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

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Update the IVI feedback path
by Vinod Koul 02 Mar '16

02 Mar '16

This patch series fixes the blob query direction and then uses that in IVI feedback path in nau88125_ssm4567 machine. Jeeja KP (3): ASoC: Intel: Skylake: correct direction in blob query ASoC: Intel: Skylake: Add IVI feedback path in nau88125_ssm4567 machine ASoC: Intel: Skylake: remove IVI codec loop in skl_nau88l25_ssm4567 machine sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 21 +++------------------ sound/soc/intel/skylake/skl-topology.c | 15 ++++++++------- 2 files changed, 11 insertions(+), 25 deletions(-) -- 1.9.1

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Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Liam Girdwood 02 Mar '16

02 Mar '16

On Tue, 2016-02-16 at 21:47 +0530, Jeeja KP wrote: > We have no mechanism to configure hw_params for DAIs which are hostless. > Like a loop or DAIs connected to internal sinks/sources. > Fwiw, a few years back Peter, Sebastien and I were considering a similar mechanism for the ABE on OMAP4. We never got around to implementing or upstreaming though.... One comment below. > Problem Statement: > During a BE-BE loop of DSP or a SRC->SINK path, we cannot program > hw_params. There are no hw_params from user as these are hostless. > This scenario can happens in below 2 cases: > > Case 1: BE-BE loop aka hostless stream through SoC DSP > Hostless stream can be defined by BE to BE connection and in this case > the path is enabled or disabled by the state of the DAPM graph. This > usually means there is a mixer control that can be used to connect or > disconnect the path between both DAIs. > Example: > |--------------/----------------| > | | > | | > codec out modem_in > | | > | | > ssp0_TX ssp1_RX > | | > | | > AIFI_Playback dummy_capture > > In the above case (SSP0,codec playback,SSp1 dai needs to be configured > and turned on with the selected hw_params). > > Case 2: SoC Internal Sink or Source > Example: > WOV SINK (DAPM Sink) > | > | > | > Switch > | > | > w1 > | > dmic rx (BE dai widget) > | > Capture(codec dai widget) > | > DMIC AIF > | > DMic > | > Soc DMic (MIC Widget) (SRC) > > In the above case BE (dmic RX) dai hw_params needs to applied. > > Approach: > 1. Allow params to be defined for other BE dai link also. this > currently applicable only for codec_codec link. > > 2. when params is defined for BE dai link, create params kcontrols > and associate it with the dai widgets. this will allow params to be > selected via Kcontrol for each dai defined in the dai link. > The only problem we have with using a kcontrol to define params is that we already have an existing kernel ioctl and public C API to define SW/HW params. This would add another mechanism, unless each new kcontrol allows modification of a single param ? Liam > 3. Before powering on the dai widget and if the dai is not powered > with the stream event, then dai_ops(hw_params) for the dai widget will > be called.This will apply the selected hw_params. > > 4. with the above, fix_up can be removed and can use .params to > select the parameter. > > Core Changes: > 1. params definition needs to changed, currently this is defined to used > only for codec codec link. define new flag to identifying codec > codec link. > > 2. if params is defined for any BE dai link, then create kcontrol for > each dai defined in the dai_link and associated the kcontrol to dai > widget. > > 3. In the powering sequence check if the dai widget and if the event is > not a stream event, SND_SOC_DAPM_STREAM_NOP , then call dai ops with > the selected params. This will apply the hw_params for dai. > > 1. params definition: > - Add new flag in struct snd_soc_dai_link. > - Based on this flag create, the codec codec link instead of > params > struct snd_soc_dai_link { > const struct snd_soc_pcm_stream *params; > unsigned int num_params; > > + /* flag to create codec codec based on the flag*/ > + unsigned int codec_link:1; > + > unsigned int dai_fmt; /* format to set on init */ > > enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ > > - creation of dai link based on the codec_link flag > static int soc_probe_link_dais(struct snd_soc_card *card, > int num, int order) > } > } else { > > - if (!dai_link->params) { > + if (!dai_link->codec_link) { > /* create the pcm */ > ret = soc_new_pcm(rtd, num); > if (ret < 0) { > > 2. when params is defined for BE dai link, create params kcontrols and > associate it with the dai widgets. this will allow params to be selected > via Kcontrol for each dai defined in the dai link. > a. Move the param control creation from snd_soc_dapm_new_pcm and add new > API to created kcontrols with the params. > +snd_soc_dapm_create_param_control(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > { > /* create kcontrol based on the params */ > } > > - modify the snd_soc_dapm_new_pcm to create the param control > int snd_soc_dapm_new_pcm(struct snd_soc_card *card, > const struct snd_soc_pcm_stream *params, > unsigned int num_params, > struct snd_soc_dapm_widget *source, > struct snd_soc_dapm_widget *sink) > { > > template.kcontrol_news = snd_soc_dapm_create_param_control(card, > params, num_params); > if (!template.kcontrol_news) { > dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > link_name); > ret = -ENOMEM; > - goto outfree_private_value; > + goto outfree_link_name; > } > > > b. add new API snd_soc_dapm_new_dai_params() to create kcontrol for dai and associated it to dai. > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w) > +{ > + struct snd_kcontrol_new *kcontrol_news; > + > + kcontrol_news = snd_soc_dapm_create_param_control(card, params, > + num_params); > + if (!kcontrol_news) { > + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > + dai_w->name); > + return -EINVAL; > + } > + > + dai_w->num_kcontrols = 1; > + dai_w->kcontrol_news = kcontrol_news; > + > + dai_w->params = params; > + dai_w->num_params = num_params; > + > + return 0; > + > +} > + > +++ b/include/sound/soc-dapm.h > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w); > > > c. if the be dai link has .params defined, call new API snd_soc_dapm_new_dai_params > > +static int soc_link_be_params_widget(struct snd_soc_card *card, > + struct snd_soc_dai *dai, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > +{ > + int ret; > + struct snd_soc_dapm_widget *w; > + > + if (!strcmp(dai->name, "snd-soc-dummy")) > + return 0; > + > + if (dai->playback_widget) { > + w = dai->playback_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + if (dai->capture_widget) { > + w = dai->capture_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + return 0; > +} > + > +static int soc_link_be_params_dai_widgets(struct snd_soc_card *card, > + struct snd_soc_dai_link *dai_link, > + struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + int ret, i; > + > + for (i = 0; i < rtd->num_codecs; i++) { > + struct snd_soc_dai *codec = rtd->codec_dais[i]; > + > + ret = soc_link_be_params_widget(card, codec, > + dai_link->params, > + dai_link->num_params); > + > + if (!ret) > + return ret; > + } > + > + return soc_link_be_params_widget(card, cpu_dai, dai_link->params, > + dai_link->num_params); > +} > + > > @@ -1651,6 +1714,13 @@ static int soc_probe_link_dais(struct snd_soc_card *card, > dai_link->stream_name, ret); > return ret; > } > + if (dai_link->params && dai_link->no_pcm) { > + /* create dai link controls */ > + ret = soc_link_be_params_dai_widgets(card, > + dai_link, rtd); > + if (ret) > + return ret; > + } > } else { > INIT_DELAYED_WORK(&rtd->delayed_work, > codec2codec_close_delayed_work); > > > 3. Applying the hw_params for dai widget > > --- a/sound/soc/soc-dapm.c > +++ b/sound/soc/soc-dapm.c > @@ -1398,9 +1398,64 @@ static void dapm_seq_check_event(struct snd_soc_card * > card, > } > } > > +static int snd_soc_dai_params(struct snd_soc_dapm_widget *w, int event) > +{ > + struct snd_soc_dai *dai; > + const struct snd_soc_pcm_stream *config = w->params + w->params_select; > + struct snd_pcm_substream substream; > + struct snd_pcm_hw_params *params = NULL; > + u64 fmt; > + int ret = 0; > + > + if (WARN_ON(!config)) > + return -EINVAL; > + > + dai = w->priv; > + > + if (config->formats) { > + fmt = ffs(config->formats) - 1; > + } else { > + dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", > + config->formats); > + fmt = 0; > + } > + > + /* Currently very limited parameter selection */ > + params = kzalloc(sizeof(*params), GFP_KERNEL); > + if (!params) { > + ret = -ENOMEM; > + goto out; > + } > + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = > + config->rate_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = > + config->rate_max; > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min > + = config->channels_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max > + = config->channels_max; > + > + memset(&substream, 0, sizeof(substream)); > + if (w->id == snd_soc_dapm_dai_in) > + substream.stream = SNDRV_PCM_STREAM_CAPTURE; > + else > + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; > + > + ret = soc_dai_hw_params(&substream, params, dai); > + if (ret < 0) > + goto out; > +out: > + kfree(params); > + return ret; > +} > + > /* Apply the coalesced changes from a DAPM sequence */ > static void dapm_seq_run_coalesced(struct snd_soc_card *card, > - struct list_head *pending) > + struct list_head *pending, > + int event) > { > struct snd_soc_dapm_context *dapm; > struct snd_soc_dapm_widget *w; > @@ -1413,6 +1468,14 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm = w->dapm; > > list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_PRE_PMU); > + } > + } > + > + list_for_each_entry(w, pending, power_list) { > WARN_ON(reg != w->reg || dapm != w->dapm); > w->power = w->new_power; > > @@ -1447,6 +1510,13 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); > } > + list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_POST_PMD); > + } > + } > } > > /* Apply a DAPM power sequence. > @@ -1482,7 +1552,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > if (sort[w->id] != cur_sort || w->reg != cur_reg || > w->dapm != cur_dapm || w->subseq != cur_subseq) { > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > @@ -1545,7 +1615,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > } > > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > -- > 1.7.9.5 > > Comments welcome, based on discussion we will send patches. > > --Jeeja

