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[alsa-devel] [PATCH] ASoC: samsung: request memory region in driver probe()
by Sylwester Nawrocki 08 Dec '14

08 Dec '14

The memory mapped registers region is common for both DAIs so request it in the I2S platform device driver probe for the platform device corresponding to the primary DAI, rather than in the ASoC DAI probe callback. While at it switch to devm_ioremap_resource(). This also drops the hard coded (0x100) register region size in the driver. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Tested on Odroid X2. sound/soc/samsung/i2s.c | 44 +++++++------------------------------------- 1 file changed, 7 insertions(+), 37 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 26d1a1a..fff429da 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -59,10 +59,8 @@ struct samsung_i2s_dai_data { struct i2s_dai { /* Platform device for this DAI */ struct platform_device *pdev; - /* IOREMAP'd SFRs */ + /* Memory mapped SFR region */ void __iomem *addr; - /* Physical base address of SFRs */ - u32 base; /* Rate of RCLK source clock */ unsigned long rclk_srcrate; /* Frame Clock */ @@ -980,12 +978,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) goto probe_exit; } - i2s->addr = ioremap(i2s->base, 0x100); - if (i2s->addr == NULL) { - dev_err(&i2s->pdev->dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->clk = clk_get(&i2s->pdev->dev, "iis"); if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); @@ -1002,7 +994,6 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { - other->addr = i2s->addr; other->clk = i2s->clk; } @@ -1044,8 +1035,6 @@ static int samsung_i2s_dai_remove(struct snd_soc_dai *dai) clk_disable_unprepare(i2s->clk); clk_put(i2s->clk); - - iounmap(i2s->addr); } i2s->clk = NULL; @@ -1164,7 +1153,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) u32 regs_base, quirks = 0, idma_addr = 0; struct device_node *np = pdev->dev.of_node; const struct samsung_i2s_dai_data *i2s_dai_data; - int ret = 0; /* Call during Seconday interface registration */ i2s_dai_data = samsung_i2s_get_driver_data(pdev); @@ -1229,16 +1217,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "Unable to get I2S SFR address\n"); - return -ENXIO; - } + pri_dai->addr = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(pri_dai->addr)) + return PTR_ERR(pri_dai->addr); - if (!request_mem_region(res->start, resource_size(res), - "samsung-i2s")) { - dev_err(&pdev->dev, "Unable to request SFR region\n"); - return -EBUSY; - } regs_base = res->start; pri_dai->dma_playback.dma_addr = regs_base + I2STXD; @@ -1247,7 +1229,6 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; - pri_dai->base = regs_base; pri_dai->quirks = quirks; pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs; @@ -1258,8 +1239,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { dev_err(&pdev->dev, "Unable to alloc I2S_sec\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.ch_name = "tx-sec"; @@ -1271,7 +1251,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_size = 4; - sec_dai->base = regs_base; + sec_dai->addr = pri_dai->addr; sec_dai->quirks = quirks; sec_dai->idma_playback.dma_addr = idma_addr; sec_dai->pri_dai = pri_dai; @@ -1280,8 +1260,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (i2s_pdata && i2s_pdata->cfg_gpio && i2s_pdata->cfg_gpio(pdev)) { dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err; + return -EINVAL; } devm_snd_soc_register_component(&pri_dai->pdev->dev, @@ -1293,17 +1272,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) samsung_asoc_dma_platform_register(&pdev->dev); return 0; -err: - if (res) - release_mem_region(regs_base, resource_size(res)); - - return ret; } static int samsung_i2s_remove(struct platform_device *pdev) { struct i2s_dai *i2s, *other; - struct resource *res; i2s = dev_get_drvdata(&pdev->dev); other = i2s->pri_dai ? : i2s->sec_dai; @@ -1313,9 +1286,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) other->sec_dai = NULL; } else { pm_runtime_disable(&pdev->dev); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res) - release_mem_region(res->start, resource_size(res)); } i2s->pri_dai = NULL; -- 1.7.9.5

1 1

[alsa-devel] Nodes, Widgets, ELD and Digital Surround 5.1 (HDMI) Output?
by bencoxdev＠gmail.com 08 Dec '14

08 Dec '14

Hi All, I have an Intel HDA compatible device that supports 5.1 digital audio output over HDMI. It reports two nodes and two widgets:- Root(0) | AFG(1) / \ / \ / \ Audio output(2) Pin Complex(3) This setup produces a “Digital Stereo (HDMI) Output” device at user level in the operating system. The codec reports its self to the operating system as follows (from /proc/asound/card0/codec#1) -------------------------------------------------------------------- AFG Function Id: 0x1 (unsol 1) No Modem Function Group found Default PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 -------------------------------------------------------------------- Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Device: name="S3 HDMI 1", type="Audio", device=0 Converter: stream=0, channel=0 Digital: Digital category: 0x0 -------------------------------------------------------------------- Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000090: OUT HDMI Pin Default 0x185650f0: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Red DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 -------------------------------------------------------------------- the ELD from the receiving device is:- monitor_present 1 eld_valid 1 monitor_name EP-HDMI-RX connection_type HDMI eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0x1017 product_id 0x901 port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0x4f] FL/FR LFE FC RL/RR RLC/RRC sad_count 2 sad0_coding_type [0x2] AC-3 sad0_channels 6 sad0_rates [0xe0] 32000 44100 48000 sad0_max_bitrate 640000 sad1_coding_type [0x7] DTS sad1_channels 7 sad1_rates [0xc0] 44100 48000 sad1_max_bitrate 1536000 Does anyone know what changes would be required to this setup in order for the device to be recognised as “Digital Surround 5.1 (HDMI) Output” by alsa? TIA ---------------------------------------------------------------------- Regards Ben

4 11

[alsa-devel] [GIT PULL] ASoC updates for v3.19
by Mark Brown 08 Dec '14

08 Dec '14

The following changes since commit b2776bf7149bddd1f4161f14f79520f17fc1d71d: Linux 3.18 (2014-12-07 14:21:05 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19 for you to fetch changes up to 1810afd3e1ded09c53d4e966dddce3c7d484521f: Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next (2014-12-08 13:12:36 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.19 Lots and lots of changes this time around, the usual set of driver updates and a huge bulk of cleanups from Lars-Peter. Probably the most interesting thing for most users is the Intel driver updates which will (with some more machine integration work) enable support for newer x86 laptops. - Conversion of AC'97 drivers to use regmap, bringing us closer to the removal of the ASoC level I/O code. - Clean up a lot of old drivers that were open coding things that have subsequently been implemented in the core. - Some DAPM performance improvements. - Removal of the now seldom used CODEC mutex. - Lots of updates for the newer Intel SoC support, including support for the DSP and some Cherrytrail and Braswell machine drivers. - Support for Samsung boards using rt5631 as the CODEC. - Removal of the obsolete AFEB9260 machine driver. - Driver support for the TI TS3A227E headset driver used in some Chrombeooks. ---------------------------------------------------------------- My plan for the next cycle is to try to send one of these each time I send fixes, I meant to do that this time around during the second half of the cycle but ended up not sending fixes (or perhaps sending one lot of fixes at the same time as I'd applied a bunch of changes I wanted to cook in -next, can't remember exactly). Alban Bedel (1): ASoC: tegra: Read and use the GPIO flags of the headphone detect Alexandre Courbot (1): ASoC: jack: update calls to gpiod_get*() Anatol Pomozov (4): ASoC: rt5677: Add option to configure gpio as floating/pullup/pulldown ASoC: rt5677: Print more information if setting DAI clock failed ASoC: rt5677: fix build when kernel compiled without GPIOLIB support ASoC: rt5677: add build dependency to spi Andreas Ruprecht (1): ASoC: rockchip: i2s: Fix Kconfig for I2S device driver Andrew Jackson (1): ASoC: dwc: Allocate resources with devm_ioremap_resource Anil Kumar (1): ASoC: davinvi-mcasp: Balance pm_runtime_enable() on probe failure Axel Lin (2): ASoC: cs4265: Remove unused *dev field from struct cs4265_private ASoC: cs4271: Split SPI and I2C code into different modules Bard Liao (14): ASoC: rt286: reduce power consumption ASoC: rt5645: make bias level more reasonale ASoC: rt286: set combo jack by dmi ASoC: rt286: fix comment style ASoC: rt5645: Add JD function support ASoC: rt5645: Add ASRC support ASoC: rt5670: add rt5672 codec support ASoC: rt5645: remove unused rt5645_clk_sel_put ASoC: rt5645: add register setting for TDM ASoC: rt5645: two jacks for hp and mic ASoC: rt5670: make bias level more reasonable ASoC: rt5645: multiple JD mode support ASoC: rt5670: check if asrc is useable ASoC: rt5670: add DMIC ASRC support Ben Zhang (2): ASoC: rt5677: Include gpio driver header ASoC: rt5677: fix rt5677 spi driver build Charles Keepax (3): ASoC: arizona: Add support for 768kHz DMIC operation ASoC: wm_adsp: Use vmalloc to allocate firmware download buffer ASoC: wm_adsp: Move core_ena to be co-located with start bit Dan Carpenter (1): ASoC: sigmadsp: uninitialized variable in sigmadsp_activate_ctrl() Daniel Mack (1): ASoC: soc-compress: consolidate two identical branches Dmitry Eremin-Solenikov (3): ARM: pxa: spitz: register spitz-audio device ASoC: pxa: Convert spitz to use snd_soc_register_card() ASoC: pxa: prepare/unprepare clocks in pxa-ssp Dylan Reid (3): ASoC: max98090: Different comp tables for different pclks ASoC: add TI ts3a227e headset chip driver ASoC: rt5677: make volume TLV closer to reality Fabio Estevam (15): ASoC: fsl: imx-wm8962: Delete unneeded test before of_node_put ASoC: fsl: imx-sgtl5000: Delete unneeded test before of_node_put ASoC: fsl: imx-spdif: Delete unneeded test before of_node_put ASoC: fsl: eukrea-tlv320: Delete unneeded test before of_node_put ASoC: sgtl5000: Cleanup the comments ASoC: simple-card: Delete unneeded test before of_node_put ASoC: sgtl5000: Use the preferred form for passing a size of a struct ASoC: wm8962: Use the preferred form for passing a size of a struct ASoC: wm8731: Use the preferred form for passing a size of a struct ASoC: mxs: mxs-saif: Register the irq with the device name ASoC: fsl_ssi: Remove comment about SSI running only in slave mode ASoC: sgtl5000: Remove MCLK restriction ASoC: sgtl5000: Allow 8kHz playback in codec slave mode ASoC: mxs-sgtl5000: Remove MCLK restriction ASoC: fsl_ssi: Fix module unbound Fang, Yang A (1): ASoC: Intel: use correct firmware name Fengguang Wu (1): ASoC: Intel: mrfld: fix semicolon.cocci warnings Geert Uytterhoeven (2): ASoC: sh: fsi: Document SoC-specific bindings ASoC: rsnd: Document SoC-specific bindings JS Park (1): ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs Jarkko Nikula (3): ASoC: max98090: Fix digital microphone ASoC: max98090: Fix ill-defined sidetone route ASoC: max98090: Fix right sidetone connection Jean-Francois Moine (4): ASoC: simple-card: Remove useless casts ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: Remove 'const' from the device_node pointers Jean-Michel Hautbois (2): ASoC: sgtl5000: Add MicBias resistor support in DT ASoC: sgtl5000: Add MicBias voltage support Jiada Wang (1): ASoC: fsl_ssi: fix error path in probe Jianqun (2): ASoC: rockchip: i2s: add text after tristate for SND_SOC_ROCKCHIP_I2S ASoC: simple-card: add "invert" property for detect GPIOs Jie Yang (9): ASoC: Intel: Fix block is enabled multiple times issue ASoC: Intel: Work around to fix HW D3 potential crash issue ASoC: Intel: Add jack detection for Broadwell ASoC: Intel: Correct a macro for FW message ASoC: Intel: Fix the driver data not set issue ASoC: Intel: Fix stream volume set no effect issue on Broadwell ASoC: Intel: Remove useless loopback volume control for Broadwell ASoC: Intel: Correct the xmax volume ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell Jyri Sarha (3): ASoC: hdmi: Mark the maximum significant bits to HDMI codec ASoC: hdmi: HDMI codec doesn't benefit from pmdown delay ASoC: tlv320aic31xx: Fix off by one error in the loop stucture. Krishna Mohan Dani (5): ASoC: rt5631: Add device tree binding documentation ASoC: rt5631: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631/RT5631 codec driver ASoC: rt5631: Fixing compilation warning when DT is disabled ASoC: Samsung: Add arndale_rt5631 machine driver and binding Kuninori Morimoto (30): ASoC: rsnd: tidyup debug information when read/write ASoC: rsnd: tidyup RSND_DVC_VOLUME_NUM to RSND_DVC_CHANNELS ASoC: rsnd: add struct rsnd_dvc_cfg and control DVC settings ASoC: simple-card: add asoc_simple_card_parse_daifmt() ASoC: rsnd: control DVC_DVUCR under rsnd_dvc_volume_update() ASoC: rsnd: move DVC_DVUER settings under rsnd_dvc_volume_update() ASoC: rsnd: enable multiple DVC valume settings ASoC: rsnd: enable single DVC valume settings ASoC: rsnd: enable enumerated DVC valume settings ASoC: rsnd: tidyup debug message format and timing ASoC: rsnd: fallback to PIO mode if DMA mode was failed ASoC: rsnd: Add Volume Ramp support ASoC: rsnd: add .fallback callback ASoC: rsnd: add callback status check method ASoC: rsnd: rsnd_src_ssiu_stop() stops SSIU compulsorily ASoC: rsnd: tidyup PIO/DMA mode settings method ASoC: rsnd: tidyup SSI interrupt enable/disable method ASoC: rsnd: care SSIWSR register in rsnd_ssi_hw_start() ASoC: rsnd: clear status register when HW start ASoC: rsnd: synchronize SSI start/stop sequence between PIO/DMA mode ASoC: rsnd: show master clock rate when ADG probe ASoC: rsnd: move snd_kcontrol_new fucntions to core.c ASoC: rsnd: tidyup rsnd_io_to_runtime() macro ASoC: rsnd: use rsnd_src_convert_rate() once on rsnd_src_set_convert_rate_gen2() ASoC: rsnd: initialize SRC on rsnd_src_init() ASoC: rsnd: set SRC_ROUTE_MODE0 on each rsnd_src_set_convert_rate() ASoC: rsnd: share SSI starting method between PIO/DMA mode ASoC: rsnd: remove un-necessary parameter from rsnd_src_start/stop() ASoC: rsnd: add salvage support for under/over flow error on SSI ASoC: rsnd: rename SSI function name of PIO Lars-Peter Clausen (145): ASoC: ak4535: Cleanup manual bias level transitions ASoC: ak4535: Use table based setup for controls ASoC: ak4641: Cleanup manual bias level transitions ASoC: ak4642: Cleanup manual bias level transitions ASoC: ak4671: Cleanup manual bias level transitions ASoC: max98088: Cleanup manual bias level transitions ASoC: max98095: Cleanup manual bias level transitions ASoC: max9850: Cleanup manual bias level transitions ASoC: sta32x: Cleanup manual bias level transitions ASoC: sta350: Cleanup manual bias level transitions ASoC: sta529: Cleanup manual bias level transitions ASoC: stac9766: Cleanup manual bias level transitions ASoC: dapm: Reduce number of checked paths in dapm_widget_in_card_paths() ASoC: dapm: Remove always true path source/sink checks ASoC: dapm: Remove special DAI widget power check functions ASoC: dapm: Remove path 'walked' flag ASoC: dapm: Only mark paths dirty when the connection status changed ASoC: dapm: Do not add un-muxed paths to MUX control ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets ASoC: dapm: Introduce toplevel widget categories ASoC: dapm: Add a flag to mark paths connected to supply widgets ASoC: dapm: Mark endpoints instead of IO widgets dirty during suspend/resume ASoC: dapm: Add a few supply widget sanity checks ASoC: dapm: Use more aggressive caching ASoC: Use generic control handlers for S8 control ASoC: ad1980: Remove unused header ASoC: ad1980: Cleanup printk usage ASoC: ad1980: Use table based control setup ASoC: stac9766: Cleanup printk usage ASoC: stac9766: Use table based control setup ASoC: wm9705: Cleanup printk usage ASoC: wm9705: Use table based control setup ASoC: wm9712: Cleanup printk usage ASoC: wm9712: Use table based control setup ASoC: wm9713: Cleanup printk usage ASoC: wm9713: Use table based control setup ASoC: wm9713: Move driver state struct allocation to driver probe ASoC: wm9713: Use virtual control instead of virtual register ASoC: Remove snd_soc_platform_driver suspend/resume callbacks ASoC: Remove platform field from snd_soc_dai ASoC: mioa701_wm9713: Don't opencode CODEC register access ASoC: Remove card field from snd_soc_dai struct ASoC: Consolidate CPU and CODEC DAI probe ASoC: sn95031: Use table based control setup ASoC: tas2552: Use table based DAPM setup ASoC: wl1273: Use table based control setup ASoC: wm8737: Use table based DAPM and control setup ASoC: wm8961: Use table based DAPM and control setup ASoC: wm8995: Use table based DAPM and control setup ASoC: ad193x: Keep DAC output stage active in idle ASoC: ssm4567: Add DAC high-pass-filter control ASoC: ssm4567: Add support for setting the DAI format and TDM configuration ASoC: ssm4567: Add support for disabling the boost stage ASoC: wm9712/wm9713: Replace virtual registers with custom put/get callbacks ASoC: Forward calls to snd_soc_cache_sync() to regcache_sync() ASoC: Remove snd_soc_cache_sync() implementation ASoC: ab8500-codec: Move control lock to the driver level ASoC: max98095: Move mutex to the driver level ASoC: wm5102: Move ultrasonic response settings lock to the driver level ASoC: wm8731: Move the deemph lock to the driver level ASoC: wm8903: Move the deemph lock to the driver level ASoC: wm8958: Move DSP firmware lock to driver level ASoC: wm8962: Move DSP enable lock to the driver level ASoC: Remove CODEC mutex ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: adau1373: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: adau17x1: Replace w->codec with snd_soc_dapm_to_codec(w->dapm) ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device ASoC: mpc5200_dma: Don't overwrite ac97 device private_data ASoC: Properly handle AC'97 device lifetime management ASoC: Move AC'97 support to its own file ASoC: ac97: Use static ac97_bus ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() ASoC: ac97: Drop support for setting platform data via the CPU DAI ASoC: ac97: Drop delayed device registration ASoC: Drop ac97_control initialization from CODEC driver DAIs ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control ASoC: ac97: Push snd_ac97 pointer to the driver level ASoC: Add helper functions for deferred regmap setup ASoC: ad1980: Convert to regmap ASoC: ad1980: Remove ac97_read/ac97_write wrappers ASoC: sigmadsp: Refuse to load firmware files with a non-supported version ASoC: sigmadsp: Drop support support SIGMA_ACTION_DELAY ASoC: sigmadsp: Restructure in preparation for fw v2 support ASoC: sigmadsp: Add support for fw v2 ASoC: adau1701: Implement sigmadsp safeload ASoC: sirf-audio-codec: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98088: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98090: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: max98095: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: adau17x1: Mark DSP parameter memory as readable and precious ASoC: sigmadsp: Fix endianness conversion ASoC: cs42l51: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: cs42l73: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: uda134x: Remove is_powered_on_standby from platform data ASoC: uda134x: Cleanup manual bias level transitions ASoC: uda1380: Cleanup manual bias level transitions ASoC: alc5623: Cleanup bias level transitions ASoC: alc5632: Cleanup bias level transitions ASoC: rt5631: Cleanup bias level transitions ASoC: wm8350: Cleanup manual bias level transitions ASoC: wm8400: Cleanup manual bias level transitions ASoC: wm8510: Cleanup manual bias level transitions ASoC: wm8523: Cleanup manual bias level transitions ASoC: wm8580: Cleanup manual bias level transitions ASoC: wm8711: Cleanup manual bias level transitions ASoC: wm8728: Cleanup manual bias level transitions ASoC: wm8731: Cleanup manual bias level transitions ASoC: wm8737: Cleanup manual bias level transitions ASoC: wm8750: Cleanup manual bias level transitions ASoC: wm8776: Cleanup manual bias level transitions ASoC: wm8804: Cleanup manual bias level transitions ASoC: wm8900: Cleanup manual bias level transitions ASoC: wm8903: Cleanup manual bias level transitions ASoC: wm8940: Cleanup manual bias level transitions ASoC: wm8955: Cleanup manual bias level transitions ASoC: wm8960: Cleanup manual bias level transitions ASoC: wm8961: Cleanup manual bias level transitions ASoC: wm8983: Cleanup manual bias level transitions ASoC: wm8978: Cleanup manual bias level transitions ASoC: wm8974: Cleanup manual bias level transitions ASoC: wm8985: Cleanup manual bias level transitions ASoC: wm8988: Cleanup manual bias level transitions ASoC: wm8990: Cleanup manual bias level transitions ASoC: wm8991: Cleanup manual bias level transitions ASoC: wm8993: Cleanup manual bias level transitions ASoC: wm8994: Cleanup manual bias level transitions ASoC: wm8995: Cleanup manual bias level transitions ASoC: wm9081: Cleanup manual bias level transitions ASoC: wm9090: Cleanup manual bias level transitions ASoC: wm9712: Cleanup manual bias level transitions ASoC: wm9713: Cleanup manual bias level transitions ASoC: qi_lb60: Pass flags to gpiod_get() ASoC: Disable regmap helpers if regmap is disabled ASoC: cq93vc: Remove unused state struct ASoC: cq93vc: Cleanup manual bias level transitions ASoC: lm49453: Cleanup manual bias level transitions ASoC: sn95031: Cleanup bias level transitions ASoC: tlv320aic23: Cleanup manual bias level transitions ASoC: tlv320aix31xx: Cleanup manual bias level transitions ASoC: tlv320aic32x4: Cleanup manual bias level transitions ASoC: tlv320aic3x: Cleanup manual bias level transitions ASoC: tlv320dac33: Cleanup manual bias level transitions ASoC: twl4030: Cleanup manual bias level transitions ASoC: twl6040: Cleanup manual bias level transitions Liam Girdwood (11): ASoC: Intel: Add TDM support to HSW/BDW SSP port ASoC: Intel: Add 4 channel support to DSP. ASoC: Intel: Make HSW/BDW pointer debug verbose ASoC: Intel: Make ADSP memory block allocation more generic ASoC: Intel: Add generic support for DSP wake, sleep and stall ASoC: Intel: Add PM support to the HSW/BDW DSP core. ASoC: Intel: Add PM support to HSW/BDW IPC driver ASoC: Intel: Add PM support to HSW/BDW PCM driver ASoC: Intel: Add debug output when boot fails. ASoC: Intel: fix build with runtime PM disabled. ASoC: Intel: Fix build with CONFIG_SLEEP enabled. Mark Brown (46): Merge branch 'fix/sgtl5000' of git://git.kernel.org/.../broonie/sound into asoc-sgtl5000 Merge tag 'v3.18-rc2' into asoc-sgtl5000 ASoC: Intel: Add dependency on DesignWare DMA controller ASoC: core: Split ops out of soc-core.c Merge tag 'v3.18-rc3' into asoc-sgtl5000 Merge branch 'topic/dapm' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'fix/max98090' of git://git.kernel.org/.../broonie/sound into asoc-max98090 Merge branches 'topic/wm9712' and 'topic/wm9713' of git://git.kernel.org/.../broonie/sound into asoc-wm971x Merge branch 'fix/davinci' of git://git.kernel.org/.../broonie/sound into asoc-davinci Merge branches 'topic/ad1980', 'topic/wm9705' and 'topic/wm971x' of git://git.kernel.org/.../broonie/sound into asoc-ac97 regmap: ac97: Add generic AC'97 callbacks Merge branch 'topic/ac97' of git://git.kernel.org/.../broonie/regmap into asoc-ac97 Merge branch 'topic/regmap' of git://git.kernel.org/.../broonie/sound into asoc-ac97 ASoC: samsung: Fix error handling for clock lookup ASoC: samsung: Fix non-DT use of I2S controller ASoC: wm5102: Initialize dac_comp_lock mutex Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/davinci', 'asoc/fix/max98090', 'asoc/fix/sigmadsp' and 'asoc/fix/tlv320aic31xx' into asoc-linus Merge remote-tracking branch 'asoc/topic/cache' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ad193x', 'asoc/topic/adau1373' and 'asoc/topic/adau17x1' into asoc-next Merge remote-tracking branches 'asoc/topic/adav80x', 'asoc/topic/adsp', 'asoc/topic/ak4535', 'asoc/topic/ak4641' and 'asoc/topic/ak4642' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623', 'asoc/topic/alc5632', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/codec-mutex', 'asoc/topic/compress' and 'asoc/topic/cq93vc' into asoc-next Merge remote-tracking branches 'asoc/topic/cs4265', 'asoc/topic/cs4271', 'asoc/topic/cs42l51' and 'asoc/topic/cs42l73' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/doc', 'asoc/topic/dpcm', 'asoc/topic/dwc' and 'asoc/topic/fsi' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl', 'asoc/topic/fsl-card', 'asoc/topic/fsl-dt' and 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/intel', 'asoc/topic/jack', 'asoc/topic/jz4740' and 'asoc/topic/lm49453' into asoc-next Merge remote-tracking branches 'asoc/topic/max98088', 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max9850' and 'asoc/topic/mop500' into asoc-next Merge remote-tracking branches 'asoc/topic/multi-codec', 'asoc/topic/mxs-saif', 'asoc/topic/mxs-sgtl5000', 'asoc/topic/omap' and 'asoc/topic/pxa' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rockchip', 'asoc/topic/rt286' and 'asoc/topic/rt5631' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5670', 'asoc/topic/rt5677', 'asoc/topic/samsung' and 'asoc/topic/sgtl5000' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/sigmadsp', 'asoc/topic/simple', 'asoc/topic/sirf' and 'asoc/topic/sn95031' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm4567', 'asoc/topic/sta32x', 'asoc/topic/sta350', 'asoc/topic/sta529' and 'asoc/topic/stac9766' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tegra', 'asoc/topic/tfa9879', 'asoc/topic/tlv320aic23' and 'asoc/topic/tlv320aic31xx' into asoc-next Merge remote-tracking branches 'asoc/topic/tlv320aic32x4', 'asoc/topic/tlv320aic3x', 'asoc/topic/tlv320dac33', 'asoc/topic/ts3a227e' and 'asoc/topic/twl4030' into asoc-next Merge remote-tracking branches 'asoc/topic/twl6040', 'asoc/topic/uda134x', 'asoc/topic/uda1380' and 'asoc/topic/wl1273' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8350', 'asoc/topic/wm8400', 'asoc/topic/wm8510', 'asoc/topic/wm8523' and 'asoc/topic/wm8580' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8711', 'asoc/topic/wm8728', 'asoc/topic/wm8731', 'asoc/topic/wm8737' and 'asoc/topic/wm8750' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8776', 'asoc/topic/wm8804', 'asoc/topic/wm8900', 'asoc/topic/wm8903' and 'asoc/topic/wm8940' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8955', 'asoc/topic/wm8960', 'asoc/topic/wm8961', 'asoc/topic/wm8962' and 'asoc/topic/wm8974' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8978', 'asoc/topic/wm8983', 'asoc/topic/wm8985', 'asoc/topic/wm8988' and 'asoc/topic/wm8990' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8991', 'asoc/topic/wm8993', 'asoc/topic/wm8994', 'asoc/topic/wm8995' and 'asoc/topic/wm9081' into asoc-next Merge remote-tracking branches 'asoc/topic/wm9090', 'asoc/topic/wm9712' and 'asoc/topic/wm9713' into asoc-next Markus Elfring (3): ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_disable" ASoC: mop500: Deletion of unnecessary checks before the function call "of_node_put" Masanari Iida (1): ASoC: jack: Fix warning while make htmldocs caused by soc-jack.c Max Filippov (2): ASoC: tlv320aic23: make codecs selectable in Kconfig ASoC: tlv320aic23: add dependencies on I2C/SPI_MASTER Mengdong Lin (4): ASoC: rt5670 : Add ACPI match ID for Intel CHT/BSW platforms ASoC: Intel: add missing ACPI device table ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672 ASoC: Intel: add support for Cherrytrail and Braswell in SST driver Misael Lopez Cruz (3): ASoC: davinci-mcasp: Active slots depend on active serializers ASoC: tlv320aic3x: Add output driver pop reduction controls ASoC: davinci-mcasp: Add overrun/underrun event handling Mythri P K (1): ASoC: Intel: use lock when changing SST state. Nicolas Ferre (2): ASoC: snd-soc-afeb9260: delete driver as board has just been removed ASoC: Kconfig: remove not used SND_AT91_SOC_AFEB9260 option Nicolin Chen (11): ASoC: fsl_esai: Add indentation for binding doc to increase readability ASoC: fsl_spdif: Add indentation for binding doc to increase readability ASoC: fsl_sai: Add indentation for binding doc to increase readability ASoC: eukrea-tlv320: Add indentation for binding doc to increase readability ASoC: imx-audmux: Add indentation for binding doc to increase readability ASoC: imx-sgtl5000: Add indentation for binding doc to increase readability ASoC: imx-spdif: Add indentation for binding doc to increase readability ASoC: imx-wm8962: Add indentation for binding doc to increase readability ASoC: fsl-asoc-card: Don't bypass settings if cpu-dai is Master ASoC: fsl_esai: Use dynamic slot width as default ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai Oder Chiou (12): ASoC: rt5645: Add the workqueue of the jack detect function for the debouncing ASoC: rt5677: Support DSP function for VAD application ASoC: rt5677: add GPIO IRQ support ASoC: rt5677: Add TDM channel mapping function ASoC: rt5677: Minor coding style and typo fix ASoC: rt5677: Use specific r/w function for DSP mode ASoC: rt5677: Add TDM channel mux in DAC side of IF1 and IF2 ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function ASoC: rt5677: Align the reg_default table with tab character ASoC: rt5677: Follow the gpio naming rule to rename the irq function ASoC: rt5677: Set the slow charge of the vref in the end of the power sequences ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed Padmavathi Venna (3): ASoC: Samsung: Add quirk for internal DMA ASoC: samsung: add support for exynos7 I2S controller ASoC: samsung: ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk() Pavel Machek (1): ASoC: omap: enable sound support on n900 on devicetree-based boot Peter Rosin (3): ASoC: atmel_ssc_dai: Match the CMR divider only in full duplex. ASoC: tfa9879: New driver for NXP Semiconductors TFA9879 amplifier. ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing Peter Ujfalusi (10): ASoC: davinci-mcasp: Correct TX start sequence ASoC: davinci-mcasp: Correct RX start sequence ASoC: davinci-mcasp: When stopping TX/RX stop the AFIFO as the last step ASoC: davinci-mcasp: Move the AFIFO related code under start_tx/rx functions ASoC: tlv320aic3x: Add TDM support ASoC: davinci-mcasp: Symmetric sample bits for IIS mode ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time ASoC: davinci-mcasp: Place constraint on number of channels ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual Qiao Zhou (2): ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE ASoC: soc-pcm: do not hw_free BE if it's still used Ramesh Babu (1): ASoC: core: Call mute for cpu dais as well Randy Dunlap (1): ASoC: cs42l51: depends on I2C Rasmus Villemoes (1): ASoC: dapm: Remove redundant cast Ricardo Neri (2): ASoC: Intel: Add new dependency for Broadwell machine ASoC: Intel: Add new dependency for Haswell machine Richard Fitzgerald (1): ASoC: arizona: Add DSP_B and LEFT_J mode support Subhransu S. Prusty (12): ASoC: Intel: mrfld: Replace pci_id with unique device id ASoC: Intel: mrfld - Define ipc_info structure ASoC: Intel: mrfld: Define sst_res_info for acpi ASoC: Intel: mrfld: Fix runtime pm calls in sst_open_pcm_stream ASoC: mfld-compress: implement .power callback ASoC: Intel: mfld-pcm: Fix to Store device context in sst_data ASoC: Intel: move the driver wq init to a routine ASoC: Intel: move the lock and wq initialization to routine ASoC: Intel: move the driver context allocation to routine ASoC: Intel: modularize driver probe and remove ASoC: Intel: more probe modularization for sst ASoC: Intel: add BYTCR machine driver with RT5640 Sudip Mukherjee (2): ASoC: Intel: fix missing mutex ASoC: rt286: build warning of section mismatch Takashi Iwai (2): ASoC: eukrea-tlv320: Fix of_node_put() call with uninitialized object ASoC: core: Use snd_ctl_enum_info() Thomas Petazzoni (1): ASoC: cs42l51: make driver user-selectable Vinod Koul (27): ASoC: Intel: mrfld: add the gain controls ASoC: Intel: mrfld: add DSP core controls ASoC: Intel: mrfld: add the DSP DAPM widgets ASoC: Intel: mfld-pcm: add FE and BE ops ASoC: Intel: mrfld: add the DSP mixers ASoC: intel: turn off COMPILE_TEST for medfield ASoC: Intel: mrfld - add the dsp sst driver ASoC: Intel: mrfld - Add DSP load and management ASoC: Intel: sst - add pcm ops handling ASoC: Intel: sst: Add IPC handling ASoC: Intel: sst: add stream operations ASoC: Intel: sst: Add some helper functions ASoC: Intel: sst: Add makefile and kconfig changes ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: intel: log an error on double free ASoC: intel: fix the kernldoc comment ASoC: intel: explain why block not found isn't error always ASoC: intel: use __iowrite32_copy for 32 bit copy ASoC: Intel: sst: add runtime power management handling ASoC: Intel: sst: load firmware using async callback ASoC: Intel: sst - add compressed ops handling ASoC: Intel: mrfld - remove unnecessary check for pointer ASoC: Intel: mrfld - create separate module for pci part ASoC: Intel: mrfld - add shim save restore ASoC: Intel: mrfld - remove non static definition ASoC: Intel: mrfld- add ACPI module ASoC: Intel: cleanup runtime_pm initialization Zidan Wang (2): ASoC: wm8960: Add device tree support ASoC: wm8960: Move register initialisation to I2C driver probe() kbuild test robot (5): ASoC: rt5677: rt5677_irq_init() can be static ASoC: Intel: dw_pdata can be static ASoC: Intel: byt_rvp_platform_data can be static ASoC: Intel: chv_platform_data can be static ASoC: sigmadsp: fix simple_return.cocci warnings .../devicetree/bindings/sound/arndale.txt | 24 + .../bindings/sound/davinci-mcasp-audio.txt | 2 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 15 +- .../devicetree/bindings/sound/fsl,esai.txt | 44 +- .../devicetree/bindings/sound/fsl,spdif.txt | 37 +- .../devicetree/bindings/sound/fsl-sai.txt | 66 +- .../bindings/sound/imx-audio-sgtl5000.txt | 61 +- .../devicetree/bindings/sound/imx-audio-spdif.txt | 22 +- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 45 +- .../devicetree/bindings/sound/imx-audmux.txt | 22 +- .../devicetree/bindings/sound/max98090.txt | 2 + .../devicetree/bindings/sound/renesas,fsi.txt | 9 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 10 +- Documentation/devicetree/bindings/sound/rt5631.txt | 48 + Documentation/devicetree/bindings/sound/rt5677.txt | 17 + .../devicetree/bindings/sound/samsung-i2s.txt | 15 +- .../devicetree/bindings/sound/sgtl5000.txt | 13 + .../devicetree/bindings/sound/ts3a227e.txt | 26 + Documentation/devicetree/bindings/sound/wm8960.txt | 31 + MAINTAINERS | 6 + arch/arm/mach-pxa/spitz.c | 9 + arch/x86/include/asm/platform_sst_audio.h | 62 + drivers/base/regmap/Kconfig | 5 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-ac97.c | 114 ++ include/linux/mfd/arizona/core.h | 1 + include/linux/mfd/davinci_voicecodec.h | 7 - include/linux/platform_data/asoc-s3c.h | 1 + include/linux/regmap.h | 7 + include/sound/rcar_snd.h | 8 +- include/sound/rt5645.h | 4 + include/sound/rt5677.h | 10 + include/sound/soc-dai.h | 7 +- include/sound/soc-dapm.h | 9 +- include/sound/soc.h | 116 +- include/sound/uda134x.h | 12 - include/trace/events/asoc.h | 25 - sound/soc/Makefile | 6 +- sound/soc/atmel/Kconfig | 9 - sound/soc/atmel/Makefile | 1 - sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/atmel/snd-soc-afeb9260.c | 151 -- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 - sound/soc/cirrus/Kconfig | 3 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/Kconfig | 47 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 32 +- sound/soc/codecs/ac97.c | 18 +- sound/soc/codecs/ad193x.c | 14 +- sound/soc/codecs/ad1980.c | 212 +-- sound/soc/codecs/ad1980.h | 26 - sound/soc/codecs/adau1373.c | 6 +- sound/soc/codecs/adau1701.c | 86 +- sound/soc/codecs/adau1761.c | 25 +- sound/soc/codecs/adau1781.c | 33 +- sound/soc/codecs/adau17x1.c | 71 +- sound/soc/codecs/adau17x1.h | 10 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4535.c | 31 +- sound/soc/codecs/ak4641.c | 33 +- sound/soc/codecs/ak4642.c | 16 - sound/soc/codecs/ak4671.c | 13 - sound/soc/codecs/alc5623.c | 22 +- sound/soc/codecs/alc5632.c | 22 +- sound/soc/codecs/arizona.c | 34 +- sound/soc/codecs/cq93vc.c | 33 - sound/soc/codecs/cs4265.c | 2 - sound/soc/codecs/cs4271-i2c.c | 62 + sound/soc/codecs/cs4271-spi.c | 55 + sound/soc/codecs/cs4271.c | 155 +- sound/soc/codecs/cs4271.h | 11 + sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l73.c | 6 +- sound/soc/codecs/hdmi.c | 2 + sound/soc/codecs/lm49453.c | 8 - sound/soc/codecs/max98088.c | 31 +- sound/soc/codecs/max98090.c | 201 ++- sound/soc/codecs/max98090.h | 8 + sound/soc/codecs/max98095.c | 23 +- sound/soc/codecs/max9850.c | 22 +- sound/soc/codecs/rt286.c | 231 ++- sound/soc/codecs/rt5631.c | 38 +- sound/soc/codecs/rt5645.c | 200 ++- sound/soc/codecs/rt5645.h | 19 +- sound/soc/codecs/rt5670.c | 136 +- sound/soc/codecs/rt5670.h | 6 + sound/soc/codecs/rt5677-spi.c | 130 ++ sound/soc/codecs/rt5677-spi.h | 21 + sound/soc/codecs/rt5677.c | 1198 ++++++++++++- sound/soc/codecs/rt5677.h | 162 +- sound/soc/codecs/sgtl5000.c | 111 +- sound/soc/codecs/sigmadsp-i2c.c | 81 +- sound/soc/codecs/sigmadsp-regmap.c | 46 +- sound/soc/codecs/sigmadsp.c | 711 +++++++- sound/soc/codecs/sigmadsp.h | 59 +- sound/soc/codecs/sirf-audio-codec.c | 6 +- sound/soc/codecs/sn95031.c | 15 +- sound/soc/codecs/ssm4567.c | 128 ++ sound/soc/codecs/sta32x.c | 21 +- sound/soc/codecs/sta350.c | 21 +- sound/soc/codecs/sta529.c | 35 +- sound/soc/codecs/stac9766.c | 60 +- sound/soc/codecs/tas2552.c | 10 +- sound/soc/codecs/tfa9879.c | 328 ++++ sound/soc/codecs/tfa9879.h | 202 +++ sound/soc/codecs/tlv320aic23.c | 21 +- sound/soc/codecs/tlv320aic31xx.c | 31 +- sound/soc/codecs/tlv320aic32x4.c | 24 +- sound/soc/codecs/tlv320aic3x.c | 228 ++- sound/soc/codecs/tlv320aic3x.h | 1 + sound/soc/codecs/tlv320dac33.c | 2 - sound/soc/codecs/ts3a227e.c | 314 ++++ sound/soc/codecs/ts3a227e.h | 17 + sound/soc/codecs/twl4030.c | 2 - sound/soc/codecs/twl6040.c | 23 +- sound/soc/codecs/uda134x.c | 32 +- sound/soc/codecs/uda1380.c | 20 +- sound/soc/codecs/wl1273.c | 10 +- sound/soc/codecs/wm5102.c | 18 +- sound/soc/codecs/wm8350.c | 21 +- sound/soc/codecs/wm8400.c | 34 +- sound/soc/codecs/wm8510.c | 26 +- sound/soc/codecs/wm8523.c | 29 +- sound/soc/codecs/wm8580.c | 4 - sound/soc/codecs/wm8711.c | 27 +- sound/soc/codecs/wm8728.c | 34 +- sound/soc/codecs/wm8731.c | 37 +- sound/soc/codecs/wm8737.c | 49 +- sound/soc/codecs/wm8750.c | 25 +- sound/soc/codecs/wm8776.c | 31 +- sound/soc/codecs/wm8804.c | 3 - sound/soc/codecs/wm8900.c | 8 - sound/soc/codecs/wm8903.c | 43 +- sound/soc/codecs/wm8940.c | 22 +- sound/soc/codecs/wm8955.c | 33 +- sound/soc/codecs/wm8958-dsp2.c | 12 +- sound/soc/codecs/wm8960.c | 113 +- sound/soc/codecs/wm8961.c | 34 +- sound/soc/codecs/wm8962.c | 11 +- sound/soc/codecs/wm8974.c | 25 +- sound/soc/codecs/wm8978.c | 10 - sound/soc/codecs/wm8983.c | 27 +- sound/soc/codecs/wm8985.c | 28 +- sound/soc/codecs/wm8988.c | 27 +- sound/soc/codecs/wm8990.c | 24 +- sound/soc/codecs/wm8991.c | 32 +- sound/soc/codecs/wm8993.c | 12 - sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8994.h | 2 + sound/soc/codecs/wm8995.c | 17 +- sound/soc/codecs/wm9081.c | 7 - sound/soc/codecs/wm9090.c | 32 +- sound/soc/codecs/wm9705.c | 46 +- sound/soc/codecs/wm9712.c | 219 ++- sound/soc/codecs/wm9713.c | 230 +-- sound/soc/codecs/wm_adsp.c | 97 +- sound/soc/davinci/davinci-mcasp.c | 339 +++- sound/soc/davinci/davinci-mcasp.h | 17 + sound/soc/dwc/designware_i2s.c | 46 +- sound/soc/fsl/eukrea-tlv320.c | 5 +- sound/soc/fsl/fsl-asoc-card.c | 19 +- sound/soc/fsl/fsl_esai.c | 12 +- sound/soc/fsl/fsl_ssi.c | 17 +- sound/soc/fsl/imx-sgtl5000.c | 6 +- sound/soc/fsl/imx-spdif.c | 3 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/imx-wm8962.c | 6 +- sound/soc/fsl/mpc5200_dma.c | 3 - sound/soc/fsl/mpc5200_psc_ac97.c | 6 +- sound/soc/generic/simple-card.c | 160 +- sound/soc/intel/Kconfig | 44 +- sound/soc/intel/Makefile | 7 + sound/soc/intel/broadwell.c | 64 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 230 +++ sound/soc/intel/cht_bsw_rt5672.c | 285 ++++ sound/soc/intel/haswell.c | 14 +- sound/soc/intel/sst-atom-controls.c | 1208 ++++++++++++- sound/soc/intel/sst-atom-controls.h | 428 +++++ sound/soc/intel/sst-baytrail-dsp.c | 24 +- sound/soc/intel/sst-dsp-priv.h | 136 +- sound/soc/intel/sst-dsp.c | 31 + sound/soc/intel/sst-dsp.h | 28 +- sound/soc/intel/sst-firmware.c | 937 +++++++++-- sound/soc/intel/sst-haswell-dsp.c | 295 +++- sound/soc/intel/sst-haswell-ipc.c | 413 ++++- sound/soc/intel/sst-haswell-ipc.h | 25 +- sound/soc/intel/sst-haswell-pcm.c | 419 ++++- sound/soc/intel/sst-mfld-platform-compress.c | 8 +- sound/soc/intel/sst-mfld-platform-pcm.c | 154 +- sound/soc/intel/sst-mfld-platform.h | 5 + sound/soc/intel/sst/Makefile | 7 + sound/soc/intel/sst/sst.c | 437 +++++ sound/soc/intel/sst/sst.h | 546 ++++++ sound/soc/intel/sst/sst_acpi.c | 383 +++++ sound/soc/intel/sst/sst_drv_interface.c | 686 ++++++++ sound/soc/intel/sst/sst_ipc.c | 373 +++++ sound/soc/intel/sst/sst_loader.c | 456 +++++ sound/soc/intel/sst/sst_pci.c | 209 +++ sound/soc/intel/sst/sst_pvt.c | 449 +++++ sound/soc/intel/sst/sst_stream.c | 437 +++++ sound/soc/jz4740/qi_lb60.c | 11 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 7 - sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/omap/Kconfig | 8 +- sound/soc/omap/mcbsp.c | 3 +- sound/soc/pxa/mioa701_wm9713.c | 7 +- sound/soc/pxa/pxa-ssp.c | 16 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/pxa/spitz.c | 52 +- sound/soc/rockchip/Kconfig | 9 +- sound/soc/samsung/Kconfig | 8 +- sound/soc/samsung/Makefile | 2 + sound/soc/samsung/ac97.c | 4 +- sound/soc/samsung/arndale_rt5631.c | 150 ++ sound/soc/samsung/i2s-regs.h | 10 +- sound/soc/samsung/i2s.c | 244 ++- sound/soc/samsung/odroidx2_max98090.c | 4 +- sound/soc/sh/fsi.c | 9 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/adg.c | 2 +- sound/soc/sh/rcar/core.c | 236 ++- sound/soc/sh/rcar/dvc.c | 215 +-- sound/soc/sh/rcar/gen.c | 30 +- sound/soc/sh/rcar/rsnd.h | 92 +- sound/soc/sh/rcar/src.c | 101 +- sound/soc/sh/rcar/ssi.c | 233 ++- sound/soc/soc-ac97.c | 256 +++ sound/soc/soc-cache.c | 149 -- sound/soc/soc-compress.c | 11 +- sound/soc/soc-core.c | 1768 +++----------------- sound/soc/soc-dapm.c | 755 +++++---- sound/soc/soc-jack.c | 11 +- sound/soc/soc-ops.c | 952 +++++++++++ sound/soc/soc-pcm.c | 23 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 6 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/ux500/mop500.c | 8 +- 244 files changed, 17569 insertions(+), 5861 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt create mode 100644 drivers/base/regmap/regmap-ac97.c delete mode 100644 sound/soc/atmel/snd-soc-afeb9260.c delete mode 100644 sound/soc/codecs/ad1980.h create mode 100644 sound/soc/codecs/cs4271-i2c.c create mode 100644 sound/soc/codecs/cs4271-spi.c create mode 100644 sound/soc/codecs/cs4271.h create mode 100644 sound/soc/codecs/rt5677-spi.c create mode 100644 sound/soc/codecs/rt5677-spi.h create mode 100644 sound/soc/codecs/tfa9879.c create mode 100644 sound/soc/codecs/tfa9879.h create mode 100644 sound/soc/codecs/ts3a227e.c create mode 100644 sound/soc/codecs/ts3a227e.h create mode 100644 sound/soc/intel/bytcr_dpcm_rt5640.c create mode 100644 sound/soc/intel/cht_bsw_rt5672.c create mode 100644 sound/soc/intel/sst/Makefile create mode 100644 sound/soc/intel/sst/sst.c create mode 100644 sound/soc/intel/sst/sst.h create mode 100644 sound/soc/intel/sst/sst_acpi.c create mode 100644 sound/soc/intel/sst/sst_drv_interface.c create mode 100644 sound/soc/intel/sst/sst_ipc.c create mode 100644 sound/soc/intel/sst/sst_loader.c create mode 100644 sound/soc/intel/sst/sst_pci.c create mode 100644 sound/soc/intel/sst/sst_pvt.c create mode 100644 sound/soc/intel/sst/sst_stream.c create mode 100644 sound/soc/samsung/arndale_rt5631.c create mode 100644 sound/soc/soc-ac97.c create mode 100644 sound/soc/soc-ops.c

