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

Re: [alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Johan Aires Rastén 11 Sep '15

11 Sep '15

Hi Piotr, great to see someone else also working on getting the U7 working! Unfortunately I don't have the answer you're looking for as I don't know that much about ALSA actually. Most of what I've done was just comparing the UAC2 specification with what the driver does, and some light USB protocol sniffing. I'm not actively working on the driver since I bought a new sound card for my PC and am using the U7 with my Windows laptop, but send me a message on Google Hangouts if you think I can help with something. When I connect the U7 it starts with "mic" as input, but I have to switch it to line and back to mic before it works. Does this happen to you too? //Johan On 10 September 2015 at 16:47, Piotr SQ9FK <sq9fk(a)wp.pl> wrote: > Hello Takashi,Johan > My patches are unessesary since following last patch was applied to the git: > ALSA: usb-audio: Change internal PCM order > > Capture working but there are no volume sliders for Line-In Capture or > Mic-In Capture (I can only toggle on off). Probably the RANGES are > broken?. From ASUS Windows driver I can see following ranges: > > Master Volume: (Phisycal Knob on Asus) > from 100 to 1 correspond 0dB to -70dB, 0 is -127dB > Range from 1 to 100 -> 0dB to -70dB (logarythmic) > > Mic-In: > from 100 to 1 correspond 30dB to -10dB, 0 is -192dB > Range from 1 to 10 -> -10dB to 10dB (linear?) > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Line-In: > from 100 to 1 correspond 0dB to -40dB, 0 is -192dB > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Actual Situation: > Input Gain Pad Control - 3 steps [0 - 50 - 100] > Double Slider. In Capture and Playback Categories, both of them are > controlled in the same time. > Controls Playback Volume only, Capture not. > > "Mic" and "Line" Switches [on/off] > PCM Cature Source [Line/Mic] > Electret Mic doesn't work. With all possible configurations - Silent. > When I pluged Line level signal, I can record the signal without problems. > PCM Capture Source [Line] - Line [on] Mic [off] - Sound > PCM Capture Source [Line] - Line [on] Mic [on] - Sound > PCM Capture Source [Line] - Line [off] Mic [off] - No Sound > PCM Capture Source [Line] - Line [off] Mic [on] - No Sound > The same with selected [Mic]. There is no Capture Volume control. > > PCM 1 [mute/on] > I think this is SPDIF switch. It doesn't work. When I switch it on -> > Blue Led on Xonar doesn't light up. > > I attached screen of alsamixer, usbmix from proc and my mixer_map from > my old patch. > I think there are is still bugs in parsing. Please point me how I can > add manually Volume control for Capture? > > Piotr Gaska > > W dniu 2015-07-27 o 14:10, Takashi Iwai pisze: > >> On Sun, 26 Jul 2015 10:00:12 +0200, >> Piotr G. wrote: >>> >>> Hello, >>> This is my first patch to alsa-devel, and I'm not very skilled >>> programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS >>> in mixer_map.c I don't know why the driver by default doesn't parse all >>> info correctly. With this patch you can see Line Switch and Mic Switch >>> in alsamixer. I can record my voice in Audacity with all supported bit >>> and sampling rates in Mono or Stereo >>> This is temporary solution. Please correct me if the patch is not >>> properly made. >>> >>> What doesn't work: >>> - Cannot change Capture Volume for Line/Mic. No mixer control in >>> alsamixer >>> - Switching between Line/Mic doesn't work (Mute doesn't work) >>> - SPDIF not tested >>> As what I wrote at beginning I'm not skilled programmer so any help >>> is >>> aprreciated >>> >>> >>> proc-usbmixer -> cat /proc/asound/*/usbmix >>> usb.debug -> lsusb -vv (only Asus Xonar U7) >> >> The changes in mixer_maps.c look good, but what's the reason for >> uac2_audio_feature_info[]? It seems adding "Control" suffix to each >> entry, which doesn't look correct. >> >> >> thanks, >> >> Takashi >> >> >>> --- >>> sound/usb/mixer.c | 23 +++++++++++++++++++++-- >>> sound/usb/mixer_maps.c | 48 >>> ++++++++++++++++++++++++++++++++++++++++++++++++ >>> 2 files changed, 69 insertions(+), 2 deletions(-) >>> >>> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c >>> index 6b3acba..027d160 100644 >>> --- a/sound/usb/mixer.c >>> +++ b/sound/usb/mixer.c >>> @@ -818,6 +818,25 @@ static struct usb_feature_control_info >>> audio_feature_info[] = { >>> { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> }; >>> >>> +static struct usb_feature_control_info uac2_audio_feature_info[] = { >>> + /* UAC2 specific */ >>> + { "Mute Control", USB_MIXER_INV_BOOLEAN }, >>> + { "Volume Control", USB_MIXER_S16 }, >>> + { "Bass Control", USB_MIXER_S8 }, >>> + { "Mid Control", USB_MIXER_S8 }, >>> + { "Treble Control", USB_MIXER_S8 }, >>> + { "Graphic Equalizer Control", USB_MIXER_S8 }, >>> + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, >>> + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in >>> UAC2 */ >>> + { "Bass Boost Control", USB_MIXER_BOOLEAN }, >>> + { "Loudness Control", USB_MIXER_BOOLEAN }, >>> + { "Input Gain Control", USB_MIXER_S16 }, >>> + { "Input Gain Pad Control", USB_MIXER_S16 }, >>> + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> + { "Underflow Control", USB_MIXER_BOOLEAN }, >>> + { "Overflow Control", USB_MIXER_BOOLEAN }, >>> +}; >>> + >>> /* private_free callback */ >>> void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) >>> { >>> @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); >>> cval->control = control; >>> cval->cmask = ctl_mask; >>> - cval->val_type = audio_feature_info[control-1].type; >>> + cval->val_type = uac2_audio_feature_info[control-1].type; >>> if (ctl_mask == 0) { >>> cval->channels = 1; /* master channel */ >>> cval->master_readonly = readonly_mask; >>> @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> break; >>> default: >>> if (!len) >>> - strlcpy(kctl->id.name, >>> audio_feature_info[control-1].name, >>> + strlcpy(kctl->id.name, >>> uac2_audio_feature_info[control-1].name, >>> sizeof(kctl->id.name)); >>> break; >>> } >>> diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c >>> index e5000da..07f510e 100644 >>> --- a/sound/usb/mixer_maps.c >>> +++ b/sound/usb/mixer_maps.c >>> @@ -341,6 +341,48 @@ static const struct usbmix_name_map >>> scms_usb3318_map[] = { >>> { 0 } >>> }; >>> >>> +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC >>> Stereo CS4398 */ >>> +/* Topology: >>> + >>> +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] >>> + >>> +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] >>> + >>> +Mic_IN[4]------->FU[16]---+ >>> + SU[20]--CS[22]--->USB_OUT[11] >>> +Line_IN[5]------>FU[17]---+ >>> + >>> +*/ >>> + >>> +static const struct usbmix_name_map xonar_u7_map[] = { >>> + /* 1: IT PCM Analog (USB Streaming) */ >>> + /* 2: IT PCM2 Digital (USB Streaming) */ >>> + /* 4: IT Microphone */ >>> + /* 5: IT Line IN */ >>> + /* 7: OT Speaker */ >>> + /* 8: OT SPDIF */ >>> + /* 11 OT Capture Playback (USB Streaming) */ >>> + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume >>> */ >>> + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ >>> + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in >>> Mute/Volume */ >>> + { 17, "Line Capture" }, /* 17 FU Capture Line-in >>> Mute/Volume */ >>> + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ >>> + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ >>> + { 20, "Capture Source" }, /* 20 SU Capture Source Selector >>> */ >>> + /* 22 CS Line-in and Mic-in Clock Source */ >>> + { 0 } >>> +}; >>> + >>> +static struct usbmix_selector_map xonar_u7_selectors[] = { >>> +{ >>> + .id = 20, >>> + .count = 2, >>> + .names = (const char*[]) {"Mic", "Line"} >>> + }, >>> + { 0 } /* terminator */ >>> + >>> +}; >>> + >>> /* >>> * Control map entries >>> */ >>> @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { >>> .id = USB_ID(0x25c4, 0x0003), >>> .map = scms_usb3318_map, >>> }, >>> + { >>> + /* Asus Xonar U7 */ >>> + .id = USB_ID(0x1043, 0x857c), >>> + .map = xonar_u7_map, >>> + .selector_map = xonar_u7_selectors, >>> + }, >>> { 0 } /* terminator */ >>> }; >>> >>> -- >>> 2.3.6 >>> >