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Re: [alsa-devel] [PATCH v2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 02 Mar '16

02 Mar '16

It looks like you replied me off the maillist. So I just add the maillist and related TO/CCs back. On Tue, Mar 01, 2016 at 05:43:22AM +0000, Zidan Wang wrote: > On Fri, Feb 26, 2016 at 11:42:42AM +0800, Zidan Wang wrote: > > This is the initial imx-wm8958 device-tree-only machine driver working > > with fsl_sai driver. This sound card has three dai link, HIFI, VOICE > > and BT dai. HIFI dai link will support codec master and slave mode. > > VOICE an BT dai link have dummy cpu dai, and just support codec master > > mode. > > Why VOICE and BT are using dummy cpu dai? Where can I find the schematics > for wm8958 from NXP website? > [Wang Zidan-B50113] I have attached the schematics for wm8958. In our > release, we can't support VOICE and BT, and just using AIF1. But in > order to upstream, I add the voice and bt dai. The way to support the other two AIFs are not very graceful -- looks like you will have to hard code the CPU DAIs in the driver. I personally suggest you to use fsl-asoc-card. Since the Codec IC provides three DAIs. And in the schematics, you do have three DAI links. It sounds plausible to me that we might treat these three DAI links as three different cards. Then use different compatible names: "imx-audio-wm8958-aif1" and so on. And each one can also has ASRC back-end support. If you really have big trouble to integrate it into fsl-asoc-card (not 100% sure if the 3 cards methodology is totally feasible), I will be okay with a dedicated driver for wm8958. But you will need to refine the DT binding to support three CPU DAIs in my opinion. > > +static u32 imx_wm8958_rates[] = { 8000, 16000, 32000, 48000 }; > > + > > +static struct snd_pcm_hw_constraint_list imx_wm8958_rate_constraints = { > > + .count = ARRAY_SIZE(imx_wm8958_rates), > > + .list = imx_wm8958_rates, > > +}; > > Where are these rate constraints coming from? > [Wang Zidan-B50113] these rates are restricted by the SAI MCLK. If the SAI MCLK runs at a different rate, it would be invalid. So there should be at least two constraints depending on the MCLK rate. And I think it might be better to have this in the SAI driver as a refinement since the current driver only errors out for those unsupported rates. > > + if (id == HIFI_DAI) { > > + /* > > + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 > > + * using shared LRCLK from DACLRCK1. > > + */ > > + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); > > I don't see any reverse operation against this GPIO configuration here. > So is it really an operation that has to be done every hw_params()? > > > + } else if (id == VOICE_DAI) { > > + /* > > + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 > > + * using shared LRCLK from DACLRCK2. > > + */ > > + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); > > Same here. Actually I just took a look at the DT binding doc of WM8994. It supports GPIO configurations in the codec DT nodes. So you may not need to do these over here at all. > > +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, > > + struct snd_soc_dapm_context *dapm, > > + enum snd_soc_bias_level level) { > > + switch (level) { > > + case SND_SOC_BIAS_OFF: > > + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) > > + for (i = 0; i < WM8958_MCLK_MAX; i++) { > > + if (!IS_ERR(data->mclk[i])) > > + clk_disable_unprepare(data->mclk[i]); > > The set_bias() funcs are merely en/disabling mclk. > Would be possible to do it in the codec driver? > [Wang Zidan-B50113] It can be moved to codec driver, > but the wm8994 driver can support wm8958 wm8994 wm1811 codec, > it's hard for me to do like that, and I also can't test them. Sending a patch to the maillist is a way to let people be aware of the change and then test it. The driver maintainer will review it and give you feedback. If it really causes compatible issues, you can then limit the code for wm8958 only. > > +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { > > + [BT_DAI] = { > > + .name = "Bluetooth", > > + .stream_name = "Bluetooth", > > + .cpu_dai_name = "snd-soc-dummy-dai", > > + .codec_name = "wm8994-codec", > > + .codec_dai_name = "wm8994-aif3", > > + .platform_name = "snd-soc-dummy", > > + .ignore_pmdown_time = 1, > > + }, > > A little curious...how is the BT gonna work for you? > [Wang Zidan-B50113] I have not test the BT dai. The BT is directly > connect to AIF3, when AIF3 receive date from BT, it will route the > date to AIF1 or AIF2. And the date from AIF1 and AIF2 can also route > to AIF3 to BT. So we just need to set the amixer setting to open the > audio route. Sounds interesting. So the Codec totally becomes a router. If this design can not be supported by using three cards via the fsl-asoc-card, you may continue this driver as I can see the situation here. But I encourage you to try that first so as to avoid duplicated code/driver.

