May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 03 May '16

03 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 81 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 70 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] iMX6UL EVK with kernel 4.6
by Petr Kulhavy 02 May '16

02 May '16

Hi, I'm trying to run the latest mainline kernel 4.6 on the iMX6UL EVK. The board mostly works and the only problem is audio. Even though the sound card is detected and the ALSA mixer can be controlled there is no MCLK coming out of the SAI2 interface. All the related peripherals in the DTS (pinmux, SAI2, sound card, codec) seem to be set-up correctly, I also tried out various things like disabling the ASRC, but without success and I'm slowly running out of ideas. Could somebody give me a hint please, what is different to the Freescale kernel branch where the MCLK works? By the way, why aren't the Freescale patches merged to the mainline kernel? Thanks Petr

2 14

Re: [alsa-devel] bytcr_rt5640.c:(.text+0x1593b4): undefined reference to `sst_acpi_find_name_from_hid'
by Randy Dunlap 02 May '16

02 May '16

On 03/29/16 07:21, Borislav Petkov wrote: > Hi guys, > > I'm seeing a bunch of those errors today on randconfig builds: > > sound/built-in.o: In function `snd_byt_rt5640_mc_probe': > bytcr_rt5640.c:(.text+0x1593b4): undefined reference to `sst_acpi_find_name_from_hid' > make: *** [vmlinux] Error 1 > > ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! > make[1]: *** [__modpost] Error 1 > make: *** [modules] Error 2 > make: *** Waiting for unfinished jobs.... > > Some of the failing configs attached. > I have 2 patches for this. I'll send them right away. -- ~Randy

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Add support for TLV based binary controls
by Charles Keepax 02 May '16

02 May '16

This patch adds support for the arbitrary length TLV based binary controls. This allows users to properly access controls that are more than 512 bytes in length. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 128 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 108 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5699a81..630ebcd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -160,6 +160,8 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define ADSP_MAX_STD_CTRL_SIZE 512 + struct wm_adsp_buf { struct list_head list; void *buf; @@ -435,6 +437,7 @@ struct wm_coeff_ctl { size_t len; unsigned int set:1; struct snd_kcontrol *kcontrol; + struct soc_bytes_ext bytes_ext; unsigned int flags; }; @@ -711,10 +714,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } +static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) +{ + return container_of(ext, struct wm_coeff_ctl, bytes_ext); +} + static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -763,7 +773,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -780,6 +792,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, + const unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (copy_from_user(ctl->cache, bytes, size)) { + ret = -EFAULT; + } else { + ctl->set = 1; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, ctl->cache, size); + } + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, void *buf, size_t len) { @@ -822,7 +857,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -845,12 +882,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, + unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { + if (ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + else + ret = -EPERM; + } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + } + + if (!ret && copy_to_user(bytes, ctl->cache, size)) + ret = -EFAULT; + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + struct wmfw_ctl_work { struct wm_adsp *dsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; +static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) +{ + unsigned int out, rd, wr, vol; + + if (len > ADSP_MAX_STD_CTRL_SIZE) { + rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ; + wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } else { + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + } + + if (in) { + if (in & WMFW_CTL_FLAG_READABLE) + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; @@ -868,19 +965,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; - kcontrol->private_value = (unsigned long)ctl; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol->tlv.c = snd_soc_bytes_tlv_callback; + kcontrol->private_value = (unsigned long)&ctl->bytes_ext; - if (ctl->flags) { - if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE; - if (ctl->flags & WMFW_CTL_FLAG_READABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ; - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } else { - kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE; - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + + kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len); ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1); if (ret < 0) @@ -1039,11 +1132,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->flags = flags; ctl->offset = offset; - if (len > 512) { - adsp_warn(dsp, "Truncating control %s from %d\n", - ctl->name, len); - len = 512; - } ctl->len = len; ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { -- 2.1.4

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[alsa-devel] Volume control with mute in the wrong end
by Peter Rosin 02 May '16

