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

[alsa-devel] [ANNOUNCE] Audio Mini Summit 2014, Oct. 14, Düsseldorf, Germany
by Takashi Iwai 24 Apr '15

24 Apr '15

Hi, like previous years, we're going to hold a meeting to discuss lowlevel audio on Linux. This time it's along with Linux Plumbers conference, but for avoiding the conflicts with other LPC mini conferences, it'll be held on Tuesday October 14, at the same venue as LPC, Congress Center Düsseldorf, Germany. If anyone is interested, feel free to sign up in the attendee list below. It's a public accessible Google doc, so you can edit freely. http://goo.gl/VbXLPW (or https://docs.google.com/spreadsheets/d/1wBAiAyZSnMjVFJbxdXE5JoVdKGIBIWt-4tu…) The topics that have been currently raised are found in the URL below. If you have any more topics to discuss there, just put your favorite one in the URL below as well. http://goo.gl/JSasQ9 (or https://docs.google.com/spreadsheets/d/1cXzON65QYNrTXi31AIQHffUZ9NWwJuNEJAC…) Thanks! Takashi

3 2

[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

2 8

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

2 2

[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(-)

3 40

[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

5 32

[alsa-devel] [PATCH v2 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 27 Jan '15

27 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [Add driver data to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..6e3f175 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,32 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + .compatible = "wlf,wm8918", + .data = &wm8904_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

4 9

[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

7 19

[alsa-devel] [PATCH v2 1/1] alsa-lib: Add snd_pcm_start_at.
by Tim Cussins 19 Jan '15

19 Jan '15

Add notion of TSTAMP_CLASS (SYSTEM/AUDIO) as per Pierre's suggestion. Add snd_pcm_start_at() Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..c57edca 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -323,13 +323,25 @@ typedef enum _snd_pcm_tstamp { SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_ENABLE } snd_pcm_tstamp_t; +typedef enum _snd_pcm_tstamp_class { + SND_PCM_TSTAMP_CLASS_SYSTEM = 0, + SND_PCM_TSTAMP_CLASS_AUDIO, + SND_PCM_TSTAMP_CLASS_LAST = SND_PCM_TSTAMP_CLASS_AUDIO, +} snd_pcm_tstamp_class_t; + typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /** gettimeofday equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /** monotonic_raw (no NTP) */ SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +typedef enum _snd_pcm_audio_tstamp_type { + SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SND_PCM_AUDIO_TSTAMP_TYPE_LINK, + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK, +} snd_pcm_audio_tstamp_t; + /** Unsigned frames quantity */ typedef unsigned long snd_pcm_uframes_t; /** Signed frames quantity */ @@ -478,6 +490,7 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t* start_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f23cd6..1592160 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -466,12 +466,30 @@ struct snd_xfern { }; enum { + SNDRV_PCM_TSTAMP_CLASS_SYSTEM = 0, + SNDRV_PCM_TSTAMP_CLASS_AUDIO, + SNDRV_PCM_TSTAMP_CLASS_LAST = SNDRV_PCM_TSTAMP_CLASS_AUDIO, +}; + +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, }; +enum { + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, +}; + +struct snd_start_at { + int tstamp_class; + int tstamp_type; + struct timespec start_time; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -550,6 +568,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..40a4689 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,37 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param tstamp_class specifies the class of tstamp_type + * \param tstamp_type specifies the clock with which to interpret \p start_time + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec, tstamp_class or tstamp_type is invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * The timer is unconditionally cancelled upon any \a attempt to change the stream + * state e.g. drop, drain, start, start_at. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, tstamp_class, tstamp_type, start_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c34b766..0190787 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -620,6 +620,26 @@ static int snd_pcm_hw_start(snd_pcm_t *pcm) return 0; } +static int snd_pcm_hw_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time) +{ + snd_pcm_hw_t *hw = pcm->private_data; + int err; + + struct snd_start_at start_at = { + .tstamp_class = tstamp_class, + .tstamp_type = tstamp_type, + .start_time = *start_time + }; + + sync_ptr(hw, 0); + if (ioctl(hw->fd, SNDRV_PCM_IOCTL_START_AT, &start_at) < 0) { + err = -errno; + SYSMSG("SNDRV_PCM_IOCTL_START_AT failed (%i)", err); + return err; + } + return 0; +} + static int snd_pcm_hw_drop(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; @@ -1336,6 +1356,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .prepare = snd_pcm_hw_prepare, .reset = snd_pcm_hw_reset, .start = snd_pcm_hw_start, + .start_at = snd_pcm_hw_start_at, .drop = snd_pcm_hw_drop, .drain = snd_pcm_hw_drain, .pause = snd_pcm_hw_pause, diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 394505f..5cdfd3a 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -154,6 +154,7 @@ typedef struct { int (*prepare)(snd_pcm_t *pcm); int (*reset)(snd_pcm_t *pcm); int (*start)(snd_pcm_t *pcm); + int (*start_at)(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time); int (*drop)(snd_pcm_t *pcm); int (*drain)(snd_pcm_t *pcm); int (*pause)(snd_pcm_t *pcm, int enable); -- 1.7.10.4

3 10

[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);

3 9

[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

3 6