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

[alsa-devel] SoundBlaster X-fi HD SPDIF fix
by Richard Körber 05 Apr '15

05 Apr '15

Hi! I own a Creative USB SoundBlaster X-fi HD and tried to use its SPDIF optical output. However it didn't work with ALSA out of the box. The reason is that for SPDIF output device #1 must be used. The fix is: In /usr/share/alsa/cards/USB-Audio.conf, change the iec958_device block like this: USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 } Digital output works fine for me after that. BTW, I tried this line in /etc/asound.conf cards.USB-Audio.pcm.iec958_device."USB Sound Blaster HD" 1 but it didn't have any effect. Why was that? Thank you! Regards Richard Körber

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Re: [alsa-devel] ASoC: Intel: mrfld- add ACPI module
by Dan Carpenter 18 Mar '15

18 Mar '15

Hello Vinod Koul, The patch 336cfbb05edf: "ASoC: Intel: mrfld- add ACPI module" from Nov 11, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_acpi.c:313 sst_acpi_probe() info: ignoring unreachable code. sound/soc/intel/sst/sst_acpi.c 304 ret = sst_context_init(ctx); 305 if (ret < 0) 306 return ret; 307 308 /* need to save shim registers in BYT */ 309 ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), 310 GFP_KERNEL); 311 if (!ctx->shim_regs64) { 312 return -ENOMEM; ^^^^^^^^^^^^^^^ 313 goto do_sst_cleanup; ^^^^^^^^^^^^^^^^^^^^ 314 } 315 316 sst_configure_runtime_pm(ctx); 317 platform_set_drvdata(pdev, ctx); 318 return ret; 319 320 do_sst_cleanup: 321 sst_context_cleanup(ctx); 322 platform_set_drvdata(pdev, NULL); 323 dev_err(ctx->dev, "failed with %d\n", ret); 324 return ret; 325 } regards, dan carpenter

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[alsa-devel] [PATCH 0/20] ASoC: rsnd: add Synchronous SRC and U/O error salvage support
by Kuninori Morimoto 24 Feb '15

24 Feb '15

Hi Mark. These patches adds [1] Synchronous SRC, [2] under/over run error salvage support to Renesas sound driver. 1) - 16) : prepare cleanup/tidyup for 17) - 19) 17) : Synchronous SRC 18)- 19) : salvage support for under/over run error 20) : cleanup for 18) - 19) Current Renesas sound can use Asynchronous/Synchronous sampling rate convert. Asynchronous sampling rate convert is already supported on upstream. ("feature" is supported, "DT" bindings is not yet supported) 17) adds Synchronous sampling rate convert. Renesas sound has HW bug (?) which switches L/R channel if under/over run error had occurred. 18) - 19) adds salvage supported for it. 20) is cleanup patch which is based on 18)-19) patch Kuninori Morimoto (20): 1. ASoC: rsnd: add .fallback callback 2. ASoC: rsnd: add callback status check method 3. ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily 4. ASoC: rsnd: tidyup PIO/DMA mode settings method 5. ASoC: rsnd: tidyup SSI interrupt enable/disable method 6. ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() 7. ASoC: rsnd: clear status register when HW start 8. ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode 9. ASoC: rsnd: show master clock rate when ADG probe 10. ASoC: rsnd: move snd_kcontrol_new fucntions to core.c 11. ASoC: rsnd: tidyup rsnd_io_to_runtime() macro 12. ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() 13. ASoC: rsnd: initialize SRC on rsnd_src_init() 14. ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() 15. ASoC: rsnd: share SSI starting method between PIO/DMA mode 16. ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() 17. ASoC: rsnd: add Synchronous SRC mode 18. ASoC: rsnd: add salvage support for under/over flow error on SSI 19. ASoC: rsnd: add salvage support for under/over flow error on SRC 20. ASoC: rsnd: rename SSI function name of PIO include/sound/rcar_snd.h | 9 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 166 +++++++++++++++++++- sound/soc/sh/rcar/dvc.c | 177 ++-------------------- sound/soc/sh/rcar/gen.c | 15 ++ sound/soc/sh/rcar/rsnd.h | 94 +++++++++++- sound/soc/sh/rcar/src.c | 378 +++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/ssi.c | 205 +++++++++++++++---------- 8 files changed, 736 insertions(+), 310 deletions(-)

