388 };

2 4

Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Takashi Iwai 09 Feb '15

09 Feb '15

At Sat, 10 Jan 2015 13:10:30 +0100 (CET), Jan Engelhardt wrote: > > > On Wednesday 2014-12-31 09:51, Takashi Iwai wrote: > >> > > >> >Ah, I see you disable the automute. So the headphone plug shouldn't > >> >matter. > >> > >> Changing the "Auto-Mute" control does not do anything; the system always > >> exhibits Automute behavior (if that's a bug, it's one which I did not > >> care about so far). > > > >Oh, that's really interesting. I didn't expect such a > >hardware-controlled automute nowadays. We need to investigate this > >more. > >[...] > >[Volumes] On 3.19, this was renamed differently, so that "PCM" can > >appear again by a softvol item like before. > > On 3.19-rc3+ [eb74926920cfa756087a82e0b081df837177cb95] I confirm > all the volume bars/switches that went missing for me are back again. > > In particular, 3.19 shows a new "Speaker+LO" no-switch volume bar > control, and a "Speaker" no-volume switch control. Works for me, > though I wonder if the change was necessary. (In 3.17, the switch and > volume were combined in one control.) > > As said, all is fine here. Sorry, I've forgotten this pending issue. Could you confirm that the patch below works for you? Then I'll queue up for 3.20-rc1 and backport appropriately. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Fix lost sound on Thinkpad X240 The commit [7c21539c45ea: ALSA: hda - Add ultra dock support for Thinkpad X240] caused a regression, the sound got lost. The machine seems requiring the fixed binding between DACs and pins, and the commit above resulted in the change of binding. As a fix, this patch adds the binding list to ensure to connect to the correct DAC. Reported-by: Jan Engelhardt <jengelh(a)inai.de> Fixes: 7c21539c45ea ('ALSA: hda - Add ultra dock support for Thinkpad X240.') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 040306194e6d..dba31bafcd1c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4226,6 +4226,22 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, } } +/* TP X240 seems to need fixed DAC binding; otherwise sound gets lost */ +static void alc_fixup_tpx240_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static hda_nid_t preferred_pairs[] = { + 0x14, 0x03, + 0x15, 0x02, + 0x16, 0x02, + 0 + }; + struct alc_spec *spec = codec->spec; + spec->gen.preferred_dacs = preferred_pairs; + } +} + static void alc283_hp_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) { @@ -4411,6 +4427,7 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, + ALC292_FIXUP_TPX240_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -4856,6 +4873,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, + [ALC292_FIXUP_TPX240_DOCK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpx240_dacs, + .chained = true, + .chain_id = ALC292_FIXUP_TPT440_DOCK, + }, [ALC283_FIXUP_BXBT2807_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -5026,7 +5049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPX240_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 2.2.2

2 1

[alsa-devel] Can we remove un-used Sound driver ?
by Kuninori Morimoto 09 Feb '15

09 Feb '15

Hi SH ML, ALSA ML We have below sound drivers in upstream kernel. But, these are not used from upstreamed board. And, these are for very old SoC. I guess Renesas doesn't/can't support these SoC/board anymore. Can we remove these sound drivers ? Because maintenance cost for these is not zero... - ${LINUX}/sound/soc/sh/ssi.c - SH7760/SH7780 - CONFIG_SND_SOC_SH4_SSI - ${LINUX}/linux/sound/soc/sh/hac.c - SH7760/SH7780 - CONFIG_SND_SOC_SH4_HAC - ${LINUX}/sound/soc/sh/sh7760-ac97.c - SH7760 - CONFIG_SND_SH7760_AC97 - ${LINUX}/sound/soc/sh/dma-sh7760.c - SH7760 - CONFIG_SND_SOC_PCM_SH7760 - ${LINUX}/sound/soc/sh/siu_dai.c *1 ${LINUX}/sound/soc/sh/siu_pcm.c ${LINUX}/sound/soc/sh/siu.h - SH7343, SH7722 - CONFIG_SND_SOC_SH4_SIU - ${LINUX}/sound/soc/sh/migor.c *2 - SH7722 - CONFIG_SND_SIU_MIGOR *1 it is registered from ${LINUX}/arch/sh/kernel/cpu/sh4a/setup-sh7722.c but, no one enabled CONFIG_SND_SOC_SH4_SIU in defconfig *2 it uses *1, but no one enabled CONFIG_SND_SIU_MIGOR Best regards --- Kuninori Morimoto