2 4

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai/trivial: typo fix
by Nicolas Ferre 08 Dec '14

08 Dec '14

From: Matthieu Crapet <mcrapet(a)gmail.com> Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f399808a..775a2badd6a4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -267,7 +267,7 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, if (!ssc_p->dir_mask) { if (ssc_p->initialized) { /* Shutdown the SSC clock. */ - pr_debug("atmel_ssc_dau: Stopping clock\n"); + pr_debug("atmel_ssc_dai: Stopping clock\n"); clk_disable(ssc_p->ssc->clk); free_irq(ssc_p->ssc->irq, ssc_p); -- 2.1.3

2 1

[alsa-devel] [PATCH v3] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 08 Dec '14

08 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 56 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 54 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..ea5f0b7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,38 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1119,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 7

[alsa-devel] [PATCH] ASoC: wm5102: Initialize dac_comp_lock mutex
by Mark Brown 08 Dec '14

08 Dec '14

Commit d74bcaaeb6682 (ASoC: wm5102: Move ultrasonic response settings lock to the driver level) created a driver local mutex for protecting the ultrasonic response settings but neglected to initialize that mutex, causing loud complaints from lockep and potential runtime failures. Fix this by initializing the mutex. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5102.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8c99b7cf47e0..f439ae052128 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1900,6 +1900,8 @@ static int wm5102_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5102); + mutex_init(&arizona->dac_comp_lock); + wm5102->core.arizona = arizona; wm5102->core.num_inputs = 6; -- 2.1.3

2 1

[alsa-devel] [PATCH 00/15 v4] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 08 Dec '14

08 Dec '14

This patchset is the rest of patches to enable ALSA Dice driver to handle more models and to improve OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084791.h… - move mutex operations to a condition in pcm.hw_params (dice/oxfw) - fix naming regression (oxfw) == Supplemental patch for userspace library This patchset adds Hwdep functionality for OXFW driver, thus userspace library needs a patch for entries of snd_hwdep_iface_t. Please apply below patch. [alsa-devel] [PATCH alsa-lib] hwdep: add OXFW driver support http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084824.h… Takashi Sakamoto (15): ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 10 +- sound/firewire/dice/Makefile | 4 +- sound/firewire/dice/dice-midi.c | 194 ++++++++++ sound/firewire/dice/dice-pcm.c | 252 +++++++++---- sound/firewire/dice/dice-stream.c | 332 ++++++++++++---- sound/firewire/dice/dice-transaction.c | 5 - sound/firewire/dice/dice.c | 37 +- sound/firewire/dice/dice.h | 33 +- sound/firewire/oxfw/Makefile | 3 +- sound/firewire/oxfw/oxfw-command.c | 153 ++++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++++++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++++++++ sound/firewire/oxfw/oxfw-pcm.c | 425 +++++++++++++++------ sound/firewire/oxfw/oxfw-proc.c | 113 ++++++ sound/firewire/oxfw/oxfw-stream.c | 667 +++++++++++++++++++++++++++++++-- sound/firewire/oxfw/oxfw.c | 181 +++++++-- sound/firewire/oxfw/oxfw.h | 104 ++++- 19 files changed, 2539 insertions(+), 377 deletions(-) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/oxfw/oxfw-command.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-proc.c -- 2.1.0

2 19

[alsa-devel] [PATCH] sound: pci: asihpi: hpifunc.c: Remove some unused functions
by Rickard Strandqvist 07 Dec '14

07 Dec '14

Removes some functions that are not used anywhere: hpi_subsys_get_version_ex() hpi_instream_wait_start() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/hpi.h | 4 ---- sound/pci/asihpi/hpifunc.c | 27 --------------------------- 2 files changed, 31 deletions(-) diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 2088724..8338b6e 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -1164,8 +1164,6 @@ u16 hpi_stream_estimate_buffer_size(struct hpi_format *pF, /* SubSystem */ /*************/ -u16 hpi_subsys_get_version_ex(u32 *pversion_ex); - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters); u16 hpi_subsys_get_adapter(int iterator, u32 *padapter_index, @@ -1289,8 +1287,6 @@ u16 hpi_instream_read_buf(u32 h_instream, u8 *pb_read_buf, u32 bytes_to_read); u16 hpi_instream_start(u32 h_instream); -u16 hpi_instream_wait_start(u32 h_instream); - u16 hpi_instream_stop(u32 h_instream); u16 hpi_instream_reset(u32 h_instream); diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 510e56c..8fd02dd 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -93,18 +93,6 @@ static inline void hpi_send_recvV1(struct hpi_message_header *m, hpi_send_recv((struct hpi_message *)m, (struct hpi_response *)r); } -u16 hpi_subsys_get_version_ex(u32 *pversion_ex) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_SUBSYSTEM, - HPI_SUBSYS_GET_VERSION); - hpi_send_recv(&hm, &hr); - *pversion_ex = hr.u.s.data; - return hr.error; -} - u16 hpi_subsys_get_num_adapters(int *pn_num_adapters) { struct hpi_message hm; @@ -950,21 +938,6 @@ u16 hpi_instream_start(u32 h_instream) return hr.error; } -u16 hpi_instream_wait_start(u32 h_instream) -{ - struct hpi_message hm; - struct hpi_response hr; - - hpi_init_message_response(&hm, &hr, HPI_OBJ_ISTREAM, - HPI_ISTREAM_WAIT_START); - if (hpi_handle_indexes(h_instream, &hm.adapter_index, &hm.obj_index)) - return HPI_ERROR_INVALID_HANDLE; - - hpi_send_recv(&hm, &hr); - - return hr.error; -} - u16 hpi_instream_stop(u32 h_instream) { struct hpi_message hm; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 07 Dec '14

07 Dec '14

sound/soc/xtensa/xtfpga-i2s.c:666:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- xtfpga-i2s.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -663,7 +663,6 @@ static struct platform_driver xtfpga_i2s .remove = xtfpga_i2s_remove, .driver = { .name = "xtfpga-i2s", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(xtfpga_i2s_of_match), .pm = &xtfpga_i2s_pm_ops, },

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Don't resubmit pending URBs at MIDI error recovery
by Takashi Iwai 06 Dec '14

06 Dec '14

In snd_usbmidi_error_timer(), the driver tries to resubmit MIDI input URBs to reactivate the MIDI stream, but this causes the error when some of URBs are still pending like: WARNING: CPU: 0 PID: 0 at ../drivers/usb/core/urb.c:339 usb_submit_urb+0x5f/0x70() URB ef705c40 submitted while active CPU: 0 PID: 0 Comm: swapper/0 Not tainted 3.16.6-2-desktop #1 Hardware name: FOXCONN TPS01/TPS01, BIOS 080015 03/23/2010 c0984bfa f4009ed4 c078deaf f4009ee4 c024c884 c09a135c f4009f00 00000000 c0984bfa 00000153 c061ac4f c061ac4f 00000009 00000001 ef705c40 e854d1c0 f4009eec c024c8d3 00000009 f4009ee4 c09a135c f4009f00 f4009f04 c061ac4f Call Trace: [<c0205df6>] try_stack_unwind+0x156/0x170 [<c020482a>] dump_trace+0x5a/0x1b0 [<c0205e56>] show_trace_log_lvl+0x46/0x50 [<c02049d1>] show_stack_log_lvl+0x51/0xe0 [<c0205eb7>] show_stack+0x27/0x50 [<c078deaf>] dump_stack+0x45/0x65 [<c024c884>] warn_slowpath_common+0x84/0xa0 [<c024c8d3>] warn_slowpath_fmt+0x33/0x40 [<c061ac4f>] usb_submit_urb+0x5f/0x70 [<f7974104>] snd_usbmidi_submit_urb+0x14/0x60 [snd_usbmidi_lib] [<f797483a>] snd_usbmidi_error_timer+0x6a/0xa0 [snd_usbmidi_lib] [<c02570c0>] call_timer_fn+0x30/0x130 [<c0257442>] run_timer_softirq+0x1c2/0x260 [<c0251493>] __do_softirq+0xc3/0x270 [<c0204732>] do_softirq_own_stack+0x22/0x30 [<c025186d>] irq_exit+0x8d/0xa0 [<c0795228>] smp_apic_timer_interrupt+0x38/0x50 [<c0794a3c>] apic_timer_interrupt+0x34/0x3c [<c0673d9e>] cpuidle_enter_state+0x3e/0xd0 [<c028bb8d>] cpu_idle_loop+0x29d/0x3e0 [<c028bd23>] cpu_startup_entry+0x53/0x60 [<c0bfac1e>] start_kernel+0x415/0x41a For avoiding these errors, check the pending URBs and skip resubmitting such ones. Reported-and-tested-by: Stefan Seyfried <stefan.seyfried(a)googlemail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index d3d49525a16b..5bfb695547f8 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -365,6 +365,8 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { + if (atomic_read(&in->urbs[j]->use_count)) + continue; in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } -- 2.2.0

2 2

[alsa-devel] [PATCH] ASoC: samsung: Fix error handling for clock lookup
by Mark Brown 05 Dec '14

05 Dec '14

Return the error code we got from clk_get() and check to make sure that clk_prepare_enable() worked. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/i2s.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 25919e090983..500b9ba5ebaa 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) { struct i2s_dai *i2s = to_info(dai); struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai; + int ret; if (other && other->clk) { /* If this is probe on secondary */ samsung_asoc_init_dma_data(dai, &other->sec_dai->dma_playback, @@ -989,9 +990,14 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (IS_ERR(i2s->clk)) { dev_err(&i2s->pdev->dev, "failed to get i2s_clock\n"); iounmap(i2s->addr); - return -ENOENT; + return PTR_ERR(i2s->clk); + } + + ret = clk_prepare_enable(i2s->clk); + if (ret != 0) { + dev_err(&i2s->pdev->dev, "failed to enable clock: %d\n", ret); + return ret; } - clk_prepare_enable(i2s->clk); samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); -- 2.1.3

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Clocks must be prepared before enabling them. Do this in one step. Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. This fixes the following warning: ------------[ cut here ]------------ WARNING: CPU: 0 PID: 1 at drivers/clk/clk.c:895 __clk_enable+0x24/0x9c() Modules linked in: CPU: 0 PID: 1 Comm: swapper Tainted: G W 3.18.0-rc7+ #245 [<c000dce8>] (unwind_backtrace) from [<c000bcf0>] (show_stack+0x10/0x14) [<c000bcf0>] (show_stack) from [<c001664c>] (warn_slowpath_common+0x60/0x80) [<c001664c>] (warn_slowpath_common) from [<c00166fc>] (warn_slowpath_null+0x18/0x20) [<c00166fc>] (warn_slowpath_null) from [<c02fd7ac>] (__clk_enable+0x24/0x9c) [<c02fd7ac>] (__clk_enable) from [<c02fdbb4>] (clk_enable+0x18/0x2c) [<c02fdbb4>] (clk_enable) from [<c0322688>] (atmel_ac97c_probe+0x154/0x694) [<c0322688>] (atmel_ac97c_probe) from [<c0235e08>] (platform_drv_probe+0x48/0x94) [<c0235e08>] (platform_drv_probe) from [<c02345f8>] (driver_probe_device+0x138/0x350) [<c02345f8>] (driver_probe_device) from [<c02348bc>] (__driver_attach+0x68/0x8c) [<c02348bc>] (__driver_attach) from [<c0232bd0>] (bus_for_each_dev+0x70/0x84) [<c0232bd0>] (bus_for_each_dev) from [<c0233cd8>] (bus_add_driver+0xfc/0x1f8) [<c0233cd8>] (bus_add_driver) from [<c0234f0c>] (driver_register+0x9c/0xe0) [<c0234f0c>] (driver_register) from [<c0008ac4>] (do_one_initcall+0x110/0x1c8) [<c0008ac4>] (do_one_initcall) from [<c053cd58>] (kernel_init_freeable+0xf8/0x1b8) [<c053cd58>] (kernel_init_freeable) from [<c03c0414>] (kernel_init+0x8/0xe4) [<c03c0414>] (kernel_init) from [<c00096d0>] (ret_from_fork+0x14/0x24) ---[ end trace cb88537fdc8fa201 ]--- atmel_ac97c fffa0000.sound: AC'97 0 does not respond - RESET atmel_ac97c fffa0000.sound: AC'97 0 access is not valid [0xffffffff], removing mixer. ------------[ cut here ]------------ Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 86f07e9..fa283d3 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 2

[alsa-devel] [PATCH v7] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 05 Dec '14

05 Dec '14

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 154 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 7 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3a4d7da..9b62457 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -533,6 +534,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130..db60701 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -942,6 +942,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56..f4956e3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1292,7 +1292,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1304,7 +1305,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9c..7e961ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -754,6 +754,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* skip control creation for links with 1 config */ + if (w->num_params == 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* reset 'walked' bit for each dapm path */ static void dapm_clear_walk_output(struct snd_soc_dapm_context *dapm, struct list_head *sink) @@ -2721,6 +2751,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3206,6 +3239,9 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, source = source_p->source->priv; sink = sink_p->sink->priv; + /* Select the configuration set by alsa control */ + config = &config[w->params_select]; + /* Be a little careful as we don't want to overflow the mask array */ if (config->formats) { fmt = ffs(config->formats) - 1; @@ -3274,8 +3310,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3283,14 +3350,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } + snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3298,6 +3403,28 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3305,15 +3432,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + kfree(w); +outfree_kcontrol_news: + kfree(template.kcontrol_news); +outfree_private_value: + kfree((void *)private_value); +outfree_link_name: + kfree(link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + kfree(w_param_text[count]); + kfree(w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] usb-audio: I'd like to change two things in usb audio, but I'm not sure if it's right
by Johan Aires Rastén 05 Dec '14

05 Dec '14

So I'll try again, this time with the proper cc:s and a second question. I've come across two issues in the usb audio module that are causing problems with my sound card, but I'm thinking that changing them might break something for some other card. In sound/usb/mixer.c function check_input_term(...), for the case UAC_INPUT_TERMINAL and UAC_VERSION_2: The recursive call to get clock selectors messes up the naming of my terminals. At this stage the terminals are already properly named Mic and Line but the call overwrites them both with the unnamed clock selector "Unit 22". Are other UAC2 cards different, so that the clock selector contains the correct terminal name, or is there some other reasoning here that I'm not seeing? Could the call be removed, or perhaps only done if no terminal name has been found yet? In sound/usb/stream.c function snd_usb_add_audio_stream(...), near the end of the function: A newly created pcm is added with "list_add(&as->list, &chip->pcm_list);" which causes devices to be listed in reverse order, i.e. device 0 will be at the end of the list, and this leads to that capture streams end up in the wrong device under certain circumstances. The problem arises when the code under "look for an empty stream" earlier in the same function executes. It will attach the capture stream to the first compatible existing stream, but since they are reversed it will find the highest numbered device. In my case analog input will be grouped with digital out (device 1) instead of analog out (device 0). This is assuming that playback streams are defined before capture streams but it could happen in the other direction as long as there are several streams of one type. I also suspect that this is what causes the M-Audio Audiophile USB to have swapped inputs, for which there's a fix in USB-Audio.conf ( http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/USB-Aud… ). Thanks to debianuser on IRC for helping me find it. I would like to change this call to use list_add_tail for adding new items instead, but I'm worried that there are quirks or other fixes that expect the current behaviour. There are very few other references to pcm_list and none of them seem to care about ordering. Sorry for being bad at referencing code, I'm new at this. //Johan Rastén

2 2

[alsa-devel] [PATCH 1/2] ALSA: sound/atmel/ac97c.c: Add missing clock prepare
by Alexander Stein 05 Dec '14

05 Dec '14

Replace clk_enable with clk_prepare_enable and clk_disable with clk_disable_unprepare. Signed-off-by: Alexander Stein <alexanders83(a)web.de> --- sound/atmel/ac97c.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b2a3225..ff2f743 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -989,7 +989,7 @@ static int atmel_ac97c_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); } - clk_enable(pclk); + clk_prepare_enable(pclk); retval = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, @@ -1167,7 +1167,7 @@ err_ioremap: err_request_irq: snd_card_free(card); err_snd_card_new: - clk_disable(pclk); + clk_disable_unprepare(pclk); clk_put(pclk); return retval; } @@ -1184,7 +1184,7 @@ static int atmel_ac97c_suspend(struct device *pdev) if (test_bit(DMA_TX_READY, &chip->flags)) dw_dma_cyclic_stop(chip->dma.tx_chan); } - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); return 0; } @@ -1194,7 +1194,7 @@ static int atmel_ac97c_resume(struct device *pdev) struct snd_card *card = dev_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - clk_enable(chip->pclk); + clk_prepare_enable(chip->pclk); if (cpu_is_at32ap7000()) { if (test_bit(DMA_RX_READY, &chip->flags)) dw_dma_cyclic_start(chip->dma.rx_chan); @@ -1222,7 +1222,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) ac97c_writel(chip, COMR, 0); ac97c_writel(chip, MR, 0); - clk_disable(chip->pclk); + clk_disable_unprepare(chip->pclk); clk_put(chip->pclk); iounmap(chip->regs); free_irq(chip->irq, chip); -- 2.2.0

2 3

[alsa-devel] DMA buffer alignment
by Carlo Caione 05 Dec '14

05 Dec '14

Hi, I'm working with a SoC sound hw that requires that all the data written into the DMA memory is aligned a 64 byte boundary. Actually the alignment is required since the format of the data into the DMA memory is (for a 2 channel 16bit audio): 16 samples for the left channel followed by 16 samples for the right channel. If I try to write anything that is not aligned to the 64 bytes I got audio corruption (high pitched noise). Several user-space programs (like aplay) work well with my hw, since the buffer sent from the user-space is always (padded) of period-size bytes, that is aligned to my 64 bytes boundary. Others (like speaker-test) are not working because sometimes they snd_pcm_writei() a buffer that is not multiple of 64 bytes. In general every time I write into the DMA area a buffer that is not aligned, my hw pointer is not aligned anymore with the 64 bytes and the noise starts. IIUC it is not possible to force the user-space programs to send buffers with a fixed size so the problem should be solved in kernel-space (am I wrong on this point?). Any suggestion on how to tackle this problem? Thanks, -- Carlo Caione

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[alsa-devel] [PATCH v3] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 05 Dec '14

05 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Only use the resolution supported, ignoring the LSB of the volume register. This results in half the steps and 0.75dB per step, but reports accurate levels through TLV. Update the masks to reflect that these are registers have the LSB ignored. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 22 +++++++++++----------- sound/soc/codecs/rt5677.h | 24 ++++++++++++------------ 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 133010d..81fe146 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -763,9 +763,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); @@ -817,13 +817,13 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5677_DAC1_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC2 Playback Volume", RT5677_DAC2_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC3 Playback Volume", RT5677_DAC3_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), SOC_DOUBLE_TLV("DAC4 Playback Volume", RT5677_DAC4_DIG_VOL, - RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 175, 0, dac_vol_tlv), + RT5677_L_VOL_SFT, RT5677_R_VOL_SFT, 87, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5677_IN1, RT5677_BST_SFT1, 8, 0, bst_tlv), @@ -842,19 +842,19 @@ static const struct snd_kcontrol_new rt5677_snd_controls[] = { RT5677_L_MUTE_SFT, RT5677_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC1 Capture Volume", RT5677_STO1_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC2 Capture Volume", RT5677_STO2_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC3 Capture Volume", RT5677_STO3_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("ADC4 Capture Volume", RT5677_STO4_ADC_DIG_VOL, - RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 127, 0, + RT5677_STO1_ADC_L_VOL_SFT, RT5677_STO1_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5677_MONO_ADC_DIG_VOL, - RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 127, 0, + RT5677_MONO_ADC_L_VOL_SFT, RT5677_MONO_ADC_R_VOL_SFT, 63, 0, adc_vol_tlv), /* Sidetone Control */ diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index dbd9ffd..c0a625f 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -306,10 +306,10 @@ #define RT5677_R_MUTE_SFT 7 #define RT5677_VOL_R_MUTE (0x1 << 6) #define RT5677_VOL_R_SFT 6 -#define RT5677_L_VOL_MASK (0x3f << 8) -#define RT5677_L_VOL_SFT 8 -#define RT5677_R_VOL_MASK (0x3f) -#define RT5677_R_VOL_SFT 0 +#define RT5677_L_VOL_MASK (0x7f << 9) +#define RT5677_L_VOL_SFT 9 +#define RT5677_R_VOL_MASK (0x7f << 1) +#define RT5677_R_VOL_SFT 1 /* LOUT1 Control (0x01) */ #define RT5677_LOUT1_L_MUTE (0x1 << 15) @@ -447,16 +447,16 @@ #define RT5677_SEL_DAC2_R_SRC_SFT 0 /* Stereo1 ADC Digital Volume Control (0x1c) */ -#define RT5677_STO1_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_STO1_ADC_L_VOL_SFT 8 -#define RT5677_STO1_ADC_R_VOL_MASK (0x7f) -#define RT5677_STO1_ADC_R_VOL_SFT 0 +#define RT5677_STO1_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_STO1_ADC_L_VOL_SFT 9 +#define RT5677_STO1_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_STO1_ADC_R_VOL_SFT 1 /* Mono ADC Digital Volume Control (0x1d) */ -#define RT5677_MONO_ADC_L_VOL_MASK (0x7f << 8) -#define RT5677_MONO_ADC_L_VOL_SFT 8 -#define RT5677_MONO_ADC_R_VOL_MASK (0x7f) -#define RT5677_MONO_ADC_R_VOL_SFT 0 +#define RT5677_MONO_ADC_L_VOL_MASK (0x3f << 9) +#define RT5677_MONO_ADC_L_VOL_SFT 9 +#define RT5677_MONO_ADC_R_VOL_MASK (0x3f << 1) +#define RT5677_MONO_ADC_R_VOL_SFT 1 /* Stereo 1/2 ADC Boost Gain Control (0x1e) */ #define RT5677_STO1_ADC_L_BST_MASK (0x3 << 14) -- 2.2.0.rc0.207.ga3a616c

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[alsa-devel] 'No audio' bug on lenovo ultra dock
by Alexander Lochmann 05 Dec '14

