April 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to suspend and resume all codecs on the hda bus. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error. snd_hda_suspend & snd_hda_resume will no longer be called by azx_bus_reset. So snd_hda_resume will only be used in system resume and may delay resuming a codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..a05af8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,26 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs, i.e. suspend and resume the codecs + * @bus: the HDA bus + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

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Re: [alsa-devel] [RFC] revoke(2) and generic handling of things like remove_proc_entry()
by Clemens Ladisch 08 Apr '13

08 Apr '13

Al Viro wrote: > BTW, snd_card_disconnect() doesn't do anything to existing mappings; smells > like a bug, and there we do have ones with non-trivial ->mmap(). Could > ALSA folks comment? I don't know of any hotplug sound driver that maps memory from a device. All hotplug buses (PCIe, USB, FireWire) involve DMAing from/to main memory. When a sound device is unplugged, sample transfers just stop, but the buffer itself stays as long as the device node is kept opened. (In the case of a capture device, that actually makes sense.) Regards, Clemens

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Re: [alsa-devel] [PATCH] ASoC: Ux500: remove test for undefined Kconfig macro
by Mark Brown 08 Apr '13

08 Apr '13

On Fri, Mar 29, 2013 at 01:28:21PM +0100, Paul Bolle wrote: > A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there > never was a corresponding Kconfig symbol so this test has always > evaluated to true. And since AB5500 support was removed in v3.5 it > appears safe to remove this test and a few lines of code. Applied, thanks.

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[alsa-devel] [PATCH] ALSA: hda: fix mute led on HP Envy 15
by Dan Savilonis 08 Apr '13

08 Apr '13

This laptop reports the string HP_Mute_LED_P_G in DMI, but the guess made in hp_set_mute_led() is wrong. So, hard-code the GPIO id. Signed-off-by: Dan Savilonis <djs(a)n-cube.org> --- sound/pci/hda/patch_sigmatel.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..3a6b111 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, + STAC_HP_ENVY_15, STAC_92HD83XXX_MODELS }; @@ -2092,6 +2093,15 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ } +static void stac92hd83xxx_fixup_hp_envy_15(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gpio_led = 0x01; /* GPIO0 */ +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2101,6 +2111,7 @@ static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, spec->headset_jack = 1; } + static const struct hda_fixup stac92hd83xxx_fixups[] = { [STAC_92HD83XXX_REF] = { .type = HDA_FIXUP_PINS, @@ -2169,6 +2180,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_HP_ENVY_15] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_envy_15, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2184,6 +2201,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = { { .id = STAC_92HD83XXX_HP_MIC_LED, .name = "hp-mic-led" }, { .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" }, { .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" }, + { .id = STAC_HP_ENVY_15, .name = "hp-envy-15" }, {} }; @@ -2269,6 +2287,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Pavilion dv5", STAC_92HD83XXX_HP_INV_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1688, + "HP Envy 15", STAC_HP_ENVY_15), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), {} /* terminator */ }; @@ -3968,6 +3988,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_add_verbs(codec, stac92hd83xxx_core_init); + snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, stac92hd83xxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); -- 1.8.1.2

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[alsa-devel] [PATCH 0/8] More dmaengine PCM cleanups
by Lars-Peter Clausen 07 Apr '13

07 Apr '13

Hi, The first few patches in this series are a few unrelated cleanups to different dmaengine PCM drivers and one fix for a small compile error I introduced in the last dmaengine PCM cleanup series. The last four patches introduce a common DAI DMA data struct, which will be used by tegra, omap and fsl/imx. Having a common DAI DMA data struct allows us to factor out some redundant code from these drivers into a common helper function in the dmaengine PCM library. This should hopefully be the last set of cleanups before the generic dmaengine PCM driver. - Lars Lars-Peter Clausen (8): ASoC: ep93xx_pcm: Fix compile error ASoC: tegra: Don't claim to support PCM pause and resume ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: omap: Use common DAI DMA data include/sound/dmaengine_pcm.h | 24 ++++++++++++++++++ sound/soc/atmel/atmel-pcm-dma.c | 2 -- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++------ sound/soc/fsl/imx-pcm-dma.c | 25 ++++++------------- sound/soc/fsl/imx-pcm-fiq.c | 4 +-- sound/soc/fsl/imx-pcm.h | 16 ++++-------- sound/soc/fsl/imx-ssi.c | 21 +++++++++------- sound/soc/fsl/imx-ssi.h | 7 ++++-- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 ++++++----- sound/soc/omap/mcbsp.h | 7 +++--- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 22 ++++++++--------- sound/soc/omap/omap-hdmi.c | 24 ++++++++---------- sound/soc/omap/omap-mcbsp.c | 6 ++--- sound/soc/omap/omap-mcpdm.c | 31 +++++++++++------------- sound/soc/omap/omap-pcm.c | 42 ++++---------------------------- sound/soc/omap/omap-pcm.h | 39 ------------------------------ sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 1 - sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/soc-dmaengine-pcm.c | 39 ++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 13 +++++----- sound/soc/tegra/tegra20_ac97.h | 4 +-- sound/soc/tegra/tegra20_i2s.c | 13 +++++----- sound/soc/tegra/tegra20_i2s.h | 4 +-- sound/soc/tegra/tegra20_spdif.c | 7 +++--- sound/soc/tegra/tegra20_spdif.h | 4 +-- sound/soc/tegra/tegra30_ahub.c | 8 +++--- sound/soc/tegra/tegra30_ahub.h | 8 +++--- sound/soc/tegra/tegra30_i2s.c | 13 +++++----- sound/soc/tegra/tegra30_i2s.h | 4 +-- sound/soc/tegra/tegra_pcm.c | 40 +++--------------------------- sound/soc/tegra/tegra_pcm.h | 7 ------ sound/soc/ux500/ux500_pcm.c | 51 ++++++++++++--------------------------- sound/soc/ux500/ux500_pcm.h | 14 ----------- 40 files changed, 226 insertions(+), 322 deletions(-) delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.8.0

