February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: usb-audio: Add a quirck for B&W PX headphones
by erik.veijola＠gmail.com 24 Feb '18

24 Feb '18

From: Erik Veijola <erik.veijola(a)gmail.com> The capture interface doesn't work and the playback interface only supports 48 kHz sampling rate even though it advertises more rates. Signed-off-by: Erik Veijola <erik.veijola(a)gmail.com> --- sound/usb/quirks-table.h | 47 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 50252046b01d..754e632a27bd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3325,4 +3325,51 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +{ + /* + * Bower's & Wilkins PX headphones only support the 48 kHz sample rate + * even though it advertises more. The capture interface doesn't work + * even on windows. + */ + USB_DEVICE(0x19b5, 0x0021), + .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_MIXER, + }, + /* Capture */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_FILL_MAX | + UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 2.11.0

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[alsa-devel] [RFC v3] ALSA: hda: Add a power_save blacklist
by Hans de Goede 23 Feb '18

23 Feb '18

On some boards setting power_save to a non 0 value leads to clicking / popping sounds when ever we enter/leave powersaving mode. Ideally we would figure out how to avoid these sounds, but that is not always feasible. This commit adds a blacklist for devices where powersaving is known to cause problems and disables it on these devices. Note I tried to put this blacklist in userspace first: https://github.com/systemd/systemd/pull/8128 But the systemd maintainers rightfully pointed out that it would be impossible to then later remove entries once we actually find a way to make power-saving work on listed boards without issues. Having this list in the kernel will allow removal of the blacklist entry in the same commit which fixes the clicks / plops. The blacklist only applies to the default power_save module-option value, if a user explicitly sets the module-option then the blacklist is not used. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=198611 Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Changes in RFC v2: -Only use the blacklist if the power_save module-option is not explicitly set by the user Changes in RFC v2: -Fix logic to work on boards with more then 1 HDA device --- sound/pci/hda/hda_intel.c | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c71dcacea807..9d67b6b0ec95 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -181,7 +181,7 @@ static const struct kernel_param_ops param_ops_xint = { }; #define param_check_xint param_check_int -static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int power_save = -1; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -2186,6 +2186,22 @@ static int azx_probe(struct pci_dev *pci, return err; } +/* On some boards setting power_save to a non 0 value leads to clicking / + * popping sounds when ever we enter/leave powersaving mode. Ideally we would + * figure out how to avoid these sounds, but that is not always feasible. + * So we keep a list of devices where we disable powersaving as its known + * to causes problems on these devices. + */ +static struct snd_pci_quirk power_save_blacklist[] = { + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ + SND_PCI_QUIRK(0x17aa, 0x2227, "Lenovo X1 Carbon 3rd Gen", 0), + {} +}; + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2197,7 +2213,9 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; + const struct snd_pci_quirk *q; int dev = chip->dev_index; + int val; int err; hda->probe_continued = 1; @@ -2278,7 +2296,18 @@ static int azx_probe_continue(struct azx *chip) chip->running = 1; azx_add_card_list(chip); - snd_hda_set_power_save(&chip->bus, power_save * 1000); + + val = power_save; + if (val == -1) { + val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } + snd_hda_set_power_save(&chip->bus, val * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_autosuspend(&pci->dev); -- 2.14.3

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[alsa-devel] [PATCHv4 0/4] Motorola Droid 4 Audio Support
by Sebastian Reichel 23 Feb '18

23 Feb '18

Hi, This adds audio support to Motorola Droid 4. I dropped the regulator from the DT binding as requested by Mark. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (needs different DAI/I2S config) Changes since PATCHv3: * Drop regulator from DT binding, request VAUDIO from global regulator namespace instead Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (4): dt-bindings: sound: add motorola,cpcap-audio-codec ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: omap4-droid4: add soundcard .../bindings/sound/motorola,cpcap-audio-codec.txt | 17 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 5 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1588 ++++++++++++++++++++ 6 files changed, 1701 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.15.1

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[alsa-devel] [PATCH v1 0/3] Enable HDA Codec support on Intel Platforms (Series1)
by Rakesh Ughreja 23 Feb '18