05 Dec '14

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! The current linux kernel (3.17 as well as 3.18-rc7) does not support the audio output on a Lenovo Ultra Dock in conjunction with the X240. The issue is discussed in [1], which provides a patch as well. The patch is available at [2]. I'm asking you for integrating it into your linux tree. So it might be merged into the mainline kernel during the current or the next development phase. Thanks in advance! Regards, Alex [1] https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1385680 [2] https://launchpadlibrarian.net/188362011/lenovo-ultradock.patch - -- Technische Universität Dortmund Alexander Lochmann PGP key: 0xBC3EF6FD Otto-Hahn-Str. 16 phone: +49.231.7556141 D-44227 Dortmund fax: +49.231.7556116 http://ess.cs.tu-dortmund.de/Staff/al -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJUgXibAAoJEFk+7QW8Pvb9zSEP/0yezuJXBK9P8ejhfrmH7VxK PPNhsALQmgMa5vTPsepzBsbj5060yiyI8BTkt76h5d6QlLy3eKq8+AE4+hkGJNie sZtvbWPr8C0GP+8lnfRz6EFOdnWdXbEc4x1hSd7JVDgpNynzHvYZQgISVnxA1rFe NTqngYzFM8yBhUEKc0DYzvuCTw9J0VODWro9eD6tjuSI7/yyQmrLln7XihcLs3/3 VWcNYVBDzQOqjwOTJM7wPfprGgalr7vOBWo3WyshBnKcXmsnahswB+HSsiVWqtMd SZBoEfm811ocvtBFGfsuMkWJALtUfw3bBEPm2iV5jzKmK1CoCxCAuqTcUkiNEzCz m6f76LEtGzq10vpXsc9FmX1Mlbrt9mavE2a5cxoaxVEPJ2OVpwAU3ufgguwUkIrt L8E4RjBEHwttDbe4CPz1YSNz4j+OD4bxsRVjQGfpk8mVGTh0ljtno1chqxn0HZ5J +9Hto37DqDzywJnc4gYyhxTcAZ669qvdMow0avyPuEwGK///C9qgkGCgrm0W36Oq u/HU2lz9A6Vqxtzr8t4bVVEhiX+J2+HQAyf2OPx7o4iMyLAFoxvvWRZUJ3qvP1ZQ OlhXIHg22xTiy9WsVLiStSbT8xd6lV1UP/HotUwOjEoQLzv8pVh/5DH3fwTX5cIw jvRMnfYW+6Ai6Nn07Frj =Y9Kt -----END PGP SIGNATURE-----

1 0

[alsa-devel] [PATCH v2 0/2] fsl_ssi misc fix
by Jiada Wang 05 Dec '14

05 Dec '14

Hi, Changes in v2: - fix error path in probe - replace irq_of_parse_and_map with platform_get_irq Changes in v1: - free IRQ before irq_dispose_mapping Jiada Wang (2): ASoC: fsl_ssi: fix error path in probe ASoC: fsl_ssi: use platform_get_irq instead of irq_of_parse_and_map sound/soc/fsl/fsl_ssi.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) -- 1.9.3

2 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix module unbound
by Fabio Estevam 05 Dec '14

05 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Trying to remove the snd-soc-fsl-ssi module leads to the following warning: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- The cause of problem and solution are well explained by Lars-Peter: "The driver creates the mapping by calling irq_of_parse_and_map(), so it also has to dispose the mapping. But the easy way out is to simply use platform_get_irq() instead of irq_of_parse_map(). In this case the mapping is not managed by the device but by the of core, so the device has not to dispose the mapping." Tested on a imx6q-sabresd board. Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 32a31d9..c528f16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1361,7 +1361,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(ssi_private->regs); } - ssi_private->irq = irq_of_parse_and_map(np, 0); + ssi_private->irq = platform_get_irq(pdev, 0); if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; @@ -1387,7 +1387,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ssi_private->soc->imx) { ret = fsl_ssi_imx_probe(pdev, ssi_private, iomem); if (ret) - goto error_irqmap; + return ret; } ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, @@ -1458,10 +1458,6 @@ error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -error_irqmap: - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return ret; } @@ -1478,9 +1474,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); - if (ssi_private->use_dma) - irq_dispose_mapping(ssi_private->irq); - return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: soc-pcm: do not hw_free BE if it's still used
by Qiao Zhou 05 Dec '14

05 Dec '14

Do not free BE hw if it's still used by other FE during dpcm runtime shutdown. Otherwise the BE runtime state will be STATE_HW_FREE and won't be updated to STATE_CLOSE when shutdown ends, because BE dai shutdown function won't close pcm when detecting BE is still under use. With STATE_HW_FREE, BE can't be triggered start again. This corner case can easily appear when one BE is used by two FE, without this patch "ASoC: dpcm: Fix race between FE/BE updates and trigger"(ea9d0d771fcd32cd56070819749477d511ec9117). One FE tries to shutdown but it's raced against xrun on another FE. It improves the be dai hw_free logic. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 57277dd..70e8088 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1664,6 +1664,10 @@ int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) continue; + /* do not free hw if this BE is used by other FE */ + if (be->dpcm[stream].users > 1) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Augment existing card DAPM routes in snd_soc_of_parse_audio_routing
by Peter Rosin 05 Dec '14

05 Dec '14

From: Peter Rosin <peda(a)axentia.se> If a snd_soc_card has any DAPM routes when it calls snd_soc_of_parse_audio_routing, those are clobbered without this change. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-core.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b60ff56ebc0f..377db486852e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4585,7 +4585,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { struct device_node *np = card->dev->of_node; - int num_routes; + int num_routes, old_routes; struct snd_soc_dapm_route *routes; int i, ret; @@ -4603,7 +4603,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } - routes = devm_kzalloc(card->dev, num_routes * sizeof(*routes), + old_routes = card->num_dapm_routes; + routes = devm_kzalloc(card->dev, + (old_routes + num_routes) * sizeof(*routes), GFP_KERNEL); if (!routes) { dev_err(card->dev, @@ -4611,9 +4613,11 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } + memcpy(routes, card->dapm_routes, old_routes * sizeof(*routes)); + for (i = 0; i < num_routes; i++) { ret = of_property_read_string_index(np, propname, - 2 * i, &routes[i].sink); + 2 * i, &routes[old_routes + i].sink); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4621,7 +4625,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, return -EINVAL; } ret = of_property_read_string_index(np, propname, - (2 * i) + 1, &routes[i].source); + (2 * i) + 1, &routes[old_routes + i].source); if (ret) { dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", @@ -4630,7 +4634,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; + card->num_dapm_routes += num_routes; card->dapm_routes = routes; return 0; -- 1.7.10.4

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Add ACPI device probing
by Ben Zhang 04 Dec '14

04 Dec '14

The rt5677 codec driver looks for ACPI device ID "RT5677CE", which is specified in coreboot. This patch allows platform data to be obtained via ACPI Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/rt5677.c | 52 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 50 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d317c68..384281d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -20,6 +20,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <linux/firmware.h> #include <linux/gpio.h> #include <sound/core.h> @@ -4525,6 +4526,43 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np) return 0; } +#ifdef CONFIG_ACPI + +static unsigned long long rt5677_parse_acpi_entry(struct device *dev, + acpi_string name) +{ + acpi_handle handle = ACPI_HANDLE(dev); + unsigned long long val; + acpi_status status; + + status = acpi_evaluate_integer(handle, name, NULL, &val); + if (ACPI_FAILURE(status)) { + dev_err(dev, "Failed to parse ACPI entry %s, default to 0: %d\n", + name, status); + return 0; + } + return val; +} + +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ + rt5677->pdata.dmic2_clk_pin = (enum rt5677_dmic2_clk) + rt5677_parse_acpi_entry(dev, "DCLK"); + rt5677->pdata.in1_diff = (bool)rt5677_parse_acpi_entry(dev, "IN1"); + rt5677->pdata.in2_diff = (bool)rt5677_parse_acpi_entry(dev, "IN2"); + rt5677->pdata.lout1_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT1"); + rt5677->pdata.lout2_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT2"); + rt5677->pdata.lout3_diff = (bool)rt5677_parse_acpi_entry(dev, "OUT3"); + rt5677->pdata.jd1_gpio = rt5677_parse_acpi_entry(dev, "JD1"); + rt5677->pdata.jd2_gpio = rt5677_parse_acpi_entry(dev, "JD2"); + rt5677->pdata.jd3_gpio = rt5677_parse_acpi_entry(dev, "JD3"); +} +#else +static void rt5677_parse_acpi(struct rt5677_priv *rt5677, struct device *dev) +{ +} +#endif + static struct regmap_irq rt5677_irqs[] = { [RT5677_IRQ_JD1] = { .reg_offset = 0, @@ -4604,6 +4642,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (pdata) rt5677->pdata = *pdata; + rt5677->pow_ldo2 = -EINVAL; if (i2c->dev.of_node) { ret = rt5677_parse_dt(rt5677, i2c->dev.of_node); if (ret) { @@ -4611,8 +4650,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, ret); return ret; } - } else { - rt5677->pow_ldo2 = -EINVAL; + } else if (ACPI_HANDLE(&i2c->dev)) { + rt5677_parse_acpi(rt5677, &i2c->dev); } if (gpio_is_valid(rt5677->pow_ldo2)) { @@ -4708,10 +4747,19 @@ static int rt5677_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5677_acpi_id[] = { + { "RT5677CE", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, rt5677_acpi_id); +#endif + static struct i2c_driver rt5677_i2c_driver = { .driver = { .name = "rt5677", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5677_acpi_id), }, .probe = rt5677_i2c_probe, .remove = rt5677_i2c_remove, -- 2.1.0.rc2.206.gedb03e5

6 35

[alsa-devel] [PATCH v2] ASoC: add xtensa xtfpga I2S interface and platform
by Max Filippov 04 Dec '14

04 Dec '14

XTFPGA boards provides an audio subsystem that consists of TI CDCE706 clock synthesizer, I2S transmitter and TLV320AIC23 audio codec. I2S transmitter has MMIO-based interface that resembles that of the OpenCores I2S transmitter. I2S transmitter is always a master on I2S bus. There's no specialized audio DMA, sample data are transferred to I2S transmitter FIFO by CPU through memory-mapped queue interface. Signed-off-by: Max Filippov <jcmvbkbc(a)gmail.com> --- Changes v1->v2: - fix module name in the sound/soc/xtensa/Makefile; - split PCM data transfer code and I2S interface setup code; - move clock enabling/disabling to runtime PM code; - document I2S interface RATIO field; - rewrite I2S interface LEVEL field calculation to avoid overflows; - drop select statements from Kconfig; - document RCU usage for tx_substream; - clean up code running in RCU read-side critical section; - use threaded IRQ handler instead of IRQ + tasklet; - remove empty PCM callback handlers; - add const to static structures where applicable; - use platform_set_drvdata instead of dev_set_drvdata; - drop redundant error reporting code; - use platform_get_irq instead of platform_get_resource; - clean up xtfpga_i2s_remove; - add comments. .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + MAINTAINERS | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 676 +++++++++++++++++++++ 7 files changed, 707 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c diff --git a/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt new file mode 100644 index 0000000..befd125 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt @@ -0,0 +1,18 @@ +Bindings for I2S controller built into xtfpga Xtensa bitstreams. + +Required properties: +- compatible: shall be "cdns,xtfpga-i2s". +- reg: memory region (address and length) with device registers. +- interrupts: interrupt for the device. +- clocks: phandle to the clk used as master clock. I2S bus clock + is derived from it. + +Examples: + + i2s0: xtfpga-i2s@0d080000 { + #sound-dai-cells = <0>; + compatible = "cdns,xtfpga-i2s"; + reg = <0x0d080000 0x40>; + interrupts = <2 1>; + clocks = <&cdce706 4>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index acb4523..d7bfeec 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10204,6 +10204,7 @@ M: Max Filippov <jcmvbkbc(a)gmail.com> L: linux-xtensa(a)linux-xtensa.org S: Maintained F: drivers/spi/spi-xtensa-xtfpga.c +F: sound/soc/xtensa/xtfpga-i2s.c YAM DRIVER FOR AX.25 M: Jean-Paul Roubelat <jpr(a)f6fbb.org> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 0e96233..9ca12e7 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -56,6 +56,7 @@ source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" +source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 534714a..99d7222 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -33,3 +33,4 @@ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ +obj-$(CONFIG_SND_SOC) += xtensa/ diff --git a/sound/soc/xtensa/Kconfig b/sound/soc/xtensa/Kconfig new file mode 100644 index 0000000..c201beb --- /dev/null +++ b/sound/soc/xtensa/Kconfig @@ -0,0 +1,7 @@ +config SND_SOC_XTFPGA_I2S + tristate "XTFPGA I2S master" + select REGMAP_MMIO + help + Say Y or M if you want to add support for codecs attached to the + I2S interface on XTFPGA daughter board. You will also need to select + the drivers for the rest of XTFPGA audio subsystem. diff --git a/sound/soc/xtensa/Makefile b/sound/soc/xtensa/Makefile new file mode 100644 index 0000000..15efbf9 --- /dev/null +++ b/sound/soc/xtensa/Makefile @@ -0,0 +1,3 @@ +snd-soc-xtfpga-i2s-objs := xtfpga-i2s.o + +obj-$(CONFIG_SND_SOC_XTFPGA_I2S) += snd-soc-xtfpga-i2s.o diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c new file mode 100644 index 0000000..bd66be2 --- /dev/null +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -0,0 +1,676 @@ +/* + * Xtfpga I2S controller driver + * + * Copyright (c) 2014 Cadence Design Systems Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/io.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define DRV_NAME "xtfpga-i2s" + +#define XTFPGA_I2S_VERSION 0x00 +#define XTFPGA_I2S_CONFIG 0x04 +#define XTFPGA_I2S_INT_MASK 0x08 +#define XTFPGA_I2S_INT_STATUS 0x0c +#define XTFPGA_I2S_CHAN0_DATA 0x10 +#define XTFPGA_I2S_CHAN1_DATA 0x14 +#define XTFPGA_I2S_CHAN2_DATA 0x18 +#define XTFPGA_I2S_CHAN3_DATA 0x1c + +#define XTFPGA_I2S_CONFIG_TX_ENABLE 0x1 +#define XTFPGA_I2S_CONFIG_INT_ENABLE 0x2 +#define XTFPGA_I2S_CONFIG_LEFT 0x4 +#define XTFPGA_I2S_CONFIG_RATIO_BASE 8 +#define XTFPGA_I2S_CONFIG_RATIO_MASK 0x0000ff00 +#define XTFPGA_I2S_CONFIG_RES_BASE 16 +#define XTFPGA_I2S_CONFIG_RES_MASK 0x003f0000 +#define XTFPGA_I2S_CONFIG_LEVEL_BASE 24 +#define XTFPGA_I2S_CONFIG_LEVEL_MASK 0x0f000000 +#define XTFPGA_I2S_CONFIG_CHANNEL_BASE 28 + +#define XTFPGA_I2S_INT_UNDERRUN 0x1 +#define XTFPGA_I2S_INT_LEVEL 0x2 +#define XTFPGA_I2S_INT_VALID 0x3 + +#define XTFPGA_I2S_FIFO_SIZE 8192 + +/* + * I2S controller operation: + * + * Enabling TX: output 1 period of zeros (starting with left channel) + * and then queued data. + * + * Level status and interrupt: whenever FIFO level is below FIFO trigger, + * level status is 1 and an IRQ is asserted (if enabled). + * + * Underrun status and interrupt: whenever FIFO is empty, underrun status + * is 1 and an IRQ is asserted (if enabled). + */ +struct xtfpga_i2s { + struct device *dev; + struct clk *clk; + struct regmap *regmap; + void __iomem *regs; + + /* current playback substream. NULL if not playing. + * + * Access to that field is synchronized between the interrupt handler + * and userspace through RCU. + * + * Interrupt handler (threaded part) does PIO on substream data in RCU + * read-side critical section. Trigger callback sets and clears the + * pointer when the playback is started and stopped with + * rcu_assign_pointer. When userspace is about to free the playback + * stream in the pcm_close callback it synchronizes with the interrupt + * handler by means of synchronize_rcu call. + */ + struct snd_pcm_substream *tx_substream; + unsigned (*tx_fn)(struct xtfpga_i2s *i2s, + struct snd_pcm_runtime *runtime, + unsigned tx_ptr); + unsigned tx_ptr; /* next frame index in the sample buffer */ + + /* current fifo level estimate. + * Doesn't have to be perfectly accurate, but must be not less than + * the actual FIFO level in order to avoid stall on push attempt. + */ + unsigned tx_fifo_level; + + /* FIFO level at which level interrupt occurs */ + unsigned tx_fifo_low; + + /* maximal FIFO level */ + unsigned tx_fifo_high; +}; + +static bool xtfpga_i2s_wr_reg(struct device *dev, unsigned int reg) +{ + return reg >= XTFPGA_I2S_CONFIG; +} + +static bool xtfpga_i2s_rd_reg(struct device *dev, unsigned int reg) +{ + return reg < XTFPGA_I2S_CHAN0_DATA; +} + +static bool xtfpga_i2s_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == XTFPGA_I2S_INT_STATUS; +} + +static const struct regmap_config xtfpga_i2s_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = XTFPGA_I2S_CHAN3_DATA, + .writeable_reg = xtfpga_i2s_wr_reg, + .readable_reg = xtfpga_i2s_rd_reg, + .volatile_reg = xtfpga_i2s_volatile_reg, + .cache_type = REGCACHE_FLAT, +}; + +/* Generate functions that do PIO from TX DMA area to FIFO for all supported + * stream formats. + * Functions will be called xtfpga_pcm_tx_<channels>x<sample bits>, e.g. + * xtfpga_pcm_tx_2x16 for 16-bit stereo. + * + * FIFO consists of 32-bit words, one word per channel, always 2 channels. + * If I2S interface is configured with smaller sample resolution, only + * the LSB of each word is used. + */ +#define xtfpga_pcm_tx_fn(channels, sample_bits) \ +static unsigned xtfpga_pcm_tx_##channels##x##sample_bits( \ + struct xtfpga_i2s *i2s, struct snd_pcm_runtime *runtime, \ + unsigned tx_ptr) \ +{ \ + const u##sample_bits (*p)[channels] = \ + (void *)runtime->dma_area; \ +\ + for (; i2s->tx_fifo_level < i2s->tx_fifo_high; \ + i2s->tx_fifo_level += 2) { \ + iowrite32(p[tx_ptr][0], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + iowrite32(p[tx_ptr][channels - 1], \ + i2s->regs + XTFPGA_I2S_CHAN0_DATA); \ + if (++tx_ptr >= runtime->buffer_size) \ + tx_ptr = 0; \ + } \ + return tx_ptr; \ +} + +xtfpga_pcm_tx_fn(1, 16) +xtfpga_pcm_tx_fn(2, 16) +xtfpga_pcm_tx_fn(1, 32) +xtfpga_pcm_tx_fn(2, 32) + +#undef xtfpga_pcm_tx_fn + +static bool xtfpga_pcm_push_tx(struct xtfpga_i2s *i2s) +{ + struct snd_pcm_substream *tx_substream; + bool tx_active; + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + tx_active = tx_substream && snd_pcm_running(tx_substream); + if (tx_active) { + unsigned tx_ptr = ACCESS_ONCE(i2s->tx_ptr); + unsigned new_tx_ptr = i2s->tx_fn(i2s, tx_substream->runtime, + tx_ptr); + + cmpxchg(&i2s->tx_ptr, tx_ptr, new_tx_ptr); + } + rcu_read_unlock(); + + return tx_active; +} + +static void xtfpga_pcm_refill_fifo(struct xtfpga_i2s *i2s) +{ + unsigned int_status; + unsigned i; + + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + for (i = 0; i < 2; ++i) { + bool tx_active = xtfpga_pcm_push_tx(i2s); + + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + if (tx_active) + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, + &int_status); + + if (!tx_active || + !(int_status & XTFPGA_I2S_INT_LEVEL)) + break; + + /* After the push the level IRQ is still asserted, + * means FIFO level is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + if (!(int_status & XTFPGA_I2S_INT_LEVEL)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_VALID); + else if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, + XTFPGA_I2S_INT_UNDERRUN); + + if (!(int_status & XTFPGA_I2S_INT_UNDERRUN)) + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE); + else + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_INT_ENABLE | + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); +} + +static irqreturn_t xtfpga_i2s_threaded_irq_handler(int irq, void *dev_id) +{ + struct xtfpga_i2s *i2s = dev_id; + struct snd_pcm_substream *tx_substream; + unsigned config, int_status, int_mask; + + regmap_read(i2s->regmap, XTFPGA_I2S_CONFIG, &config); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_MASK, &int_mask); + regmap_read(i2s->regmap, XTFPGA_I2S_INT_STATUS, &int_status); + + if (!(config & XTFPGA_I2S_CONFIG_INT_ENABLE) || + !(int_status & int_mask & XTFPGA_I2S_INT_VALID)) + return IRQ_NONE; + + /* Update FIFO level estimate in accordance with interrupt status + * register. + */ + if (int_status & XTFPGA_I2S_INT_UNDERRUN) { + i2s->tx_fifo_level = 0; + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_TX_ENABLE, 0); + } else { + /* The FIFO isn't empty, but is below tx_fifo_low. Estimate + * it as tx_fifo_low. + */ + i2s->tx_fifo_level = i2s->tx_fifo_low; + } + + rcu_read_lock(); + tx_substream = rcu_dereference(i2s->tx_substream); + + if (tx_substream && snd_pcm_running(tx_substream)) { + snd_pcm_period_elapsed(tx_substream); + if (int_status & XTFPGA_I2S_INT_UNDERRUN) + dev_dbg_ratelimited(i2s->dev, "%s: underrun\n", + __func__); + } + rcu_read_unlock(); + + /* Refill FIFO, update allowed IRQ reasons, enable IRQ if FIFO is + * not empty. + */ + xtfpga_pcm_refill_fifo(i2s); + + return IRQ_HANDLED; +} + +static int xtfpga_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_set_dma_data(dai, substream, i2s); + return 0; +} + +static int xtfpga_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct xtfpga_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned srate = params_rate(params); + unsigned channels = params_channels(params); + unsigned period_size = params_period_size(params); + unsigned sample_size = snd_pcm_format_width(params_format(params)); + unsigned freq, ratio, level; + int err; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RES_MASK, + sample_size << XTFPGA_I2S_CONFIG_RES_BASE); + + freq = 256 * srate; + err = clk_set_rate(i2s->clk, freq); + if (err < 0) + return err; + + /* ratio field of the config register controls MCLK->I2S clock + * derivation: I2S clock = MCLK / (2 * (ratio + 2)). + * + * So with MCLK = 256 * sample rate ratio is 0 for 32 bit stereo + * and 2 for 16 bit stereo. + */ + ratio = (freq - (srate * sample_size * 8)) / + (srate * sample_size * 4); + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_RATIO_MASK, + ratio << XTFPGA_I2S_CONFIG_RATIO_BASE); + + i2s->tx_fifo_low = XTFPGA_I2S_FIFO_SIZE / 2; + + /* period_size * 2: FIFO always gets 2 samples per frame */ + for (level = 1; + i2s->tx_fifo_low / 2 >= period_size * 2 && + level < (XTFPGA_I2S_CONFIG_LEVEL_MASK >> + XTFPGA_I2S_CONFIG_LEVEL_BASE); ++level) + i2s->tx_fifo_low /= 2; + + i2s->tx_fifo_high = 2 * i2s->tx_fifo_low; + + regmap_update_bits(i2s->regmap, XTFPGA_I2S_CONFIG, + XTFPGA_I2S_CONFIG_LEVEL_MASK, + level << XTFPGA_I2S_CONFIG_LEVEL_BASE); + + dev_dbg(i2s->dev, + "%s srate: %u, channels: %u, sample_size: %u, period_size: %u\n", + __func__, srate, channels, sample_size, period_size); + dev_dbg(i2s->dev, "%s freq: %u, ratio: %u, level: %u\n", + __func__, freq, ratio, level); + + return 0; +} + +static int xtfpga_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + return -EINVAL; + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) + return -EINVAL; + + return 0; +} + +/* PCM */ + +static const struct snd_pcm_hardware xtfpga_pcm_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 2, + .period_bytes_max = XTFPGA_I2S_FIFO_SIZE / 2 * 8, + .periods_min = 2, + .periods_max = XTFPGA_I2S_FIFO_SIZE * 8 / 2, + .buffer_bytes_max = XTFPGA_I2S_FIFO_SIZE * 8, + .fifo_size = 16, +}; + +static int xtfpga_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + void *p; + + snd_soc_set_runtime_hwparams(substream, &xtfpga_pcm_hardware); + p = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + runtime->private_data = p; + + return 0; +} + +static int xtfpga_pcm_close(struct snd_pcm_substream *substream) +{ + synchronize_rcu(); + return 0; +} + +static int xtfpga_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + unsigned channels = params_channels(hw_params); + + switch (channels) { + case 1: + case 2: + break; + + default: + return -EINVAL; + + } + + switch (params_format(hw_params)) { + case SNDRV_PCM_FORMAT_S16_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x16 : + xtfpga_pcm_tx_2x16; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + i2s->tx_fn = (channels == 1) ? + xtfpga_pcm_tx_1x32 : + xtfpga_pcm_tx_2x32; + break; + + default: + return -EINVAL; + } + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + return ret; +} + +static int xtfpga_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ACCESS_ONCE(i2s->tx_ptr) = 0; + rcu_assign_pointer(i2s->tx_substream, substream); + xtfpga_pcm_refill_fifo(i2s); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + rcu_assign_pointer(i2s->tx_substream, NULL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static snd_pcm_uframes_t xtfpga_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct xtfpga_i2s *i2s = runtime->private_data; + snd_pcm_uframes_t pos = ACCESS_ONCE(i2s->tx_ptr); + + return pos < runtime->buffer_size ? pos : 0; +} + +static int xtfpga_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + size_t size = xtfpga_pcm_hardware.buffer_bytes_max; + + return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, size, size); +} + +static void xtfpga_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static const struct snd_pcm_ops xtfpga_pcm_ops = { + .open = xtfpga_pcm_open, + .close = xtfpga_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = xtfpga_pcm_hw_params, + .trigger = xtfpga_pcm_trigger, + .pointer = xtfpga_pcm_pointer, +}; + +static const struct snd_soc_platform_driver xtfpga_soc_platform = { + .pcm_new = xtfpga_pcm_new, + .pcm_free = xtfpga_pcm_free, + .ops = &xtfpga_pcm_ops, +}; + +static const struct snd_soc_component_driver xtfpga_i2s_component = { + .name = DRV_NAME, +}; + +static const struct snd_soc_dai_ops xtfpga_i2s_dai_ops = { + .startup = xtfpga_i2s_startup, + .hw_params = xtfpga_i2s_hw_params, + .set_fmt = xtfpga_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver xtfpga_i2s_dai[] = { + { + .name = "xtfpga-i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &xtfpga_i2s_dai_ops, + }, +}; + +static int xtfpga_i2s_runtime_suspend(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + return 0; +} + +static int xtfpga_i2s_runtime_resume(struct device *dev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(i2s->clk); + if (ret) { + dev_err(dev, "clk_prepare_enable failed: %d\n", ret); + return ret; + } + return 0; +} + +static int xtfpga_i2s_probe(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s; + struct resource *mem; + int err, irq; + + i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + if (!i2s) { + err = -ENOMEM; + goto err; + } + platform_set_drvdata(pdev, i2s); + i2s->dev = &pdev->dev; + dev_dbg(&pdev->dev, "dev: %p, i2s: %p\n", &pdev->dev, i2s); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + i2s->regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(i2s->regs)) { + err = PTR_ERR(i2s->regs); + goto err; + } + + i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->regs, + &xtfpga_i2s_regmap_config); + if (IS_ERR(i2s->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + err = PTR_ERR(i2s->regmap); + goto err; + } + + i2s->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(i2s->clk)) { + dev_err(&pdev->dev, "couldn't get clock\n"); + err = PTR_ERR(i2s->clk); + goto err; + } + + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, + (0x1 << XTFPGA_I2S_CONFIG_CHANNEL_BASE)); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, XTFPGA_I2S_INT_VALID); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, XTFPGA_I2S_INT_UNDERRUN); + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_err(&pdev->dev, "No IRQ resource\n"); + err = irq; + goto err; + } + err = devm_request_threaded_irq(&pdev->dev, irq, NULL, + xtfpga_i2s_threaded_irq_handler, + IRQF_SHARED | IRQF_ONESHOT, + pdev->name, i2s); + if (err < 0) { + dev_err(&pdev->dev, "request_irq failed\n"); + goto err; + } + + err = snd_soc_register_platform(&pdev->dev, &xtfpga_soc_platform); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register platform\n"); + goto err; + } + err = devm_snd_soc_register_component(&pdev->dev, + &xtfpga_i2s_component, + xtfpga_i2s_dai, + ARRAY_SIZE(xtfpga_i2s_dai)); + if (err < 0) { + dev_err(&pdev->dev, "couldn't register component\n"); + goto err_unregister_platform; + } + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + err = xtfpga_i2s_runtime_resume(&pdev->dev); + if (err) + goto err_pm_disable; + } + return 0; + +err_pm_disable: + pm_runtime_disable(&pdev->dev); +err_unregister_platform: + snd_soc_unregister_platform(&pdev->dev); +err: + dev_err(&pdev->dev, "%s: err = %d\n", __func__, err); + return err; +} + +static int xtfpga_i2s_remove(struct platform_device *pdev) +{ + struct xtfpga_i2s *i2s = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + if (i2s->regmap && !IS_ERR(i2s->regmap)) { + regmap_write(i2s->regmap, XTFPGA_I2S_CONFIG, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_MASK, 0); + regmap_write(i2s->regmap, XTFPGA_I2S_INT_STATUS, + XTFPGA_I2S_INT_VALID); + } + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + xtfpga_i2s_runtime_suspend(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id xtfpga_i2s_of_match[] = { + { .compatible = "cdns,xtfpga-i2s", }, + {}, +}; +MODULE_DEVICE_TABLE(of, xtfpga_i2s_of_match); +#endif + +static const struct dev_pm_ops xtfpga_i2s_pm_ops = { + SET_RUNTIME_PM_OPS(xtfpga_i2s_runtime_suspend, + xtfpga_i2s_runtime_resume, NULL) +}; + +static struct platform_driver xtfpga_i2s_driver = { + .probe = xtfpga_i2s_probe, + .remove = xtfpga_i2s_remove, + .driver = { + .name = "xtfpga-i2s", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(xtfpga_i2s_of_match), + .pm = &xtfpga_i2s_pm_ops, + }, +}; + +module_platform_driver(xtfpga_i2s_driver); + +MODULE_AUTHOR("Max Filippov <jcmvbkbc(a)gmail.com>"); +MODULE_DESCRIPTION("xtfpga I2S controller driver"); +MODULE_LICENSE("GPL v2"); -- 1.8.1.4

1 0

[alsa-devel] [RFC 0/2] get rid of remaining users of attach_adapter
by Wolfram Sang 04 Dec '14

04 Dec '14

The attach_adapter mechanism of the I2C framework is deprecated for years. There are two users left, drivers for old Macintosh computers. I recently got the idea of replacing this mechanism with a custom one with the help of deferred probing. Because I don't have the hardware, I worked on a seperate branch where I modified the windtunnel driver to be runnable on my Renesas Lager board (ARM). I first verified that the attach_adapter method was used, then the driver rightfully failed on detecting the i2c clients. Using my custom mechanism, the same happens: all busses get scanned, then the clients cannot be detected. Since I didn't change the actual detection code, I assume that it should be working on those Macintosh devices as well. The keywest driver is only compile-tested. That being said, I'd be more than happy, if we could find someone willing to test these patches. If they could be applied, we can _finally_ get rid of this legacy mechanism and clean up the i2c core. The branch I used for ARM compilation is here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/attach_adapter_removal_with_arm_compile-experimental Thanks, Wolfram Wolfram Sang (2): macintosh: therm_windtunnel: drop using attach adapter sound: ppc: keywest: drop using attach adapter drivers/macintosh/therm_windtunnel.c | 25 +++++++++++++++++++++++-- sound/ppc/keywest.c | 21 ++++++++++++++++++--- 2 files changed, 41 insertions(+), 5 deletions(-) -- 2.1.3

2 4

[alsa-devel] [PATCH v2] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 04 Dec '14

04 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. If platform data is not supplied, we will get MCLK from device tree and check the return code to handle probe deferral. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. If we use PLL to generate sysclk, wm8960_set_dai_pll() will enable PLL. But PLL doesn't power down after using codec. So power down PLL in suspend(). Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 59 ++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 57 insertions(+), 3 deletions(-) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..2c7a97a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -978,7 +980,7 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; -static void wm8960_set_pdata_from_of(struct i2c_client *i2c, +static int wm8960_set_pdata_from_of(struct i2c_client *i2c, struct wm8960_data *pdata) { const struct device_node *np = i2c->dev.of_node; @@ -988,6 +990,15 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); + + if (IS_ERR(pdata->mclk)) { + if (PTR_ERR(pdata->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + return 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1008,8 +1019,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (pdata) memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); - else if (i2c->dev.of_node) - wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + else if (i2c->dev.of_node) { + ret = wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (ret != 0) + return ret; + } ret = wm8960_reset(wm8960->regmap); if (ret != 0) { @@ -1041,6 +1055,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1070,41 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->pdata.mclk)) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + /* Power down PLL to save power */ + regmap_update_bits(wm8960->regmap, WM8960_POWER2, 0x1, 0); + + if (!IS_ERR(wm8960->pdata.mclk)) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1122,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 2

[alsa-devel] [PATCH 4/5] ASoC: dwc: Add devicetree support for Designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