4 12

[alsa-devel] [PATCH 0/2] ASoC: rsnd: tidyup .remove method
by Kuninori Morimoto 09 Feb '15

09 Feb '15

Hi Mark These patches call missing remove/unregister functions on rsnd driver. Kuninori Morimoto (2): ASoC: rsnd: call missing snd_ctl_remove() ASoC: rsnd: call missing snd_soc_unregiter_component/platform() sound/soc/sh/rcar/core.c | 14 +++++++++++++- sound/soc/sh/rcar/dvc.c | 15 +++++++++++++++ sound/soc/sh/rcar/rsnd.h | 5 +++++ 3 files changed, 33 insertions(+), 1 deletion(-) BTW, we found ALSA SoC bind/unbind issue on latest upstream kernel. ALSA SoC is using like this. [CPU] - [CARD] - [CODEC] If we unbind CPU here, it still has CARD infomation. And, kernel doesn't call initialize function when re-bind, and it will panic when playback ------------------------- In our case, CPU : rsnd CARD : simple-card CODEC : ak4642 Current list / # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Unbind CPU echo "ec500000.rcar_sound" > /sys/bus/platform/drivers/rcar_sound/unbind But, it still has unbinded card / # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: rsnddai0ak4642h [rsnd-dai.0-ak4642-hifi], device 0: rsnd-dai.0-ak4642-hifi ak4642-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Re-bind. .probe function was called, but no mapping message "asoc-simple-card sound: ak4642-hifi <-> ec500000.rcar_sound mapping ok" echo "ec500000.rcar_sound" > /sys/bus/platform/drivers/rcar_sound/bind rcar_sound ec500000.rcar_sound: probed Kernel panic when playback after rebind / # aplay /home/xxx.wav ASoC: ak4642-hifi <-> No matching rates Unable to handle kernel paging request at virtual address ee196100 pgd = eebbc000 [ee196100] *pgd=6e01141e(bad) Internal error: Oops: 8000000d [#1] SMP ARM CPU: 0 PID: 936 Comm: aplay Tainted: G W 3.19.0-rc6-02875-gf8eac9a #1541 Hardware name: Generic R8A7790 (Flattened Device Tree) task: edc2adc0 ti: ee08a000 task.ti: ee08a000 PC is at 0xee196100 LR is at soc_pcm_open+0x6e8/0x7ac pc : [<ee196100>] lr : [<c0386e18>] psr: a00f0013 sp : ee08bc90 ip : ee08bc90 fp : ee08bcec r10: eeb30800 r9 : eeae7a00 r8 : ffffffea r7 : eeb30800 r6 : ee947480 r5 : ee328a00 r4 : edbf5010 r3 : ee196100 r2 : ee0dc290 r1 : ee947480 r0 : ee328a00 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 10c5307d Table: 6ebbc06a DAC: 00000015 Process aplay (pid: 936, stack limit = 0xee08a238) Stack: (0xee08bc90 to 0xee08c000) bc80: 00000001 00000002 edbf5010 00000010 bca0: ee91d520 edbf501c 0000000c 00000000 00000000 00000001 ffffffff eeae7a00 bcc0: ee08bcf4 00000000 ee08bd10 eea3ad00 ee328ce8 ee328d00 00000000 00000001 bce0: ee08bd0c ee08bcf0 c0376a60 c038673c c0045098 ee328a00 00000000 ee328c00 bd00: ee08bd54 ee08bd10 c0376b38 c0376a18 ee08bd2c 00000000 edc2adc0 c0045098 bd20: ee328d04 ee328d04 c024dd10 ee328c00 eea3ad00 eebb3638 00000000 c0661848 bd40: eea3ad00 eea3ad00 ee08bd6c ee08bd58 c0376d04 c0376ab4 c04f179c eea3ad00 bd60: ee08bd8c ee08bd70 c0367f18 c0376ccc c0367e9c 00000000 eebb3638 ee13bec0 bd80: ee08bdbc ee08bd90 c00c8b94 c0367ea8 00000000 00000010 eea3ad00 eebb3638 bda0: ee08be88 00000000 c00c8a5c eea3ad08 ee08bde4 ee08bdc0 c00c303c c00c8a68 bdc0: ee08bed0 eea3ad00 ee08be88 00000000 ee3f0000 ee08be84 