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[alsa-devel] Mixer control appears twice in alsamixer
by James Cameron 11 Sep '15

11 Sep '15

G'day, I've a mixer control for "Analog Mic Boost" which appears twice in alsamixer --view all, and once in amixer. How do I cause it to appear only once in alsamixer? amixer shows both Playback and Capture channels for the control: -- Simple mixer control 'Analog Mic Boost',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 8 Mono: 8 [100%] -- The control is declared with: /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { TLV_DB_RANGE_HEAD(7), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), }; static const struct snd_kcontrol_new rt5631_snd_controls[] = { ... SOC_SINGLE_TLV("Analog Mic Boost", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), ... } I'm working on alc5631/rt5631 in a local branch of either 3.0 or 3.5 for the OLPC XO-1.75 and XO-4. But the declaration seems no different in 4.2, apart from the name of the control. https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… -- James Cameron http://quozl.linux.org.au/

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[alsa-devel] 'control ... is already present' message
by Ran Shalit 11 Sep '15

11 Sep '15

Hello, I'm using 2 codecs (tlv320aic3x-codec) each which both are registered into same sound card, each as a single device. After boot I have the following devices: root@dm814x:~# cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture But I get the following error messeges during boot: usbcore: registered new interface driver snd-usb-audio Registered tvp5158 audio codec PM: Adding info for platform:soc-audio.0 PM: Adding info for No Bus:TLV320AIC3X_2 asoc: tlv320aic3x-hifi <-> davinci-mcasp.2 mapping ok snd_soc_instantiate_card 7 tlv320aic3x-codec 2-0019: tlv320aic3x-codec.2-0019: Failed to add PCM Playback Volume: -16 asoc: failed to add dapm kcontrol Right HPCOM Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left HPCOM Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Right HP Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left HP Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Mono Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Right Line Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left Line Mixer Line2L Bypass Switch: -16 asoc: failed to add kcontrol Right Line2R Mux asoc: failed to add kcontrol Right Line1R Mux asoc: failed to add kcontrol Right Line1L Mux asoc: failed to add dapm kcontrol Right PGA Mixer Line1R Switch: -16 asoc: failed to add kcontrol Left Line2L Mux asoc: failed to add kcontrol Left Line1R Mux asoc: failed to add kcontrol Left Line1L Mux asoc: failed to add dapm kcontrol Left PGA Mixer Line1L Switch: -16 asoc: failed to add kcontrol Right HPCOM Mux asoc: failed to add kcontrol Right DAC Mux asoc: failed to add kcontrol Left HPCOM Mux asoc: failed to add kcontrol Left DAC Mux PM: Adding info for No Bus:TLV320AIC3X_3 asoc: tlv320aic3x-hifi <-> davinci-mcasp.3 mapping ok PM: Adding info for No Bus:card0 PM: Adding info for No Bus:pcmC0D1p PM: Adding info for No Bus:pcmC0D1c PM: Adding info for No Bus:pcmC0D0p PM: Adding info for No Bus:pcmC0D0c PM: Adding info for No Bus:controlC0 ALSA device list: #0: TI81XX SOUND0 What is the reason for the "asoc: failed to add dapm kcontrol" and the " 'control ... already present" messeges ? Is it becuase both codec are registered on same card (single card) ? 1. If I move each to be registered to its own card, will it resolve this issue ? 2. Should the following patch resolve this issue: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033593.h… Regards, Ran