23 Feb '18

Enhanced HD Audio capabilities introduced in HD Audio controllers are added in backward compatible way i.e. it does not change the behavior of the controller with respect to HDA Bus and HDA devices. Since there is no change in the hardware with respect to the HDA bus and HDA device it is more appropriate to represent the same in software as well. In order to align software representation with hardware it makes more sense to have common data structures across device, bus and driver for both enhanced HDA controllers and legacy HDA controllers. This patch series removes the hdac_ext_device structure, hdac_ext_bus and hdac_ext_driver data structures so that legacy and enhanced HDaudio capabilities can be handled in a backward-compatible way without separate definitions. Once this clean-up is complete, additional patches will allow for HDaudio codec support in an ASoC framework without the need to develop new codec drivers, thereby enabling the use of the Intel DSP on more platforms (currently limited to hdmi). Rakesh Ughreja (3): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead include/sound/hdaudio.h | 20 ++ include/sound/hdaudio_ext.h | 119 +++------- sound/hda/ext/hdac_ext_bus.c | 76 +++--- sound/hda/ext/hdac_ext_controller.c | 55 +++-- sound/hda/ext/hdac_ext_stream.c | 104 ++++----- sound/soc/codecs/hdac_hdmi.c | 416 ++++++++++++++++----------------- sound/soc/intel/skylake/skl-messages.c | 50 ++-- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 112 +++++---- sound/soc/intel/skylake/skl-topology.c | 18 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 178 +++++++------- sound/soc/intel/skylake/skl.h | 7 +- 13 files changed, 521 insertions(+), 648 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH V3 0/2 RESEND] ARM: dts: bcm283x: Fix probing of bcm2835-i2s
by Stefan Wahren 22 Feb '18

22 Feb '18

This small series fixes the probing of bcm2835-i2s, which is broken since 517e7a1537a ("ASoC: bcm2835: move to use the clock framework"). Changes in V3: * split up patch into dt-binding and functional part Changes in V2: * add Fixes, Reviewed-by and Tested-by tags Stefan Wahren (2): dt-bindings: bcm283x: Fix register ranges of bcm2835-i2s ARM: dts: bcm283x: Fix probing of bcm2835-i2s Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt | 4 ++-- Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 9 ++++----- arch/arm/boot/dts/bcm283x.dtsi | 4 ++-- 3 files changed, 8 insertions(+), 9 deletions(-) -- 2.7.4

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[alsa-devel] [asoc:topic/generic-dmaengine 1/1] sound/soc/soc-generic-dmaengine-pcm.c:435:5: error: 'struct dmaengine_pcm' has no member named 'component'
by kbuild test robot 22 Feb '18

