August 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Choosing the sysclk in simple-card looks broken to me.
by jonsmirl＠gmail.com 11 Aug '14

11 Aug '14

The problem is in asoc_simple_card_sub_parse_of() in the else case.. } else { clk = of_clk_get(node, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } This is picking up the input clocks to the devices. On the cpu-dai you want to pick up the ouput mclk (my first input clock is an 80Mhz bus clock). My IIS output clock is: clock-output-names = "mclk0"; Setting clocks="" doesn't work either, it also picks up the input clock. if (of_property_read_bool(np, "clocks")) { clk = of_clk_get(np, 0); if (IS_ERR(clk)) { ret = PTR_ERR(clk); return ret; When the codec-dai bit runs it finds the input clock on the codec, which is really my cpu-dai's output clock and it gets the correct value. But then simple goes and tries to set both system clocks - my 80Mhz bus clock and the 22.5 true sysclk. And my cpu-dai barfs when it is passed a 80Mhz sysclk. How to fix? Is anyone using this feature? This needs to be fixed to pick up output clocks, not input ones. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH 2/2] mfd: arizona: Update DT binding to support INn_MODE
by Inha Song 11 Aug '14

11 Aug '14

This patch update DT binding to support INn_MODE init_data Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- Documentation/devicetree/bindings/mfd/arizona.txt | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/Documentation/devicetree/bindings/mfd/arizona.txt b/Documentation/devicetree/bindings/mfd/arizona.txt index 5c7e723..c21a30e 100644 --- a/Documentation/devicetree/bindings/mfd/arizona.txt +++ b/Documentation/devicetree/bindings/mfd/arizona.txt @@ -42,6 +42,11 @@ Optional properties: the chip default will be used. If present exactly five values must be specified. + - wlf,inmode : A list of INn_MODE register values. If absent, INn_MODE register + set to 0 by default. The size of this list depends on ARIZONA_MAX_INPUT values. + ARIZONA_MAX_INPUT is defined in include/linux/mfd/arizona/pdata.h. If present, + exactly ARIZONA_MAX_INPUT values must be specified. + - DCVDD-supply, MICVDD-supply : Power supplies, only need to be specified if they are being externally supplied. As covered in Documentation/devicetree/bindings/regulator/regulator.txt -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH alsa-lib 1/2] Sync include/sound/asound.h with 3.17-rc1 kernel
by Takashi Iwai 11 Aug '14

11 Aug '14

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/asound.h | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 519452466317..32168f7ffce3 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -93,9 +93,12 @@ enum { SNDRV_HWDEP_IFACE_SB_RC, /* SB Extigy/Audigy2NX remote control */ SNDRV_HWDEP_IFACE_HDA, /* HD-audio */ SNDRV_HWDEP_IFACE_USB_STREAM, /* direct access to usb stream */ + SNDRV_HWDEP_IFACE_FW_DICE, /* TC DICE FireWire device */ + SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ + SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_USB_STREAM + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_BEBOB }; struct snd_hwdep_info { @@ -214,8 +217,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* 8 1-bit samples in 1 byte */ -#define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* 16 1-bit samples in 2 bytes */ +#define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ +#define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ #define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE #ifdef SNDRV_LITTLE_ENDIAN @@ -461,7 +464,7 @@ struct snd_xfern { enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; @@ -820,6 +823,8 @@ typedef int __bitwise snd_ctl_elem_iface_t; #define SNDRV_CTL_POWER_D3hot (SNDRV_CTL_POWER_D3|0x0000) /* Off, with power */ #define SNDRV_CTL_POWER_D3cold (SNDRV_CTL_POWER_D3|0x0001) /* Off, without power */ +#define SNDRV_CTL_ELEM_ID_NAME_MAXLEN 44 + struct snd_ctl_elem_id { unsigned int numid; /* numeric identifier, zero = invalid */ snd_ctl_elem_iface_t iface; /* interface identifier */ -- 2.0.4

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[alsa-devel] [PATCH] Fix DSD formats userland usability
by Jurgen Kramer 11 Aug '14

11 Aug '14

Support for DSD sample formats has been added a while ago. This patch makes those sample formats beter usable from userland (e.g. aplay). --- include/sound/asound.h | 4 +++- src/pcm/pcm_misc.c | 4 ++++ 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index c819df4..5194524 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* 8 1-bit samples in 1 byte */ +#define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* 16 1-bit samples in 2 bytes */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index d52160c..44bb89c 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -195,11 +195,13 @@ int snd_pcm_format_width(snd_pcm_format_t format) switch (format) { case SNDRV_PCM_FORMAT_S8: case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_DSD_U8: return 8; case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U16_LE: return 16; case SNDRV_PCM_FORMAT_S18_3LE: case SNDRV_PCM_FORMAT_S18_3BE: @@ -253,11 +255,13 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) switch (format) { case SNDRV_PCM_FORMAT_S8: case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_DSD_U8: return 8; case SNDRV_PCM_FORMAT_S16_LE: case SNDRV_PCM_FORMAT_S16_BE: case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U16_LE: return 16; case SNDRV_PCM_FORMAT_S18_3LE: case SNDRV_PCM_FORMAT_S18_3BE: -- 1.9.3

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[alsa-devel] Disabled jacks on motherboard 'Gigabyte G1 Sniper Z87' (CA0132-based)
by Swen Wenzel 11 Aug '14