ee08bdf4 ee08bde8 bde0: c00c3ff8 c00c2ea0 ee08be6c ee08bdf8 c00d0c78 c00c3fb8 00000101 ee4264c8 be00: ee03e890 ee425770 eebb26b8 00000026 00000000 ee425770 ee08bee8 00000000 be20: 00000802 ee08bf5c 00000000 00000000 ee08a000 eebb3638 ee03e890 ee425770 be40: c00cef24 ee08bed0 eea3ad00 ee08be90 ee08bf5c ee3f0000 ee08be88 00000000 be60: ee08bec4 ee08be70 c00d10d4 c00d0400 ee08be84 ee9a1380 ee08bf0c ee08be88 be80: c036cddc 00000000 ee03e890 ee4274c8 ee08bec4 ee08bea0 c01aa17c ee08bf5c bea0: 00000001 ee3f0000 ffffff9c c000eb24 ee08a000 00000000 ee08bf4c ee08bec8 bec0: c00d1474 c00d0f00 00000041 ee08bed8 ee03e890 ee4274c8 474edc20 00000008 bee0: ee3f0019 eea3a700 ee03e7d0 ee4264c8 eebb3638 00000101 00000000 00000034 bf00: 00000000 00000000 00000000 c00de444 00080802 00080802 be8b4ea4 ee3f0000 bf20: ffffff9c c000eb24 ee08a000 00000000 00080802 00000004 ee3f0000 ffffff9c bf40: ee08bf94 ee08bf50 c00c4430 c00d144c c00d38e0 c00def38 eea3a700 00000802 bf60: be8b0000 00000026 00000100 00000001 be8b466c be8b4ea4 00000002 00000005 bf80: c000eb24 ee08a000 ee08bfa4 ee08bf98 c00c44f8 c00c431c 00000000 ee08bfa8 bfa0: c000e980 c00c44dc be8b466c be8b4ea4 be8b4794 00080802 be8b4628 00080802 bfc0: be8b466c be8b4ea4 00000002 00000005 001255f8 00129264 000dd264 00124478 bfe0: 00000000 be8b4634 00049257 00083036 600f0030 be8b4794 03433c61 0717419f Backtrace: [<c0386730>] (soc_pcm_open) from [<c0376a60>] (snd_pcm_open_substream+0x54/0x9c) r10:00000001 r9:00000000 r8:ee328d00 r7:ee328ce8 r6:eea3ad00 r5:ee08bd10 r4:00000000 [<c0376a0c>] (snd_pcm_open_substream) from [<c0376b38>] (snd_pcm_open+0x90/0x1b4) r5:ee328c00 r4:00000000 [<c0376aa8>] (snd_pcm_open) from [<c0376d04>] (snd_pcm_playback_open+0x44/0x64) r10:eea3ad00 r9:eea3ad00 r8:c0661848 r7:00000000 r6:eebb3638 r5:eea3ad00 r4:ee328c00 [<c0376cc0>] (snd_pcm_playback_open) from [<c0367f18>] (snd_open+0x7c/0x90) r5:eea3ad00 r4:c04f179c [<c0367e9c>] (snd_open) from [<c00c8b94>] (chrdev_open+0x138/0x164) r6:ee13bec0 r5:eebb3638 r4:00000000 r3:c0367e9c [<c00c8a5c>] (chrdev_open) from [<c00c303c>] (do_dentry_open.isra.17+0x1a8/0x2b8) r9:eea3ad08 r8:c00c8a5c r7:00000000 r6:ee08be88 r5:eebb3638 r4:eea3ad00 [<c00c2e94>] (do_dentry_open.isra.17) from [<c00c3ff8>] (vfs_open+0x4c/0x50) r9:ee08be84 r8:ee3f0000 r7:00000000 r6:ee08be88 r5:eea3ad00 r4:ee08bed0 [<c00c3fac>] (vfs_open) from [<c00d0c78>] (do_last.isra.47+0x884/0xb00) [<c00d03f4>] (do_last.isra.47) from [<c00d10d4>] (path_openat+0x1e0/0x54c) r10:00000000 r9:ee08be88 r8:ee3f0000 r7:ee08bf5c r6:ee08be90 r5:eea3ad00 r4:ee08bed0 [<c00d0ef4>] (path_openat) from [<c00d1474>] (do_filp_open+0x34/0x80) r10:00000000 r9:ee08a000 r8:c000eb24 r7:ffffff9c r6:ee3f0000 r5:00000001 r4:ee08bf5c [<c00d1440>] (do_filp_open) from [<c00c4430>] (do_sys_open+0x120/0x1c0) r7:ffffff9c r6:ee3f0000 r5:00000004 r4:00080802 [<c00c4310>] (do_sys_open) from [<c00c44f8>] (SyS_open+0x28/0x2c) r9:ee08a000 r8:c000eb24 r7:00000005 r6:00000002 r5:be8b4ea4 r4:be8b466c [<c00c44d0>] (SyS_open) from [<c000e980>] (ret_fast_syscall+0x0/0x34) Code: 00000000 00000000 00000000 00000000 (0000004f) ---[ end trace bd44069e52a540ae ]--- Segmentation fault Best regards --- Kuninori Morimoto