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[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table
by Cyrus Harmon 03 Feb '15

03 Feb '15

Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

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[alsa-devel] [PATCH 0/3] ARM: edma: Correct header file usage
by Peter Ujfalusi 26 Jan '15

26 Jan '15

Hi, The linux/platform_data/edma.h file was used for API definition as well, which is not correct since the header should only contain platform data related structures, defines, etc. Mark: I think this series can go via arm since it is just changing include stuff under ASoC. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-evm: Do not include edma headers ARM: edma: Rename header file for dmaengine filter function definition ARM: edma: Split up header file to platform_data and API file arch/arm/common/edma.c | 3 +- arch/arm/mach-davinci/devices.c | 1 + arch/arm/mach-davinci/include/mach/da8xx.h | 1 + drivers/dma/edma.c | 2 +- drivers/mmc/host/davinci_mmc.c | 3 +- drivers/spi/spi-davinci.c | 2 +- include/linux/edma-dmaengine.h | 29 ++++++ include/linux/edma.h | 152 ++++++++++++++++++++++++++--- include/linux/platform_data/edma.h | 148 ++-------------------------- sound/soc/davinci/davinci-evm.c | 3 - sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/edma-pcm.c | 2 +- 12 files changed, 182 insertions(+), 165 deletions(-) create mode 100644 include/linux/edma-dmaengine.h -- 2.1.3

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[alsa-devel] [PATCH 1/2] ALSA: firewire-lib: remove rx_blocks_for_midi quirk
by Clemens Ladisch 16 Jan '15

16 Jan '15

There are several devices that expect to receive MIDI data only in the first eight data blocks of a packet. If the driver restricts the data rate to the allowed rate (as mandated by the specification, but not yet implemented by this driver), this happens naturally. Therefore, there is no reason to ever try to use more data packets with any device. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/firewire/amdtp.c | 10 +++++++--- sound/firewire/amdtp.h | 3 --- sound/firewire/bebob/bebob_stream.c | 7 ------- sound/firewire/fireworks/fireworks_stream.c | 5 ----- 4 files changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 3badc70..17548d8 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -21,6 +21,12 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* + * Several devices look only at the first eight data blocks. + * In any case, this is more than enough for the MIDI data rate. + */ +#define MAX_MIDI_RX_BLOCKS 8 + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 Âµs */ /* isochronous header parameters */ @@ -78,8 +84,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->callbacked = false; s->sync_slave = NULL; - s->rx_blocks_for_midi = UINT_MAX; - return 0; } EXPORT_SYMBOL(amdtp_stream_init); @@ -474,7 +478,7 @@ static void amdtp_fill_midi(struct amdtp_stream *s, b = (u8 *)&buffer[s->midi_position]; port = (s->data_block_counter + f) % 8; - if ((f >= s->rx_blocks_for_midi) || + if ((f >= MAX_MIDI_RX_BLOCKS) || (s->midi[port] == NULL) || (snd_rawmidi_transmit(s->midi[port], b + 1, 1) <= 0)) b[0] = 0x80; diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index e6e8926..cd4c4df 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -152,9 +152,6 @@ struct amdtp_stream { /* quirk: fixed interval of dbc between previos/current packets. */ unsigned int tx_dbc_interval; - /* quirk: the first count of data blocks in an rx packet for MIDI */ - unsigned int rx_blocks_for_midi; - bool callbacked; wait_queue_head_t callback_wait; struct amdtp_stream *sync_slave; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 1aab0a32..0ebcabf 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,13 +484,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) amdtp_stream_destroy(&bebob->rx_stream); destroy_both_connections(bebob); } - /* - * The firmware for these devices ignore MIDI messages in more than - * first 8 data blocks of an received AMDTP packet. - */ - if (bebob->spec == &maudio_fw410_spec || - bebob->spec == &maudio_special_spec) - bebob->rx_stream.rx_blocks_for_midi = 8; end: return err; } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index b985fc5..4f440e1 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -179,11 +179,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) destroy_stream(efw, &efw->tx_stream); goto end; } - /* - * Fireworks ignores MIDI messages in more than first 8 data - * blocks of an received AMDTP packet. - */ - efw->rx_stream.rx_blocks_for_midi = 8; /* set IEC61883 compliant mode (actually not fully compliant...) */ err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration
by Peter Ujfalusi 14 Jan '15