11 Aug '14

Hello guys, I own the motherboard mentioned in the subject and experienced problems when I tried to get it to work. After some conversation <http://pastebin.com/HnmxuU1Q> in IRC we found out, that probably the jack I used was deactivated. I had to use the lineout jack instead of the headphone/speaker jack. However the headphone/speaker jack could be enabled to give the same output as the lineout jack with 'hda-verb'. Here is my alsa information: http://www.alsa-project.org/db/?f=8f57f0b72d65046469b8b6ce8a57ff6fe0371769 Maybe you can have a look at it. You can also always contact me if you have further questions or want me to try something out. I am also a programmer, and studied something related to computer science so you don't have to fear completely useless answers or too dumb questions. :D kind regards Swen

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[alsa-devel] [AMD OSS Driver Bug] Only 2 Channel Audio Detected on HDMI
by Mathew Nichols 10 Aug '14

10 Aug '14

Hello, Using Ubuntu Mini Kernel v.3.15.3-generic (OSS drivers) I cannot pass 7.1 LPCM and cannot bitstream HD Audio formats. XBMC only detects 2 channels. The xbmc.log only reports m_channel=FL,FR. I can also bitstream AC3 / DTS. I have worked closely with fritsch. We believe this is a kernel audio error with my GPU (AMD 5670.) The setup is dual boot and the Windows partition works correctly, as does the older Ubuntu kernel with AMD proprietary drivers. Debug information below. • Kernel log (dmesg with drm.debug=0xe):http://paste.ubuntu.com/7811317 • Driver Codec: http://paste.ubuntu.com/7811321 • xbmc.log: http://paste.ubuntu.com/7805000 Using Ubuntu Mini Kernel v.3.13.0-32-generic (AMD proprietary drivers 14.4) I CAN pass 7.1 LPCM and I can then bitstream HD Audio formats. Windows 7 also detects 7.1 LPCM @ 192khz and will bitstream all HD Audio formats. The xbmc.log reports all 8 channels for m_channels on both Ubuntu Mini and Windows 7. Debug information below. • Kernel log (dmesg with drm.debug=0xe):http://paste.ubuntu.com/7811473 • Driver Codec: http://paste.ubuntu.com/7811478 • xbmc.log: http://paste.ubuntu.com/7811488 Windows 7 with (AMD proprietary drivers 13.6): • xbmc.log: http:// <http://pastebin.com/HkFPpCiR>pastebin.com <http://pastebin.com/HkFPpCiR>/ <http://pastebin.com/HkFPpCiR>HkFPpCiR <http://pastebin.com/HkFPpCiR> Thank you for any help you can provide!

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[alsa-devel] [PATCH] ALSA: hda/ca0132 - Don't try loading firmware at resume when already failed
by Takashi Iwai 10 Aug '14

10 Aug '14

CA0132 driver tries to reload the firmware at resume. Usually this works since the firmware loader core caches the firmware contents by itself. However, if the driver failed to load the firmwares (e.g. missing files), reloading the firmware at resume goes through the actual file loading code path, and triggers a kernel WARNING like: WARNING: CPU: 10 PID:11371 at drivers/base/firmware_class.c:1105 _request_firmware+0x9ab/0x9d0() For avoiding this situation, this patch makes CA0132 skipping the f/w loading at resume when it failed at probe time. Reported-and-tested-by: Janek Kozicki <cosurgi(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 4f3aba78f720..5d8455e2dacd 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4376,6 +4376,9 @@ static void ca0132_download_dsp(struct hda_codec *codec) return; /* NOP */ #endif + if (spec->dsp_state == DSP_DOWNLOAD_FAILED) + return; /* don't retry failures */ + chipio_enable_clocks(codec); spec->dsp_state = DSP_DOWNLOADING; if (!ca0132_download_dsp_images(codec)) @@ -4552,7 +4555,8 @@ static int ca0132_init(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - spec->dsp_state = DSP_DOWNLOAD_INIT; + if (spec->dsp_state != DSP_DOWNLOAD_FAILED) + spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; snd_hda_power_up(codec); @@ -4663,6 +4667,7 @@ static int patch_ca0132(struct hda_codec *codec) codec->spec = spec; spec->codec = codec; + spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; spec->mixers[0] = ca0132_mixer; -- 2.0.4

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[alsa-devel] [PATCH] ALSA: usb-audio: fix BOSS ME-25 MIDI regression
by Clemens Ladisch 10 Aug '14

10 Aug '14

The BOSS ME-25 turns out not to have any useful descriptors in its MIDI interface, so its needs a quirk entry after all. Reported-and-tested-by: Kees van Veen <kees.vanveen(a)gmail.com> Fixes: 8e5ced83dd1c ("ALSA: usb-audio: remove superfluous Roland quirks") Cc: <stable(a)vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/usb/quirks-table.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..223c47b 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1581,6 +1581,35 @@ } }, { + /* BOSS ME-25 */ + USB_DEVICE(0x0582, 0x0113), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ /* only 44.1 kHz works at the moment */ USB_DEVICE(0x0582, 0x0120), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {

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[alsa-devel] [PATCH 0/2] ALSA: hda - More pop noise fixes for Dell XPS 13 9333
by Gabriele Mazzotta 10 Aug '14

10 Aug '14

Fix the pop noises that happen when the laptop is booting and when it's rebooting. Gabriele Mazzotta (2): ALSA: hda - Set internal mic as default input source on Dell XPS 13 9333 ALSA: hda - Fix pop noises on reboot for Dell XPS 13 9333 sound/pci/hda/patch_realtek.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) -- 2.1.0.rc1

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[alsa-devel] fast additive copy and morph functions
by Joël Krähemann 10 Aug '14

10 Aug '14

Does someone how to do fast additive copy and morph functions? This is the header of the corresponding *.c file https://sourceforge.net/p/ags/code/HEAD/tree/src/ags/audio/ags_audio_signal… https://sourceforge.net/p/ags/code/HEAD/tree/src/ags/audio/ags_audio_signal… Your contribution will be noted in these files! best regards Joël

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