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[alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Piotr G. 10 Sep '15

10 Sep '15

Hello, This is my first patch to alsa-devel, and I'm not very skilled programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS in mixer_map.c I don't know why the driver by default doesn't parse all info correctly. With this patch you can see Line Switch and Mic Switch in alsamixer. I can record my voice in Audacity with all supported bit and sampling rates in Mono or Stereo This is temporary solution. Please correct me if the patch is not properly made. What doesn't work: - Cannot change Capture Volume for Line/Mic. No mixer control in alsamixer - Switching between Line/Mic doesn't work (Mute doesn't work) - SPDIF not tested As what I wrote at beginning I'm not skilled programmer so any help is aprreciated proc-usbmixer -> cat /proc/asound/*/usbmix usb.debug -> lsusb -vv (only Asus Xonar U7) --- sound/usb/mixer.c | 23 +++++++++++++++++++++-- sound/usb/mixer_maps.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..027d160 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -818,6 +818,25 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; +static struct usb_feature_control_info uac2_audio_feature_info[] = { + /* UAC2 specific */ + { "Mute Control", USB_MIXER_INV_BOOLEAN }, + { "Volume Control", USB_MIXER_S16 }, + { "Bass Control", USB_MIXER_S8 }, + { "Mid Control", USB_MIXER_S8 }, + { "Treble Control", USB_MIXER_S8 }, + { "Graphic Equalizer Control", USB_MIXER_S8 }, + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in UAC2 */ + { "Bass Boost Control", USB_MIXER_BOOLEAN }, + { "Loudness Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16 }, + { "Input Gain Pad Control", USB_MIXER_S16 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Underflow Control", USB_MIXER_BOOLEAN }, + { "Overflow Control", USB_MIXER_BOOLEAN }, +}; + /* private_free callback */ void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + cval->val_type = uac2_audio_feature_info[control-1].type; if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, break; default: if (!len) - strlcpy(kctl->id.name, audio_feature_info[control-1].name, + strlcpy(kctl->id.name, uac2_audio_feature_info[control-1].name, sizeof(kctl->id.name)); break; } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da..07f510e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -341,6 +341,48 @@ static const struct usbmix_name_map scms_usb3318_map[] = { { 0 } }; +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC Stereo CS4398 */ +/* Topology: + +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] + +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] + +Mic_IN[4]------->FU[16]---+ + SU[20]--CS[22]--->USB_OUT[11] +Line_IN[5]------>FU[17]---+ + +*/ + +static const struct usbmix_name_map xonar_u7_map[] = { + /* 1: IT PCM Analog (USB Streaming) */ + /* 2: IT PCM2 Digital (USB Streaming) */ + /* 4: IT Microphone */ + /* 5: IT Line IN */ + /* 7: OT Speaker */ + /* 8: OT SPDIF */ + /* 11 OT Capture Playback (USB Streaming) */ + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume */ + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in Mute/Volume */ + { 17, "Line Capture" }, /* 17 FU Capture Line-in Mute/Volume */ + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ + { 20, "Capture Source" }, /* 20 SU Capture Source Selector */ + /* 22 CS Line-in and Mic-in Clock Source */ + { 0 } +}; + +static struct usbmix_selector_map xonar_u7_selectors[] = { +{ + .id = 20, + .count = 2, + .names = (const char*[]) {"Mic", "Line"} + }, + { 0 } /* terminator */ + +}; + /* * Control map entries */ @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Asus Xonar U7 */ + .id = USB_ID(0x1043, 0x857c), + .map = xonar_u7_map, + .selector_map = xonar_u7_selectors, + }, { 0 } /* terminator */ }; -- 2.3.6

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[alsa-devel] [PATCH 1/2] ASoC: wm8960: correct the min gain value of some PGA
by Zidan Wang 10 Sep '15

10 Sep '15

The min gain is the corresponding gain value when the register value is 0 instead of 1, just correct it. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c..6163474 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -211,11 +211,11 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, return wm8960_set_deemph(codec); } -static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); -static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); -static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform
by John Lin 10 Sep '15

