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

[alsa-devel] [PATCH] Bugfix: Fix resampling when client and slave both use format float
by Maarten Baert 26 Feb '14

26 Feb '14

I found a bug in libasound. When both the client and the slave (in my case, the JACK plugin) try to use the float format, but the sample rate or channel count does not match, libasound *should* insert linear conversion plugins to convert from float to linear, then resample/remap channels, and then convert back to float (because apparently the resamplers and channel remapper don't support floating point, only 'linear' i.e. integers). Currently this doesn't work, snd_pcm_plug_change_format doesn't know what to do and simply returns EINVAL. As a result, snd_pcm_hw_params fails even though the HW params were perfectly valid (it indicates that both the float format and any sample rate are supported). In my test, this broke audio for WINE (and any other application that tries to use float, such as aplay with the right settings) when I wanted to use the JACK plugin (which only supports the float format). This patch fixes this bug by doing a conversion to S16 and back when resampling or remapping is needed. And while I was at it, I also removed a check that had no effect because the exact same check is already being done at the start of the function. I still think it's a bit silly that libasound requires integers for resampling, because both libsamplerate and libspeex use float internally for resampling. So now ALSA is literally doing a float-to-s16-to-float-to-s16-to-float conversion. But changing that would have been a lot harder. Maarten Baert

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[alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices
by Takashi Sakamoto 26 Feb '14

26 Feb '14

This series of patch is to update patches which I've posted below. [RFC v2][PATCH 00/38] Enhancement for support of some firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070424.h… [RFC][PATCH v2 00/13] speakers: Add support for capture/playback of PCM/MIDI http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071071.ht… This series may be last RFC. Repository: https://github.com/takaswie/sound For backport to Linux kernel 3.11 or later: https://github.com/takaswie/snd-firewire-improve For maintainers: I have 4 items. 1.expected impacts to user-land Refer to my last message. 2. Address allocation for Fireworks drivers Related to this suggestion: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071120.ht… The alternative way is to implement: 1.'AV/C SIGNAL FORMAT' commanids defined in 'AV/C Digital Interface Command Set General Specification Version 4.2 (Sep 2004, 1394TA)' - 'snd-firewire-lib' has these commands 2.'AV/C DESCRIPTOR' mechanism defined in 'Enhancements to the AV/C General Specification 3.0 Version 1.1 (Oct 2000, 1394TA)' 3.'AV/C SIGNAL SOURCE' commands defined in 'AV/C Music Subunit 1.0' - I don't have this specification. They cost higher than actual effects. Especially, 2 is a bit complicated and need much codes. Actually FFADO has all of them. And snd-fireworks gives hwdep interface for application to command/response. There is currently several ways in user-land so I think it good to keep the address non-relocatable yet. 3.Support for devices with fatal quirk M-Audio 'Firewire 1814' and 'ProjectMix I/O' has BeBoB firmware which is heavily customized. See last patch. They have a time not to respond transaction. As long as I experienced, the time may be 300 msec per 1 sec. As a result, snd-bebob often fail to handle them. Especially, they can ignore transactions for resource management of connection/channel/bandwidth. This is a fatal for host system. But there is no user-land driver for these devices and I received some requests from FFADO users to support them. I hope to get your opinion about this support. 4.Rest of my work I've finished almost work for them. If I receive no big problems from developers and testers, I have a plan to post merge-request. Updates since previous patches: snd-firewire-lib: - remove wrong line to regain 'stream_running' member to 'struct fwspk' - allow drivers to indicate timeout value - remove 'amdtp_stream_midi_running' snd-bebob: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - remove 'device' member fron 'struct snd_bebob' - update device entry snd-fireworks: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - remove 'device' member from 'struct snd_fireworks' snd-firewire-speakers: - rename to snd-oxfw and obsolete snd-oxfw: - add reference counter for MIDI substreams - improve 'mutex_lock()/mutex_unlock()' for streaming - fix lock mechanism of streaming by hwdep - fix wrong formation for Lacie FWSpeaker and Griffin Firewave - remove 'fw_unit_get()/fw_unit_put()' because helper function do this Confirm to work with: snd-fireworks: - Echo Audio AudioFire4 - Echo Audio AudioFire8 (by tester) - Echo Audio AudioFirePre8 - Echo Audio AudioFire12 snd-bebob: - Edirol FA-66 (by tester) - M-Audio Ozonic - M-Audio Firewire Solo - M-Audio Firewire Audiophile - M-Audio Firewire 410 - M-Audio Firewire 1814 (by tester) - PreSonus FIREBOX - PreSonus FP10 (former known as FIREPOD, by tester) - Yamaha GO44 - Yamaha GO46 snd-oxfw: - Behringer F-Control Audio 202 Takashi Sakamoto (52): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series speakers: Rename to oxfw oxfw: Move to its own directory oxfw: Split stream functionality to a new file and add a header file oxfw: Split PCM functionality to a new file oxfw: Split control functionality to a new file oxfw: Change the way to name card oxfw: Change the way to make PCM rules/constraints oxfw: Add proc interface for debugging purpose oxfw: Change the way to start stream oxfw: Add some AV/C commands to get stream formation and supported sample rate oxfw: Add support for Behringer/Mackie devices oxfw: Add support AMDTP in-stream and PCM capture oxfw: Add support for capture/playback MIDI messages oxfw: Add hwdep interface bebob: Add support for M-Audio special Firewire series include/uapi/sound/asound.h | 5 +- include/uapi/sound/firewire.h | 23 +- sound/firewire/Kconfig | 76 +- sound/firewire/Makefile | 5 +- sound/firewire/amdtp.c | 905 ++++++++++++++++------ sound/firewire/amdtp.h | 161 +++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 453 +++++++++++ sound/firewire/bebob/bebob.h | 262 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 922 +++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 178 +++++ sound/firewire/bebob/bebob_pcm.c | 419 ++++++++++ sound/firewire/bebob/bebob_proc.c | 185 +++++ sound/firewire/bebob/bebob_stream.c | 873 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 217 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 151 ++++ sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 330 ++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 299 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 186 +++++ sound/firewire/fireworks/fireworks_pcm.c | 443 +++++++++++ sound/firewire/fireworks/fireworks_proc.c | 206 +++++ sound/firewire/fireworks/fireworks_stream.c | 356 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw.c | 264 +++++++ sound/firewire/oxfw/oxfw.h | 146 ++++ sound/firewire/oxfw/oxfw_command.c | 162 ++++ sound/firewire/oxfw/oxfw_control.c | 283 +++++++ sound/firewire/oxfw/oxfw_hwdep.c | 195 +++++ sound/firewire/oxfw/oxfw_midi.c | 176 +++++ sound/firewire/oxfw/oxfw_pcm.c | 399 ++++++++++ sound/firewire/oxfw/oxfw_proc.c | 60 ++ sound/firewire/oxfw/oxfw_stream.c | 582 ++++++++++++++ sound/firewire/speakers.c | 830 -------------------- 45 files changed, 10576 insertions(+), 1173 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h create mode 100644 sound/firewire/oxfw/oxfw_command.c create mode 100644 sound/firewire/oxfw/oxfw_control.c create mode 100644 sound/firewire/oxfw/oxfw_hwdep.c create mode 100644 sound/firewire/oxfw/oxfw_midi.c create mode 100644 sound/firewire/oxfw/oxfw_pcm.c create mode 100644 sound/firewire/oxfw/oxfw_proc.c create mode 100644 sound/firewire/oxfw/oxfw_stream.c delete mode 100644 sound/firewire/speakers.c -- 1.8.3.2 Regards Takashi Sakamoto o-takashi(a)sakamocchi.jp