22 Feb '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/generic-dmaengine head: f0b3bdbd5c63c3f69f3b6a869355c676d22fd348 commit: f0b3bdbd5c63c3f69f3b6a869355c676d22fd348 [1/1] ASoC: soc-generic-dmaengine-pcm: Add a DMA debugfs_prefix entry config: i386-randconfig-x004-201807 (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout f0b3bdbd5c63c3f69f3b6a869355c676d22fd348 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/soc-generic-dmaengine-pcm.c: In function 'snd_dmaengine_pcm_register': >> sound/soc/soc-generic-dmaengine-pcm.c:435:5: error: 'struct dmaengine_pcm' has no member named 'component' pcm->component.debugfs_prefix = "dma"; ^~ vim +435 sound/soc/soc-generic-dmaengine-pcm.c 417 418 /** 419 * snd_dmaengine_pcm_register - Register a dmaengine based PCM device 420 * @dev: The parent device for the PCM device 421 * @config: Platform specific PCM configuration 422 * @flags: Platform specific quirks 423 */ 424 int snd_dmaengine_pcm_register(struct device *dev, 425 const struct snd_dmaengine_pcm_config *config, unsigned int flags) 426 { 427 struct dmaengine_pcm *pcm; 428 int ret; 429 430 pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); 431 if (!pcm) 432 return -ENOMEM; 433 434 #ifdef CONFIG_DEBUG_FS > 435 pcm->component.debugfs_prefix = "dma"; 436 #endif 437 pcm->config = config; 438 pcm->flags = flags; 439 440 ret = dmaengine_pcm_request_chan_of(pcm, dev, config); 441 if (ret) 442 goto err_free_dma; 443 444 ret = snd_soc_add_platform(dev, &pcm->platform, 445 &dmaengine_pcm_platform); 446 if (ret) 447 goto err_free_dma; 448 449 return 0; 450 451 err_free_dma: 452 dmaengine_pcm_release_chan(pcm); 453 kfree(pcm); 454 return ret; 455 } 456 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); 457 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v3 00/17] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Change log== v3 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to make RX and TX more clearly defined + Replaced "bool dir" with "int dir" in PATCH-04 + Replaced "!dir" with "int adir" in PATCH-05 + Put CBM_CFS behind the baudclk check to keep the same program flow in PATCH-14 + Removed all cpu_dai_drv changes in PATCH-15 v2 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to keep all ssi->i2s_net updated + Replaced bool tx with bool dir in PATCH-03 and PATCH-06 + Moved all initial register configurations from dai probe() to platform probe() so as to let AC97 CODEC successfully probe. * Added Tested-by from Caleb for TDM test cases. ==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Caleb has tested v1 with TDM lookback tests on i.MX6. Example of uncovered tests: AC97, PowerPC and FIQ. Nicolin Chen (17): ASoC: fsl_ssi: Redefine RX and TX macros ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Add bool synchronous to mark synchronous mode ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 733 ++++++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 3 - 2 files changed, 370 insertions(+), 366 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 00/16] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Change log== v2 * Reworked the series by taking suggestions from Maciej + Added PATCH-01 to keep all ssi->i2s_net updated + Replaced bool tx with bool dir in PATCH-03 and PATCH-06 + Moved all initial register configurations from dai probe() to platform probe() so as to let AC97 CODEC successfully probe. * Added Tested-by from Caleb for TDM test cases. ==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Caleb has tested v1 with TDM lookback tests on i.MX6. Example of uncovered tests: AC97, PowerPC and FIQ. Nicolin Chen (16): ASoC: fsl_ssi: Keep ssi->i2s_net updated ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to probe() ASoC: fsl_ssi: Setup AC97 in fsl_ssi_hw_init() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Remove cpu_dai_drv from fsl_ssi structure ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 740 ++++++++++++++++++++++++------------------------ 1 file changed, 369 insertions(+), 371 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v1 00/15] ASoC: fsl_ssi: Clean up - program flow level
by Nicolin Chen 22 Feb '18

22 Feb '18

==Background== The fsl_ssi driver was designed for PPC originally and then it has been updated to support different modes for i.MX Series, including SDMA, I2S Master mode, AC97 and older i.MXs with FIQ, by different contributors for different use cases in different coding styles. Additionally, in order to fix/work-around hardware bugs and design flaws, the driver made a lot of compromise so now its program flow looks very complicated and it's getting hard to maintain or update. So I am going to clean up the driver on both coding style level and program flow level. ==Introduction== This series of patches is the second set to clean up fsl_ssi driver in the program flow level. Any patch here may impact a fundamental test case like playback or record. ==Verification== This series of patches require fully tested. I have done such tests on i.MX6SoloX with WM8962 using imx_v6_v7_defconfig as: - Playback via I2S Master and Slave mode - Record via I2S Master and Slave mode - Simultaneous playback and record via I2S Master and Slave mode - Background playback with foreground record (starting at different time) via I2S Master and Slave mode - Background record with foreground playback (starting at different time) via I2S Master and Slave mode * All tests above by hacking offline_config to true in imx51. Example of uncovered tests: TDM, AC97, PowerPC and FIQ. Nicolin Chen (15): ASoC: fsl_ssi: Clean up set_dai_tdm_slot() ASoC: fsl_ssi: Maintain a mask of active streams ASoC: fsl_ssi: Rename fsl_ssi_disable_val macro ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() ASoC: fsl_ssi: Clean up helper functions of trigger() ASoC: fsl_ssi: Add DAIFMT define for AC97 ASoC: fsl_ssi: Clean up fsl_ssi_setup_regvals() ASoC: fsl_ssi: Set xFEN0 and xFEN1 together ASoC: fsl_ssi: Use snd_soc_init_dma_data instead ASoC: fsl_ssi: Move one-time configurations to dai_probe() ASoC: fsl_ssi: Setup AC97 in dai_probe() ASoC: fsl_ssi: Clean up _fsl_ssi_set_dai_fmt() ASoC: fsl_ssi: Remove cpu_dai_drv from fsl_ssi structure ASoC: fsl_ssi: Move DT related code to a separate probe() ASoC: fsl_ssi: Use ssi->streams instead of reading register sound/soc/fsl/fsl_ssi.c | 710 ++++++++++++++++++++++++------------------------ 1 file changed, 348 insertions(+), 362 deletions(-) -- 2.7.4