Convert to driver to use either platform_data or device-tree for configuration of the device. When using device-tree, the I2S block's configuration is read from the relevant registers: this reduces the amount of information required in the device tree. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- .../devicetree/bindings/sound/designware-i2s.txt | 32 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 238 ++++++++++++++++---- 3 files changed, 222 insertions(+), 49 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/designware-i2s.txt b/Documentation/devicetree/bindings/sound/designware-i2s.txt new file mode 100644 index 0000000..cdee591 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/designware-i2s.txt @@ -0,0 +1,32 @@ +DesignWare I2S controller + +Required properties: + - compatible : Must be "snps,designware-i2s" + - reg : Must contain I2S core's registers location and length + - clocks : Pairs of phandle and specifier referencing the controller's clocks. + The controller expects two clocks, the clock used for the APB interface and + the clock used as the sampling rate reference clock sample. + - clock-names : "apb_plck" for the clock to the APB interface, "i2sclk" for the sample + rate reference clock. + - dmas: Pairs of phandle and specifier for the DMA channels that are used by + the core. The core expects one or two dma channels, one for transmit and one for + receive. + - dma-names : "tx" for the transmit channel, "rx" for the receive channel. + +For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties +please check: + * resource-names.txt + * clock/clock-bindings.txt + * dma/dma.txt + +Example: + + soc_i2s: i2s@7ff90000 { + compatible = "snps,designware-i2s"; + reg = <0x0 0x7ff90000 0x0 0x1000>; + clocks = <&scpi_i2sclk 0>, <&soc_refclk100mhz>; + clock-names = "i2sclk", "apb_pclk"; + #sound-dai-cells = <0>; + dmas = <&dma0 5>; + dma-names = "tx"; + }; diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index e334900..d50e085 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,6 +1,7 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" depends on CLKDEV_LOOKUP + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index c497ada..083779d 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -22,6 +22,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/dmaengine_pcm.h> /* common register for all channel */ #define IER 0x000 @@ -54,19 +55,46 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +/* + * Component parameter register fields - define the I2S block's + * configuration. + */ +#define COMP1_TX_WORDSIZE_3(r) (((r) & GENMASK(27, 25)) >> 25) +#define COMP1_TX_WORDSIZE_2(r) (((r) & GENMASK(24, 22)) >> 22) +#define COMP1_TX_WORDSIZE_1(r) (((r) & GENMASK(21, 19)) >> 19) +#define COMP1_TX_WORDSIZE_0(r) (((r) & GENMASK(18, 16)) >> 16) +#define COMP1_TX_CHANNELS(r) (((r) & GENMASK(10, 9)) >> 9) +#define COMP1_RX_CHANNELS(r) (((r) & GENMASK(8, 7)) >> 7) +#define COMP1_RX_ENABLED(r) (((r) & BIT(6)) >> 6) +#define COMP1_TX_ENABLED(r) (((r) & BIT(5)) >> 5) +#define COMP1_MODE_EN(r) (((r) & BIT(4)) >> 4) +#define COMP1_FIFO_DEPTH_GLOBAL(r) (((r) & GENMASK(3, 2)) >> 2) +#define COMP1_APB_DATA_WIDTH(r) (((r) & GENMASK(1, 0)) >> 0) + +#define COMP2_RX_WORDSIZE_3(r) (((r) & GENMASK(12, 10)) >> 10) +#define COMP2_RX_WORDSIZE_2(r) (((r) & GENMASK(9, 7)) >> 7) +#define COMP2_RX_WORDSIZE_1(r) (((r) & GENMASK(5, 3)) >> 3) +#define COMP2_RX_WORDSIZE_0(r) (((r) & GENMASK(2, 0)) >> 0) + #define MAX_CHANNEL_NUM 8 #define MIN_CHANNEL_NUM 2 +union snd_dma_data { + struct i2s_dma_data pd; + struct snd_dmaengine_dai_dma_data dt; +}; + struct dw_i2s_dev { void __iomem *i2s_base; struct clk *clk; int active; unsigned int capability; struct device *dev; + bool using_pd; /* data related to DMA transfers b/w i2s and DMAC */ - struct i2s_dma_data play_dma_data; - struct i2s_dma_data capture_dma_data; + union snd_dma_data play_dma_data; + union snd_dma_data capture_dma_data; struct i2s_clk_config_data config; int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; @@ -153,7 +181,7 @@ static int dw_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); - struct i2s_dma_data *dma_data = NULL; + union snd_dma_data *dma_data = NULL; if (!(dev->capability & DWC_I2S_RECORD) && (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) @@ -227,8 +255,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - /* Iterate over set of channels - independently controlled. - */ + /* Iterate over set of channels - independently controlled. */ do { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), @@ -251,13 +278,18 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, config->sample_rate = params_rate(params); - if (!dev->i2s_clk_cfg) - return -EINVAL; + if (dev->using_pd) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk; - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; + /* TODO: Validate sample rate against permissible set */ + bitclk = config->sample_rate * config->data_width * 2; + clk_set_rate(dev->clk, bitclk); } return 0; @@ -331,6 +363,31 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) #define dw_i2s_resume NULL #endif +/* Maximum resolution of a channel - not uniformly spaced */ +static const u32 fifo_width[] = { + 12, 16, 20, 24, 32, 0, 0, 0 +}; + +/* Width of (DMA) bus */ +static const u32 bus_widths[] = { + DMA_SLAVE_BUSWIDTH_1_BYTE, + DMA_SLAVE_BUSWIDTH_2_BYTES, + DMA_SLAVE_BUSWIDTH_4_BYTES, + DMA_SLAVE_BUSWIDTH_UNDEFINED +}; + +/* PCM format to support channel resolution */ +static const u32 formats[] = { + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S16_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, + SNDRV_PCM_FMTBIT_S32_LE, + 0, + 0, + 0 +}; + static int dw_i2s_probe(struct platform_device *pdev) { const struct i2s_platform_data *pdata = pdev->dev.platform_data; @@ -340,11 +397,6 @@ static int dw_i2s_probe(struct platform_device *pdev) unsigned int cap; struct snd_soc_dai_driver *dw_i2s_dai; - if (!pdata) { - dev_err(&pdev->dev, "Invalid platform data\n"); - return -EINVAL; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); @@ -373,47 +425,114 @@ static int dw_i2s_probe(struct platform_device *pdev) return PTR_ERR(dev->i2s_base); } - cap = pdata->cap; - dev->capability = cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (pdata) { + dev->using_pd = true; + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock device\n"); + return -ENODEV; + } - /* Set DMA slaves info */ + /* Set DMA slaves info */ + + dev->play_dma_data.pd.data = pdata->play_dma_data; + dev->capture_dma_data.pd.data = pdata->capture_dma_data; + dev->play_dma_data.pd.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.pd.addr = res->start + I2S_RXDMA; + dev->play_dma_data.pd.max_burst = 16; + dev->capture_dma_data.pd.max_burst = 16; + dev->play_dma_data.pd.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.pd.addr_width = + DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.pd.filter = pdata->filter; + dev->capture_dma_data.pd.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " designware: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } - dev->play_dma_data.data = pdata->play_dma_data; - dev->capture_dma_data.data = pdata->capture_dma_data; - dev->play_dma_data.addr = res->start + I2S_TXDMA; - dev->capture_dma_data.addr = res->start + I2S_RXDMA; - dev->play_dma_data.max_burst = 16; - dev->capture_dma_data.max_burst = 16; - dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; - dev->play_dma_data.filter = pdata->filter; - dev->capture_dma_data.filter = pdata->filter; + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "designware: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + } else { + /* + * Read component parameter registers to extract + * the I2S block's configuration. + */ + u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); + u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); + u32 bus_width = bus_widths[COMP1_APB_DATA_WIDTH(comp1)]; + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); + u32 max_size; + + dev->using_pd = false; + + dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + /* + * Code presumes all channels are configured with the same + * word size. + */ + if (COMP1_TX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "playback capable\n"); + + dev->capability |= DWC_I2S_PLAY; + max_size = COMP1_TX_WORDSIZE_0(comp1); + dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; + dev->play_dma_data.dt.addr_width = bus_width; + dev->play_dma_data.dt.chan_name = "TX"; + dev->play_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size]) >> 8; + dev->play_dma_data.dt.maxburst = 16; + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = + 1 << (COMP1_TX_CHANNELS(comp1) + 1); + dw_i2s_dai->playback.formats = formats[max_size]; + dw_i2s_dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + } + if (COMP1_RX_ENABLED(comp1)) { + dev_dbg(&pdev->dev, "record capable\n"); + + dev->capability |= DWC_I2S_RECORD; + max_size = COMP2_RX_WORDSIZE_0(comp2); + dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; + dev->capture_dma_data.dt.addr_width = bus_width; + dev->capture_dma_data.dt.chan_name = "RX"; + dev->capture_dma_data.dt.fifo_size = fifo_depth * + (fifo_width[max_size] >> 8); + dev->capture_dma_data.dt.maxburst = 16; + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = + 1 << (COMP1_RX_CHANNELS(comp1) + 1); + dw_i2s_dai->capture.formats = formats[max_size]; + dw_i2s_dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + } + } - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); + ret = clk_prepare(dev->clk); + if (ret < 0) + goto err_clk_put; ret = clk_enable(dev->clk); if (ret < 0) goto err_clk_put; - if (cap & DWC_I2S_PLAY) { - dev_dbg(&pdev->dev, " designware: play supported\n"); - dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->playback.channels_max = pdata->channel; - dw_i2s_dai->playback.formats = pdata->snd_fmts; - dw_i2s_dai->playback.rates = pdata->snd_rates; - } - - if (cap & DWC_I2S_RECORD) { - dev_dbg(&pdev->dev, "designware: record supported\n"); - dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; - dw_i2s_dai->capture.channels_max = pdata->channel; - dw_i2s_dai->capture.formats = pdata->snd_fmts; - dw_i2s_dai->capture.rates = pdata->snd_rates; - } - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, @@ -423,6 +542,15 @@ static int dw_i2s_probe(struct platform_device *pdev) goto err_clk_disable; } + if (!dev->using_pd) { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, + "Could not register PCM: %d\n", ret); + return ret; + } + } + return 0; err_clk_disable: @@ -443,12 +571,24 @@ static int dw_i2s_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id dw_i2s_of_match[] = { + { .compatible = "snps,designware-i2s", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, dw_i2s_of_match); +#endif + static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = dw_i2s_of_match, +#endif }, }; -- 1.7.1

4 12

[alsa-devel] [PATCH 2/5] ASoC: dwc: Don't allow negative use counts
by Andrew Jackson 04 Dec '14

04 Dec '14

Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 08f0229..f8946bd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -280,7 +280,8 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dev->active--; + if (dev->active > 0) + dev->active--; i2s_stop(dev, substream); break; default: -- 1.7.1

3 6

[alsa-devel] [PATCH 1/5] ASoC: dwc: Allocate resources with devm_ioremap_resource
by Andrew Jackson 04 Dec '14

04 Dec '14

Prepare for the introduction of device-tree support by re-ordering some of the allocations and using devm_iomap_resource to simplify IO mapping. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 46 ++++++++++++++++----------------------- 1 files changed, 19 insertions(+), 27 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e961388..08f0229 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -338,31 +338,34 @@ static int dw_i2s_probe(struct platform_device *pdev) return -EINVAL; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "no i2s resource defined\n"); - return -ENODEV; - } - - if (!devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name)) { - dev_err(&pdev->dev, "i2s region already claimed\n"); - return -EBUSY; - } - dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { dev_warn(&pdev->dev, "kzalloc fail\n"); return -ENOMEM; } - dev->i2s_base = devm_ioremap(&pdev->dev, res->start, - resource_size(res)); - if (!dev->i2s_base) { - dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); return -ENOMEM; } + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + dev->i2s_base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(dev->i2s_base)) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return PTR_ERR(dev->i2s_base); + } + cap = pdata->cap; dev->capability = cap; dev->i2s_clk_cfg = pdata->i2s_clk_cfg; @@ -388,13 +391,6 @@ static int dw_i2s_probe(struct platform_device *pdev) if (ret < 0) goto err_clk_put; - dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); - if (!dw_i2s_dai) { - dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); - ret = -ENOMEM; - goto err_clk_disable; - } - if (cap & DWC_I2S_PLAY) { dev_dbg(&pdev->dev, " designware: play supported\n"); dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; @@ -411,10 +407,6 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai->capture.rates = pdata->snd_rates; } - dw_i2s_dai->ops = &dw_i2s_dai_ops; - dw_i2s_dai->suspend = dw_i2s_suspend; - dw_i2s_dai->resume = dw_i2s_resume; - dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &dw_i2s_component, -- 1.7.1

3 5

[alsa-devel] [PATCH 3/5] ASoC: dwc: Iterate over all channels
by Andrew Jackson 04 Dec '14

04 Dec '14

On the Designware core, the channels are independent and not combined in higher registers. So as more channels are added, more registers need to be updated. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 32 +++++++++++++++++++------------- 1 files changed, 19 insertions(+), 13 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index f8946bd..c497ada 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -227,19 +227,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_disable_channels(dev, substream->stream); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); - } else { - i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); - irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); - i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); - } + /* Iterate over set of channels - independently controlled. + */ + do { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), + xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + } while (ch_reg-- > 0); i2s_write_reg(dev->i2s_base, CCR, ccr); -- 1.7.1

3 4

[alsa-devel] [PATCH 0/5] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 04 Dec '14

04 Dec '14

This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Andrew Jackson (5): ASoC: dwc: Allocate resources with devm_ioremap_resource ASoC: dwc: Don't allow negative use counts ASoC: dwc: Iterate over all channels ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap .../devicetree/bindings/sound/designware-i2s.txt | 32 ++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 313 ++++++++++++++------ 3 files changed, 260 insertions(+), 86 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

2 2

[alsa-devel] [PATCH 0/2] rockchip: i2s: patches to add optional clock output from
by Jianqun Xu 04 Dec '14

04 Dec '14

The clock output from rockchip SoCs' CRU to chip outside is from the same source as hclk which is for i2s controller. We make it optional because some audio codec outside not need the master clock from SoC but from other clock source. Jianqun Xu (2): ASoC: rockchip-i2s: dt: add an optional property "i2s_clk_out" ASoC: rockchip: i2s: add support for grabbing output clock to codec Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 8 ++++++-- sound/soc/rockchip/rockchip_i2s.c | 11 +++++++++++ 2 files changed, 17 insertions(+), 2 deletions(-) -- 1.9.1

6 16

[alsa-devel] [PATCH v2] ASoC: rt5677: make volume TLV closer to reality
by Dylan Reid 03 Dec '14

03 Dec '14

The volume blocks have an step of 0.375dB, but TLV uses 0.01dB for units. Switch to use TLV_DB_MINMAX as this allows the minimum to be off by only 0.005dB. This is an improvement from the ~500dB is was misreported by before. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 16aa4d9..a4c71f9 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -538,9 +538,9 @@ static bool rt5677_readable_register(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -6562, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); -static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_MINMAX(adc_vol_tlv, -1762, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(st_vol_tlv, -4650, 150, 0); -- 2.1.2.330.g565301e

3 6

[alsa-devel] [PATCH 0/2] ALSA: ctxfi: Delete an unnecessary check before kfree()
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 20:35:41 +0100 Another update suggestion was taken into account after a patch was applied from static source code analysis. Markus Elfring (2): Deletion of an unnecessary check before the function call "kfree" One function call less in get_daio_rsc() after error detection sound/pci/ctxfi/ctdaio.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) -- 2.1.3

3 10

[alsa-devel] [PATCH 1/1] ASoC: mop500: Deletion of unnecessary checks before the function call "of_node_put"
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 17:15:11 +0100 The of_node_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/ux500/mop500.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 7f7093b..4e0c0e5 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -63,10 +63,8 @@ static void mop500_of_node_put(void) int i; for (i = 0; i < 2; i++) { - if (mop500_dai_links[i].cpu_of_node) - of_node_put(mop500_dai_links[i].cpu_of_node); - if (mop500_dai_links[i].codec_of_node) - of_node_put(mop500_dai_links[i].codec_of_node); + of_node_put(mop500_dai_links[i].cpu_of_node); + of_node_put(mop500_dai_links[i].codec_of_node); } } -- 2.1.3

2 2

[alsa-devel] [PATCH 1/1] ASoC: fsi: Deletion of unnecessary checks before the function call "clk_disable"
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 14:34:30 +0100 The clk_disable() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/sh/fsi.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 88e5df4..8869971 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -842,12 +842,9 @@ static int fsi_clk_disable(struct device *dev, return -EINVAL; if (1 == clock->count--) { - if (clock->xck) - clk_disable(clock->xck); - if (clock->ick) - clk_disable(clock->ick); - if (clock->div) - clk_disable(clock->div); + clk_disable(clock->xck); + clk_disable(clock->ick); + clk_disable(clock->div); } return 0; -- 2.1.3

2 1

[alsa-devel] Analog sound (ALC262) lost on linux (any distro and kernel) on a dual boot system
by Ploumistos Alexandros 03 Dec '14

03 Dec '14

Hello, A while back I had installed Ubuntu on a friend's laptop -a Sony Vaio VGN-FW21M- alongside windows Vista. Everything worked fine for a few years, until at some point last year linux went mute, while in windows both the analog and the digital outputs continued working. She couldn't remember doing something "out of the ordinary", but she was sure it was not after an update. Several months later, she upgraded to windows 7 but that didn't change anything. When I got my hands on the laptop, I checked for muted controls and other usual suspects, deprecated quirks (there were none) and all inputs and outputs of the system. HDMI audio works fine and so does the internal mic. Speakers, headphones and the external mic input do not work, even though pulse audio volume meter acts as if there were actually sounds being played. I followed all of Ubuntu's troubleshooting procedures, I removed and reinstalled alsa and pulse, I purged all configuration files, I created a new user and I even installed the latest alsa-packages from the "ALSA daily build snapshots" repository and nothing worked. I also tried booting into windows, setting all volume levels to the max and disabled power saving on all sound and multimedia devices. Next, I tried retasking the jacks one by one to no avail. I remembered that I had performed the installation with an Ubuntu 11.10 live CD, so I booted the computer off of that, but this time I got no sound. I tried with all the live CDs of a multitude of distros I had lying around, with kernels ranging from 2.6.35 to 3.16.6 and none of them worked, so I figured the issue must have something to do with windows. Is it possible that windows somehow started locking the audio chip on shutdown? If that is the case, can it be deduced from the logs and more importantly, can it be fixed? I am attaching the output of alsa-info and lspci as well as kernel logs. Thank you for your time.

2 7

[alsa-devel] [PATCH 5/5] ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap
by Andrew Jackson 03 Dec '14

03 Dec '14

If the FIFOs aren't flushed, the left/right channels may be swapped: this may occur if the FIFOs are not empty when the streams start. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 083779d..932abfd 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -258,6 +258,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, /* Iterate over set of channels - independently controlled. */ do { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TXFFR, 1); i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); @@ -265,6 +266,7 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); } else { + i2s_write_reg(dev->i2s_base, RXFFR, 1); i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); -- 1.7.1

2 1

[alsa-devel] [PATCH] sound: pci: cmipci.c: Cleaning up function that are not used anywhere
by Rickard Strandqvist 03 Dec '14

03 Dec '14

Remove function snd_cmipci_pll_rmn() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/cmipci.c | 41 ----------------------------------------- 1 file changed, 41 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 85ed403..e33edae 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -615,47 +615,6 @@ static unsigned int snd_cmipci_rate_freq(unsigned int rate) #ifdef USE_VAR48KRATE /* - * Determine PLL values for frequency setup, maybe the CMI8338 (CMI8738???) - * does it this way .. maybe not. Never get any information from C-Media about - * that <werner(a)suse.de>. - */ -static int snd_cmipci_pll_rmn(unsigned int rate, unsigned int adcmult, int *r, int *m, int *n) -{ - unsigned int delta, tolerance; - int xm, xn, xr; - - for (*r = 0; rate < CM_MAXIMUM_RATE/adcmult; *r += (1<<5)) - rate <<= 1; - *n = -1; - if (*r > 0xff) - goto out; - tolerance = rate*CM_TOLERANCE_RATE; - - for (xn = (1+2); xn < (0x1f+2); xn++) { - for (xm = (1+2); xm < (0xff+2); xm++) { - xr = ((CM_REFFREQ_XIN/adcmult) * xm) / xn; - - if (xr < rate) - delta = rate - xr; - else - delta = xr - rate; - - /* - * If we found one, remember this, - * and try to find a closer one - */ - if (delta < tolerance) { - tolerance = delta; - *m = xm - 2; - *n = xn - 2; - } - } - } -out: - return (*n > -1); -} - -/* * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen * at the register CM_REG_FUNCTRL1 (0x04). -- 1.7.10.4

3 4

[alsa-devel] [RFC] alsa-lib: Add snd_pcm_start_at
by Tim Cussins 03 Dec '14

03 Dec '14

snd_pcm_start_at() - alsa-lib The headlines: - Provide userspace snd_pcm_start_at, leverage new ioctl - Provide some decent doxygen guff diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..bbe6e79 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -325,9 +325,10 @@ typedef enum _snd_pcm_tstamp { 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_AUDIO_WALLCLOCK, /** audio hw clock */ + SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, } snd_pcm_tstamp_type_t; /** Unsigned frames quantity */ @@ -478,6 +479,7 @@ 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, 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 941d32f..bcc49dc 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -467,8 +467,9 @@ struct snd_xfern { 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_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, /* audio hw clock */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, }; /* channel positions */ @@ -549,6 +550,8 @@ 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, snd_htimestamp_t) + /***************************************************************************** * * diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..61158cb 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,38 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \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 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. + * + * \p start_time is interpreted in the context of the stream snd_pcm_tstamp_type, + * set using \link ::snd_pcm_sw_params_set_tstamp_type() \endlink. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, 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, start_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code @@ -1693,6 +1725,7 @@ static const char *const snd_pcm_tstamp_type_names[] = { TSTAMP_TYPE(GETTIMEOFDAY), TSTAMP_TYPE(MONOTONIC), TSTAMP_TYPE(MONOTONIC_RAW), + TSTAMP_TYPE(AUDIO_WALLCLOCK), }; #endif diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c34b766..aa0c7ba 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -447,6 +447,11 @@ static int snd_pcm_hw_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t * params) hw->mmap_control->avail_min = params->avail_min; return sync_ptr(hw, 0); } + if (params->tstamp_type == SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK && + hw->version < SNDRV_PROTOCOL_VERSION(2, 0, 12)) { + SYSMSG("Kernel doesn't support SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK"); + return -EINVAL; + } if (params->tstamp_type == SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW && hw->version < SNDRV_PROTOCOL_VERSION(2, 0, 12)) { SYSMSG("Kernel doesn't support SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW"); @@ -620,6 +625,20 @@ static int snd_pcm_hw_start(snd_pcm_t *pcm) return 0; } +static int snd_pcm_hw_start_at(snd_pcm_t *pcm, const snd_htimestamp_t *start_time) +{ + snd_pcm_hw_t *hw = pcm->private_data; + int err; + + sync_ptr(hw, 0); + if (ioctl(hw->fd, SNDRV_PCM_IOCTL_START_AT, start_time) < 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 +1355,7 @@ 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..6829750 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, 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

1 0

[alsa-devel] [RFC] kernel: Add snd_pcm_start_at
by Tim Cussins 03 Dec '14

03 Dec '14

snd_pcm_start_at() - kernel-side The headlines: - Provide snd_pcm_start_at ioctl - Add SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK - Use the stream tstamp_type to interpret the start_at timespec - snd_pcm_start_at_ops defines handler for a TSTAMP_TYPE - handler requirements clearly defined - If posix clock, use a high-res timer implementation (thx Nick Stoughton!) - If audio wallclock, delegate to the pcm driver if possible, otherwise error - Further implementations easily added - Cancel pending timer on: - Subsequent call to snd_pcm_start_at() - Any /attempt/ to change the stream state Major points for discussion: - snd_pcm_start_at_ops requirements aren't clear enough - callback must not do cleanup - cancel is guaranteed to be called - do cleanup here - snd_pcm_gettime() can't currently return a value when tstamp_type is AUDIO_WALLCLOCK - start_at timer callbacks *should* probably lock the stream, contrary to Nick's note - The user story for start_at cancellation from userspace is a bit weird... - Cancellation on state-change seems necessary, but not sufficient - Maybe add snd_pcm_start_at_cancel()? Future thoughts: - Perhaps snd_pcm_start_at should be a blocking call (EAGAIN when stream is non-blocking) - Maybe tstamp_types and startat_types should be different things I've become convinced that exposing the nature of the pcm clock is troublesome, and unnecessary, and fell back to Takashi's suggestion of a DEVICE_SPECIFIC tstamp_type. I note that AUDIO_WALLCLOCK seems to capture the concept of 'pcm time', so I've called it that. Because I'm using SND_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK instead of SNDRV_PCM_TSTAMP_TYPE_PTP, there's no additional payload required for snd_pcm_sw_set_tstamp_type(). Comments and suggestions welcome :) Tim diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1e7f74a..b8bfa9b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -82,6 +82,8 @@ struct snd_pcm_ops { unsigned long offset); int (*mmap)(struct snd_pcm_substream *substream, struct vm_area_struct *vma); int (*ack)(struct snd_pcm_substream *substream); + int (*wall_clock_start_at)(struct snd_pcm_substream *substream, const struct timespec *ts); + int (*wall_clock_start_at_cancel)(struct snd_pcm_substream *substream); }; /* @@ -210,6 +212,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE #endif +#define SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL (SNDRV_PCM_TSTAMP_TYPE_LAST+1) + struct snd_pcm_file { struct snd_pcm_substream *substream; int no_compat_mmap; @@ -364,6 +368,10 @@ struct snd_pcm_runtime { #ifdef CONFIG_SND_PCM_XRUN_DEBUG struct snd_pcm_hwptr_log *hwptr_log; #endif + + int start_at_tstamp_type; /* start_at timer tstamp_type Set to + SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL if not active */ + void *start_at_timer_data; /* start_at timer data */ }; struct snd_pcm_group { /* keep linked substreams */ @@ -1069,6 +1077,7 @@ static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime, case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: getrawmonotonic(tv); break; + case SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK: default: getnstimeofday(tv); break; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 941d32f..dfe2bac 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -467,8 +467,9 @@ struct snd_xfern { 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_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, /* audio wallclock timestamp */ + SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK, }; /* channel positions */ @@ -549,6 +550,7 @@ 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 timespec) /***************************************************************************** * * diff --git a/sound/core/pcm.c b/sound/core/pcm.c index cfc56c8..483f85d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1003,6 +1003,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; + runtime->start_at_tstamp_type = SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL; substream->runtime = runtime; substream->private_data = pcm->private_data; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 095d957..352921d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -35,6 +35,9 @@ #include <sound/timer.h> #include <sound/minors.h> #include <asm/io.h> +#if defined(CONFIG_HIGH_RES_TIMERS) +#include <linux/hrtimer.h> +#endif /* * Compatibility @@ -67,6 +70,8 @@ static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, #endif static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream); +static int snd_pcm_start_at_cancel(struct snd_pcm_substream *substream); + /* * */ @@ -265,6 +270,7 @@ static const char * const snd_pcm_hw_param_names[] = { }; #endif + int snd_pcm_hw_refine(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -834,6 +840,11 @@ static int snd_pcm_action_group(struct action_ops *ops, struct snd_pcm_substream *s1; int res = 0, depth = 1; + /* Any attempt to change state cancels a pending start_at timer */ + res = snd_pcm_start_at_cancel(substream); + if (res < 0) + return res; + snd_pcm_group_for_each_entry(s, substream) { if (do_lock && s != substream) { if (s->pcm->nonatomic) @@ -888,6 +899,11 @@ static int snd_pcm_action_single(struct action_ops *ops, int state) { int res; + + /* Any attempt to change state cancels a pending start_at timer */ + res = snd_pcm_start_at_cancel(substream); + if (res < 0) + return res; res = ops->pre_action(substream, state); if (res < 0) @@ -1015,6 +1031,234 @@ static struct action_ops snd_pcm_action_start = { .post_action = snd_pcm_post_start }; +static inline clockid_t snd_pcm_get_clockid(struct snd_pcm_substream* substream) +{ + switch(substream->runtime->tstamp_type) + { + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC: + return CLOCK_MONOTONIC; + case SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW: + return CLOCK_MONOTONIC_RAW; + default: + return CLOCK_REALTIME; + } +} + +/* snd_pcm_start_at_ops + * There are various mechanisms for supporting snd_pcm_start_at. Posix clocks may + * use hires timers if available. + * + * schedule() may use the runtime member 'start_at_timer_data' to store enough + * information to cancel the timer. When the timer fires, it must be single-shot, + * and must not cancel/delete the timer: cancel() is guaranteed to be called + * before the timer is reused, so resources should be freed within cancel(), not + * by the timer callback. + * + * cancel() returns when the start_at timer callback is guaranteed to be both + * cancelled and not currently running. +*/ + +struct snd_pcm_start_at_ops { + int (*schedule)(struct snd_pcm_substream *substream, const struct timespec *start_time); + int (*cancel)(struct snd_pcm_substream *substream); +}; + +/* To support snd_pcm_start_at for posix tstamp_types, we use high-res timers, if + * kernel is configured appropriately. + */ + +#ifdef CONFIG_HIGH_RES_TIMERS +/* + * hrtimer interface + */ + +struct hrtimer_pcm { + struct hrtimer timer; + struct snd_pcm_substream *substream; +}; + +/* + * called from a hard irq context - no need for locks. + * only problem is that the caller might have gone away and closed the substream + * before the timer expires. + */ +enum hrtimer_restart snd_pcm_do_start_time(struct hrtimer *timer) +{ + struct hrtimer_pcm *pcm_timer; + struct snd_pcm_substream *substream; + int ret; + + pcm_timer = container_of(timer, struct hrtimer_pcm, timer); + substream = pcm_timer->substream; + + ret = snd_pcm_do_start(substream, SNDRV_PCM_STATE_RUNNING); + if (ret == 0) { + snd_pcm_post_start(substream, SNDRV_PCM_STATE_RUNNING); + } + return HRTIMER_NORESTART; +} +#endif + + + +static int start_at_posix_schedule(struct snd_pcm_substream *substream, const struct timespec *start_time) +{ +#ifdef CONFIG_HIGH_RES_TIMERS + struct hrtimer_pcm *pcm_timer; + struct timespec now; + int ret; + + /* Get time now and check if start_time is in the past */ + snd_pcm_gettime(substream->runtime, &now); + if (timespec_compare(&now, start_time) >= 0) { + return -ETIME; + } + + /* Allocate a hrtimer to handle the start_at */ + pcm_timer = kmalloc(sizeof(*pcm_timer), GFP_KERNEL); + if (!pcm_timer) + return -ENOMEM; + + hrtimer_init(&pcm_timer->timer, snd_pcm_get_clockid(substream), HRTIMER_MODE_ABS); + + /* Setup timer */ + pcm_timer->timer.function = snd_pcm_do_start_time; + pcm_timer->substream = substream; + + /* Store timer in start_at info */ + substream->runtime->start_at_timer_data = pcm_timer; + + /* Pre start */ + ret = snd_pcm_pre_start(substream, SNDRV_PCM_STATE_PREPARED); + if (ret < 0) + goto error; + + ret = hrtimer_start(&pcm_timer->timer, timespec_to_ktime(*start_time), HRTIMER_MODE_ABS); + if (ret < 0 ) + goto error; + + return 0; +error: + kfree(pcm_timer); + return ret; +#else + return -ENOSYS; +#endif +} + +static int start_at_posix_cancel(struct snd_pcm_substream *substream) +{ +#ifdef CONFIG_HIGH_RES_TIMERS + struct hrtimer_pcm *pcm_timer = substream->runtime->start_at_timer_data; + hrtimer_cancel(&pcm_timer->timer); /* Cancel existing timer. (NOP if it's not running) */ + kfree(pcm_timer); + return 0; +#else + return -ENOSYS; +#endif +} + +static int start_at_wallclock_schedule(struct snd_pcm_substream *substream, const struct timespec *start_time) +{ + if (substream->ops->wall_clock_start_at) + return substream->ops->wall_clock_start_at(substream, start_time); + else + return -ENOSYS; +} + +static int start_at_wallclock_cancel(struct snd_pcm_substream *substream) +{ + if (substream->ops->wall_clock_start_at_cancel) + return substream->ops->wall_clock_start_at_cancel(substream); + else + return -ENOSYS; +} + +static struct snd_pcm_start_at_ops start_at_ops[SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL+1] = { + [SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY] = { + .schedule = start_at_posix_schedule, + .cancel = start_at_posix_cancel + }, + [SNDRV_PCM_TSTAMP_TYPE_MONOTONIC] = { + .schedule = start_at_posix_schedule, + .cancel = start_at_posix_cancel + }, + [SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW] = { /* hrtimers can't handle CLOCK_MONOTONIC_RAW */ + .schedule = NULL, + .cancel = NULL, + }, + [SNDRV_PCM_TSTAMP_TYPE_AUDIO_WALLCLOCK] = { /* delegate start_at to pcm driver */ + .schedule = start_at_wallclock_schedule, + .cancel = start_at_wallclock_cancel, + }, + [SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL] = { /* null handler - required for handling first cancellation */ + .schedule = NULL, + .cancel = NULL, + }, +}; + +/* snd_pcm_start_at_cancel() allows state-transition code to conveniently cancel the pending timer */ +static int snd_pcm_start_at_cancel(struct snd_pcm_substream *substream) +{ + struct snd_pcm_start_at_ops *ops = &start_at_ops[substream->runtime->start_at_tstamp_type]; + int ret = 0; + + /* If ops->cancel is NULL, it's not an error. */ + if (ops->cancel) { + ret = ops->cancel(substream); + if (ret == 0) + substream->runtime->start_at_tstamp_type = SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL; + } + + return ret; +} + +int snd_pcm_start_at(struct snd_pcm_substream *substream, + struct timespec __user *_start_time) +{ + struct timespec start_time; + int new_tstamp_type; + struct snd_pcm_start_at_ops *ops; + int ret; + + if (copy_from_user(&start_time, _start_time, sizeof(start_time))) + return -EFAULT; + + if (!timespec_valid(&start_time)) + return -EINVAL; + + /* If not a playback substream, give up */ + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + /* Cancel any existing timer */ + ret = snd_pcm_start_at_cancel(substream); + if (ret < 0) + return ret; + + /* Get new tstamp_type */ + new_tstamp_type = substream->runtime->tstamp_type; + + /* Save current start_at tstamp_type. This way, it's valid before + schedule() is called */ + substream->runtime->start_at_tstamp_type = new_tstamp_type; + + /* Get apprpriate start_at ops */ + ops = &start_at_ops[new_tstamp_type]; + + /* Schedule start_at. If it doesn't exist, that's an error. */ + if (ops->schedule) { + ret = ops->schedule(substream, &start_time); + /* If successful, mark timer as cancelled */ + if (ret < 0) + /* If schedule() failed, reset tstamp type */ + substream->runtime->start_at_tstamp_type = SNDRV_PCM_START_AT_TSTAMP_TYPE_NULL; + return ret; + } + else + return -ENOSYS; +} + /** * snd_pcm_start - start all linked streams * @substream: the PCM substream instance @@ -2721,6 +2965,8 @@ static int snd_pcm_common_ioctl1(struct file *file, return snd_pcm_action_lock_irq(&snd_pcm_action_start, substream, SNDRV_PCM_STATE_RUNNING); case SNDRV_PCM_IOCTL_LINK: return snd_pcm_link(substream, (int)(unsigned long) arg); + case SNDRV_PCM_IOCTL_START_AT: + return snd_pcm_start_at(substream, arg); case SNDRV_PCM_IOCTL_UNLINK: return snd_pcm_unlink(substream); case SNDRV_PCM_IOCTL_RESUME: -- 1.9.1

1 0

[alsa-devel] [PATCH 00/30 v3] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 03 Dec '14

03 Dec '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch. == Updates from my previous post: [alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/082844.ht… - Add a patch to fix some minor bugs (dice) - Improve handling config-rom strings for loud models (oxfw) - Fix a bug of mutual exclusive lock for AMDTP streams (oxfw) - Obsolete atomic_t for reference counters (dice/oxfw) Takashi Sakamoto (30): ALSA: dice: suppress checkpatch.pl warnings ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 7 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 194 ++++ sound/firewire/dice/dice-pcm.c | 424 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 407 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 361 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 207 ++++ sound/firewire/oxfw/oxfw-pcm.c | 425 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 685 +++++++++++++ sound/firewire/oxfw/oxfw.c | 315 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4944 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile 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 create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 2.1.0

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[alsa-devel] [alsa-dev] Variable latency issue in alsa-jack-plugins
by Corfu, Valentin (Wind River) 03 Dec '14

03 Dec '14

Hello guys, I recently encountered a problem about variable latency in alsa-jack-plugin. I have successfully checked the latency with jack_iodelay in following way: $ jack_iodelay & $ alsaloop -Pjackplug -Cjackplug & $ jack_connect alsa-jack.jackP.998.0:out_000 jack_delay:in $ jack_connect jack_delay:out alsa-jack.jackC.998.1:in_000 But total roundtrip latency, seems to be variable after each start-up. Q1: What can be the root cause of this variable latency issue? Q2: How can I find theoretical latency of the alsa-jack-plugin in both directions (playback - capture) ? Thanks and Best Regards, Valentin Intel GmbH Dornacher Strasse 1 85622 Feldkirchen/Muenchen, Deutschland Sitz der Gesellschaft: Feldkirchen bei Muenchen Geschaeftsfuehrer: Christian Lamprechter, Hannes Schwaderer, Douglas Lusk Registergericht: Muenchen HRB 47456 Ust.-IdNr./VAT Registration No.: DE129385895 Citibank Frankfurt a.M. (BLZ 502 109 00) 600119052

1 0

[alsa-devel] [PATCH] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 03 Dec '14

03 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK and power down PLL. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+) diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..d7e84ed 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -16,6 +16,7 @@ #define WM8960_DRES_MAX 3 struct wm8960_data { + struct clk *mclk; bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..6a529b1 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -988,6 +990,8 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + pdata->mclk = devm_clk_get(&i2c->dev, NULL); } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1011,6 +1015,14 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + /* Mark the mclk pointer to NULL if no mclk assigned */ + if (IS_ERR(wm8960->pdata.mclk)) { + /* But do not ignore the request for probe defer */ + if (PTR_ERR(wm8960->pdata.mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + wm8960->pdata.mclk = NULL; + } + ret = wm8960_reset(wm8960->regmap); if (ret != 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); @@ -1041,6 +1053,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1068,41 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (wm8960->pdata.mclk) { + ret = clk_prepare_enable(wm8960->pdata.mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + /* Power down PLL to save power */ + regmap_update_bits(wm8960->regmap, WM8960_POWER2, 0x1, 0); + + if (wm8960->pdata.mclk) + clk_disable_unprepare(wm8960->pdata.mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1120,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Define the DCAPS preset for the old Intel chipsets
by Takashi Iwai 03 Dec '14