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[alsa-devel] [PATCH] ASoC: da732x: Mark DC offset control registers volatile
by Mark Brown 26 Feb '14

26 Feb '14

From: Mark Brown <broonie(a)linaro.org> The driver reads from the DC offset control registers during callibration but since the registers are marked as volatile and there is a register cache the values will not be read from the hardware after the first reading rendering the callibration ineffective. It appears that the driver was originally written for the ASoC level register I/O code but converted to regmap prior to merge and this issue was missed during the conversion as the framework level volatile register functionality was not being used. Signed-off-by: Mark Brown <broonie(a)linaro.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/da732x.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 8e2c74d..798eb43 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, -- 1.9.0.rc3

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[alsa-devel] Fwd: i.MX audio support
by jwoodward 26 Feb '14

26 Feb '14

-------- Original Message -------- Subject: i.MX audio support Date: Wed, 26 Feb 2014 09:33:06 +0000 From: jwoodward <jwoodward(a)tensor.co.uk> To: s.hauer(a)pengutronix.de Hi Sascha, I am having trouble implimenting the wm8731 codec with the i.MX21. I have read your article https://lwn.net/Articles/363406/ which does not fill me with confidence on this. It is at the stage of finding the i2c device, and setting up the transmit line for AC97, I am not very experienced with audio and would appreciate any advise possible? Here are a few traces I have added: [ 2.935589] wm8731_i2c_probe: i2c_set_clientdata [ 2.989318] imx_ssi_init [ 2.997084] imx_ssi_probe [ 3.007482] imx_ssi_probe [ 3.015846] setup_channel_to_ac97 [ 3.029115] imx-fiq-pcm-audio imx-fiq-pcm-audio.1: failed to claim fiq: -16 [ 3.049930] imx-fiq-pcm-audio: probe of imx-fiq-pcm-audio.1 failed with error -16 [ 3.075173] ALSA device list: [ 3.084401] No soundcards found. Attached is a schematic of our interface. Many thanks, Jon -- Regards, __| __| __| Jonathan Woodward __| __| __| Embedded Developer __| __| __| Tensor plc jwoodward(a)tensor.co.uk Tel: +44 (0)1480 215530 http://www.tensor.co.uk Fax: +44 (0)1480 223966 ---------------------------------------------------------------------------------- Important Email Information The information in this email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and unlawful. If you are not the intended addressee please contact the sender and dispose of this email. Any opinions expressed in this e-mail are those of the individual and not necessarily those of Tensor plc or any companies within the Tensor group. Any prices for the supply of goods or services are only valid if supported by a formal written quotation. This communication is for information only and shall not create or change any contractual relationship. Every effort has been made to ensure that any attachment to this mail does not contain a virus. Whilst we have taken every reasonable precaution to minimise this risk, neither it nor the sender can accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checks before opening the attachment. ---------------------------------------------------------------------------------- Tensor plc, Hail Weston House, Hail Weston, St. Neots, Cambs, PE19 5JY, UK Registered in England number 03670909 - VAT Number 761 546 815