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[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by Dennis Kadioglu 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ ---

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[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by denk 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ --- Signed-off-by: Dennis Kadioglu <denk(a)post.com>

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[alsa-devel] [PATCH 1/1] alsabat: truncate sample frames for faster FFT analysis
by han.lu＠intel.com 01 Mar '16

01 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Truncate the sample frames to powers of 2, since the FFTW algorithm runs especially fast in this case, and other sizes may be computed by means of a slow, general-purpose algorithm. In my test environment applying the patch, a sound clip of 33072 frames is cut off to 32768 frames before analysis, and the time cost is reduced from 6.128s to 0.224s. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/analyze.c b/bat/analyze.c index 60e2d1c..5cfdac3 100644 --- a/bat/analyze.c +++ b/bat/analyze.c @@ -256,6 +256,20 @@ static int reorder_data(struct bat *bat) return 0; } +/* truncate sample frames for faster FFT analysis process */ +static int truncate_frames(struct bat *bat) +{ + int shift = SHIFT_MAX; + + for (; shift > SHIFT_MIN; shift--) + if (bat->frames & (1 << shift)) { + bat->frames = 1 << shift; + return 0; + } + + return -EINVAL; +} + int analyze_capture(struct bat *bat) { int err = 0; @@ -263,6 +277,13 @@ int analyze_capture(struct bat *bat) int c; struct analyze a; + err = truncate_frames(bat); + if (err < 0) { + fprintf(bat->err, _("Invalid frame number for analysis: %d\n"), + bat->frames); + return err; + } + fprintf(bat->log, _("\nBAT analysis: signal has %d frames at %d Hz,"), bat->frames, bat->rate); fprintf(bat->log, _(" %d channels, %d bytes per sample.\n"), diff --git a/bat/common.h b/bat/common.h index b8f6670..ff03fc1 100644 --- a/bat/common.h +++ b/bat/common.h @@ -68,6 +68,12 @@ #define FOUND_DC (1<<1) #define FOUND_WRONG_PEAK (1<<0) +/* Truncate sample frames to (1 << N), for faster FFT analysis process. The + * valid range of N is (SHIFT_MIN, SHIFT_MAX). When N increases, the analysis + * will be more time-consuming, and the result will be more accurate. */ +#define SHIFT_MAX (sizeof(int) * 8 - 2) +#define SHIFT_MIN 8 + struct wav_header { unsigned int magic; /* 'RIFF' */ unsigned int length; /* file len */ -- 2.5.0

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[alsa-devel] [PATCH] ALSA: hda - hdmi_find_pcm_slot return value bug fix
by libin.yang＠linux.intel.com 01 Mar '16