02 May '16

Hi! I'm writing a driver for the max9860 codec https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf and it has a digital sidetone control where it can mix in the left channel ADC into the (mono) DAC output. The register that controls the sidetone level makes me wonder how I should wire up DAPM for it. You set the gain for the sidetone in the same register field as you enable it. I have a problem though, since the mute value is at the wrong and of the scale, i.e. you turn the volume up and up and up, and when you reach "max" it is off. I did try to declare a TLV range like so: /* The dvst field has its mute in the wrong end. Sigh. */ static const DECLARE_TLV_DB_RANGE(dvst_tlv, 0, 30, TLV_DB_SCALE_ITEM(-6000, 200, 0), 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); and then hook it that up with a dapm mixer: static const struct snd_kcontrol_new max9860_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("Sidetone Volume", MAX9860_DACGAIN, MAX9860_DVST_SHIFT, 31, 1, dvst_tlv), }; static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 0, max9860_mixer_controls, ARRAY_SIZE(max9860_mixer_controls)), SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), SND_SOC_DAPM_OUTPUT("OUT"), }; static const struct snd_soc_dapm_route max9860_dapm_routes[] = { { "ADCL", NULL, "MICL" }, { "ADCR", NULL, "MICR" }, { "AIFOUTL", NULL, "ADCL" }, { "AIFOUTR", NULL, "ADCR" }, { "OUT", NULL, "DAC" }, { "DAC", NULL, "Mixer" }, { "Mixer", "Sidetone Volume", "ADCL" }, { "Mixer", NULL, "AIFINL" }, { "Mixer", NULL, "AIFINR" }, }; But DAPM and some tools (e.g. alsamixer) do not properly recognize the mute value at the wrong end on the scale for the sidetone. So, the sidetone path is never off (in the dapm sense). I can use amixer to set the special mute value, and it states -99999.99dB, so I think the tlv declaration is correct, at least in theory. But as I said alsamixer is thoroughly confused by the control and behaves rather strange... If I change the tlv declaration so that DAPM thinks the mute is in the natural end of the scale, the paths are off in the dapm sense when the sidetone is muted (but the sidetone is not in fact muted, of course). Is there anything I can do to make DAPM handle this insane register field better? Cheers, Peter

1 0

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix overwriting of ahclkx
by J.D. Schroeder 02 May '16

02 May '16

From: Jim Lodes <jim.lodes(a)garmin.com> The mcasp davinci_mcasp_set_dai_fmt function was overriding ahclkx input/output status that had already been set by the davinci_mcasp_set_sysclk function. This commit removes clearing of the ahclkx input/output status from davinci_mcasp_set_dai_fmt. Signed-off-by: Jim Lodes <jim.lodes(a)garmin.com> Signed-off-by: J.D. Schroeder <jay.schroeder(a)garmin.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e132498..a1197ad 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + ACLKX | AFSX | ACLKR | AHCLKR | AFSR); mcasp->bclk_master = 0; break; default: -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake: Wov optimizations
by Vinod Koul 02 May '16

02 May '16

This series starts with two minor fixes on driver to fix a memory leak and one NULL pointer exception and then adds power optimization to driver for WOV case by suspending the stream DMA and keeping only DSP pipe and endpoint On during that case Dharageswari.R (2): ASoC: Intel: Skylake: Fix the NULL pointer exception in dsp_clean up ASoC: Intel: Skylake: Prevent sending Set DMA Control IPC if the widget is "On" Jeeja KP (2): ASoC: Intel: Skylake: Suspend PCMs when marked as active suspend ASoC: Intel: Boards: remove ignore_suspend for WoV streams Mousumi Jana (1): ASoC: Intel: Skylake: Fix memory leak during init instance sound/soc/intel/boards/skl_nau88l25_max98357a.c | 1 - sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 1 - sound/soc/intel/boards/skl_rt286.c | 1 - sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 32 ++++++++++++++++--------- sound/soc/intel/skylake/skl-sst-dsp.c | 2 -- sound/soc/intel/skylake/skl-sst.c | 4 ++++ 7 files changed, 26 insertions(+), 17 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by Enric Balletbo Serra 02 May '16