03 Dec '14

Just for improving readability. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fc7aff0eb562..53e43d111a3b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -276,6 +276,10 @@ enum { (((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10) #define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10) +/* quirks for old Intel chipsets */ +#define AZX_DCAPS_INTEL_ICH \ + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_BUFSIZE) + /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ @@ -2054,31 +2058,30 @@ static const struct pci_device_id azx_ids[] = { /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, - /* ICH */ + /* ICH6 */ { PCI_DEVICE(0x8086, 0x2668), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH6 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH7 */ { PCI_DEVICE(0x8086, 0x27d8), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH7 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ESB2 */ { PCI_DEVICE(0x8086, 0x269a), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ESB2 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH8 */ { PCI_DEVICE(0x8086, 0x284b), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH8 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH9 */ { PCI_DEVICE(0x8086, 0x293e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH9 */ { PCI_DEVICE(0x8086, 0x293f), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH9 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a3e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, + /* ICH10 */ { PCI_DEVICE(0x8086, 0x3a6e), - .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | - AZX_DCAPS_BUFSIZE }, /* ICH10 */ + .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, /* Generic Intel */ { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- 2.1.3

2 3

[alsa-devel] [PATCH] ALSA: i2sbus: Deletion of unnecessary checks before the function call "release_and_free_resource"
by SF Markus Elfring 03 Dec '14

03 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 22:50:24 +0100 The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/aoa/soundbus/i2sbus/core.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index a80d5ea..4e2b4fb 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -79,8 +79,7 @@ static void i2sbus_release_dev(struct device *dev) if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma); if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) - if (i2sdev->allocated_resource[i]) - release_and_free_resource(i2sdev->allocated_resource[i]); + release_and_free_resource(i2sdev->allocated_resource[i]); free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring); free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring); for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) @@ -323,8 +322,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (dev->out.dbdma) iounmap(dev->out.dbdma); if (dev->in.dbdma) iounmap(dev->in.dbdma); for (i=0;i<3;i++) - if (dev->allocated_resource[i]) - release_and_free_resource(dev->allocated_resource[i]); + release_and_free_resource(dev->allocated_resource[i]); mutex_destroy(&dev->lock); kfree(dev); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH] sound: pci: azt3328.c: Cleaning up function that are not used anywhere
by Rickard Strandqvist 03 Dec '14

03 Dec '14

Remove function snd_azf3328_codec_outl() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/azt3328.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5a69e26..82da7b9 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -379,15 +379,6 @@ snd_azf3328_codec_inw(const struct snd_azf3328_codec_data *codec, unsigned reg) } static inline void -snd_azf3328_codec_outl(const struct snd_azf3328_codec_data *codec, - unsigned reg, - u32 value -) -{ - outl(value, codec->io_base + reg); -} - -static inline void snd_azf3328_codec_outl_multi(const struct snd_azf3328_codec_data *codec, unsigned reg, const void *buffer, int count ) -- 1.7.10.4

2 1

Re: [alsa-devel] ASoC: core: Call mute for cpu dais as well
by Dan Carpenter 02 Dec '14

02 Dec '14

Hello Ramesh Babu, This is a semi-automatic email about new static checker warnings. The patch ae11601b80b9: "ASoC: core: Call mute for cpu dais as well" from Oct 15, 2014, leads to the following Smatch complaint: sound/soc/soc-pcm.c:777 soc_pcm_prepare() error: we previously assumed 'cpu_dai->driver->ops' could be null (see line 755) sound/soc/soc-pcm.c 754 755 if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ^^^^^^^^^^^^^^^^^^^^ I don't think this can be NULL. Probably we should remove this check. 756 ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); 757 if (ret < 0) { 758 dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", 759 ret); 760 goto out; 761 } 762 } 763 764 /* cancel any delayed stream shutdown that is pending */ 765 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && 766 rtd->pop_wait) { 767 rtd->pop_wait = 0; 768 cancel_delayed_work(&rtd->delayed_work); 769 } 770 771 snd_soc_dapm_stream_event(rtd, substream->stream, 772 SND_SOC_DAPM_STREAM_START); 773 774 for (i = 0; i < rtd->num_codecs; i++) 775 snd_soc_dai_digital_mute(rtd->codec_dais[i], 0, 776 substream->stream); 777 snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream); ^^^^^^^ But if it were NULL then this newly introduced function call would Oops. 778 779 out: regards, dan carpenter

3 2

[alsa-devel] [PATCH 1/1] ALSA: trident: Deletion of a check before snd_util_memhdr_free()
by SF Markus Elfring 02 Dec '14

02 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 18:52:21 +0100 The snd_util_memhdr_free() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/trident/trident_main.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index da875dc..57cd757 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3702,8 +3702,7 @@ static int snd_trident_free(struct snd_trident *trident) free_irq(trident->irq, trident); if (trident->tlb.buffer.area) { outl(0, TRID_REG(trident, NX_TLBC)); - if (trident->tlb.memhdr) - snd_util_memhdr_free(trident->tlb.memhdr); + snd_util_memhdr_free(trident->tlb.memhdr); if (trident->tlb.silent_page.area) snd_dma_free_pages(&trident->tlb.silent_page); vfree(trident->tlb.shadow_entries); -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: echoaudio: Deletion of a check before release_and_free_resource()
by SF Markus Elfring 02 Dec '14

02 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 18:34:45 +0100 The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 60e4003..21228ad 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -1875,8 +1875,7 @@ static int snd_echo_free(struct echoaudio *chip) if (chip->dsp_registers) iounmap(chip->dsp_registers); - if (chip->iores) - release_and_free_resource(chip->iores); + release_and_free_resource(chip->iores); pci_disable_device(chip->pci); -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: asihpi: Deletion of an unnecessary check before the function call "vfree"
by SF Markus Elfring 02 Dec '14

02 Dec '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 2 Dec 2014 18:05:32 +0100 The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/asihpi/hpioctl.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index e457eb8..6aa677e 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -547,8 +547,7 @@ void asihpi_adapter_remove(struct pci_dev *pci_dev) if (pa->irq) free_irq(pa->irq, pa); - if (pa->p_buffer) - vfree(pa->p_buffer); + vfree(pa->p_buffer); if (1) dev_info(&pci_dev->dev, -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: snd-soc-afeb9260: delete driver as board has just been removed
by Nicolas Ferre 02 Dec '14

02 Dec '14

During the removal of all AT91 !DT boards, we removed the AFEB9260. This driver is !DT and was only used by this board, so we delete it as well. Reported-by: Paul Bolle <pebolle(a)tiscali.nl> Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- sound/soc/atmel/Makefile | 1 - sound/soc/atmel/snd-soc-afeb9260.c | 151 ------------------------------------- 2 files changed, 152 deletions(-) delete mode 100644 sound/soc/atmel/snd-soc-afeb9260.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 5baabc8bde3a..466a821da98c 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o -obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c deleted file mode 100644 index 9579799ace54..000000000000 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ /dev/null @@ -1,151 +0,0 @@ -/* - * afeb9260.c -- SoC audio for AFEB9260 - * - * Copyright (C) 2009 Sergey Lapin <slapin(a)ossfans.org> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/platform_device.h> - -#include <linux/atmel-ssc.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/hardware.h> -#include <linux/gpio.h> - -#include "../codecs/tlv320aic23.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - -#define CODEC_CLOCK 12000000 - -static int afeb9260_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int err; - - /* Set the codec system clock for DAC and ADC */ - err = - snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); - - if (err < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return err; - } - - return err; -} - -static struct snd_soc_ops afeb9260_ops = { - .hw_params = afeb9260_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route afeb9260_audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link afeb9260_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "atmel-ssc-dai.0", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &afeb9260_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_machine_afeb9260 = { - .name = "AFEB9260", - .owner = THIS_MODULE, - .dai_link = &afeb9260_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = afeb9260_audio_map, - .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map), -}; - -static struct platform_device *afeb9260_snd_device; - -static int __init afeb9260_soc_init(void) -{ - int err; - struct device *dev; - - if (!(machine_is_afeb9260())) - return -ENODEV; - - - afeb9260_snd_device = platform_device_alloc("soc-audio", -1); - if (!afeb9260_snd_device) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260); - err = platform_device_add(afeb9260_snd_device); - if (err) - goto err1; - - dev = &afeb9260_snd_device->dev; - - return 0; -err1: - platform_device_put(afeb9260_snd_device); - return err; -} - -static void __exit afeb9260_soc_exit(void) -{ - platform_device_unregister(afeb9260_snd_device); -} - -module_init(afeb9260_soc_init); -module_exit(afeb9260_soc_exit); - -MODULE_AUTHOR("Sergey Lapin <slapin(a)ossfans.org>"); -MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); -MODULE_LICENSE("GPL"); - -- 2.1.3

3 4

[alsa-devel] [PATCH 2/2] ARM: tegra: Enable the mic-detect gpio on Acer Chromebook 13
by Dylan Reid 02 Dec '14

02 Dec '14

Enables the gpio-base mic detection on the Acer Chromebook 13. This gpio is set by the jack-detection chip when it notices either of the TRRS type headsets with a microphone. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- arch/arm/boot/dts/tegra124-nyan-big.dts | 2 ++ 1 file changed, 2 insertions(+) diff --git a/arch/arm/boot/dts/tegra124-nyan-big.dts b/arch/arm/boot/dts/tegra124-nyan-big.dts index 7d0784c..1c68ff9 100644 --- a/arch/arm/boot/dts/tegra124-nyan-big.dts +++ b/arch/arm/boot/dts/tegra124-nyan-big.dts @@ -1130,6 +1130,8 @@ clock-names = "pll_a", "pll_a_out0", "mclk"; nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-det-gpios = + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; }; }; -- 2.1.2.330.g565301e

3 2

[alsa-devel] [PATCH v7.1 00/19] Rework OMAP4+ HDMI audio support
by Jyri Sarha 02 Dec '14

02 Dec '14

The patches are based on: git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git for-next The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: https://github.com/jsarha/linux.git omap-hdmi-audio It would make the most sense to get these in trough fbdev tree. So it would be nice to get acked-bys (if the patches are Ok) for ASoC side changes from appropriate maintainers. Changes since v7: - Squash: - "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - into "OMAPDSS: Remove all references to obsolete HDMI audio callbacks" Changes since v6: - "ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support" - Fix Kconfig help text - Remove #include <sound/simple_card.h> - Use snd-soc-dummy codec instead of hdmi-audio-codec - Add: OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() - Add: OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter - Add: "OMAPDSS: omapdss.h: Remove audio_state member of struct omap_dss_device" - "OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - "OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio" - Register "omap-hdmi-audio" with PLATFORM_DEVID_AUTO - Add: "OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback" Jyri Sarha (19): OMAPDSS: hdmi_wp: Protect reserved bits in hdmi_wp_audio_config_format() OMAPDSS: hdmi5_core: Initialize mandatory sample_order parameter OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi: Remove most of OMAP[45]_DSS_HDMI_AUDIO ifdefs OMAPDSS: hdmi4_core: Remove unused hdmi4_audio_get_dma_port() OMAPDSS: hdmi_wp: Add function for getting audio dma address OMAPDSS: hdmi: Make hdmi structure public OMAPDSS: hdmi: Add pdev pointer for audio_pdev in HDMI DRV data ASoC: omap-hdmi-audio: Add platform device for OMAP HDMI audio support OMAPDSS: Kconfig: Remove HDMI audio booleans from Kconfig OMAPDSS: hdmi: Make hdmi_mode_has_audio() more user friedly OMAPDSS: hdmi.h: Add members to hdmi drvdata for audio implementation OMAPDSS: hdmi4: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi4: Register ASoC platform device for omap hdmi audio OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove all references to obsolete HDMI audio callbacks OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ----- .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 7 - drivers/video/fbdev/omap2/dss/hdmi.h | 38 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 269 +++++++------ drivers/video/fbdev/omap2/dss/hdmi4_core.c | 14 - drivers/video/fbdev/omap2/dss/hdmi4_core.h | 4 - drivers/video/fbdev/omap2/dss/hdmi5.c | 265 ++++++------- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 9 +- drivers/video/fbdev/omap2/dss/hdmi5_core.h | 2 - drivers/video/fbdev/omap2/dss/hdmi_common.c | 2 - drivers/video/fbdev/omap2/dss/hdmi_wp.c | 16 +- include/sound/omap-hdmi-audio.h | 43 +++ include/video/omapdss.h | 40 -- sound/soc/omap/Kconfig | 26 +- sound/soc/omap/Makefile | 6 +- sound/soc/omap/omap-hdmi-audio.c | 407 ++++++++++++++++++++ sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 ----------------- sound/soc/omap/omap-hdmi.h | 38 -- 20 files changed, 781 insertions(+), 1011 deletions(-) create mode 100644 include/sound/omap-hdmi-audio.h create mode 100644 sound/soc/omap/omap-hdmi-audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

5 51

[alsa-devel] [PATCHv6 1/9] ASOC: wm8971: clean for checkpatch --strict
by Xavier Hsu 02 Dec '14

02 Dec '14

Before adding wm8973 support, clean up existing wm8971 style. Signed-off-by: Xavier Hsu <xavier.hsu(a)linaro.org> Signed-off-by: Andy Green <andy.green(a)linaro.org> --- sound/soc/codecs/wm8971.c | 110 +++++++++++++++++++++++---------------------- sound/soc/codecs/wm8971.h | 30 ++++++------- 2 files changed, 72 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..01c76a4 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -29,9 +29,9 @@ #include "wm8971.h" -#define WM8971_REG_COUNT 43 +#define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; +static struct workqueue_struct *wm8971_workq; /* codec private data */ struct wm8971_priv { @@ -92,25 +92,28 @@ static const struct reg_default wm8971_reg_defaults[] = { #define wm8971_reset(c) snd_soc_write(c, WM8971_RESET, 0) /* WM8971 Controls */ -static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; -static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", - "200Hz @ 48kHz" }; -static const char *wm8971_treble[] = { "8kHz", "4kHz" }; -static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; -static const char *wm8971_ng_type[] = { "Constant PGA Gain", - "Mute ADC Output" }; -static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; -static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", - "Mono (Right)", "Digital Mono"}; -static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; -static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", - "Differential"}; -static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", - "Differential"}; -static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; -static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; -static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", - "L + R Invert"}; +static const char const *wm8971_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char const *wm8971_bass_filter[] = {"130Hz @ 48kHz", + "200Hz @ 48kHz"}; +static const char const *wm8971_treble[] = {"8kHz", "4kHz"}; +static const char const *wm8971_alc_func[] = {"Off", "Right", + "Left", "Stereo"}; +static const char const *wm8971_ng_type[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const char const *wm8971_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"}; +static const char const *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const char const *wm8971_dac_phase[] = {"Non Inverted", "Inverted"}; +static const char const *wm8971_lline_mux[] = {"Line", "NC", "NC", + "PGA", "Differential"}; +static const char const *wm8971_rline_mux[] = {"Line", "Mic", "NC", + "PGA", "Differential"}; +static const char const *wm8971_lpga_sel[] = {"Line", "NC", "NC", + "Differential"}; +static const char const *wm8971_rpga_sel[] = {"Line", "Mic", "NC", + "Differential"}; +static const char const *wm8971_adcpol[] = {"Normal", "L Invert", + "R Invert", "L + R Invert"}; static const struct soc_enum wm8971_enum[] = { SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ @@ -136,24 +139,24 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = { SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, - WM8971_ROUT1V, 7, 1, 0), + WM8971_ROUT1V, 7, 1, 0), SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, - WM8971_ROUT2V, 7, 1, 0), + WM8971_ROUT2V, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, - WM8971_LOUTM2, 4, 7, 1), + WM8971_LOUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, - WM8971_ROUTM2, 4, 7, 1), + WM8971_ROUTM2, 4, 7, 1), SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, - WM8971_MOUTM2, 4, 7, 1), + WM8971_MOUTM2, 4, 7, 1), SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, - WM8971_ROUT1V, 0, 127, 0), + WM8971_ROUT1V, 0, 127, 0), SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, - WM8971_ROUT2V, 0, 127, 0), + WM8971_ROUT2V, 0, 127, 0), SOC_ENUM("Bass Boost", wm8971_enum[0]), SOC_ENUM("Bass Filter", wm8971_enum[1]), @@ -238,14 +241,14 @@ SOC_DAPM_ENUM("Route", wm8971_enum[13]); static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_left_mixer_controls[0], - ARRAY_SIZE(wm8971_left_mixer_controls)), + &wm8971_left_mixer_controls[0], + ARRAY_SIZE(wm8971_left_mixer_controls)), SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, - &wm8971_right_mixer_controls[0], - ARRAY_SIZE(wm8971_right_mixer_controls)), + &wm8971_right_mixer_controls[0], + ARRAY_SIZE(wm8971_right_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, - &wm8971_mono_mixer_controls[0], - ARRAY_SIZE(wm8971_mono_mixer_controls)), + &wm8971_mono_mixer_controls[0], + ARRAY_SIZE(wm8971_mono_mixer_controls)), SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), @@ -260,18 +263,18 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, - &wm8971_left_pga_controls), + &wm8971_left_pga_controls), SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, - &wm8971_right_pga_controls), + &wm8971_right_pga_controls), SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_left_line_controls), + &wm8971_left_line_controls), SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, - &wm8971_right_line_controls), + &wm8971_right_line_controls), SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, - &wm8971_monomux_controls), + &wm8971_monomux_controls), SND_SOC_DAPM_OUTPUT("LOUT1"), SND_SOC_DAPM_OUTPUT("ROUT1"), @@ -431,7 +434,7 @@ static int get_coeff(int mclk, int rate) } static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) + int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -449,7 +452,7 @@ static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, - unsigned int fmt) + unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; u16 iface = 0; @@ -507,8 +510,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); @@ -553,7 +556,7 @@ static int wm8971_mute(struct snd_soc_dai *dai, int mute) } static int wm8971_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) + enum snd_soc_bias_level level) { u16 pwr_reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; @@ -580,11 +583,12 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec, } #define WM8971_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8971_dai_ops = { .hw_params = wm8971_pcm_hw_params, @@ -616,6 +620,7 @@ static void wm8971_work(struct work_struct *work) container_of(work, struct snd_soc_dapm_context, delayed_work.work); struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + wm8971_set_bias_level(codec, codec->dapm.bias_level); } @@ -637,7 +642,7 @@ static int wm8971_resume(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); } return 0; @@ -660,7 +665,7 @@ static int wm8971_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, - msecs_to_jiffies(1000)); + msecs_to_jiffies(1000)); /* set the update bits */ snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100); @@ -675,7 +680,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) return ret; } - /* power down chip */ static int wm8971_remove(struct snd_soc_codec *codec) { @@ -729,8 +733,8 @@ static int wm8971_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8971); - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8971, &wm8971_dai, 1); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8971, + &wm8971_dai, 1); return ret; } diff --git a/sound/soc/codecs/wm8971.h b/sound/soc/codecs/wm8971.h index f31c38f..dc38e21 100644 --- a/sound/soc/codecs/wm8971.h +++ b/sound/soc/codecs/wm8971.h @@ -22,25 +22,25 @@ #define WM8971_ADCDAC 0x05 #define WM8971_IFACE 0x07 #define WM8971_SRATE 0x08 -#define WM8971_LDAC 0x0a -#define WM8971_RDAC 0x0b -#define WM8971_BASS 0x0c +#define WM8971_LDAC 0x0a +#define WM8971_RDAC 0x0b +#define WM8971_BASS 0x0c #define WM8971_TREBLE 0x0d #define WM8971_RESET 0x0f -#define WM8971_ALC1 0x11 -#define WM8971_ALC2 0x12 -#define WM8971_ALC3 0x13 +#define WM8971_ALC1 0x11 +#define WM8971_ALC2 0x12 +#define WM8971_ALC3 0x13 #define WM8971_NGATE 0x14 -#define WM8971_LADC 0x15 -#define WM8971_RADC 0x16 -#define WM8971_ADCTL1 0x17 -#define WM8971_ADCTL2 0x18 -#define WM8971_PWR1 0x19 -#define WM8971_PWR2 0x1a -#define WM8971_ADCTL3 0x1b +#define WM8971_LADC 0x15 +#define WM8971_RADC 0x16 +#define WM8971_ADCTL1 0x17 +#define WM8971_ADCTL2 0x18 +#define WM8971_PWR1 0x19 +#define WM8971_PWR2 0x1a +#define WM8971_ADCTL3 0x1b #define WM8971_ADCIN 0x1f -#define WM8971_LADCIN 0x20 -#define WM8971_RADCIN 0x21 +#define WM8971_LADCIN 0x20 +#define WM8971_RADCIN 0x21 #define WM8971_LOUTM1 0x22 #define WM8971_LOUTM2 0x23 #define WM8971_ROUTM1 0x24 -- 1.7.9.5

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Re: [alsa-devel] Add new machine support for Dell
by Takashi Iwai 02 Dec '14

02 Dec '14

At Tue, 2 Dec 2014 07:03:45 +0000, Kailang wrote: > > Hi Takashi, > > Dell has new machine for ALC293. > It supports headset Mic and Headphone Mic. Thanks, applied with Cc to stable now. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-headset-mic-support-for-new-dell-machine.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] [PATCH v2] ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell/Haswell
by Jie Yang 02 Dec '14

02 Dec '14

Move capture PCM pin from PCM4 to PCM0 for Broadwell/Haswell. This will allow us to integrate with pulseaudio better for usually default device is set to 0. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 15 ++------------- sound/soc/intel/haswell.c | 14 ++------------ sound/soc/intel/sst-haswell-pcm.c | 34 ++++++++++++++++------------------ 3 files changed, 20 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 52cb764..c256764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ { .name = "System PCM", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -180,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .init = broadwell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -214,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture PCM", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ { /* SSP0 - Codec */ diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 3981982..cb8a482 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -109,7 +109,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { /* Front End DAI links */ { .name = "System", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -118,6 +118,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .init = haswell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -152,17 +153,6 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, /* Back End DAI links */ { diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5f70d81..0180b38 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -309,7 +309,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -396,8 +396,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, /* DSP stream type depends on DAI ID */ switch (rtd->cpu_dai->id) { case 0: - stream_type = SST_HSW_STREAM_TYPE_SYSTEM; - module_id = SST_HSW_MODULE_PCM_SYSTEM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + } + else { + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + } break; case 1: case 2: @@ -410,10 +416,6 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, path_id = SST_HSW_STREAM_PATH_SSP0_OUT; module_id = SST_HSW_MODULE_PCM_REFERENCE; break; - case 4: - stream_type = SST_HSW_STREAM_TYPE_CAPTURE; - module_id = SST_HSW_MODULE_PCM_CAPTURE; - break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); @@ -779,6 +781,13 @@ static struct snd_soc_dai_driver hsw_dais[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, + }, }, { /* PCM */ @@ -815,17 +824,6 @@ static struct snd_soc_dai_driver hsw_dais[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, - { - .name = "Capture Pin", - .id = HSW_PCM_DAI_ID_CAPTURE, - .capture = { - .stream_name = "Analog Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, - }, - }, }; static const struct snd_soc_dapm_widget widgets[] = { -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: fsl_ssi: free irq before irq_dispose_mapping()
by Jiada Wang 01 Dec '14

01 Dec '14

irq_dispose_mapping() in turns calls unregister_irq_proc(), which will remove irq proc entry, if IRQ is not freed before calling of irq_dispose_mapping(), then it will cause kernel warning. By free IRQ before irq_dispose_mapping(), this patch fix the following kernel warning found when remove of fsl_ssi driver: [ 31.515336] ------------[ cut here ]------------ [ 31.520091] WARNING: CPU: 2 PID: 434 at fs/proc/generic.c:521 remove_proc_entry+0x14c/0x16c() [ 31.528708] remove_proc_entry: removing non-empty directory 'irq/79', leaking at least '202c000.ss' [ 31.537911] Modules linked in: snd_soc_wm8962 snd_soc_imx_wm8962 snd_soc_fsl_ssi(-) evbug [ 31.546249] CPU: 2 PID: 434 Comm: rmmod Not tainted 3.18.0-rc6-00028-g3314bf6-dirty #1 [ 31.554235] Backtrace: [ 31.556816] [<80011ea8>] (dump_backtrace) from [<80012044>] (show_stack+0x18/0x1c) [ 31.564416] r6:80142c88 r5:00000000 r4:00000000 r3:00000000 [ 31.570267] [<8001202c>] (show_stack) from [<806980ec>] (dump_stack+0x88/0xa4) [ 31.577588] [<80698064>] (dump_stack) from [<80029d78>] (warn_slowpath_common+0x70/0x94) [ 31.585711] r5:00000009 r4:bb61fd90 [ 31.589423] [<80029d08>] (warn_slowpath_common) from [<80029e40>] (warn_slowpath_fmt+0x38/0x40) [ 31.598187] r8:bb61fdfe r7:be05d76d r6:be05d9a8 r5:00000002 r4:be05d700 [ 31.605054] [<80029e0c>] (warn_slowpath_fmt) from [<80142c88>] (remove_proc_entry+0x14c/0x16c) [ 31.613709] r3:806a79c0 r2:808229a0 [ 31.617371] [<80142b3c>] (remove_proc_entry) from [<80070380>] (unregister_irq_proc+0x94/0xb8) [ 31.625989] r10:00000000 r8:8000ede4 r7:80955f2c r6:0000004f r5:8118e738 r4:be00af00 [ 31.633952] [<800702ec>] (unregister_irq_proc) from [<80069dac>] (free_desc+0x2c/0x64) [ 31.641898] r6:0000004f r5:80955f38 r4:be00af00 [ 31.646604] [<80069d80>] (free_desc) from [<80069e68>] (irq_free_descs+0x4c/0x8c) [ 31.654092] r7:00000081 r6:00000001 r5:0000004f r4:00000001 [ 31.659863] [<80069e1c>] (irq_free_descs) from [<8006fc3c>] (irq_dispose_mapping+0x40/0x5c) [ 31.668247] r6:be17b844 r5:be17b800 r4:0000004f r3:802c5ec0 [ 31.673998] [<8006fbfc>] (irq_dispose_mapping) from [<7f004ea4>] (fsl_ssi_remove+0x58/0x70 [snd_so) [ 31.683948] r4:bb5bba10 r3:00000001 [ 31.687618] [<7f004e4c>] (fsl_ssi_remove [snd_soc_fsl_ssi]) from [<803720a0>] (platform_drv_remove) [ 31.697564] r5:7f0064f8 r4:be17b810 [ 31.701195] [<80372080>] (platform_drv_remove) from [<80370494>] (__device_release_driver+0x78/0xc) [ 31.710361] r5:7f0064f8 r4:be17b810 [ 31.713987] [<8037041c>] (__device_release_driver) from [<80370d20>] (driver_detach+0xbc/0xc0) [ 31.722631] r5:7f0064f8 r4:be17b810 [ 31.726259] [<80370c64>] (driver_detach) from [<80370304>] (bus_remove_driver+0x54/0x98) [ 31.734382] r6:00000800 r5:00000000 r4:7f0064f8 r3:bb67f500 [ 31.740149] [<803702b0>] (bus_remove_driver) from [<80371398>] (driver_unregister+0x30/0x50) [ 31.748617] r4:7f0064f8 r3:bd9f7080 [ 31.752245] [<80371368>] (driver_unregister) from [<80371f3c>] (platform_driver_unregister+0x14/0x) [ 31.761498] r4:7f00655c r3:7f005a70 [ 31.765130] [<80371f28>] (platform_driver_unregister) from [<7f005a84>] (fsl_ssi_driver_exit+0x14/) [ 31.776147] [<7f005a70>] (fsl_ssi_driver_exit [snd_soc_fsl_ssi]) from [<8008ed80>] (SyS_delete_mod) [ 31.786553] [<8008ec64>] (SyS_delete_module) from [<8000ec20>] (ret_fast_syscall+0x0/0x48) [ 31.794824] r6:00c46d18 r5:00000800 r4:00c46d18 [ 31.799530] ---[ end trace 954e8a3a15379e52 ]--- Moreover replace devm_request_irq() with request_irq() since there is no need to use it as now driver always frees IRQ manually. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> --- sound/soc/fsl/fsl_ssi.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e695517..cfdb337 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1400,9 +1400,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) } if (ssi_private->use_dma) { - ret = devm_request_irq(&pdev->dev, ssi_private->irq, - fsl_ssi_isr, 0, dev_name(&pdev->dev), - ssi_private); + ret = request_irq(ssi_private->irq, fsl_ssi_isr, 0, + dev_name(&pdev->dev), ssi_private); if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); @@ -1412,7 +1411,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); if (ret) - goto error_asoc_register; + goto error_debugfs_create; /* * If codec-handle property is missing from SSI node, we assume @@ -1453,6 +1452,10 @@ done: error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); +error_debugfs_create: + if (ssi_private->use_dma) + free_irq(ssi_private->irq, ssi_private); + error_irq: snd_soc_unregister_component(&pdev->dev); @@ -1473,6 +1476,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); + if (ssi_private->use_dma) + free_irq(ssi_private->irq, ssi_private); + if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); -- 1.9.3

7 17

[alsa-devel] [PATCH] ALSA: usb: Add support for Zoom R16/24 capture and midi interfaces
by Panu Matilainen 01 Dec '14

01 Dec '14

This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- sound/usb/quirks-table.h | 34 ++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..73621c9 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3202,6 +3202,40 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { + /* ZOOM R16/24 in audio interface mode */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Control */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback, sound is missing atm */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a5941f8..2e49193 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1160,6 +1160,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- 1.9.3

2 3

[alsa-devel] [PATCH v2] ALSA: usb: Add support for Zoom R16/24 capture and midi interfaces
by Panu Matilainen 01 Dec '14

01 Dec '14

This makes the midi interface and capture work out of the box with R16 (and presumably R24 too but untested). Playback stream would also seem to function fine except for one caveat: no sound is produced, so it is disabled for now. Mixer descriptors are garbage and will require further quirks to enable functionality, also disabled here. Signed-off-by: Panu Matilainen <pmatilai(a)laiskiainen.org> --- sound/usb/quirks-table.h | 40 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 8 ++++++++ 2 files changed, 48 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c657752..c955663 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3203,6 +3203,46 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* + * ZOOM R16/24 in audio interface mode. + * Mixer descriptors are garbage, further quirks will be needed + * to make any of it functional, thus disabled for now. + * Playback stream appears to start and run fine but no sound + * is produced, so also disabled for now. + */ + USB_DEVICE(0x1686, 0x00dd), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + /* Mixer */ + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Playback */ + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + /* Capture */ + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE, + }, + { + /* Midi */ + .ifnum = 3, + .type = QUIRK_MIDI_STANDARD_INTERFACE + }, + { + .ifnum = -1 + }, + } + } +}, + +{ + /* * Some USB MIDI devices don't have an audio control interface, * so we have to grab MIDI streaming interfaces here. */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a5941f8..2e49193 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1160,6 +1160,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, break; } } + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && + (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(1); } /* -- 1.9.3

2 1

[alsa-devel] [GIT PULL] Add sst audio firmware for baytrail platforms
by Vinod Koul 01 Dec '14

01 Dec '14

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Baytrail platforms. This firmware is using the same license as another version of the Baytrail firmware already available in linux-firmware git. The following changes since commit 0e5f63771d0df6d7859f7c4100a74d737c62ac88: are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git byt Vinod Koul (1): linux-firmware: add sst audio firmware for baytrail platforms WHENCE | 8 ++++++++ intel/fw_sst_0f28.bin | Bin 0 -> 647078 bytes 2 files changed, 8 insertions(+), 0 deletions(-) create mode 100644 intel/fw_sst_0f28.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH] Sound: drivers: virmidi: fixed code style issues
by Kyle Chamberlin 30 Nov '14

30 Nov '14

Fixed some minor code style issues and also removed some assignments inside of if conditionals. Signed-off-by: Kyle Chamberlin <KyleChamberlin(a)project20million.org> --- sound/drivers/virmidi.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index b17872429..0af88d9 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -99,30 +99,33 @@ static int snd_virmidi_probe(struct platform_device *devptr) if (midi_devs[dev] > MAX_MIDI_DEVICES) { snd_printk(KERN_WARNING - "too much midi devices for virmidi %d: " - "force to use %d\n", dev, MAX_MIDI_DEVICES); + "too much midi devices for virmidi %d: force to use %d\n", + dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { struct snd_rawmidi *rmidi; struct snd_virmidi_dev *rdev; - if ((err = snd_virmidi_new(card, idx, &rmidi)) < 0) + + err = snd_virmidi_new(card, idx, &rmidi); + if (err < 0) goto __nodev; rdev = rmidi->private_data; vmidi->midi[idx] = rmidi; strcpy(rmidi->name, "Virtual Raw MIDI"); rdev->seq_mode = SNDRV_VIRMIDI_SEQ_DISPATCH; } - + strcpy(card->driver, "VirMIDI"); strcpy(card->shortname, "VirMIDI"); sprintf(card->longname, "Virtual MIDI Card %i", dev + 1); - if ((err = snd_card_register(card)) == 0) { + err = snd_card_register(card); + if (err) { platform_set_drvdata(devptr, card); return 0; } - __nodev: +__nodev: snd_card_free(card); return err; } @@ -157,13 +160,15 @@ static int __init alsa_card_virmidi_init(void) { int i, cards, err; - if ((err = platform_driver_register(&snd_virmidi_driver)) < 0) + err = platform_driver_register(&snd_virmidi_driver); + if (err < 0) return err; cards = 0; for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; - if (! enable[i]) + + if (!enable[i]) continue; device = platform_device_register_simple(SND_VIRMIDI_DRIVER, i, NULL, 0); -- 2.1.3

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[alsa-devel] [alsa/virmidi] BUG: unable to handle kernel NULL pointer dereference at 000002a0
by Fengguang Wu 30 Nov '14