14 Jan '15

In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha <jsarha(a)ti.com> (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- 2.1.0

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[alsa-devel] "ak" vs "asahi-kasei" (was: Re: [PATCH 3/3 v3] ARM: shmobile: lager: add sound support on DT)
by Geert Uytterhoeven 13 Jan '15

13 Jan '15

Hi Morimoto-san, CC devicetree, alsa-devel On Wed, Jun 11, 2014 at 9:40 AM, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: >> On Wed, Jun 11, 2014 at 8:54 AM, Kuninori Morimoto >> <kuninori.morimoto.gx(a)renesas.com> wrote: >> > + compatible = "asahi-kasei,ak4643"; >> >> As Sergei already commented, >> Documentation/devicetree/bindings/vendor-prefixes.txt >> says >> >> ak Asahi Kasei Corp. > > I know, but, > > grep asahi-kasei sound/soc/codecs/* > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4554.c: { .compatible = "asahi-kasei,ak4554" }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > ak4642 driver can't probe by "ak"; That's bad. And all but one of these match entries have been added _after_ the addition of "ak" to vendor-prefixes.txt... Outside sound, there's also "asahi-kasei,ak8975" vs. "ak,ak8975". > My code is based on "code", not "document" Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Re: [alsa-devel] [FFADO-devel] [RFC] libhinawa: a light-weight I/O library for status/transactions to ALSA FireWire devices
by Jonathan Woithe 12 Jan '15

12 Jan '15

Hi Takashi S On Tue, Sep 23, 2014 at 08:39:34PM +0900, Takashi Sakamoto wrote: > For some reasons (I describe later), I start to develop a light-weight > I/O library for status/transactions to ALSA FireWire devices. > https://github.com/takaswie/libhinawa > > This library gives ways to use ALSA FireWire driver functionality such as > lock/unlock kernel streaming, EFW transaction. This is one of my > solutions to some issues between ALSA/FFADO. As I understand it, this library will be a thin wrapper around whatever kernel interface is used to communicate these details out to userspace (I think byte-type mixer controls are the preferred avenue at present). In principle I have no problem with this approach so long as it remains lightweight. The situation to avoid is the one we had with libraw1394; when the new stack came about the preferred way to interact with it switched back to the native kernel interface since (as I understand the situation) it was felt that libraw1394 didn't really provide a worthwhile API abstraction. Out of interest, is there any signficance in the library's name "hinawa"? > I hope to merge this library into alsa-tools package in future, and this > is a first RFC for it. It's under development and I require your > comments about its APIs and structure ... I will try to find some time to look at the current proposal in the next week or so. > Dice notification is an actual example. Dice based devices transfer > notification to a specific address on host controller. The address space > is exclusive resources on an system. For Dice notification, ALSA Dice > driver gives a way to utilize it for applications As well as Fireworks > situation, some userspace stuffs are needed to use this functionality. For sure. One thing to keep in mind is that while this is an issue for DICE devices it may not apply to others. The library API should be structured to keep in mind that not all features will be needed across the board. We don't want to burden all subsequent firewire-audio streaming drivers with baggage which is only required for a small subset. Regards jonathan

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[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

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