02 May '16

Hi John, Thanks for answer. 2016-04-30 9:00 GMT+02:00 John Keeping <john(a)keeping.me.uk>: > Hi Enric, > > On Fri, Apr 29, 2016 at 04:59:27PM +0200, Enric Balletbo Serra wrote: >> 2015-12-09 11:32 GMT+01:00 John Keeping <john(a)metanate.com>: >> > If we only clear the tx/rx state when both are disabled it is not >> > possible to start/stop one multiple times while the other is running. >> > Since the two are independently controlled, treat them as such and >> > remove the false dependency between capture and playback. >> > >> > Signed-off-by: John Keeping <john(a)metanate.com> >> > --- >> > sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- >> > 1 file changed, 32 insertions(+), 40 deletions(-) >> > >> > diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c >> > index 83b1b9c..acc6225 100644 >> > --- a/sound/soc/rockchip/rockchip_i2s.c >> > +++ b/sound/soc/rockchip/rockchip_i2s.c >> > @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >> > I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); >> > >> > regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >> > + I2S_XFER_TXS_START, >> > + I2S_XFER_TXS_START); >> > >> > i2s->tx_start = true; >> > } else { >> > @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >> > regmap_update_bits(i2s->regmap, I2S_DMACR, >> > I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); >> > >> > - if (!i2s->rx_start) { >> > - regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | >> > - I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_STOP | >> > - I2S_XFER_RXS_STOP); >> > + regmap_update_bits(i2s->regmap, I2S_XFER, >> > + I2S_XFER_TXS_START, >> > + I2S_XFER_TXS_STOP); >> > >> > - regmap_update_bits(i2s->regmap, I2S_CLR, >> > - I2S_CLR_TXC | I2S_CLR_RXC, >> > - I2S_CLR_TXC | I2S_CLR_RXC); >> > + regmap_update_bits(i2s->regmap, I2S_CLR, >> > + I2S_CLR_TXC, >> > + I2S_CLR_TXC); >> > >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > >> > - /* Should wait for clear operation to finish */ >> > - while (val) { >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > - retry--; >> > - if (!retry) { >> > - dev_warn(i2s->dev, "fail to clear\n"); >> > - break; >> > - } >> > + /* Should wait for clear operation to finish */ >> > + while (val & I2S_CLR_TXC) { >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > + retry--; >> > + if (!retry) { >> > + dev_warn(i2s->dev, "fail to clear\n"); >> > + break; >> > } >> > } >> > } >> > @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >> > I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); >> > >> > regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >> > + I2S_XFER_RXS_START, >> > + I2S_XFER_RXS_START); >> > >> > i2s->rx_start = true; >> > } else { >> > @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >> > regmap_update_bits(i2s->regmap, I2S_DMACR, >> > I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); >> > >> > - if (!i2s->tx_start) { >> > - regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | >> > - I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_STOP | >> > - I2S_XFER_RXS_STOP); >> > + regmap_update_bits(i2s->regmap, I2S_XFER, >> > + I2S_XFER_RXS_START, >> > + I2S_XFER_RXS_STOP); >> > >> > - regmap_update_bits(i2s->regmap, I2S_CLR, >> > - I2S_CLR_TXC | I2S_CLR_RXC, >> > - I2S_CLR_TXC | I2S_CLR_RXC); >> > + regmap_update_bits(i2s->regmap, I2S_CLR, >> > + I2S_CLR_RXC, >> > + I2S_CLR_RXC); >> > >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > >> > - /* Should wait for clear operation to finish */ >> > - while (val) { >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > - retry--; >> > - if (!retry) { >> > - dev_warn(i2s->dev, "fail to clear\n"); >> > - break; >> > - } >> > + /* Should wait for clear operation to finish */ >> > + while (val & I2S_CLR_RXC) { >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > + retry--; >> > + if (!retry) { >> > + dev_warn(i2s->dev, "fail to clear\n"); >> > + break; >> > } >> > } >> > } >> > -- >> > 2.6.3.462.gbe2c914 >> >> Using my Veyron Jerry Chromebook wiht latest kernel I found that the >> speakers doesn't work. Bisect points to this patch as the offending >> commit. However your changes looks reasonable the fact is that >> reverting the patch makes the audio work again on my device. I need to >> dig a bit more into the issue, but meanwhile, any idea on what is >> happening ? Can I ask which device did you test? > > I tested this on a Radxa Rock2 square. This change fixed an issue I > found when running separate processes for playback and recording (I was > testing with arecord and speaker-test IIRC) but I don't think that > should be relevant. I just read through the patch again and I can't see > anything obviously wrong so I'm afraid I don't have any idea what's > causing the problem you're seeing. > By chance, do you have a git repo containing the changes required to test the sound on Radxa Rock2? Guess this is not in mainline. Thanks, Enric > > John

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[alsa-devel] [PATCH 0/2] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 02 May '16

02 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver. One more patch to add support in codec will follow this The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are shutdown, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and second one is dependent on it so Takashi if you are okay, pls ACK it and we cna merge thru ASoC tree Vinod Koul (2): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link mangement include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 ++++++++++++++++++++ 5 files changed, 112 insertions(+), 1 deletion(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/3]: conf: topology: Add the conf for SKL-rt286
by Vinod Koul 02 May '16

02 May '16

This series adds SKL-i2c conf for simple PCM topology graph Changes in v4: - add back the pre-compiled binary blobs - remove running the binary on intsall step Changes in v3: - remove bin blobs, now they are generated in install step - hook the tool compilation to alsa-lib and run it on install - add u8 in alsa type_compat.h Shreyas NC (3): conf: topology: Add Skylake i2s conf Add u8 in type_compat.h conf: topology: Generate Private data binary blobs configure.ac | 2 + include/sound/type_compat.h | 2 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 5 + src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/data/Makefile.am | 4 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_local.h | 9 + .../topology/sklrt286/data/skl-tplg-interface.h | 232 +++ .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 28 files changed, 2502 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_local.h create mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin create mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin create mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 1.9.1

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