30 Nov '14

Hi Kyle, 0day kernel testing robot got the below dmesg and the first bad commit is git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next commit 316638a5030a04bb3259dcbca0632281001a4b24 Author: Kyle Chamberlin <kylechamberlin(a)project20million.org> AuthorDate: Fri Nov 28 13:59:56 2014 -0500 Commit: Takashi Iwai <tiwai(a)suse.de> CommitDate: Sat Nov 29 21:34:47 2014 +0100 ALSA: virmidi: fixed code style issues Fixed some minor code style issues and also removed some assignments inside of if conditionals. Signed-off-by: Kyle Chamberlin <kylechamberlin(a)project20million.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> In particular, this chunk introduced the behavior change: - if ((err = snd_card_register(card)) == 0) { + err = snd_card_register(card); + if (err) { +-------------------------------------------------+------------+------------+------------+ | | 31514bfb4a | 316638a503 | 795172aaa6 | +-------------------------------------------------+------------+------------+------------+ | boot_successes | 67 | 0 | 0 | | boot_failures | 13 | 20 | 14 | | BUG:kernel_early_hang_without_any_printk_output | 13 | 0 | 2 | | BUG:unable_to_handle_kernel | 0 | 20 | 12 | | Oops | 0 | 20 | 12 | | EIP_is_at_snd_card_free | 0 | 20 | 12 | | Kernel_panic-not_syncing:Fatal_exception | 0 | 20 | 12 | | backtrace:platform_device_unregister | 0 | 20 | 12 | | backtrace:alsa_card_virmidi_init | 0 | 20 | 12 | | backtrace:kernel_init_freeable | 0 | 20 | 12 | +-------------------------------------------------+------------+------------+------------+ [ 1.688337] usbcore: registered new interface driver line6usb [ 1.689386] dgap: dgap-1.3-16, Digi International Part Number 40002347_C [ 1.690258] dgap: For the tools package please visit http://www.digi.com [ 1.694793] BUG: unable to handle kernel NULL pointer dereference at 000002a0 [ 1.695739] IP: [<7940555f>] snd_card_free+0x36/0x7f [ 1.696409] *pde = 00000000 [ 1.696686] Oops: 0002 [#1] [ 1.696686] Modules linked in: [ 1.696686] CPU: 0 PID: 1 Comm: swapper Not tainted 3.18.0-rc5-00179-g316638a #101 [ 1.696686] Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.7.5-20140531_083030-gandalf 04/01/2014 [ 1.696686] task: 8a068000 ti: 8a064000 task.ti: 8a064000 [ 1.696686] EIP: 0060:[<7940555f>] EFLAGS: 00010246 CPU: 0 [ 1.696686] EIP is at snd_card_free+0x36/0x7f [ 1.696686] EAX: 00000000 EBX: 8a065e04 ECX: 7a354880 EDX: 798a5d21 [ 1.696686] ESI: 00000000 EDI: 00000000 EBP: 8a065e44 ESP: 8a065e04 [ 1.696686] DS: 007b ES: 007b FS: 0000 GS: 00e0 SS: 0068 [ 1.696686] CR0: 8005003b CR2: 000002a0 CR3: 01d49000 CR4: 000006d0 [ 1.696686] Stack: [ 1.696686] 00000000 00000001 dead4ead ffffffff ffffffff 7a354880 00000000 00000000 [ 1.696686] 798a5d21 00000000 00000000 8a065e30 8a065e30 006d61bc 7808440c 00000001 [ 1.696686] 8a065e4c 7943d967 8a065e54 792c0e93 8a065e68 792befa9 78084440 7808440c [ 1.696686] Call Trace: [ 1.696686] [<7943d967>] snd_virmidi_remove+0x13/0x17 [ 1.696686] [<792c0e93>] platform_drv_remove+0x14/0x16 [ 1.696686] [<792befa9>] __device_release_driver+0xc1/0x121 [ 1.696686] [<792bf028>] device_release_driver+0x1f/0x2a [ 1.696686] [<792be929>] bus_remove_device+0x166/0x17d [ 1.696686] [<792bb4ad>] device_del+0x14b/0x1d8 [ 1.696686] [<792bb4ad>] ? device_del+0x14b/0x1d8 [ 1.696686] [<792c1407>] platform_device_del+0x32/0xa3 [ 1.696686] [<792c1488>] platform_device_unregister+0x10/0x1a [ 1.696686] [<79cee7bf>] alsa_card_virmidi_init+0xd1/0x118 [ 1.696686] [<79cee6ee>] ? alsa_seq_midi_init+0x34/0x34 [ 1.696686] [<79000551>] do_one_initcall+0x186/0x288 [ 1.696686] [<79cb4400>] ? rdinit_setup+0x18/0x22 [ 1.696686] [<79060c87>] ? parse_args+0x4a6/0x579 [ 1.696686] [<7906be5f>] ? local_clock+0x2b/0x35 [ 1.696686] [<79cb44f6>] ? do_early_param+0xac/0xac [ 1.696686] [<79cb4f46>] kernel_init_freeable+0xfd/0x1bd [ 1.696686] [<79cb4f46>] ? kernel_init_freeable+0xfd/0x1bd [ 1.696686] [<7959bbfc>] kernel_init+0xe/0x13c [ 1.696686] [<795a7cc0>] ret_from_kernel_thread+0x20/0x30 [ 1.696686] [<7959bbee>] ? rest_init+0xa7/0xa7 [ 1.696686] Code: 26 00 89 c6 65 a1 14 00 00 00 89 45 f4 31 c0 c7 45 c0 00 00 00 00 8d 5d c0 b9 80 48 35 7a ba 21 5d 8a 79 8d 45 c4 e8 bc b2 c6 ff <89> 9e a0 02 00 00 89 f0 e8 80 fe ff ff 85 c0 0f 95 c2 0f b6 f2 [ 1.696686] EIP: [<7940555f>] snd_card_free+0x36/0x7f SS:ESP 0068:8a065e04 [ 1.696686] CR2: 00000000000002a0 [ 1.696686] ---[ end trace bc9b8dd3625e806d ]--- [ 1.696686] Kernel panic - not syncing: Fatal exception git bisect start 795172aaa6701c4b66b449e79b49c41e044156c6 5d01410fe4d92081f349b013a2e7a95429e4f2c9 -- git bisect bad 5d81516a74d44d9301e140ff97e06b37eaf5831d # 06:13 0- 13 Merge 'luto/userns/setgroups' into devel-lkp-hsx01-smoke-201411300558 git bisect good b5675b9e191a3981a38f5c94b6dd8b6fa4ba8091 # 06:23 20+ 3 0day base guard for 'devel-lkp-hsx01-smoke-201411300558' git bisect bad e929ef74cdd822177ad2322514a40127ad1e696c # 06:31 0- 4 Merge 'sound/for-next' into devel-lkp-hsx01-smoke-201411300558 git bisect good 2603fe21b764eb7412598c8c6cd6199fb8b1d9c5 # 06:41 20+ 10 ALSA: es18xx: Add GPO controls git bisect good 2bfb14c3b8fbc787ff4478f9d77ecee78cb922fe # 06:51 20+ 0 ALSA: usb-audio: Add Xonar U1 resume support git bisect good 37e661ee10c6d0d1310c62b3d29ae9a63073ac5d # 06:58 20+ 5 ALSA: hda - Add AZX_DCAPS_SNOOP_OFF (and refactor snoop setup) git bisect good 6eb6c81eee2a6270b39ca02a446f3ccece24b6f8 # 07:16 20+ 0 ALSA: dice: Split stream functionality into a file git bisect good 8832c5a74ba3506c51b6637ac78941fcd21afbef # 07:25 20+ 0 ALSA: speakers: Rename to oxfw and rename some members git bisect good 3713d93a6a12f8629c2660bb4a30d48b98105fca # 07:32 20+ 4 ALSA: oxfw: Split PCM functionality to a new file git bisect bad 316638a5030a04bb3259dcbca0632281001a4b24 # 07:40 0- 17 ALSA: virmidi: fixed code style issues git bisect good 31514bfb4ab8ba6f93b5ce5fcc543cb2ac4f96e5 # 07:46 20+ 1 ALSA: oxfw: Split control functionality to a new file # first bad commit: [316638a5030a04bb3259dcbca0632281001a4b24] ALSA: virmidi: fixed code style issues git bisect good 31514bfb4ab8ba6f93b5ce5fcc543cb2ac4f96e5 # 07:48 60+ 13 ALSA: oxfw: Split control functionality to a new file # extra tests on HEAD of linux-devel/devel-lkp-hsx01-smoke-201411300558 git bisect bad 795172aaa6701c4b66b449e79b49c41e044156c6 # 07:48 0- 14 0day head guard for 'devel-lkp-hsx01-smoke-201411300558' # extra tests on tree/branch sound/for-next git bisect bad 316638a5030a04bb3259dcbca0632281001a4b24 # 07:49 0- 20 ALSA: virmidi: fixed code style issues # extra tests on tree/branch linus/master git bisect good 7a5a4f978750756755dc839014e13d1b088ccc8e # 07:51 60+ 10 Merge branch 'irq-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip # extra tests on tree/branch next/master git bisect good 29340b8ee0b4def844aa8d6b09babcede80993d2 # 07:54 60+ 80 Add linux-next specific files for 20141128 This script may reproduce the error. ---------------------------------------------------------------------------- #!/bin/bash kernel=$1 initrd=quantal-core-i386.cgz wget --no-clobber https://github.com/fengguang/reproduce-kernel-bug/raw/master/initrd/$initrd kvm=( qemu-system-x86_64 -cpu kvm64 -enable-kvm -kernel $kernel -initrd $initrd -m 320 -smp 2 -net nic,vlan=1,model=e1000 -net user,vlan=1 -boot order=nc -no-reboot -watchdog i6300esb -rtc base=localtime -serial stdio -display none -monitor null ) append=( hung_task_panic=1 earlyprintk=ttyS0,115200 debug apic=debug sysrq_always_enabled rcupdate.rcu_cpu_stall_timeout=100 panic=-1 softlockup_panic=1 nmi_watchdog=panic oops=panic load_ramdisk=2 prompt_ramdisk=0 console=ttyS0,115200 console=tty0 vga=normal root=/dev/ram0 rw drbd.minor_count=8 ) "${kvm[@]}" --append "${append[*]}" ---------------------------------------------------------------------------- Thanks, Fengguang _______________________________________________ LKP mailing list LKP(a)linux.intel.com

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[alsa-devel] ALSA/ASOC new device and routing
by Eli Britstein 30 Nov '14

30 Nov '14

Hi I'm a newbie to ALSA. I'm targeting to use ALSA or ASoC solution in order to route capture/playback streams of a USB headset (that uses ALSA) to/from my own DSP device. I saw DPCM which is a part of ASoC manages dynamically connections between FE and BE streams. In my product, I just need them to be connected. No need for the "dynamically" part. My questions: 1. Can someone please recommend a step by step guide, or an example how to make my DSP "look like" a BE? Do I need to write an ALSA driver for it or a ASOC driver? 2. After I have it as a BE, how can I connect the streams? I want the transfers to be done in kernel level, without need for application interventions for each buffer. 3. I'm using kernel 2.6.10 (quite old, I know). I don't want to port my entire kernel to a new one. As my requirement is not very complex, I don't think I'd encounter too many issues. However, if someone can point the important parts I need to back port, I'd appreciate it. Alternatively, if someone sees a showstopper in this approach, I'd be grateful to be aware of it. Thanks, Best regards, Eli ________________________________ This email and any files transmitted with it are confidential material. They are intended solely for the use of the designated individual or entity to whom they are addressed. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, use, distribution or copying of this communication is strictly prohibited and may be unlawful. If you have received this email in error please immediately notify the sender and delete or destroy any copy of this message

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[alsa-devel] AKAI EIE PRO
by Loki Harfagr 28 Nov '14

28 Nov '14

(hope that's correct to post that way through email since I'm just adding an answer to a specific question in a specific thread), here we go: > Daniel Mack daniel at zonque.org > Sun Sep 7 13:39:12 CEST 2014 > Previous message: [alsa-devel] AKAI EIE PRO > Next message: [alsa-devel] AKAI EIE PRO > Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] > On 09/07/2014 01:30 PM, humbert.olivier.1 at free.fr wrote: > > if that can help, we had an user on http://www.linuxmao.org with an EIE PRO. > > Here you have his lsusb -v : > > > > Bus 001 Device 006: ID 09e8:0010 AKAI Professional M.I. Corp. > > Couldn't open device, some information will be missing > > Device Descriptor: > > bLength 18 > > bDescriptorType 1 > > bcdUSB 2.00 > > bDeviceClass 255 Vendor Specific Class > > bDeviceSubClass 255 Vendor Specific Subclass > > bDeviceProtocol 255 Vendor Specific Protocol > > According to the tech specs, this should be a USB hub, but it isn't. Or > is this just one of the sub-devices? What does 'lsusb -t' say, and does > Linux see something like a mouse or mass storage device when connected > to one of the hub ports? To test accordingly to the question a USB device (Webcam C905) is plugged in one of the EIE Pro. hub ports 1) AKAI EIE Pro *OFF*: ------------------- $ lsusb Bus 002 Device 002: ID 152d:0551 JMicron Technology Corp. / JMicron USA Technology Corp. Bus 005 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 006 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 006 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 005 Device 003: ID 05e3:0608 Genesys Logic, Inc. USB-2.0 4-Port HUB Bus 005 Device 004: ID 174c:5106 ASMedia Technology Inc. Transcend StoreJet 25M3 Bus 006 Device 003: ID 058f:6362 Alcor Micro Corp. Flash Card Reader/Writer Bus 005 Device 006: ID 046d:c318 Logitech, Inc. Illuminated Keyboard Bus 005 Device 007: ID 047d:1020 Kensington Expert Mouse Trackball Bus 005 Device 012: ID 058f:9254 Alcor Micro Corp. Hub ------------------- 2) AKAI EIE Pro *ON*: ------------------- $ lsusb Bus 002 Device 002: ID 152d:0551 JMicron Technology Corp. / JMicron USA Technology Corp. Bus 005 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub ven. nov. 28 19:27:58 CET 2014 Bus 006 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 006 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 005 Device 003: ID 05e3:0608 Genesys Logic, Inc. USB-2.0 4-Port HUB Bus 005 Device 004: ID 174c:5106 ASMedia Technology Inc. Transcend StoreJet 25M3 Bus 005 Device 018: ID 0409:005a NEC Corp. HighSpeed Hub Bus 006 Device 003: ID 058f:6362 Alcor Micro Corp. Flash Card Reader/Writer Bus 005 Device 006: ID 046d:c318 Logitech, Inc. Illuminated Keyboard Bus 005 Device 007: ID 047d:1020 Kensington Expert Mouse Trackball Bus 005 Device 012: ID 058f:9254 Alcor Micro Corp. Hub Bus 005 Device 019: ID 046d:080a Logitech, Inc. Portable Webcam C905 Bus 005 Device 020: ID 09e8:0010 AKAI Professional M.I. Corp. ------------------- ------------------- $ lsusb -t /: Bus 06.Port 1: Dev 1, Class=root_hub, Driver=ehci-pci/2p, 480M |__ Port 1: Dev 2, If 0, Class=Hub, Driver=hub/8p, 480M |__ Port 7: Dev 3, If 0, Class=Mass Storage, Driver=usb-storage, 480M /: Bus 05.Port 1: Dev 1, Class=root_hub, Driver=ehci-pci/2p, 480M |__ Port 1: Dev 2, If 0, Class=Hub, Driver=hub/6p, 480M |__ Port 1: Dev 3, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 1: Dev 6, If 0, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 1: Dev 6, If 1, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 2: Dev 7, If 0, Class=Human Interface Device, Driver=usbhid, 1.5M |__ Port 3: Dev 12, If 0, Class=Hub, Driver=hub/4p, 12M |__ Port 2: Dev 4, If 0, Class=Mass Storage, Driver=usb-storage, 480M |__ Port 4: Dev 18, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 1: Dev 19, If 0, Class=Video, Driver=uvcvideo, 480M |__ Port 1: Dev 19, If 1, Class=Video, Driver=uvcvideo, 480M |__ Port 1: Dev 19, If 2, Class=Audio, Driver=snd-usb-audio, 480M |__ Port 1: Dev 19, If 3, Class=Audio, Driver=snd-usb-audio, 480M |__ Port 4: Dev 20, If 0, Class=Vendor Specific Class, Driver=, 480M |__ Port 4: Dev 20, If 1, Class=Vendor Specific Class, Driver=, 480M /: Bus 04.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 5000M /: Bus 03.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 480M /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 5000M |__ Port 2: Dev 2, If 0, Class=Mass Storage, Driver=usb-storage, 5000M /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 480M ------------------- 3) Then, the Webcam plugged in the EIE Pro. has been unplugged: ------------------- $ lsusb -t /: Bus 06.Port 1: Dev 1, Class=root_hub, Driver=ehci-pci/2p, 480M |__ Port 1: Dev 2, If 0, Class=Hub, Driver=hub/8p, 480M |__ Port 7: Dev 3, If 0, Class=Mass Storage, Driver=usb-storage, 480M /: Bus 05.Port 1: Dev 1, Class=root_hub, Driver=ehci-pci/2p, 480M |__ Port 1: Dev 2, If 0, Class=Hub, Driver=hub/6p, 480M |__ Port 1: Dev 3, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 1: Dev 6, If 0, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 1: Dev 6, If 1, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 2: Dev 7, If 0, Class=Human Interface Device, Driver=usbhid, 1.5M |__ Port 3: Dev 12, If 0, Class=Hub, Driver=hub/4p, 12M |__ Port 2: Dev 4, If 0, Class=Mass Storage, Driver=usb-storage, 480M |__ Port 4: Dev 18, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 4: Dev 20, If 0, Class=Vendor Specific Class, Driver=, 480M |__ Port 4: Dev 20, If 1, Class=Vendor Specific Class, Driver=, 480M /: Bus 04.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 5000M /: Bus 03.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 480M /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 5000M |__ Port 2: Dev 2, If 0, Class=Mass Storage, Driver=usb-storage, 5000M /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 480M ------------------- ------------------- $ lsusb Bus 002 Device 002: ID 152d:0551 JMicron Technology Corp. / JMicron USA Technology Corp. Bus 005 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 006 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 005 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 006 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 005 Device 003: ID 05e3:0608 Genesys Logic, Inc. USB-2.0 4-Port HUB Bus 005 Device 004: ID 174c:5106 ASMedia Technology Inc. Transcend StoreJet 25M3 Bus 005 Device 018: ID 0409:005a NEC Corp. HighSpeed Hub Bus 006 Device 003: ID 058f:6362 Alcor Micro Corp. Flash Card Reader/Writer Bus 005 Device 006: ID 046d:c318 Logitech, Inc. Illuminated Keyboard Bus 005 Device 007: ID 047d:1020 Kensington Expert Mouse Trackball Bus 005 Device 012: ID 058f:9254 Alcor Micro Corp. Hub Bus 005 Device 020: ID 09e8:0010 AKAI Professional M.I. Corp. ------------------- Sincerely hoping it'll help to include full support in ALSA for the AKAI EIE Pro as it's very good stuff, nicely built and convenient (for its hub) to USB ports eaters as we're all in the music phylum :-) Cheers! -- < °/_ /| ¹

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[alsa-devel] [PATCH v3 0/2] usb: Add native DSD support for Denon/Marantz
by Jurgen Kramer 28 Nov '14

28 Nov '14

This patchset adds native DSD support for Denon/Marantz devices. A specific vendor command is needed to switch between PCM and DSD mode. This is implemented as a new quirk function. Changes from v2: - switch) -> switch ( - combine PCM and DSD mode select to a single USB ctrl msg Changes from v1: - fix compare in err check - remove unneeded brackets Jurgen Kramer (2): usb: Add native DSD support for Denon/Marantz DACs usb: Add mode select quirk for Denon/Marantz DACs sound/usb/pcm.c | 5 +++++ sound/usb/quirks.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 55 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH v2 0/2] usb: add native DSD support for Denon/Marantz
by Jurgen Kramer 28 Nov '14

28 Nov '14

This patchset adds native DSD support for Denon/Marantz devices. A specific vendor command is needed to switch between PCM and DSD mode. This is implemented as a new quirk function. Changes from v1: - fix compare in err check - remove unneeded brackets Jurgen Kramer (2): usb: Add native DSD support for Denon/Marantz DACs usb: Add mode select quirk for Denon/Marantz DACs sound/usb/pcm.c | 5 +++++ sound/usb/quirks.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 61 insertions(+) -- 1.9.3

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[alsa-devel] [PATCH] ASoC: Intel: Move capture PCM pin to PCM0 for Broadwell
by Jie Yang 28 Nov '14

28 Nov '14

Move capture PCM pin from PCM4 to PCM0 for Broadwell. This will allow us to integrate with pulseaudio better for usually default device is set to 0. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/broadwell.c | 15 ++------------- sound/soc/intel/sst-haswell-pcm.c | 34 ++++++++++++++++------------------ 2 files changed, 18 insertions(+), 31 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 52cb764..c256764 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -171,7 +171,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { /* Front End DAI links */ { .name = "System PCM", - .stream_name = "System Playback", + .stream_name = "System Playback/Capture", .cpu_dai_name = "System Pin", .platform_name = "haswell-pcm-audio", .dynamic = 1, @@ -180,6 +180,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .init = broadwell_rtd_init, .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_playback = 1, + .dpcm_capture = 1, }, { .name = "Offload0", @@ -214,18 +215,6 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, .dpcm_capture = 1, }, - { - .name = "Capture PCM", - .stream_name = "Capture", - .cpu_dai_name = "Capture Pin", - .platform_name = "haswell-pcm-audio", - .dynamic = 1, - .codec_name = "snd-soc-dummy", - .codec_dai_name = "snd-soc-dummy-dai", - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - }, - /* Back End DAI links */ { /* SSP0 - Codec */ diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 5f70d81..0180b38 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -309,7 +309,7 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ - SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 0, 0, 8, ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; @@ -396,8 +396,14 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, /* DSP stream type depends on DAI ID */ switch (rtd->cpu_dai->id) { case 0: - stream_type = SST_HSW_STREAM_TYPE_SYSTEM; - module_id = SST_HSW_MODULE_PCM_SYSTEM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + } + else { + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + } break; case 1: case 2: @@ -410,10 +416,6 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, path_id = SST_HSW_STREAM_PATH_SSP0_OUT; module_id = SST_HSW_MODULE_PCM_REFERENCE; break; - case 4: - stream_type = SST_HSW_STREAM_TYPE_CAPTURE; - module_id = SST_HSW_MODULE_PCM_CAPTURE; - break; default: dev_err(rtd->dev, "error: invalid DAI ID %d\n", rtd->cpu_dai->id); @@ -779,6 +781,13 @@ static struct snd_soc_dai_driver hsw_dais[] = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, + }, }, { /* PCM */ @@ -815,17 +824,6 @@ static struct snd_soc_dai_driver hsw_dais[] = { .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, - { - .name = "Capture Pin", - .id = HSW_PCM_DAI_ID_CAPTURE, - .capture = { - .stream_name = "Analog Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, - }, - }, }; static const struct snd_soc_dapm_widget widgets[] = { -- 1.9.1

3 3

[alsa-devel] [PATCH 0/2] usb: add native DSD support for Denon/Marantz
by Jurgen Kramer 28 Nov '14

28 Nov '14

This patchset adds native DSD support for Denon/Marantz devices. A specific vendor command is needed to switch between PCM and DSD mode. This is implemented as a new quirk function. Jurgen Kramer (2): Add native DSD support for Denon/Marantz DACs Add mode select quirk for Denon/Marantz DACs sound/usb/pcm.c | 6 ++++++ sound/usb/quirks.c | 53 +++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 3 +++ 3 files changed, 62 insertions(+) -- 1.9.3

3 5

[alsa-devel] [PATCH] ASoC: Intel: Correct the xmax volume
by Jie Yang 28 Nov '14

28 Nov '14

The xmax volume should be corrected to ARRAY_SIZE(volume_map)-1, otherwise, the xmax value will be mapped to 0 wrongly. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 20561ca..5f70d81 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -298,19 +298,19 @@ static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); static const struct snd_kcontrol_new hsw_volume_controls[] = { /* Global DSP volume */ SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8, - ARRAY_SIZE(volume_map) -1, 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_volume_get, hsw_volume_put, hsw_vol_tlv), /* Offload 0 volume */ SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Offload 1 volume */ SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, - ARRAY_SIZE(volume_map), 0, + ARRAY_SIZE(volume_map) - 1, 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), }; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: mxs-sgtl5000: Remove MCLK restriction
by Fabio Estevam 27 Nov '14

27 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> According to the sgtl5000 datasheet the MCLK frequency range restriction of 8 to 27 MHz only applies when the PLL is used - synchronous SYS_MCLK input mode. mxs-sgtl5000 machine sets the codec as slave, and mx28 generates MCLK in the range of 256*fs, 384*fs or 512*fs, which is called asynchronous SYS_MCLK input. In asynchronous SYS_MCLK we cannot have the 8 to 27 MHz check because if we want to play a 8KHz sample rate track, with a MCLK of 8k * 512 = 4.096MHz the current check would return -EINVAL, which is not correct. Remove the 8 to 27MHz frequency check, since this only applies to the synchronous SYS_MCLK input case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 7515cd4..6f1916b 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -49,13 +49,6 @@ static int mxs_sgtl5000_hw_params(struct snd_pcm_substream *substream, break; } - /* Sgtl5000 sysclk should be >= 8MHz and <= 27M */ - if (mclk < 8000000 || mclk > 27000000) { - dev_err(codec_dai->dev, "Invalid mclk frequency: %u.%03uMHz\n", - mclk / 1000000, mclk / 1000 % 1000); - return -EINVAL; - } - /* Set SGTL5000's SYSCLK (provided by SAIF MCLK) */ ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk, 0); if (ret) { -- 1.9.1

2 3

[alsa-devel] [PATCH 01/11] ASoC: cq93vc: Remove unused state struct
by Lars-Peter Clausen 27 Nov '14

27 Nov '14

While two of the fields in the cq93vc driver state struct are initialized none of them are ever acutally read again. So remove the whole struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/mfd/davinci_voicecodec.h | 7 ------- sound/soc/codecs/cq93vc.c | 8 -------- 2 files changed, 15 deletions(-) diff --git a/include/linux/mfd/davinci_voicecodec.h b/include/linux/mfd/davinci_voicecodec.h index cb01496..8e1cdbe 100644 --- a/include/linux/mfd/davinci_voicecodec.h +++ b/include/linux/mfd/davinci_voicecodec.h @@ -99,12 +99,6 @@ struct davinci_vcif { dma_addr_t dma_rx_addr; }; -struct cq93vc { - struct platform_device *pdev; - struct snd_soc_codec *codec; - u32 sysclk; -}; - struct davinci_vc; struct davinci_vc { @@ -122,7 +116,6 @@ struct davinci_vc { /* Client devices */ struct davinci_vcif davinci_vcif; - struct cq93vc cq93vc; }; #endif diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 537327c..036a877 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -62,14 +62,10 @@ static int cq93vc_mute(struct snd_soc_dai *dai, int mute) static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct davinci_vc *davinci_vc = codec->dev->platform_data; - switch (freq) { case 22579200: case 27000000: case 33868800: - davinci_vc->cq93vc.sysclk = freq; return 0; } @@ -135,10 +131,6 @@ static int cq93vc_resume(struct snd_soc_codec *codec) static int cq93vc_probe(struct snd_soc_codec *codec) { - struct davinci_vc *davinci_vc = codec->dev->platform_data; - - davinci_vc->cq93vc.codec = codec; - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- 1.8.0

3 22

[alsa-devel] [PATCH] ASoC: Intel: Remove useless loopback volume control for Broadwell
by Jie Yang 27 Nov '14

27 Nov '14

On Broadwell, the ADSP FW don't support loopback record volume tuning, so here remove this control. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 331237f..20561ca 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -308,10 +308,6 @@ static const struct snd_kcontrol_new hsw_volume_controls[] = { SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, ARRAY_SIZE(volume_map), 0, hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), - /* Loopback volume */ - SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8, - ARRAY_SIZE(volume_map), 0, - hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), /* Mic Capture volume */ SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, ARRAY_SIZE(volume_map), 0, -- 1.9.1

2 1

[alsa-devel] [patch] ASoC: sigmadsp: uninitialized variable in sigmadsp_activate_ctrl()
by Dan Carpenter 27 Nov '14

27 Nov '14

The "changed" variable should be set to false at the start. Fixes: a35daac77a03 ('ASoC: sigmadsp: Add support for fw v2') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 34fdc40..d53680a 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -661,7 +661,8 @@ static void sigmadsp_activate_ctrl(struct sigmadsp *sigmadsp, struct snd_card *card = sigmadsp->component->card->snd_card; struct snd_kcontrol_volatile *vd; struct snd_ctl_elem_id id; - bool active, changed; + bool active; + bool changed = false; active = sigmadsp_samplerate_valid(ctrl->samplerates, samplerate_mask);

3 2

[alsa-devel] [PATCH] ASoC: wm8960: Move register initialisation to I2C driver probe()
by Zidan Wang 27 Nov '14

27 Nov '14

We must ensure that the clocking configuration is valid as rapidly as possible. And do software reset before the others registers updates, or the registers will be reset to the default state. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 41 ++++++++++++++++++++--------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index bc8793cd..031a1ae 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -128,7 +128,7 @@ struct wm8960_priv { struct wm8960_data pdata; }; -#define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) +#define wm8960_reset(c) regmap_write(c, WM8960_RESET, 0) /* enumerated controls */ static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", @@ -947,31 +947,12 @@ static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct wm8960_data *pdata = &wm8960->pdata; - int ret; if (pdata->capless) wm8960->set_bias_level = wm8960_set_bias_level_capless; else wm8960->set_bias_level = wm8960_set_bias_level_out3; - ret = wm8960_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - - /* Latch the update bits */ - snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); - snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); - snd_soc_add_codec_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); @@ -1030,7 +1011,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); - if (pdata && pdata->shared_lrclk) { + ret = wm8960_reset(wm8960->regmap); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + return ret; + } + + if (wm8960->pdata.shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); if (ret != 0) { @@ -1040,6 +1027,18 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, } } + /* Latch the update bits */ + regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LADC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RADC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LDAC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_RDAC, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LOUT1, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_ROUT1, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_LOUT2, 0x100, 0x100); + regmap_update_bits(wm8960->regmap, WM8960_ROUT2, 0x100, 0x100); + i2c_set_clientdata(i2c, wm8960); ret = snd_soc_register_codec(&i2c->dev, -- 1.9.1

3 2

[alsa-devel] [patch] ALSA: hda - using uninitialized data
by Dan Carpenter 27 Nov '14

27 Nov '14

In olden times the snd_hda_param_read() function always set "*start_id" but in 2007 we introduced a new return and it causes uninitialized data bugs in a couple of the callers: print_codec_info() and hdmi_parse_codec(). Fixes: e8a7f136f5ed ('[ALSA] hda-intel - Improve HD-audio codec probing robustness') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- This is from static analysis. Untested. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b2d5899..2fe86d2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -346,8 +346,10 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, unsigned int parm; parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT); - if (parm == -1) + if (parm == -1) { + *start_id = 0; return 0; + } *start_id = (parm >> 16) & 0x7fff; return (int)(parm & 0x7fff); }

2 1

[alsa-devel] [PATCH 0/9] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 27 Nov '14

27 Nov '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a machine driver that handles the board-specific pins - a native driver that handles hardware access to the LPAIF - a CPU DAI driver for controlling the LPAIF block - a PCM MI2S platform driver Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. - Ken Kenneth Westfield (9): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Add device tree binding docs ASoC: ipq806x: add native LPAIF driver ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: ipq806x: Add machine driver for IPQ806X SOC ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../bindings/sound/qcom,ipq806x-snd-card.txt | 41 ++ .../bindings/sound/qcom,lpass-cpu-dai.txt | 20 + .../devicetree/bindings/sound/qcom,lpass-lpaif.txt | 21 + .../bindings/sound/qcom,lpass-pcm-mi2s.txt | 12 + MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 33 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/qcom/Kconfig | 43 ++ sound/soc/qcom/Makefile | 11 + sound/soc/qcom/ipq806x.c | 221 ++++++++++ sound/soc/qcom/lpass-cpu-dai.c | 307 +++++++++++++ sound/soc/qcom/lpass-lpaif.c | 488 +++++++++++++++++++++ sound/soc/qcom/lpass-lpaif.h | 181 ++++++++ sound/soc/qcom/lpass-pcm-mi2s.c | 390 ++++++++++++++++ sound/soc/qcom/lpass-pcm-mi2s.h | 40 ++ 16 files changed, 1817 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,ipq806x-snd-card.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-dai.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-lpaif.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-pcm-mi2s.txt create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/ipq806x.c create mode 100644 sound/soc/qcom/lpass-cpu-dai.c create mode 100644 sound/soc/qcom/lpass-lpaif.c create mode 100644 sound/soc/qcom/lpass-lpaif.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.h -- 1.8.2.1

7 30

[alsa-devel] [PATCH v2] ALSA: korg1212: cleanup of printk
by Sudip Mukherjee 26 Nov '14