01 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> hdmi_find_pcm_slot return -EBUSY when not no pcm slot found, not -ENODEV. So the caller should compare with -EBUSY. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d77509a..aa2d6a2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1758,7 +1758,7 @@ static void hdmi_attach_hda_pcm(struct hdmi_spec *spec, if (per_pin->pcm) return; idx = hdmi_find_pcm_slot(spec, per_pin); - if (idx == -ENODEV) + if (idx == -EBUSY) return; per_pin->pcm_idx = idx; per_pin->pcm = get_hdmi_pcm(spec, idx); -- 1.9.1

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[alsa-devel] [PATCH] teac ud-501/ud-503 usb delay
by Guillaume Fougnies 01 Mar '16

01 Mar '16

Hello, This is a small patch against sound/usb/quirks.c to correct "Teac Corp." products delay. Tested on Teac UD-501 and UD-503 products, could not harm other Teac products. It corrects the "clock source 41 is not valid, cannot use" error for both devices. And more critically on the new UD-503, a complete freeze of the teac usb interface when switching between different rate/format. (Freeze only solved by a power switch of the device...) The patch matches the problem of the "Playback Design" products. I just refactored a bit the code to include Teac products properly. Regards, Guillaume --- sound/usb/quirks.c | 58 +++++++++++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 23ea6d8..f3faaef 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1202,40 +1202,48 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) void snd_usb_set_interface_quirk(struct usb_device *dev) { /* - * "Playback Design" products need a 50ms delay after setting the + * Products needing a 50ms delay after setting the * USB interface. */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + switch (le16_to_cpu(dev->descriptor.idVendor)) { + case 0x23ba: /* Playback Design */ + case 0x0644: /* TEAC Corp. */ + mdelay(50); + } } void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) { - /* - * "Playback Design" products need a 20ms delay after each - * class compliant request - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); - - /* Marantz/Denon devices with USB DAC functionality need a delay - * after each class compliant request - */ - if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct))) - && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + if ((requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + switch (le16_to_cpu(dev->descriptor.idVendor)) { + /* + * "Playback Design"/"TEAC Corp." products need a 20ms delay after each + * class compliant request + */ + case 0x23ba: /* "Playback Design" */ + case 0x0644: /* TEAC Corp. */ + mdelay(20); + return; + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + case 0x1686: + /* restrict to 0x00dd */ + if (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) + mdelay(1); + return; + } - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && - (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)))) + mdelay(20); + } } /* -- 2.6.4

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[alsa-devel] Applied "ASoC: intel: remove unused variable" to the asoc tree
by Mark Brown 01 Mar '16

01 Mar '16

The patch ASoC: intel: remove unused variable has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dbc77eed6553362e08c1874ff30df640c96eca92 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> Date: Mon, 29 Feb 2016 22:53:46 +0530 Subject: [PATCH] ASoC: intel: remove unused variable The variable cmd_id is only assigned some value and is never used. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3dc7358828b3..8afa6fe7b0b0 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -318,7 +318,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, union ipc_header_high msg_high; u32 msg_low; struct ipc_dsp_hdr *dsp_hdr; - unsigned int cmd_id; msg_high = msg->mrfld_header.p.header_high; msg_low = msg->mrfld_header.p.header_low_payload; @@ -357,7 +356,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, return; /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; - cmd_id = dsp_hdr->cmd_id; dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, msg_high.part.drv_id, -- 2.7.0

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[alsa-devel] [PATCH] ASoC: dapm: Fix ctl value accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

snd_soc_dapm_dai_link_get() and _put() access the associated ctl values as value.integer.value[]. However, this is an enum ctl, and it has to be accessed via value.enumerated.item[]. The former is long while the latter is unsigned int, so they don't align. Fixes: c66150824b8a ('ASoC: dapm: add code to configure dai link parameters') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d3707987900..581175a51ecf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3573,7 +3573,7 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = w->params_select; + ucontrol->value.enumerated.item[0] = w->params_select; return 0; } @@ -3587,13 +3587,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, if (w->power) return -EBUSY; - if (ucontrol->value.integer.value[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == w->params_select) return 0; - if (ucontrol->value.integer.value[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= w->num_params) return -EINVAL; - w->params_select = ucontrol->value.integer.value[0]; + w->params_select = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

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