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[alsa-devel] Applied "ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config()" to the asoc tree
by Mark Brown 22 Feb '18

22 Feb '18

The patch ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2e1327403b2543bc4dd639571b84bf6174212cee Mon Sep 17 00:00:00 2001 From: Nicolin Chen <nicoleotsuka(a)gmail.com> Date: Mon, 12 Feb 2018 14:03:14 -0800 Subject: [PATCH] ASoC: fsl_ssi: Clear FIFO directly in fsl_ssi_config() The FIFO clear helper function is just one line of code now. So it could be cleaned up by removing it and calling regmap directly. Meanwhile, FIFO clear could be applied to all use cases, not confined to AC97. So this patch also moves FIFO clear in the trigger() to fsl_ssi_config() and removes the AC97 check. Note that SOR register is safe from offline_config HW limit. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Tested-by: Caleb Crome <caleb(a)crome.org> Tested-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Reviewed-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_ssi.c | 33 ++++++++++----------------------- 1 file changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0d8c800db0b3..d276b78684e4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -409,17 +409,6 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi *ssi, bool enable) } } -/** - * Clear remaining data in the FIFO to avoid dirty data or channel slipping - */ -static void fsl_ssi_fifo_clear(struct fsl_ssi *ssi, bool is_rx) -{ - bool tx = !is_rx; - - regmap_update_bits(ssi->regs, REG_SSI_SOR, - SSI_SOR_xX_CLR(tx), SSI_SOR_xX_CLR(tx)); -} - /** * Exclude bits that are used by the opposite stream * @@ -446,10 +435,11 @@ static void fsl_ssi_fifo_clear(struct fsl_ssi *ssi, bool is_rx) static void fsl_ssi_config(struct fsl_ssi *ssi, bool enable, struct fsl_ssi_regvals *vals) { - int adir = (&ssi->regvals[TX] == vals) ? RX : TX; - int dir = (&ssi->regvals[TX] == vals) ? TX : RX; + bool tx = &ssi->regvals[TX] == vals; struct regmap *regs = ssi->regs; struct fsl_ssi_regvals *avals; + int adir = tx ? RX : TX; + int dir = tx ? TX : RX; bool aactive; /* Check if the opposite stream is active */ @@ -489,7 +479,9 @@ static void fsl_ssi_config(struct fsl_ssi *ssi, bool enable, /* Online configure single direction while SSI is running */ if (enable) { - fsl_ssi_fifo_clear(ssi, vals->scr & SSI_SCR_RE); + /* Clear FIFO to prevent dirty data or channel slipping */ + regmap_update_bits(ssi->regs, REG_SSI_SOR, + SSI_SOR_xX_CLR(tx), SSI_SOR_xX_CLR(tx)); regmap_update_bits(regs, REG_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, REG_SSI_STCR, vals->stcr, vals->stcr); @@ -511,6 +503,10 @@ static void fsl_ssi_config(struct fsl_ssi *ssi, bool enable, regmap_update_bits(regs, REG_SSI_SRCR, srcr, 0); regmap_update_bits(regs, REG_SSI_STCR, stcr, 0); regmap_update_bits(regs, REG_SSI_SIER, sier, 0); + + /* Clear FIFO to prevent dirty data or channel slipping */ + regmap_update_bits(ssi->regs, REG_SSI_SOR, + SSI_SOR_xX_CLR(tx), SSI_SOR_xX_CLR(tx)); } config_done: @@ -1091,7 +1087,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi *ssi = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct regmap *regs = ssi->regs; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -1116,14 +1111,6 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, return -EINVAL; } - /* Clear corresponding FIFO */ - if (fsl_ssi_is_ac97(ssi)) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - regmap_write(regs, REG_SSI_SOR, SSI_SOR_TX_CLR); - else - regmap_write(regs, REG_SSI_SOR, SSI_SOR_RX_CLR); - } - return 0; } -- 2.16.1

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