26 Nov '14

From: Sudip Mukherjee <sudip(a)vectorindia.org> replaced all references of the debug messages via printk with dev_* macro (mostly dev_dbg). one reference was changed to pr_err as there the card might have been uninitialized. this patch will generate warning from checkpatch about broken quoted strings. but that was not fixed intentionally to improve the readability. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v2: the build warnings of v1 are fixed. hi Takashi, in your review of v1, you said about some lines which are not ending with \n. but i was not able to find them. did i miss them somewhere? sound/pci/korg1212/korg1212.c | 407 +++++++++++++++++++++++------------------- 1 file changed, 220 insertions(+), 187 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 59d21c9..faebe81 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -39,21 +39,6 @@ #include <asm/io.h> // ---------------------------------------------------------------------------- -// Debug Stuff -// ---------------------------------------------------------------------------- -#define K1212_DEBUG_LEVEL 0 -#if K1212_DEBUG_LEVEL > 0 -#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK(fmt,...) -#endif -#if K1212_DEBUG_LEVEL > 1 -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) -#else -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,...) -#endif - -// ---------------------------------------------------------------------------- // Record/Play Buffer Allocation Method. If K1212_LARGEALLOC is defined all // buffers are alocated as a large piece inside KorgSharedBuffer. // ---------------------------------------------------------------------------- @@ -530,12 +515,12 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, int rc = K1212_CMDRET_Success; if (!korg1212->outDoorbellPtr) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: CardUninitialized\n"); + pr_err("K1212_DEBUG: CardUninitialized\n"); return K1212_CMDRET_CardUninitialized; } - K1212_DEBUG_PRINTK("K1212_DEBUG: Card <- 0x%08x 0x%08x [%s]\n", - doorbellVal, mailBox0Val, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Card <- 0x%08x 0x%08x [%s]\n", + doorbellVal, mailBox0Val, stateName[korg1212->cardState]); for (retryCount = 0; retryCount < MAX_COMMAND_RETRIES; retryCount++) { writel(mailBox3Val, korg1212->mailbox3Ptr); writel(mailBox2Val, korg1212->mailbox2Ptr); @@ -562,7 +547,8 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, mailBox3Lo = readl(korg1212->mailbox3Ptr); if (mailBox3Lo & COMMAND_ACK_MASK) { if ((mailBox3Lo & DOORBELL_VAL_MASK) == (doorbellVal & DOORBELL_VAL_MASK)) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- Success\n"); + dev_dbg(korg1212->card->dev, + "Card <- Success\n"); rc = K1212_CMDRET_Success; break; } @@ -571,7 +557,7 @@ static int snd_korg1212_Send1212Command(struct snd_korg1212 *korg1212, korg1212->cmdRetryCount += retryCount; if (retryCount >= MAX_COMMAND_RETRIES) { - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Card <- NoAckFromCard\n"); + dev_dbg(korg1212->card->dev, "Card <- NoAckFromCard\n"); rc = K1212_CMDRET_NoAckFromCard; } @@ -612,20 +598,21 @@ static void snd_korg1212_timer_func(unsigned long data) korg1212->stop_pending_cnt = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: Stop ack'ed [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop ack'ed [%s]\n", + stateName[korg1212->cardState]); } else { if (--korg1212->stop_pending_cnt > 0) { /* reprogram timer */ korg1212->timer.expires = jiffies + 1; add_timer(&korg1212->timer); } else { - snd_printd("korg1212_timer_func timeout\n"); + dev_dbg(korg1212->card->dev, + "korg1212_timer_func timeout\n"); korg1212->sharedBufferPtr->cardCommand = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); - K1212_DEBUG_PRINTK("K1212_DEBUG: Stop timeout [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Stop timeout [%s]\n", + stateName[korg1212->cardState]); } } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -660,8 +647,8 @@ static inline void snd_korg1212_setCardState(struct snd_korg1212 * korg1212, enu static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: OpenCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "OpenCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (korg1212->opencnt++ == 0) { snd_korg1212_TurnOffIdleMonitor(korg1212); @@ -674,8 +661,8 @@ static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard [%s] %d\n", - stateName[korg1212->cardState], korg1212->opencnt); + dev_dbg(korg1212->card->dev, "CloseCard [%s] %d\n", + stateName[korg1212->cardState], korg1212->opencnt); mutex_lock(&korg1212->open_mutex); if (--(korg1212->opencnt)) { @@ -687,8 +674,9 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_StopPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "CloseCard - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { mutex_unlock(&korg1212->open_mutex); return 0; @@ -711,8 +699,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->setcnt); + dev_dbg(korg1212->card->dev, "SetupForPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->setcnt); if (korg1212->setcnt++) return 0; @@ -721,8 +709,8 @@ static int snd_korg1212_SetupForPlay(struct snd_korg1212 * korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_SetupPlay, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: SetupForPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "SetupForPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -734,8 +722,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "K1212_DEBUG: TriggerPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (korg1212->playcnt++) return 0; @@ -743,8 +731,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) snd_korg1212_setCardState(korg1212, K1212_STATE_PLAYING); rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_TriggerPlay, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: TriggerPlay - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "TriggerPlay - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { return 1; } @@ -754,8 +742,8 @@ static int snd_korg1212_TriggerPlay(struct snd_korg1212 * korg1212) /* spinlock already held */ static int snd_korg1212_StopPlay(struct snd_korg1212 * korg1212) { - K1212_DEBUG_PRINTK("K1212_DEBUG: StopPlay [%s] %d\n", - stateName[korg1212->cardState], korg1212->playcnt); + dev_dbg(korg1212->card->dev, "StopPlay [%s] %d\n", + stateName[korg1212->cardState], korg1212->playcnt); if (--(korg1212->playcnt)) return 0; @@ -866,8 +854,9 @@ static int snd_korg1212_SetRate(struct snd_korg1212 *korg1212, int rate) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); return 0; } @@ -902,8 +891,8 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) u16 count; unsigned long flags; - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "WriteADCSensivity [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------------------------------- // initialize things. The local init bit is always set when writing to the @@ -1026,8 +1015,9 @@ static int snd_korg1212_WriteADCSensitivity(struct snd_korg1212 *korg1212) int rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_SelectPlayMode, K1212_MODE_MonitorOn, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: WriteADCSensivity - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "WriteADCSensivity - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); } spin_unlock_irqrestore(&korg1212->lock, flags); @@ -1039,8 +1029,8 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) { int channel, rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is complete. [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is complete. [%s]\n", + stateName[korg1212->cardState]); // ---------------------------------------------------- // tell the card to boot @@ -1048,8 +1038,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_BootFromDSPPage4, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Boot from Page 4 - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Boot from Page 4 - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); msleep(DSP_BOOT_DELAY_IN_MS); // -------------------------------------------------------------------------------- @@ -1065,8 +1056,9 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Buffer Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Configure Buffer Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); udelay(INTERCOMMAND_DELAY); @@ -1079,8 +1071,8 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Configure Misc Memory - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Configure Misc Memory - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); // -------------------------------------------------------------------------------- // Initialize the routing and volume tables, then update the card's state. @@ -1100,15 +1092,16 @@ static void snd_korg1212_OnDSPDownloadComplete(struct snd_korg1212 *korg1212) ClockSourceSelector[korg1212->clkSrcRate], 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Clock Source Selector - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Set Clock Source Selector - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); rc = snd_korg1212_TurnOnIdleMonitor(korg1212); snd_korg1212_setCardState(korg1212, K1212_STATE_READY); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Set Monitor On - RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Set Monitor On - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); snd_korg1212_setCardState(korg1212, K1212_STATE_DSP_COMPLETE); } @@ -1133,9 +1126,10 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) switch (doorbellValue) { case K1212_DB_DSPDownloadDone: - K1212_DEBUG_PRINTK("K1212_DEBUG: IRQ DNLD count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DNLD count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); if (korg1212->cardState == K1212_STATE_DSP_IN_PROCESS) { korg1212->dsp_is_loaded = 1; wake_up(&korg1212->wait); @@ -1146,10 +1140,11 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // an error occurred - stop the card // ------------------------------------------------------------------------ case K1212_DB_DMAERROR: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); - snd_printk(KERN_ERR "korg1212: DMA Error\n"); + dev_dbg(korg1212->card->dev, + "IRQ DMAE count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); + dev_err(korg1212->card->dev, "DMA Error\n"); korg1212->errorcnt++; korg1212->totalerrorcnt++; korg1212->sharedBufferPtr->cardCommand = 0; @@ -1161,16 +1156,19 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) // the semaphore in case someone is waiting for this. // ------------------------------------------------------------------------ case K1212_DB_CARDSTOPPED: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n", - korg1212->irqcount, doorbellValue, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ CSTP count - %ld, %x, [%s].\n", + korg1212->irqcount, doorbellValue, + stateName[korg1212->cardState]); korg1212->sharedBufferPtr->cardCommand = 0; break; default: - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", - korg1212->irqcount, doorbellValue, - korg1212->currentBuffer, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "IRQ DFLT count - %ld, %x, cpos=%d [%s].\n", + korg1212->irqcount, doorbellValue, + korg1212->currentBuffer, + stateName[korg1212->cardState]); if ((korg1212->cardState > K1212_STATE_SETUP) || korg1212->idleMonitorOn) { korg1212->currentBuffer++; @@ -1206,8 +1204,8 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) { int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP download is starting... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "DSP download is starting... [%s]\n", + stateName[korg1212->cardState]); // --------------------------------------------------------------- // verify the state of the card before proceeding. @@ -1221,8 +1219,9 @@ static int snd_korg1212_downloadDSPCode(struct snd_korg1212 *korg1212) UpperWordSwap(korg1212->dma_dsp.addr), 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Start DSP Download RC = %d [%s]\n", - rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "Start DSP Download RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); korg1212->dsp_is_loaded = 0; wait_event_timeout(korg1212->wait, korg1212->dsp_is_loaded, HZ * CARD_BOOT_TIMEOUT); @@ -1281,20 +1280,20 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_silence pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", - dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", + dst, i); return -EFAULT; } -#endif memset((void*) dst + offset, 0, size); dst++; } @@ -1307,22 +1306,25 @@ static int snd_korg1212_copy_to(struct snd_korg1212 *korg1212, void __user *dst, struct KorgAudioFrame * src = korg1212->recordDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_to pos=%d offset=%d size=%d\n", - pos, offset, size); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to pos=%d offset=%d size=%d\n", + pos, offset, size); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_to_user(dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_to USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } src++; @@ -1337,23 +1339,26 @@ static int snd_korg1212_copy_from(struct snd_korg1212 *korg1212, void __user *sr struct KorgAudioFrame * dst = korg1212->playDataBufsPtr[0].bufferData + pos; int i, rc; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", - pos, offset, size, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from pos=%d offset=%d size=%d count=%d\n", + pos, offset, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; for (i=0; i < count; i++) { -#if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } -#endif rc = copy_from_user((void*) dst + offset, src, size); if (rc) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", src, dst, i); + dev_dbg(korg1212->card->dev, + "snd_korg1212_copy_from USER EFAULT src=%p dst=%p iter=%d\n", + src, dst, i); return -EFAULT; } dst++; @@ -1367,8 +1372,8 @@ static void snd_korg1212_free_pcm(struct snd_pcm *pcm) { struct snd_korg1212 *korg1212 = pcm->private_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_free_pcm [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_free_pcm [%s]\n", + stateName[korg1212->cardState]); korg1212->pcm = NULL; } @@ -1379,8 +1384,8 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1408,8 +1413,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_open [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_open [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_OpenCard(korg1212); @@ -1435,8 +1440,8 @@ static int snd_korg1212_playback_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_playback_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_close [%s]\n", + stateName[korg1212->cardState]); snd_korg1212_silence(korg1212, 0, K1212_MAX_SAMPLES, 0, korg1212->channels * 2); @@ -1457,8 +1462,8 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) unsigned long flags; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_capture_close [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_close [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1475,14 +1480,18 @@ static int snd_korg1212_capture_close(struct snd_pcm_substream *substream) static int snd_korg1212_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_ioctl: cmd=%d\n", cmd); + struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); + + dev_dbg(korg1212->card->dev, "snd_korg1212_ioctl: cmd=%d\n", cmd); if (cmd == SNDRV_PCM_IOCTL1_CHANNEL_INFO ) { struct snd_pcm_channel_info *info = arg; info->offset = 0; info->first = info->channel * 16; info->step = 256; - K1212_DEBUG_PRINTK("K1212_DEBUG: channel_info %d:, offset=%ld, first=%d, step=%d\n", info->channel, info->offset, info->first, info->step); + dev_dbg(korg1212->card->dev, + "channel_info %d:, offset=%ld, first=%d, step=%d\n", + info->channel, info->offset, info->first, info->step); return 0; } @@ -1498,8 +1507,8 @@ static int snd_korg1212_hw_params(struct snd_pcm_substream *substream, pid_t this_pid; pid_t other_pid; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_hw_params [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_hw_params [%s]\n", + stateName[korg1212->cardState]); spin_lock_irqsave(&korg1212->lock, flags); @@ -1546,15 +1555,16 @@ static int snd_korg1212_prepare(struct snd_pcm_substream *substream) struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_prepare [%s]\n", + stateName[korg1212->cardState]); spin_lock_irq(&korg1212->lock); /* FIXME: we should wait for ack! */ if (korg1212->stop_pending_cnt > 0) { - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_prepare - Stop is pending... [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "snd_korg1212_prepare - Stop is pending... [%s]\n", + stateName[korg1212->cardState]); spin_unlock_irq(&korg1212->lock); return -EAGAIN; /* @@ -1579,8 +1589,8 @@ static int snd_korg1212_trigger(struct snd_pcm_substream *substream, struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); int rc; - K1212_DEBUG_PRINTK("K1212_DEBUG: snd_korg1212_trigger [%s] cmd=%d\n", - stateName[korg1212->cardState], cmd); + dev_dbg(korg1212->card->dev, "snd_korg1212_trigger [%s] cmd=%d\n", + stateName[korg1212->cardState], cmd); spin_lock(&korg1212->lock); switch (cmd) { @@ -1621,8 +1631,9 @@ static snd_pcm_uframes_t snd_korg1212_playback_pointer(struct snd_pcm_substream pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1634,8 +1645,8 @@ static snd_pcm_uframes_t snd_korg1212_capture_pointer(struct snd_pcm_substream * pos = korg1212->currentBuffer * kPlayBufferFrames; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_pointer [%s] %ld\n", - stateName[korg1212->cardState], pos); + dev_dbg(korg1212->card->dev, "snd_korg1212_capture_pointer [%s] %ld\n", + stateName[korg1212->cardState], pos); return pos; } @@ -1648,8 +1659,9 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_playback_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_from(korg1212, src, pos, count, 0, korg1212->channels * 2); @@ -1662,8 +1674,8 @@ static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_playback_silence [%s]\n", - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "snd_korg1212_playback_silence [%s]\n", + stateName[korg1212->cardState]); return snd_korg1212_silence(korg1212, pos, count, 0, korg1212->channels * 2); } @@ -1676,8 +1688,9 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream, { struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_capture_copy [%s] %ld %ld\n", - stateName[korg1212->cardState], pos, count); + dev_dbg(korg1212->card->dev, + "snd_korg1212_capture_copy [%s] %ld %ld\n", + stateName[korg1212->cardState], pos, count); return snd_korg1212_copy_to(korg1212, dst, pos, count, 0, korg1212->channels * 2); } @@ -2138,7 +2151,7 @@ snd_korg1212_free(struct snd_korg1212 *korg1212) static int snd_korg1212_dev_free(struct snd_device *device) { struct snd_korg1212 *korg1212 = device->device_data; - K1212_DEBUG_PRINTK("K1212_DEBUG: Freeing device\n"); + dev_dbg(korg1212->card->dev, "Freeing device\n"); return snd_korg1212_free(korg1212); } @@ -2210,19 +2223,17 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, ioport_size = pci_resource_len(korg1212->pci, 1); iomem2_size = pci_resource_len(korg1212->pci, 2); - K1212_DEBUG_PRINTK("K1212_DEBUG: resources:\n" - " iomem = 0x%lx (%d)\n" - " ioport = 0x%lx (%d)\n" - " iomem = 0x%lx (%d)\n" - " [%s]\n", - korg1212->iomem, iomem_size, - korg1212->ioport, ioport_size, - korg1212->iomem2, iomem2_size, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "resources:\n\tiomem = 0x%lx (%d)\n\tioport = 0x%lx (%d)\n\tiomem = 0x%lx (%d)\n\t[%s]\n", + korg1212->iomem, iomem_size, + korg1212->ioport, ioport_size, + korg1212->iomem2, iomem2_size, + stateName[korg1212->cardState]); if ((korg1212->iobase = ioremap(korg1212->iomem, iomem_size)) == NULL) { - snd_printk(KERN_ERR "korg1212: unable to remap memory region 0x%lx-0x%lx\n", korg1212->iomem, - korg1212->iomem + iomem_size - 1); + dev_err(korg1212->card->dev, + "unable to remap memory region 0x%lx-0x%lx\n", + korg1212->iomem, korg1212->iomem + iomem_size - 1); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2232,7 +2243,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, KBUILD_MODNAME, korg1212); if (err) { - snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq); + dev_err(korg1212->card->dev, "unable to grab IRQ %d\n", + pci->irq); snd_korg1212_free(korg1212); return -EBUSY; } @@ -2252,40 +2264,45 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->sensRegPtr = (u16 __iomem *) (korg1212->iobase + SENS_CONTROL_OFFSET); korg1212->idRegPtr = (u32 __iomem *) (korg1212->iobase + DEV_VEND_ID_OFFSET); - K1212_DEBUG_PRINTK("K1212_DEBUG: card registers:\n" - " Status register = 0x%p\n" - " OutDoorbell = 0x%p\n" - " InDoorbell = 0x%p\n" - " Mailbox0 = 0x%p\n" - " Mailbox1 = 0x%p\n" - " Mailbox2 = 0x%p\n" - " Mailbox3 = 0x%p\n" - " ControlReg = 0x%p\n" - " SensReg = 0x%p\n" - " IDReg = 0x%p\n" - " [%s]\n", - korg1212->statusRegPtr, - korg1212->outDoorbellPtr, - korg1212->inDoorbellPtr, - korg1212->mailbox0Ptr, - korg1212->mailbox1Ptr, - korg1212->mailbox2Ptr, - korg1212->mailbox3Ptr, - korg1212->controlRegPtr, - korg1212->sensRegPtr, - korg1212->idRegPtr, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "card registers:\n" + " Status register = 0x%p\n" + " OutDoorbell = 0x%p\n" + " InDoorbell = 0x%p\n" + " Mailbox0 = 0x%p\n" + " Mailbox1 = 0x%p\n" + " Mailbox2 = 0x%p\n" + " Mailbox3 = 0x%p\n" + " ControlReg = 0x%p\n" + " SensReg = 0x%p\n" + " IDReg = 0x%p\n" + " [%s]\n", + korg1212->statusRegPtr, + korg1212->outDoorbellPtr, + korg1212->inDoorbellPtr, + korg1212->mailbox0Ptr, + korg1212->mailbox1Ptr, + korg1212->mailbox2Ptr, + korg1212->mailbox3Ptr, + korg1212->controlRegPtr, + korg1212->sensRegPtr, + korg1212->idRegPtr, + stateName[korg1212->cardState]); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%Zd bytes)\n", sizeof(struct KorgSharedBuffer)); + dev_err(korg1212->card->dev, + "can not allocate shared buffer memory (%Zd bytes)\n", + sizeof(struct KorgSharedBuffer)); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->sharedBufferPtr = (struct KorgSharedBuffer *)korg1212->dma_shared.area; korg1212->sharedBufferPhy = korg1212->dma_shared.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Shared Buffer Area = 0x%p (0x%08lx), %d bytes\n", korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, sizeof(struct KorgSharedBuffer)); + dev_dbg(korg1212->card->dev, + "Shared Buffer Area = 0x%p (0x%08lx), %lu bytes\n", + korg1212->sharedBufferPtr, korg1212->sharedBufferPhy, + sizeof(struct KorgSharedBuffer)); #ifndef K1212_LARGEALLOC @@ -2293,27 +2310,35 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_play) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate play data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(korg1212->card->dev, + "can not allocate play data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->playDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_play.area; korg1212->PlayDataPhy = korg1212->dma_play.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Play Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->playDataBufsPtr, korg1212->PlayDataPhy, korg1212->DataBufsSize); + dev_dbg(korg1212->card->dev, + "Play Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->playDataBufsPtr, korg1212->PlayDataPhy, + korg1212->DataBufsSize); if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), korg1212->DataBufsSize, &korg1212->dma_rec) < 0) { - snd_printk(KERN_ERR "korg1212: can not allocate record data buffer memory (%d bytes)\n", korg1212->DataBufsSize); + dev_err(korg1212->card->dev, + "can not allocate record data buffer memory (%d bytes)\n", + korg1212->DataBufsSize); snd_korg1212_free(korg1212); return -ENOMEM; } korg1212->recordDataBufsPtr = (struct KorgAudioBuffer *)korg1212->dma_rec.area; korg1212->RecDataPhy = korg1212->dma_rec.addr; - K1212_DEBUG_PRINTK("K1212_DEBUG: Record Data Area = 0x%p (0x%08x), %d bytes\n", - korg1212->recordDataBufsPtr, korg1212->RecDataPhy, korg1212->DataBufsSize); + dev_dbg(korg1212->card->dev, + "Record Data Area = 0x%p (0x%08x), %d bytes\n", + korg1212->recordDataBufsPtr, korg1212->RecDataPhy, + korg1212->DataBufsSize); #else // K1212_LARGEALLOC @@ -2334,22 +2359,25 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { release_firmware(dsp_code); - snd_printk(KERN_ERR "firmware not available\n"); + dev_err(korg1212->card->dev, "firmware not available\n"); snd_korg1212_free(korg1212); return err; } if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), dsp_code->size, &korg1212->dma_dsp) < 0) { - snd_printk(KERN_ERR "korg1212: cannot allocate dsp code memory (%zd bytes)\n", dsp_code->size); + dev_err(korg1212->card->dev, + "cannot allocate dsp code memory (%zd bytes)\n", + dsp_code->size); snd_korg1212_free(korg1212); release_firmware(dsp_code); return -ENOMEM; } - K1212_DEBUG_PRINTK("K1212_DEBUG: DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", - korg1212->dma_dsp.area, korg1212->dma_dsp.addr, dsp_code->size, - stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, + "DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", + korg1212->dma_dsp.area, (int)korg1212->dma_dsp.addr, + (int)dsp_code->size, stateName[korg1212->cardState]); memcpy(korg1212->dma_dsp.area, dsp_code->data, dsp_code->size); @@ -2358,7 +2386,8 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0); if (rc) - K1212_DEBUG_PRINTK("K1212_DEBUG: Reboot Card - RC = %d [%s]\n", rc, stateName[korg1212->cardState]); + dev_dbg(korg1212->card->dev, "Reboot Card - RC = %d [%s]\n", + rc, stateName[korg1212->cardState]); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, korg1212, &ops)) < 0) { snd_korg1212_free(korg1212); @@ -2372,18 +2401,22 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, if (snd_korg1212_downloadDSPCode(korg1212)) return -EBUSY; - K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " - "PlayDataPhy = %08x L[%08x]\n" - "korg1212: RecDataPhy = %08x L[%08x], " - "VolumeTablePhy = %08x L[%08x]\n" - "korg1212: RoutingTablePhy = %08x L[%08x], " - "AdatTimeCodePhy = %08x L[%08x]\n", - (int)korg1212->dma_dsp.addr, UpperWordSwap(korg1212->dma_dsp.addr), - korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy), - korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), - korg1212->VolumeTablePhy, LowerWordSwap(korg1212->VolumeTablePhy), - korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), - korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); + dev_dbg(korg1212->card->dev, "dspMemPhy = %08x U[%08x], " + "PlayDataPhy = %08x L[%08x]\n" + "korg1212: RecDataPhy = %08x L[%08x], " + "VolumeTablePhy = %08x L[%08x]\n" + "korg1212: RoutingTablePhy = %08x L[%08x], " + "AdatTimeCodePhy = %08x L[%08x]\n", + (int)korg1212->dma_dsp.addr, + UpperWordSwap(korg1212->dma_dsp.addr), + korg1212->PlayDataPhy, LowerWordSwap(korg1212->PlayDataPhy), + korg1212->RecDataPhy, LowerWordSwap(korg1212->RecDataPhy), + korg1212->VolumeTablePhy, + LowerWordSwap(korg1212->VolumeTablePhy), + korg1212->RoutingTablePhy, + LowerWordSwap(korg1212->RoutingTablePhy), + korg1212->AdatTimeCodePhy, + LowerWordSwap(korg1212->AdatTimeCodePhy)); if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) return err; @@ -2446,7 +2479,7 @@ snd_korg1212_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %d", card->shortname, korg1212->iomem, korg1212->irq); - K1212_DEBUG_PRINTK("K1212_DEBUG: %s\n", card->longname); + dev_dbg(&pci->dev, "%s\n", card->longname); if ((err = snd_card_register(card)) < 0) { snd_card_free(card); -- 1.8.1.2

3 6

[alsa-devel] [PATCH] ASoC: rt5631: Fixing compilation warning when DT is disabled
by Krishna Mohan Dani 26 Nov '14

26 Nov '14

Fixes the following compilation warning: Warning: 'rt5631_i2c_dt_ids' defined but not used - when DT is not used. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- sound/soc/codecs/rt5631.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3b7d5e4..5c65915 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,7 +1691,7 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); -static struct of_device_id rt5631_i2c_dt_ids[] = { +static struct of_device_id rt5631_i2c_dt_ids[] __maybe_unused = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } -- 1.7.9.5

3 3

[alsa-devel] [PATCH] ASoC: Samsung: Add arndale_rt5631 machine driver and binding
by Krishna Mohan Dani 26 Nov '14

26 Nov '14

Adding machine driver to instantiate I2S based realtek's ALC5631 sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec hence support for ALC5631 based machine driver is being added. This patch also documents the device tree binding for the Arndale board based machine driver. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- .../devicetree/bindings/sound/arndale.txt | 24 ++++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 150 ++++++++++++++++++++ 4 files changed, 182 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt new file mode 100644 index 0000000..0e76946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 55a3869..80b5c61 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -239,3 +239,9 @@ config SND_SOC_ODROIDX2 select SND_SAMSUNG_I2S help Say Y here to enable audio support for the Odroid-X2/U3. + +config SND_SOC_ARNDALE_RT5631_ALC5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on SND_SOC_SAMSUNG + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 91505dd..31e3dba 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -45,6 +45,7 @@ snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o snd-soc-bells-objs := bells.o snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o +snd-soc-arndale-rt5631-objs := arndale_rt5631.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o +obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c new file mode 100644 index 0000000..1e2b61c --- /dev/null +++ b/sound/soc/samsung/arndale_rt5631.c @@ -0,0 +1,150 @@ +/* + * arndale_rt5631.c + * + * Copyright (c) 2014, Insignal Co., Ltd. + * + * Author: Claude <claude(a)insginal.co.kr> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "i2s.h" + +static int arndale_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int rfs, ret; + unsigned long rclk; + + rfs = 256; + + rclk = params_rate(params) * rfs; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, + 0, SND_SOC_CLOCK_OUT); + + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops arndale_ops = { + .hw_params = arndale_hw_params, +}; + +static struct snd_soc_dai_link arndale_rt5631_dai[] = { + { + .name = "RT5631 HiFi", + .stream_name = "Primary", + .codec_dai_name = "rt5631-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ops = &arndale_ops, + }, +}; + +static struct snd_soc_card arndale_rt5631 = { + .name = "Arndale RT5631", + .dai_link = arndale_rt5631_dai, + .num_links = ARRAY_SIZE(arndale_rt5631_dai), +}; + +static int arndale_audio_probe(struct platform_device *pdev) +{ + int n, ret; + struct device_node *np = pdev->dev.of_node; + struct snd_soc_card *card = &arndale_rt5631; + + card->dev = &pdev->dev; + + for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { + if (!arndale_rt5631_dai[n].cpu_dai_name) { + arndale_rt5631_dai[n].cpu_of_node = of_parse_phandle(np, + "samsung,audio-cpu", n); + + if (!arndale_rt5631_dai[n].cpu_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } + } + if (!arndale_rt5631_dai[n].platform_name) + arndale_rt5631_dai[n].platform_of_node = + arndale_rt5631_dai[n].cpu_of_node; + + arndale_rt5631_dai[n].codec_name = NULL; + arndale_rt5631_dai[n].codec_of_node = of_parse_phandle(np, + "samsung,audio-codec", n); + if (!arndale_rt5631_dai[0].codec_of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-codec' missing or invalid\n"); + return -EINVAL; + } + } + + ret = devm_snd_soc_register_card(card->dev, card); + + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); + + return ret; +} + +static int arndale_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { + { .compatible = "samsung,arndale-rt5631", }, + { .compatible = "samsung,arndale-alc5631", }, + {}, +}; +MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match); + +static struct platform_driver arndale_audio_driver = { + .driver = { + .name = "arndale-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), + }, + .probe = arndale_audio_probe, + .remove = arndale_audio_remove, +}; + +module_platform_driver(arndale_audio_driver); + +MODULE_AUTHOR("Claude <claude(a)insignal.co.kr>"); +MODULE_DESCRIPTION("ALSA SoC Driver for Arndale Board"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: bt-sco: support wide band speech
by Barry Song 26 Nov '14

26 Nov '14

From: Yibo Cai <Yibo.Cai(a)csr.com> HFP1.6 requires support for wide band speech(16KHz) over Bluetooth. Signed-off-by: Yibo Cai <Yibo.Cai(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index c4cf069..3878491 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -52,6 +52,14 @@ static struct snd_soc_codec_driver soc_codec_dev_bt_sco = { static int bt_sco_probe(struct platform_device *pdev) { + const struct platform_device_id *id = platform_get_device_id(pdev); + + /* Check wide band SCO(16KHz) support */ + if (id && (strcmp(id->name, "bt-sco-wb") == 0)) { + bt_sco_dai.playback.rates |= SNDRV_PCM_RATE_16000; + bt_sco_dai.capture.rates |= SNDRV_PCM_RATE_16000; + } + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, &bt_sco_dai, 1); } @@ -70,6 +78,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "bt-sco", }, + { + .name = "bt-sco-wb", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); -- 2.1.3

3 2

[alsa-devel] [PATCH] ASoC: wm8960: LRCM operation should after software reset
by Zidan Wang 26 Nov '14

26 Nov '14

When shared_lrclk is set, LRCM will be enabled. But the following software reset in wm8960_probe will reset it to the default state. So LRCM operation should after software reset. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index bc8793cd..21db90c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -960,6 +960,14 @@ static int wm8960_probe(struct snd_soc_codec *codec) return ret; } + if (pdata->shared_lrclk) { + ret = snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable LRCM: %d\n", ret); + return ret; + } + } + /* Latch the update bits */ snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); @@ -1030,15 +1038,6 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, else if (i2c->dev.of_node) wm8960_set_pdata_from_of(i2c, &wm8960->pdata); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } i2c_set_clientdata(i2c, wm8960); -- 1.9.1

3 2

[alsa-devel] [PATCH V2 0/6] Add support for the WM8280 and WM8281 codecs
by Richard Fitzgerald 26 Nov '14

26 Nov '14

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Only change since V1 series is update to device tree binding documentation in patch 2 Richard Fitzgerald (6): mfd: arizona: add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 15 +++++++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 14 ++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 37 insertions(+), 8 deletions(-) -- 1.7.2.5

2 7

[alsa-devel] Where to find Alsa experienced programmer?
by Andrej Falout 26 Nov '14

26 Nov '14

Hello, I have a client that would like to hire services of an skilled programmer with Alsa related experience, for a particular project contract task. They advertised locally, but came up with nothing, which is probably not a big surprise. They have accepted remote/online working as an option. Could anyone suggest the appropriate method/place to seek such a person? Any suggestions gratefully received. Kind regards, Andrej Falout

2 1

Re: [alsa-devel] [PATCH 26/29] ALSA: oxfw: Add support AMDTP in-stream
by Clemens Ladisch 26 Nov '14

26 Nov '14

Takashi Sakamoto wrote: > On Nov 24 2014 22:54, Clemens Ladisch wrote: >> In theory, it would be possible to use atomic_t to synchronize multiple >> streams, but this requires that >> 1. two CPUs that start a stream at the same time do not both think they >> are the first, ... and >> 2. when one CPU is still starting a stream, code on all other CPUs must >> be able to work correctly, without synchronization. [...] > > If I rewrite your pseudo code, like this: > >> void start_some_substream(...) >> { >> if (atomic_inc_and_test(substreams)) { >> mutex_lock(); >> // lots of stuff to start DMA ... > > if (wait_for_first_packet() == ETIMEDOUT) { > stop_dma(); > release_streams(); > mutex_unlock(); > return ETIMEDOUT; > } >> mutex_unlock(); >> } >> ... >> } This is an even better example of why atomic_t is incorrect: when the startup fails, the second call to start_some_substream() at the same time, which just sees that substreams>0, cannot notice the error, and thinks its own substream has been started successfully. Regards, Clemens

2 1

Re: [alsa-devel] [PATCH v3] ASoC: Remove 'const' from the device_node pointers
by Mark Brown 26 Nov '14

26 Nov '14

On Tue, Nov 25, 2014 at 12:14:48PM +0100, Jean-Francois Moine wrote: > As Russell King's explained it, there should not be pointers to > struct device_node: OK, found a user that needs this so applying now. Please do try to make your changelogs clearer about why things are being done, it's often hard to figure things out which slows everything down.

1 0

[alsa-devel] Remove my address from alsa-devel@alsa-project,org
by John Houston 25 Nov '14

25 Nov '14

jbhdbr(a)gmail.com Sent from my iPhone

1 0

[alsa-devel] [PATCH] ASoC: Disable regmap helpers if regmap is disabled
by Lars-Peter Clausen 25 Nov '14

25 Nov '14

If regmap is disabled there will be no users of the ASoC regmap helpers. Furthermore regmap_exit() will no be defined causing the following compile error: sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap': sound/soc/soc-core.c:2645:2: error: implicit declaration of function 'regmap_exit' [-Werror=implicit-function-declaration] So disable the helpers if regmap is disabled. Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Fixes: 20feb881988c ASoC: Add helper functions for deferred regmap setup") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 4 ++++ sound/soc/soc-core.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 879e2b3..4cef90cc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1283,6 +1283,8 @@ void snd_soc_component_async_complete(struct snd_soc_component *component); int snd_soc_component_test_bits(struct snd_soc_component *component, unsigned int reg, unsigned int mask, unsigned int value); +#ifdef CONFIG_REGMAP + void snd_soc_component_init_regmap(struct snd_soc_component *component, struct regmap *regmap); void snd_soc_component_exit_regmap(struct snd_soc_component *component); @@ -1318,6 +1320,8 @@ static inline void snd_soc_codec_exit_regmap(struct snd_soc_codec *codec) snd_soc_component_exit_regmap(&codec->component); } +#endif + /* device driver data */ static inline void snd_soc_card_set_drvdata(struct snd_soc_card *card, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cc52ea1..7904eda 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2612,6 +2612,8 @@ static void snd_soc_component_setup_regmap(struct snd_soc_component *component) component->val_bytes = val_bytes; } +#ifdef CONFIG_REGMAP + /** * snd_soc_component_init_regmap() - Initialize regmap instance for the component * @component: The component for which to initialize the regmap instance @@ -2647,6 +2649,8 @@ void snd_soc_component_exit_regmap(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(snd_soc_component_exit_regmap); +#endif + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { if (!component->write && !component->read) { -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: qi_lb60: Pass flags to gpiod_get()
by Lars-Peter Clausen 25 Nov '14

25 Nov '14

Pass flags to gpiod_get() to automatically configure the GPIOs. This is shorter and not passing any flags to gpiod_get() will eventually no longer be supported. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/qi_lb60.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 5cb91f9..0fb7d2a 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -77,25 +77,18 @@ static int qi_lb60_probe(struct platform_device *pdev) { struct qi_lb60 *qi_lb60; struct snd_soc_card *card = &qi_lb60_card; - int ret; qi_lb60 = devm_kzalloc(&pdev->dev, sizeof(*qi_lb60), GFP_KERNEL); if (!qi_lb60) return -ENOMEM; - qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd"); + qi_lb60->snd_gpio = devm_gpiod_get(&pdev->dev, "snd", GPIOD_OUT_LOW); if (IS_ERR(qi_lb60->snd_gpio)) return PTR_ERR(qi_lb60->snd_gpio); - ret = gpiod_direction_output(qi_lb60->snd_gpio, 0); - if (ret) - return ret; - qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp"); + qi_lb60->amp_gpio = devm_gpiod_get(&pdev->dev, "amp", GPIOD_OUT_LOW); if (IS_ERR(qi_lb60->amp_gpio)) return PTR_ERR(qi_lb60->amp_gpio); - ret = gpiod_direction_output(qi_lb60->amp_gpio, 0); - if (ret) - return ret; card->dev = &pdev->dev; -- 1.7.10.4

2 1

[alsa-devel] [PATCH] alsa-lib: Fix assorted tstamp_type bugs/omissions.
by Tim Cussins 25 Nov '14

25 Nov '14

Just a couple of quick fixes related to tstamp_type. - snd_pcm_sw_params(): copy tstamp_type field - snd_pcm_tstamp_type_name(): fix argument type - snd_pcm_dump_sw_setup(): fix dumping of tstamp_type field --- src/pcm/pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 3904256..baa47c7 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -897,6 +897,7 @@ int snd_pcm_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) if (err < 0) return err; pcm->tstamp_mode = params->tstamp_mode; + pcm->tstamp_type = params->tstamp_type; pcm->period_step = params->period_step; pcm->avail_min = params->avail_min; pcm->period_event = sw_get_period_event(params); @@ -1843,7 +1844,7 @@ const char *snd_pcm_tstamp_mode_name(snd_pcm_tstamp_t mode) * \param mode PCM tstamp type * \return ascii name of PCM tstamp type setting */ -const char *snd_pcm_tstamp_type_name(snd_pcm_tstamp_t type) +const char *snd_pcm_tstamp_type_name(snd_pcm_tstamp_type_t type) { if (type > SND_PCM_TSTAMP_TYPE_LAST) return NULL; @@ -1924,7 +1925,7 @@ int snd_pcm_dump_sw_setup(snd_pcm_t *pcm, snd_output_t *out) return -EIO; } snd_output_printf(out, " tstamp_mode : %s\n", snd_pcm_tstamp_mode_name(pcm->tstamp_mode)); - snd_output_printf(out, " tstamp_type : %s\n", snd_pcm_tstamp_type_name(pcm->tstamp_mode)); + snd_output_printf(out, " tstamp_type : %s\n", snd_pcm_tstamp_type_name(pcm->tstamp_type)); snd_output_printf(out, " period_step : %d\n", pcm->period_step); snd_output_printf(out, " avail_min : %ld\n", pcm->avail_min); snd_output_printf(out, " period_event : %i\n", pcm->period_event); -- 1.7.10.4

2 1

[alsa-devel] Cache issue on BayTrail (ValleyView2) HD-audio
by Takashi Iwai 25 Nov '14

25 Nov '14

Hi Mengdong, right now I'm playing onboard HD-audio on a Baytrail (ValleyView2) machine, not DP/HDMI output, and I noticed that the snooping doesn't seem working with this. The playback sometimes stutters / gets noises. This is a typical behavior with caching mismatch and indeed it works when I disable snooping via snoop=0 option. Could you check whether this can be seen generically? Also, can anyone check whether snoop=0 for Baytrail HD-audio causes any issues? In general, snoop=0 isn't too bad (IIRC, Windows does it as default), so we can turn it off at least for Baytrail as default. thanks, Takashi

1 1

Re: [alsa-devel] [PATCH v3] ASoC: Remove 'const' from the device_node pointers
by Mark Brown 25 Nov '14

25 Nov '14

On Tue, Nov 25, 2014 at 12:14:48PM +0100, Jean-Francois Moine wrote: > As Russell King's explained it, there should not be pointers to > struct device_node: > > "struct device_node is a ref-counted structure. That means if you > store a reference to it, you should "get" it, and you should "put" > it once you've done. The act of "put"ing the pointed-to structure > involves writing to that structure, so it is totally unappropriate > to store a device_node structure as a const pointer. It forces you > to have to cast it back to a non-const pointer at various points > in time to use various OF function calls." So, we're not holding references here (we're just doing comparisons, the references need to be owned before we get into the core) and I'm not seeing anything here removing casts?