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[alsa-devel] Samsung S5PV210 I2S DMA
by Mike Dyer 07 Apr '13

07 Apr '13

Hi All, I'm attempting to play audio into a WM8960 codec attached to I2S 0 on a Samsung S5PV210. The DMA engine emits a lot of messages complaining of (I think) a read fault: dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 ... I've had audio playing using the I2S idma driver, so I know clocks + mixer settings are OK. I'd like to get the external DMA working because I also need to do capture. Any suggestions on how to debug? Cheers, Mike

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[alsa-devel] [PATCH] ALSA: hda/realtek - Add a quirk for AC700 Chromebook.
by Dylan Reid 07 Apr '13

07 Apr '13

Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1bd2d49..145d078 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2617,7 +2617,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec, }; unsigned int cfg; - if (strcmp(codec->chip_name, "ALC271X")) + if (strcmp(codec->chip_name, "ALC271X") && + strcmp(codec->chip_name, "ALC269VB")) return; cfg = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) @@ -2838,6 +2839,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC269_FIXUP_ACER_AC700, }; static const struct hda_fixup alc269_fixups[] = { @@ -3010,6 +3012,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC269_FIXUP_ACER_AC700] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x03a11c20 }, /* mic */ + { 0x1e, 0x0346101e }, /* SPDIF1 */ + { 0x21, 0x0321101f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC271_FIXUP_DMIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3034,6 +3049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), -- 1.8.1.3.605.g02339dd

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[alsa-devel] [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
by Eldad Zack 07 Apr '13

07 Apr '13

The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident> sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/mixer_quirks.c | 4 ++-- sound/usb/quirks.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 497d274..ebe9144 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, cpu_to_le16(wIndex), + 0, wIndex, &tmp, sizeof(tmp), 1000); up_read(&mixer->chip->shutdown_rwsem); @@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cpu_to_le16(wValue), cpu_to_le16(wIndex), + wValue, wIndex, NULL, 0, 1000); up_read(&mixer->chip->shutdown_rwsem); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a2ac004..68e9220 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -499,7 +499,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, - cpu_to_le16(1), 0, NULL, 0, 1000); + 1, 0, NULL, 0, 1000); if (ret < 0) return ret; -- 1.8.1.5

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[alsa-devel] [PATCH 1/2] ALSA: add DSD formats
by Daniel Mack 07 Apr '13

07 Apr '13

This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for a copy of the documentation. The two new types added by this patch describe streams that are capable of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8 or x16 data rate, respectively). DSD itself specifies samples in *bit*, while DOP and ALSA handle them as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample rare configuration, according to the following table: configured hardware 352.8kHz 705.6KHz 1411.2KHz <---- sample rate 8-bit 2.8MHz 5.6MHz 11.2MHz 16-bit 5.6MHz 11.2MHz `-----------------------------------' actual DSD sample rates Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index aa7b0a8..d957046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -181,6 +181,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) #define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U8 _SNDRV_PCM_FMTBIT(DSD_DOP_U8) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U16 _SNDRV_PCM_FMTBIT(DSD_DOP_U16) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..eb9eda8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/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_DOP_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD DOP format (x8) */ +#define SNDRV_PCM_FORMAT_DSD_DOP_U16 ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD DOP format (x16) */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_DOP_U16 #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 578327e..578a761 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_24_1B), FORMAT(G723_40), FORMAT(G723_40_1B), + FORMAT(DSD_DOP_U8), + FORMAT(DSD_DOP_U16), }; const char *snd_pcm_format_name(snd_pcm_format_t format) -- 1.8.1.4

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Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Jussi Laako 06 Apr '13

06 Apr '13

On 04/05/2013 10:59 PM, Andreas Koch wrote: > However, they are interpreting the ASIO spec for DSD differently. They > express buffer size in samples, which means that for DSD it is measured > in bits. That's how pretty much everybody, including Sony who co-authored the spec, has been interpreting it so far... Otherwise all the latency calculations become inconsistent and need special exceptions for DSD. I agree the ASIO spec is not very clear, however it always talks about number of samples and then specifically mentions that DSD samples are _packed_ into bytes and thus all buffer sizes have to multiples of eight. I have a hidden configuration flag to flip that either way as necessary, but default is that sizes are in _bits_. - Jussi

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