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[alsa-devel] I lost one SSID for HP mute led
by Kailang 26 Feb '14

26 Feb '14

Hi Takashi, I lost one SSID. I add it as attach patch. Thanks. BR, Kailang

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[alsa-devel] alsa on non-linux
by Patrick Welche 26 Feb '14

26 Feb '14

I am currently successfully running alsa-lib 1.0.22 with oss plugins on a non-linux box. The advantage is that programs written to use libasound work. I just tried to update alsa-lib, and see that now alsa-lib directly includes headers like linux/types.h, and all protection, such as #if defined(LINUX) || defined(__LINUX__) || defined(__linux__) has been removed. This means that compilation on non-linux is essentially impossible. Is it that there is now a different way of obtaining that alsa lib front end / oss back end layer? Cheers, Patrick

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[alsa-devel] [PATCH 00/10] A few more ALSA and HD-audio core updates
by Takashi Iwai 26 Feb '14

26 Feb '14

Hi, this is a series of patches to cleanup / enhance the ALSA core and HD-audio driver codes. A part of them are the patches I already submitted as RFC shortly ago (patch 4 & patch 5), another part are for better sysfs managment (part 1 & 2). The rest half are about HD-audio, for creating the device struct for each codec, followed by a cleanup patch to replace printk with dev_err() & co. Takashi

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[alsa-devel] CA0132 6 Channel Output
by Sean McCully 26 Feb '14

26 Feb '14

I have one of these sound cards that came with my motherboard, /http://www.asrock.com/microsite/intelx79/GameBlaster.html/[1] Two Channel Output from one audio Jack works perfectly, but there seems to be a problem with using the additional audio jacks for 6 channel "surround sound". http://www.alsa-project.org/db/?f=a8ae0aa3c0da4d697c1df5f403001dff50bb6e51 pasuspender -- speaker-test --channels=6 --device=surround51:0 speaker-test 1.0.27.2 Playback device is surround51:0 Stream parameters are 48000Hz, S16_LE, 6 channels Using 16 octaves of pink noise Channels count (6) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument $ aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System pulse PulseAudio Sound Server sysdefault:CARD=PCH HDA Intel PCH, CA0132 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, CA0132 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output -- -- Sean McCully -------- [1] http://www.asrock.com/microsite/intelx79/GameBlaster.html

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[alsa-devel] [PATCH] ASoC: cirrus: Remove excess dependencies on SND_SOC
by Alexander Shiyan 26 Feb '14

26 Feb '14

Configuration for Cirrus Logic audio support is included only if SND_SOC symbol selected, so no reason to check it once more. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/cirrus/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 06f938d..c872dac 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC + depends on ARCH_EP93XX || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: fsl-sai: fix Freescale SAI DAI format setting.
by Xiubo Li 26 Feb '14

26 Feb '14

o Fix some bugs of fsl_sai_set_dai_fmt_tr(). o Add SND_SOC_DAIFMT_LEFT_J support. o Add SND_SOC_DAIFMT_CBS_CFM support. o Add SND_SOC_DAIFMT_CBM_CFS support. o And SND_SOC_DAIFMT_RIGHT_J need to be done in the future. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 38 +++++++++++++++++++++++++++++--------- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index faa65af..26d9f5e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -105,35 +105,47 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, else val_cr4 |= FSL_SAI_CR4_MF; + /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - val_cr4 |= FSL_SAI_CR4_FSE; + /* Data on rising edge of bclk, frame low, 1clk before data */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high, 0clk before data */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; + case SND_SOC_DAIFMT_RIGHT_J: + /* To be done */ default: return -EINVAL; } + /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert both clocks */ + val_cr2 ^= FSL_SAI_CR2_BCP; + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_IB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert bit clock */ + val_cr2 ^= FSL_SAI_CR2_BCP; break; case SND_SOC_DAIFMT_NB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Invert frame clock */ + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_NB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Nothing to do for both normal cases */ break; default: return -EINVAL; } + /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; @@ -143,6 +155,14 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; + case SND_SOC_DAIFMT_CBS_CFM: + val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; + val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + break; default: return -EINVAL; } -- 1.8.4

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