2 28

[alsa-devel] [GIT PULL] ASoC updates for v3.20
by Mark Brown 09 Feb '15

09 Feb '15

2 1

[alsa-devel] [PATCH] ASoC: rt5670: Set use_single_rw flag for regmap
by Bard Liao 09 Feb '15

09 Feb '15

RT5670 doesn't support auto incrementing writes so driver should set the use_single_rw flag for regmap. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0632b74..6aa4ab2 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2614,6 +2614,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5670 = { static const struct regmap_config rt5670_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5670_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5670_ranges) * RT5670_PR_SPACING), .volatile_reg = rt5670_volatile_register, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2] ASoC: rt286: Add rt288 codec support
by Bard Liao 09 Feb '15

09 Feb '15

This patch adds support for rt288 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81bdd276..7a72eac 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -1168,6 +1169,7 @@ static const struct regmap_config rt286_regmap = { static const struct i2c_device_id rt286_i2c_id[] = { {"rt286", 0}, + {"rt288", 0}, {} }; MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); @@ -1210,7 +1212,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: max98357a: Fix build in !CONFIG_OF case
by Mark Brown 09 Feb '15

09 Feb '15

Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 98b9153..1806333 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -120,6 +120,7 @@ static const struct of_device_id max98357a_device_id[] = { { .compatible = "maxim," DRV_NAME, }, {} }; +MODULE_DEVICE_TABLE(of, max98357a_device_id); #endif static struct platform_driver max98357a_platform_driver = { @@ -135,4 +136,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:" DRV_NAME); -MODULE_DEVICE_TABLE(of, max98357a_device_id); -- 2.1.4

1 0