10 Sep '15

Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4972bf3..8e51242 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3205,6 +3205,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, }, + { + .ident = "Google Ultima", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: intel: Fix SSP port configuration after RTD3 resume.
by Liam Girdwood 10 Sep '15

10 Sep '15

Currently the SSP port settings are being clobbered as part of the DSP RTD3 restore logic. make sure we save the correct params and restore them at resume. The FW sadly does not save SSP settings as part of the PM context. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f6efa9d..b27f25f 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -302,6 +302,10 @@ struct sst_hsw { struct sst_hsw_ipc_dx_reply dx; void *dx_context; dma_addr_t dx_context_paddr; + enum sst_hsw_device_id dx_dev; + enum sst_hsw_device_mclk dx_mclk; + enum sst_hsw_device_mode dx_mode; + u32 dx_clock_divider; /* boot */ wait_queue_head_t boot_wait; @@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, trace_ipc_request("set device config", dev); - config.ssp_interface = dev; - config.clock_frequency = mclk; - config.mode = mode; - config.clock_divider = clock_divider; + hsw->dx_dev = config.ssp_interface = dev; + hsw->dx_mclk = config.clock_frequency = mclk; + hsw->dx_mode = config.mode = mode; + hsw->dx_clock_divider = config.clock_divider = clock_divider; if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER) config.channels = 4; else @@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) return -EIO; } - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); + /* Set ADSP SSP port settings - sadly the FW does not store SSP port + settings as part of the PM context. */ + ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk, + hsw->dx_mode, hsw->dx_clock_divider); if (ret < 0) dev_err(dev, "error: SSP re-initialization failed\n"); -- 2.1.4

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[alsa-devel] [PATCH V2 1/1] ASoC: sti: check return of of_property_read
by Arnaud Pouliquen 10 Sep '15

10 Sep '15

Add check on of_property_read to return error when DT required property is not defined. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 14 ++++++++++---- sound/soc/sti/uniperif_reader.c | 6 +++++- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index f6eefe1..843f037 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(pnode, "version", &player->ver); - if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + if (of_property_read_u32(pnode, "version", &player->ver) || + player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; } @@ -998,10 +998,16 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - of_property_read_u32(pnode, "uniperiph-id", &info->id); + if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + dev_err(dev, "uniperipheral id not defined"); + return -EINVAL; + } /* Read the device mode property */ - of_property_read_string(pnode, "mode", &mode); + if (of_property_read_string(pnode, "mode", &mode)) { + dev_err(dev, "uniperipheral mode not defined"); + return -EINVAL; + } if (strcasecmp(mode, "hdmi") == 0) info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index c502626..f791239 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(node, "version", &reader->ver); + if (of_property_read_u32(node, "version", &reader->ver) || + reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + dev_err(&pdev->dev, "Unknown uniperipheral version "); + return -EINVAL; + } /* Save the info structure */ reader->info = info; -- 1.9.1

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Re: [alsa-devel] [PATCH] ASoC: make dapm cache search depth configurable
by Mark Brown 10 Sep '15

10 Sep '15

On Thu, Sep 10, 2015 at 10:42:50AM +0100, Nikesh wrote: > Hi Mark, As I'm fairly sure I've had to tell you before don't top post. This means that when someone sees something like this: > The reason I though this will be usefull is as below: while reading they know what the "this" you are talking about is. > the widgets in wm5110_dapm_widgets[]) as defined in this macro. But if you > notice the way DSP widgets are created they use a WM_ADSP2 macro which > creates a preloader widget for every DSP widget and we want to skip past > this preloader widget when doing a cache search so increasing the cache > search depth helps here. Why is the solution to this problem to add device specific tuning? Is this extremely expensive so no other device would want it?

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[alsa-devel] [PATCH] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 10 Sep '15

10 Sep '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 56 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 11 ++++++++ 2 files changed, 58 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..a12bf57 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,8 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + for (i = 0; i < I915_MAX_PORTS; i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..11999eb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,19 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + * The 4K resolution audio issue should be fixed on CNL + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

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