2 2

[alsa-devel] [PATCH] ASoC: Intel: Fix stream volume set no effect issue on Broadwell
by Jie Yang 25 Nov '14

25 Nov '14

The volume setting control for capture stream doesn't take effect on intel Broadwell platform. Root cause it at 2 points: 1. set stream volume with channel=2 is wrongly bapassed; 2. the saved stream volume should be restored after stream is commit. Here correct these 2 items to fix the stream volume set issue. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 35 ++++++++++++++++++++++++++--------- sound/soc/intel/sst-haswell-ipc.h | 1 + sound/soc/intel/sst-haswell-pcm.c | 21 ++++++++++++--------- 3 files changed, 39 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index ffd5728..3f8c482 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1042,14 +1042,9 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, trace_ipc_request("set stream volume", stream->reply.stream_hw_id); - if (channel > 1) + if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL) return -EINVAL; - if (stream->mute[channel]) { - stream->mute_volume[channel] = volume; - return 0; - } - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); @@ -1057,9 +1052,28 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, header |= (stage_id << IPC_STG_ID_SHIFT); req = &stream->vol_req; - req->channel = channel; req->target_volume = volume; + /* set both at same time ? */ + if (channel == SST_HSW_CHANNELS_ALL) { + if (hsw->mute[0] && hsw->mute[1]) { + hsw->mute_volume[0] = hsw->mute_volume[1] = volume; + return 0; + } else if (hsw->mute[0]) + req->channel = 1; + else if (hsw->mute[1]) + req->channel = 0; + else + req->channel = SST_HSW_CHANNELS_ALL; + } else { + /* set only 1 channel */ + if (hsw->mute[channel]) { + hsw->mute_volume[channel] = volume; + return 0; + } + req->channel = channel; + } + ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0); if (ret < 0) { dev_err(hsw->dev, "error: set stream volume failed\n"); @@ -1138,8 +1152,11 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, trace_ipc_request("set mixer volume", volume); + if (channel >= 2 && channel != SST_HSW_CHANNELS_ALL) + return -EINVAL; + /* set both at same time ? */ - if (channel == 2) { + if (channel == SST_HSW_CHANNELS_ALL) { if (hsw->mute[0] && hsw->mute[1]) { hsw->mute_volume[0] = hsw->mute_volume[1] = volume; return 0; @@ -1148,7 +1165,7 @@ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, else if (hsw->mute[1]) req.channel = 0; else - req.channel = 0xffffffff; + req.channel = SST_HSW_CHANNELS_ALL; } else { /* set only 1 channel */ if (hsw->mute[channel]) { diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 387511f..138e894 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -24,6 +24,7 @@ #define SST_HSW_NO_CHANNELS 4 #define SST_HSW_MAX_DX_REGIONS 14 #define SST_HSW_DX_CONTEXT_SIZE (640 * 1024) +#define SST_HSW_CHANNELS_ALL 0xffffffff #define SST_HSW_FW_LOG_CONFIG_DWORDS 12 #define SST_HSW_GLOBAL_LOG 15 diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 7eb9afc..331237f 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -189,7 +189,8 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, if (ucontrol->value.integer.value[0] == ucontrol->value.integer.value[1]) { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume); + /* apply volume value to all channels */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, SST_HSW_CHANNELS_ALL, volume); } else { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume); @@ -255,7 +256,7 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[1]) { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); - sst_hsw_mixer_set_volume(hsw, 0, 2, volume); + sst_hsw_mixer_set_volume(hsw, 0, SST_HSW_CHANNELS_ALL, volume); } else { volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); @@ -525,7 +526,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, dev_err(rtd->dev, "error: failed to commit stream %d\n", ret); return ret; } - pcm_data->allocated = true; + + if (!pcm_data->allocated) { + /* Set previous saved volume */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 0, pcm_data->volume[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 1, pcm_data->volume[1]); + pcm_data->allocated = true; + } ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1); if (ret < 0) @@ -632,12 +641,6 @@ static int hsw_pcm_open(struct snd_pcm_substream *substream) return -EINVAL; } - /* Set previous saved volume */ - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, - 0, pcm_data->volume[0]); - sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, - 1, pcm_data->volume[1]); - mutex_unlock(&pcm_data->mutex); return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: fsl: Use dynamic slot width for ESAI.
by Nicolin Chen 25 Nov '14

25 Nov '14

This series of patches add the support of dynamic slot width for ESAI driver. To do this, the PATCH-1 removes the fixed width. In order not to break those platforms that use fixed width, there comes the PATCH-2 to apply an override of slot_width. @Shengjiu, Will you be available to test this series on Sabre Auto for both Master and Slave cases? I'd like to wait for your Test-by. Thanks. Nicolin Chen (2): ASoC: fsl_esai: Use dynamic slot width as default ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++++++ sound/soc/fsl/fsl_esai.c | 12 +++++++----- 2 files changed, 19 insertions(+), 5 deletions(-) -- 1.9.1

2 9

[alsa-devel] [PATCH v2] ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed
by Oder Chiou 25 Nov '14

25 Nov '14

After the patch "ASoC: rt5677: Use specific r/w function for DSP mode", the regmap_range "rt5677_ranges" was not registered in rt5677_regmap_physical, and it caused that the regmap_range "rt5677_ranges" cannot be accessed by the specific r/w function. The patch fixes this issue. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 1/3] ASoC: rt5670: check if asrc is useable
by Bard Liao 25 Nov '14

25 Nov '14

To use ASRC, the sysclk should be faster than 384 times sample rate of I2S1. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- *Change source->codec to snd_soc_dapm_to_codec(source->dapm) --- sound/soc/codecs/rt5670.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index b842be0..fc26742 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -576,6 +576,18 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int can_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + + if (rt5670->sysclk > rt5670->lrck[RT5670_AIF1] * 384) + return 1; + + return 0; +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, @@ -1639,8 +1651,8 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "DAC Mono Right Filter", NULL, "DAC MONO R ASRC", is_using_asrc }, { "DAC Stereo1 Filter", NULL, "DAC STO ASRC", is_using_asrc }, - { "I2S1", NULL, "I2S1 ASRC" }, - { "I2S2", NULL, "I2S2 ASRC" }, + { "I2S1", NULL, "I2S1 ASRC", can_use_asrc}, + { "I2S2", NULL, "I2S2 ASRC", can_use_asrc}, { "DMIC1", NULL, "DMIC L1" }, { "DMIC1", NULL, "DMIC R1" }, -- 1.8.1.1.439.g50a6b54

3 5

[alsa-devel] [PATCH 1/2] ASoC: rt5645: multiple JD mode support
by Bard Liao 25 Nov '14

25 Nov '14

There are 3 JD modes in RT5645. This patch configure register values according to platform data. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5645.h | 1 + sound/soc/codecs/rt5645.c | 35 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5645.h | 7 +++++++ 3 files changed, 42 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index 937f421..120d961 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -26,6 +26,7 @@ struct rt5645_platform_data { /* true if codec's jd function is used */ bool en_jd_func; + unsigned int jd_mode; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 31b5b1a..a7789a8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2241,7 +2241,8 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); - snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); + if (rt5645->pdata.jd_mode == 0) + snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync(&codec->dapm); } @@ -2545,6 +2546,38 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, RT5645_IRQ_CLK_INT, RT5645_IRQ_CLK_INT); } + if (rt5645->pdata.jd_mode) { + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_IRQ_JD_1_1_EN, RT5645_IRQ_JD_1_1_EN); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL3, + RT5645_JD_PSV_MODE, RT5645_JD_PSV_MODE); + regmap_update_bits(rt5645->regmap, RT5645_HPO_MIXER, + RT5645_IRQ_PSV_MODE, RT5645_IRQ_PSV_MODE); + regmap_update_bits(rt5645->regmap, RT5645_MICBIAS, + RT5645_MIC2_OVCD_EN, RT5645_MIC2_OVCD_EN); + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + switch (rt5645->pdata.jd_mode) { + case 1: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_0); + break; + case 2: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_1); + break; + case 3: + regmap_update_bits(rt5645->regmap, RT5645_A_JD_CTRL1, + RT5645_JD1_MODE_MASK, + RT5645_JD1_MODE_2); + break; + default: + break; + } + } + if (rt5645->i2c->irq) { ret = request_threaded_irq(rt5645->i2c->irq, NULL, rt5645_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index c72220a..a815e36 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -594,6 +594,7 @@ #define RT5645_M_DAC1_HM_SFT 14 #define RT5645_M_HPVOL_HM (0x1 << 13) #define RT5645_M_HPVOL_HM_SFT 13 +#define RT5645_IRQ_PSV_MODE (0x1 << 12) /* SPK Left Mixer Control (0x46) */ #define RT5645_G_RM_L_SM_L_MASK (0x3 << 14) @@ -1350,6 +1351,10 @@ #define RT5645_PWR_CLK25M_PU (0x1 << 4) #define RT5645_IRQ_CLK_MCLK (0x0 << 3) #define RT5645_IRQ_CLK_INT (0x1 << 3) +#define RT5645_JD1_MODE_MASK (0x3 << 0) +#define RT5645_JD1_MODE_0 (0x0 << 0) +#define RT5645_JD1_MODE_1 (0x1 << 0) +#define RT5645_JD1_MODE_2 (0x2 << 0) /* VAD Control 4 (0x9d) */ #define RT5645_VAD_SEL_MASK (0x3 << 8) @@ -1638,6 +1643,7 @@ #define RT5645_OT_P_SFT 10 #define RT5645_OT_P_NOR (0x0 << 10) #define RT5645_OT_P_INV (0x1 << 10) +#define RT5645_IRQ_JD_1_1_EN (0x1 << 9) /* IRQ Control 2 (0xbe) */ #define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) @@ -2120,6 +2126,7 @@ enum { #define RT5645_RXDP2_SEL_SFT (3) /* General Control3 (0xfc) */ +#define RT5645_JD_PSV_MODE (0x1 << 12) #define RT5645_IRQ_CLK_GATE_CTRL (0x1 << 11) #define RT5645_MICINDET_MANU (0x1 << 7) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v3] ASoC: Remove 'const' from the device_node pointers
by Jean-Francois Moine 25 Nov '14

25 Nov '14

As Russell King's explained it, there should not be pointers to struct device_node: "struct device_node is a ref-counted structure. That means if you store a reference to it, you should "get" it, and you should "put" it once you've done. The act of "put"ing the pointed-to structure involves writing to that structure, so it is totally unappropriate to store a device_node structure as a const pointer. It forces you to have to cast it back to a non-const pointer at various points in time to use various OF function calls." Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v3: change the patch comment (Mark Brown) v2: remove const from the pointers instead of modifying the OF functions (Russell King) --- include/sound/soc.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 879e2b3..13e1648 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -883,7 +883,7 @@ struct snd_soc_platform_driver { struct snd_soc_dai_link_component { const char *name; - const struct device_node *of_node; + struct device_node *of_node; const char *dai_name; }; @@ -985,7 +985,7 @@ struct snd_soc_codec_conf { * DT/OF node, but not both. */ const char *dev_name; - const struct device_node *of_node; + struct device_node *of_node; /* * optional map of kcontrol, widget and path name prefixes that are @@ -1002,7 +1002,7 @@ struct snd_soc_aux_dev { * DT/OF node, but not both. */ const char *codec_name; - const struct device_node *codec_of_node; + struct device_node *codec_of_node; /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_component *component); -- 2.1.3

1 0

[alsa-devel] [PATCH] ALSA: hda - Add AZX_DCAPS_SNOOP_OFF (and refactor snoop setup)
by Takashi Iwai 25 Nov '14

25 Nov '14

Add a new driver_caps bit, AZX_DCAPS_SNOOP_OFF, to set the snoop off as default. This new bit is used for the checks in azx_check_snoop_available(). Most of case-switches are replaced with the new dcaps in each entry. While working on it, for avoiding to spend more bits, combine three bits AZX_DCAPS_SNOOP_SCH, AZX_DCAPS_SNOOP_ATI and AZX_DCAPS_SNOOP_NVIDIA bits into a flat type of two bits. This reduces the bits usages, and assign AZX_DCAPS_OFF to this empty bit now. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 71 +++++++++++++++++++++++++---------------------- sound/pci/hda/hda_priv.h | 12 ++++++-- 2 files changed, 47 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91fa959d05fe..b1b4a72cf29c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -271,42 +271,51 @@ enum { AZX_NUM_DRIVERS, /* keep this as last entry */ }; +#define azx_get_snoop_type(chip) \ + (((chip)->driver_caps & AZX_DCAPS_SNOOP_MASK) >> 10) +#define AZX_DCAPS_SNOOP_TYPE(type) ((AZX_SNOOP_TYPE_ ## type) << 10) + /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_NOPM \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_REVERSE_ASSIGN) + (AZX_DCAPS_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + AZX_DCAPS_REVERSE_ASSIGN | AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_PCH \ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) #define AZX_DCAPS_INTEL_HASWELL \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME | \ - AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ - (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_ALIGN_BUFSIZE | \ - AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_PM_RUNTIME | \ - AZX_DCAPS_I915_POWERWELL) + (AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ + AZX_DCAPS_SNOOP_TYPE(SCH)) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ - (AZX_DCAPS_ATI_SNOOP | AZX_DCAPS_NO_TCSEL | \ - AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB) + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_SNOOP_TYPE(ATI)) /* quirks for ATI/AMD HDMI */ #define AZX_DCAPS_PRESET_ATI_HDMI \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB) +/* quirks for ATI HDMI with snoop off */ +#define AZX_DCAPS_PRESET_ATI_HDMI_NS \ + (AZX_DCAPS_PRESET_ATI_HDMI | AZX_DCAPS_SNOOP_OFF) + /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ - AZX_DCAPS_ALIGN_BUFSIZE | AZX_DCAPS_NO_64BIT |\ - AZX_DCAPS_CORBRP_SELF_CLEAR) + (AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI | AZX_DCAPS_ALIGN_BUFSIZE |\ + AZX_DCAPS_NO_64BIT | AZX_DCAPS_CORBRP_SELF_CLEAR |\ + AZX_DCAPS_SNOOP_TYPE(NVIDIA)) #define AZX_DCAPS_PRESET_CTHDA \ - (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_4K_BDLE_BOUNDARY) + (AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB |\ + AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) /* * VGA-switcher support @@ -435,6 +444,8 @@ static void update_pci_byte(struct pci_dev *pci, unsigned int reg, static void azx_init_pci(struct azx *chip) { + int snoop_type = azx_get_snoop_type(chip); + /* Clear bits 0-2 of PCI register TCSEL (at offset 0x44) * TCSEL == Traffic Class Select Register, which sets PCI express QOS * Ensuring these bits are 0 clears playback static on some HD Audio @@ -449,7 +460,7 @@ static void azx_init_pci(struct azx *chip) /* For ATI SB450/600/700/800/900 and AMD Hudson azalia HD audio, * we need to enable snoop. */ - if (chip->driver_caps & AZX_DCAPS_ATI_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_ATI) { dev_dbg(chip->card->dev, "Setting ATI snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, @@ -458,7 +469,7 @@ static void azx_init_pci(struct azx *chip) } /* For NVIDIA HDA, enable snoop */ - if (chip->driver_caps & AZX_DCAPS_NVIDIA_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_NVIDIA) { dev_dbg(chip->card->dev, "Setting Nvidia snoop: %d\n", azx_snoop(chip)); update_pci_byte(chip->pci, @@ -473,7 +484,7 @@ static void azx_init_pci(struct azx *chip) } /* Enable SCH/PCH snoop if needed */ - if (chip->driver_caps & AZX_DCAPS_SCH_SNOOP) { + if (snoop_type == AZX_SNOOP_TYPE_SCH) { unsigned short snoop; pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); if ((!azx_snoop(chip) && !(snoop & INTEL_SCH_HDA_DEVC_NOSNOOP)) || @@ -1360,8 +1371,8 @@ static void azx_check_snoop_available(struct azx *chip) { bool snoop = chip->snoop; - switch (chip->driver_type) { - case AZX_DRIVER_VIA: + if (azx_get_snoop_type(chip) == AZX_SNOOP_TYPE_NONE && + chip->driver_type == AZX_DRIVER_VIA) { /* force to non-snoop mode for a new VIA controller * when BIOS is set */ @@ -1371,17 +1382,11 @@ static void azx_check_snoop_available(struct azx *chip) if (!(val & 0x80) && chip->pci->revision == 0x30) snoop = false; } - break; - case AZX_DRIVER_ATIHDMI_NS: - /* new ATI HDMI requires non-snoop */ - snoop = false; - break; - case AZX_DRIVER_CTHDA: - case AZX_DRIVER_CMEDIA: - snoop = false; - break; } + if (chip->driver_caps & AZX_DCAPS_SNOOP_OFF) + snoop = false; + if (snoop != chip->snoop) { dev_info(chip->card->dev, "Force to %s mode\n", snoop ? "snoop" : "non-snoop"); @@ -2112,13 +2117,13 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0xaa98), .driver_data = AZX_DRIVER_ATIHDMI | AZX_DCAPS_PRESET_ATI_HDMI }, { PCI_DEVICE(0x1002, 0x9902), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaaa0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaaa8), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaab0), - .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI }, + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA }, @@ -2165,7 +2170,7 @@ static const struct pci_device_id azx_ids[] = { /* CM8888 */ { PCI_DEVICE(0x13f6, 0x5011), .driver_data = AZX_DRIVER_CMEDIA | - AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB }, + AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB | AZX_DCAPS_SNOOP_OFF }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index 949cd437eeb2..602536c2147d 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -152,9 +152,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; /* bits 0-7 are used for indicating driver type */ #define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ #define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ -#define AZX_DCAPS_ATI_SNOOP (1 << 10) /* ATI snoop enable */ -#define AZX_DCAPS_NVIDIA_SNOOP (1 << 11) /* Nvidia snoop enable */ -#define AZX_DCAPS_SCH_SNOOP (1 << 12) /* SCH/PCH snoop enable */ +#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ +#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ #define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ #define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ @@ -172,6 +171,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ +enum { + AZX_SNOOP_TYPE_NONE , + AZX_SNOOP_TYPE_SCH, + AZX_SNOOP_TYPE_ATI, + AZX_SNOOP_TYPE_NVIDIA, +}; + /* HD Audio class code */ #define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 -- 2.1.3

1 0

Re: [alsa-devel] ASoC: Intel: sst: Add IPC handling
by Dan Carpenter 25 Nov '14

25 Nov '14

Hello Vinod Koul, The patch ea12aa4acd70: "ASoC: Intel: sst: Add IPC handling" from Oct 16, 2014, leads to the following static checker warning: sound/soc/intel/sst/sst_ipc.c:340 sst_process_reply_mrfld() warn: we tested 'drv_id' before and it was 'true' sound/soc/intel/sst/sst_ipc.c 330 drv_id = msg_high.part.drv_id; 331 332 /* Check for async messages first */ 333 if (drv_id == SST_ASYNC_DRV_ID) { ^^^^^^^^^^^^^^^^ This is zero. 334 /*FW sent async large message*/ 335 process_fw_async_msg(sst_drv_ctx, msg); 336 return; 337 } 338 339 /* FW sent short error response for an IPC */ 340 if (msg_high.part.result && drv_id && !msg_high.part.large) { ^^^^^^ So this is non-zero. The thinking behind these warnings is that maybe a different test was intended. What does a non-zero drv_id mean which is different from checking SST_ASYNC_DRV_ID? The intent is not clear. In other words, would it be more readable to test "drv_id != SST_ASYNC_DRV_ID"? But in that case, we would just leave the test out since we obviously tested for that earlier. 341 /* 32-bit FW error code in msg_low */ 342 dev_err(sst_drv_ctx->dev, "FW sent error response 0x%x", msg_low); 343 sst_wake_up_block(sst_drv_ctx, msg_high.part.result, 344 msg_high.part.drv_id, 345 msg_high.part.msg_id, NULL, 0); 346 return; 347 } 348 regards, dan carpenter

1 0

[alsa-devel] [PATCH] ASoC: rockchip: i2s: Add dependency to I2S device driver
by Andreas Ruprecht 24 Nov '14

24 Nov '14

Currently, CONFIG_SND_SOC_ROCKCHIP_I2S could also be selected without having CONFIG_SND_SOC_ROCKCHIP enabled. As this makes no sense (and breaks the build, when the option CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM - which would be selected by CONFIG_SND_SOC_ROCKCHIP - is not enabled), a Kconfig dependency is added to CONFIG_SND_SOC_ROCKCHIP_I2S. This will make the item visible only if CONFIG_SND_SOC_ROCKCHIP is enabled. Signed-off-by: Andreas Ruprecht <rupran(a)einserver.de> Reported-by: Jim Davis <jim.epost(a)gmail.com> --- sound/soc/rockchip/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index b1fc0ca..74380b4 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -9,7 +9,7 @@ config SND_SOC_ROCKCHIP config SND_SOC_ROCKCHIP_I2S tristate "Rockchip I2S Device Driver" - depends on CLKDEV_LOOKUP + depends on CLKDEV_LOOKUP && SND_SOC_ROCKCHIP help Say Y or M if you want to add support for I2S driver for Rockchip I2S device. The device supports upto maximum of -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Fix off by one error in the loop stucture.
by Jyri Sarha 24 Nov '14

24 Nov '14

Fix off by one read beyond the end of a table. Reported-by: David Binderman <dcb314(a)hotmail.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 145fe5b..93de5dd 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -911,12 +911,13 @@ static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, } aic31xx->p_div = i; - for (i = 0; aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) { - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", - __func__, freq); - return -EINVAL; - } + for (i = 0; i < ARRAY_SIZE(aic31xx_divs) && + aic31xx_divs[i].mclk_p != freq/aic31xx->p_div; i++) + ; + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; } /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH 01/32] ASoC: wm8350: Cleanup manual bias level transitions
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8350.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 628ec77..87f664b 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1242,19 +1242,6 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8350_suspend(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - -static int wm8350_resume(struct snd_soc_codec *codec) -{ - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} - static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) @@ -1565,9 +1552,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); - - wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } @@ -1596,8 +1580,6 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) * wait for its completion */ flush_delayed_work(&codec->dapm.delayed_work); - wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); - wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); return 0; @@ -1613,10 +1595,9 @@ static struct regmap *wm8350_get_regmap(struct device *dev) static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .probe = wm8350_codec_probe, .remove = wm8350_codec_remove, - .suspend = wm8350_suspend, - .resume = wm8350_resume, .get_regmap = wm8350_get_regmap, .set_bias_level = wm8350_set_bias_level, + .suspend_bias_off = true, .controls = wm8350_snd_controls, .num_controls = ARRAY_SIZE(wm8350_snd_controls), -- 1.8.0

3 64

[alsa-devel] [PATCH 1/3] ASoC: alc5623: Cleanup bias level transitions
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

Set the CODEC driver's suspend_bias_off flag rather than manually going to SND_SOC_BIAS_OFF in suspend and SND_SOC_BIAS_STANDBY in resume. This makes the code a bit shorter and cleaner. Also remove the manual sequencing back to SND_SOC_BIAS_ON in resume as this is already handled by the ASoC core. Since the ASoC core now takes care of setting the bias level to SND_SOC_BIAS_OFF when removing the CODEC there is no need to do it manually anymore either. The manual transition to SND_SOC_BIAS_STANDBY at the end of CODEC probe() can also be removed as the core will automatically do this after the CODEC has been probed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/alc5623.c | 22 +--------------------- 1 file changed, 1 insertion(+), 21 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9d0755a..bdf8c5a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(alc5623->regmap, true); return 0; @@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec) return ret; } - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge alc5623 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->dapm.bias_level); - } - return 0; } @@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) alc5623_reset(codec); - /* power on device */ - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); @@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5623_remove(struct snd_soc_codec *codec) -{ - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, - .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config alc5623_regmap = { -- 1.8.0

2 5

[alsa-devel] [PATCH 1/3] ASoC: uda134x: Remove is_powered_on_standby from platform data
by Lars-Peter Clausen 24 Nov '14

24 Nov '14

According to its documentation the is_powered_on_standby field of the uda134x platform data is supposed to prevent the the driver from shutting down the ADC and DAC in standby mode. This behavior was broken in commit commit f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") almost 5 years ago and all the flag does now is cause the driver to go to SND_SOC_BIAS_ON in probe, just for the ASoC core to put it back into SND_SOC_BIAS_STANDBY right after probe. Apparently the intended behavior has not been missed, so just remove is_powered_on_standby from the platform data struct. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/uda134x.h | 12 ------------ sound/soc/codecs/uda134x.c | 5 +---- 2 files changed, 1 insertion(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index e475659..509efb0 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,18 +18,6 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; - /* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you unset is_powered_on_standby the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - set is_powered_on_standby, you will have slightly higher power - consumption. Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 32b2f78..54240f1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -518,10 +518,7 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) uda134x_reset(codec); - if (pd->is_powered_on_standby) - uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); - else - uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (pd->model == UDA134X_UDA1341) { widgets = uda1341_dapm_widgets; -- 1.8.0

2 3

[alsa-devel] [PATCH 1/3] ASoC: max98090: Fix digital microphone
by Jarkko Nikula 24 Nov '14

24 Nov '14

Commit e409dfbfccf9 ("ASoC: dapm: Add a few supply widget sanity checks") broke digital microphone support in max98090.c: max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICL_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICL_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICL_ENA -> DMIC -> DMIC Mux max98090 i2c-193C9890:00: Conditional paths are not supported for supply widgets (DMICR_ENA -> [DMIC] -> DMIC Mux) max98090 i2c-193C9890:00: ASoC: no dapm match for DMICR_ENA --> DMIC --> DMIC Mux max98090 i2c-193C9890:00: ASoC: Failed to add route DMICR_ENA -> DMIC -> DMIC Mux Problem is partially caused by commit f69e3caa9e18 ("ASoC: max98090: Enable both DMIC channels also when using mono configuration") which connects "DMICL_ENA" and "DMICR_ENA" supply widgets to "DMIC Mux". Fix the breakage by reverting f69e3caa9e18 and then by adding additional "DMICR_ENA" to "DMICL" and "DMICL_ENA" to "DMICR" cross-connections. This disconnects these supply widgets from the mux and makes sure that both DMIC data channels are still enabled together. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98090.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 2ad381c4ec57..19258a910974 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,6 +1311,10 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, + {"DMICL", NULL, "DMICL_ENA"}, + {"DMICL", NULL, "DMICR_ENA"}, + {"DMICR", NULL, "DMICL_ENA"}, + {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1368,8 +1372,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, - {"DMIC Mux", "DMIC", "DMICL_ENA"}, - {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, -- 2.1.3

2 5

[alsa-devel] [PATCH] ASoC: rt5677: Fix the issue that the regmap_range "rt5677_ranges" cannot be accessed
by Oder Chiou 24 Nov '14

24 Nov '14

After the commit "19ba484d7b15c8650b30377aad6e65b34d3cf3d5", it will cause that the regmap_range "rt5677_ranges" cannot be accessed. The patch fixes this issue. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Arndale: Add machine device tree binding documentation for Arndale board
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

Document the device tree binding for the Arndale board based machine driver. Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- .../devicetree/bindings/sound/arndale.txt | 24 ++++++++++++++++++++ 1 file changed, 24 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt new file mode 100644 index 0000000..0e76946 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/arndale.txt @@ -0,0 +1,24 @@ +Audio Binding for Arndale boards + +Required properties: +- compatible : Can be the following, + "samsung,arndale-rt5631" + +- samsung,audio-cpu: The phandle of the Samsung I2S controller +- samsung,audio-codec: The phandle of the audio codec + +Optional: +- samsung,model: The name of the sound-card + +Arndale Boards has many audio daughter cards, one of them is +rt5631/alc5631. Below example shows audio bindings for rt5631/ +alc5631 based codec. + +Example: + +sound { + compatible = "samsung,arndale-rt5631"; + + samsung,audio-cpu = <&i2s0> + samsung,audio-codec = <&rt5631>; +}; -- 1.7.9.5

2 1

[alsa-devel] TLV320AIC3104 Sample rate problem
by Catalin Crenguta 24 Nov '14

24 Nov '14

Hello, I'm using an Audio Cape Rev. B connected to BeagleBone Black (MCLK=12MHz) and I'm trying to record audio at sample rate 8KHz. Recording and playback using ALSA is OK, but when playing the recorded samples on PC you can hear that the actual sample rate is lower (the pitch shifted to a lower tone). I have added debug messages throughout the alsa driver tlv320aic3x.c, and I found out that the PLL parameters are computed correctly as in TLV320AIC3104 Datasheet Table found at page 27. (p: 1 j: 8 d: 1920 r: 1). Has anyone else encountered a problem like this? Can someone with more experience check the driver? Kernel: https://github.com/beagleboard/linux , commit 2bcc89c, 3.14.19+. Best regards, Catalin Crenguta

1 0

[alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 24 Nov '14

24 Nov '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch and all of my work for Linux 3.19. == Features: Currently, ALSA Dice driver is limited for playback-only devices. The aim of this patchset is to add supports for: - Full-duplex streams with synchronization - Both PCM capture and playback - Both MIDI capture and playback Currently, ALSA firewire-speakers driver is only for two playback-only models. And this driver has a bug for one of them. The aim of this patchset is to fix the bug and to add supports for more models with: - Both PCM capture and playback - Both MIDI capture and playback The renew drivers doesn't support: - Synchronization between multiple devices on the same bus For this purpose, please join in user-space driver project (FFADO) - Any control functionality for internal DSP For this purpose, please use ffado-dbus-server/ffado-mixer, or write your own application == Updates from my previous post: [alsa-devel] [PATCH 00/14] ALSA: Dice enhancement http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081890.ht… - Fix typo - Minor code change [alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076581.html - Support more Mackie (Loud) models by name detection - Fix a bug to handle FireWave - Use AV/C Stream Format Information command to set sampling rate == Remarks: - When the Dice driver is loaded, user-space driver fails to register notification address, thus cannot handle device for streaming. See: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081731.… - OXFW chipset seems to transmit wrong values for timestamp, thus the driver don't apply full-duplex streams with synchronization == Supported Dice-based models (as long as I know): TC Electronic - Desktop Konnekt 6 - Impact Twin - Konnekt 8/24D/Live - Studio Konnekt 48 PreSonus - FireStudio 26x26/Mobile/Project/Tube/LightPipe - StudioLive 1602/1642/2442 - FireStation Mackie - Onyx Blackbird - Onyx i-series (latter models) Focusrite - Saffire Pro 14/24/24 DSP/26/40/56 M-Audio - Profire 610/2626 Alessis - IO 24/26 - MultiMix 12/16 Weiss - ADC2 - Vesta - Mineva - AFI 1 - TAG DAC 1 - INT 202 - DAC 202 - Flex 5000 DnR - Axus FireWire I/O card 16x16 Lexicon - I-ONIX FW 810S Allen and Heath - Zed R16 Midas - Venice F32 == Supported OXFW-based models (as long as I know): Griffin: - FireWave LaCie: - Firewire Speakers Behringer: - F-Control Audio 202 Mackie (Loud): - Onyx-i series (former models) - Onyx Satellite - Tapco Link.Firewire - d.2 pro/d.4 pro - U.420/U.420d Takashi Sakamoto (29): ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 6 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 176 ++++ sound/firewire/dice/dice-pcm.c | 404 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 423 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 359 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 189 ++++ sound/firewire/oxfw/oxfw-pcm.c | 399 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 702 +++++++++++++ sound/firewire/oxfw/oxfw.c | 310 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4887 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile 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 create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 1.9.1

4 42

[alsa-devel] [PATCH] ASoC: bt-sco: support wide band speech
by Barry Song 24 Nov '14

24 Nov '14

From: Yibo Cai <yibo.cai(a)csr.com> HFP1.6 requires support for wide band speech(16KHz) over Bluetooth. Signed-off-by: Yibo Cai <yibo.cai(a)csr.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/bt-sco.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index c4cf069..fc0664b 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -31,14 +31,14 @@ static struct snd_soc_dai_driver bt_sco_dai = { .stream_name = "Playback", .channels_min = 1, .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -- 2.1.1

2 3

[alsa-devel] [PATCH] ASoC: rt5631: Fixing compilation warning when DT is disabled
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

Fixes the following compilation warning: Warning: 'rt5631_i2c_dt_ids' defined but not used - when DT is not used. Signed-off-by: Claude Youn <claude.youn(a)gmail.com> Signed-off-by: Krishna Mohan Dani <krishna.md(a)samsung.com> --- sound/soc/codecs/rt5631.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3b7d5e4..9425545 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1691,12 +1691,14 @@ static const struct i2c_device_id rt5631_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5631_i2c_id); +#ifdef CONFIG_OF static struct of_device_id rt5631_i2c_dt_ids[] = { { .compatible = "realtek,rt5631"}, { .compatible = "realtek,alc5631"}, { } }; MODULE_DEVICE_TABLE(of, rt5631_i2c_dt_ids); +#endif static const struct regmap_config rt5631_regmap_config = { .reg_bits = 8, -- 1.7.9.5

4 4

[alsa-devel] [PATCH 0/4 V3] Arndale: Adding Sound Machine Driver for ALC5631 Codec
by Krishna Mohan Dani 24 Nov '14

24 Nov '14

These patches add machine driver to instantiate I2S based realtek's ALC5631 based sound card on Arndale board. There are other variants of Audio Daughter Cards for Arndale Board for which support already exists but there is no support for Realtek's alc5631 codec based card hence support for ALC5631 based machine driver is being added. Most of the code has been pulled from insignal's git for Arndale and modified to suit the maintain syntax. Machine driver has been made DT compatible. Changes done: - Removed ifdef for of_device_id structure as commented by Mark Brown - Supporting both variants of device name:alc5631 and rt5631 as commented by Mark Brown - Added "return -EINVAL" if cpu_of_node or codec_of_node is not found in dtb, as commented by Anil Kumar - Removed i2s specific changes - Added signed off of claude - Removed CONFIG_OF check from rt5631.c codec driver and other changes Krishna Mohan Dani (4): ASoC: ALC5631/RT5631: Add device tree binding documentation ASoC: Samsung: Add arndale_rt5631 machine driver Sound: Kconfig: Adding the description of the codec ASoC: rt5631: Adding Device Tree compatibility to Realtek's ALC5631/RT5631 codec driver Documentation/devicetree/bindings/sound/rt5631.txt | 48 +++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/rt5631.c | 9 ++ sound/soc/samsung/Kconfig | 6 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/arndale_rt5631.c | 150 ++++++++++++++++++++ 6 files changed, 217 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt create mode 100644 sound/soc/samsung/arndale_rt5631.c -- 1.7.9.5

3 9

[alsa-devel] [PATCH] ASoC: soc-pcm: skip dpcm path checking with incapable/unready FE
by Qiao Zhou 24 Nov '14

24 Nov '14

Skip dpcm path checking for playback or capture, if corresponding FE doesn't support playback or capture, or currently is not ready. It can reduce the unnecessary cost to search connected widgets. Signed-off-by: Qiao Zhou <zhouqiao(a)marvell.com> --- sound/soc/soc-pcm.c | 16 ++++++++++++++-- 1 file changed, 14 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 002311a..3b068d4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2248,7 +2248,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) fe->dai_link->name); /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min) + if (!fe->cpu_dai->driver->playback.channels_min + || !fe->codec_dai->driver->playback.channels_min) + goto capture; + + /* skip if FE doesn't do playback */ + if (!fe->cpu_dai->playback_active + || !fe->codec_dai->playback_active) goto capture; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); @@ -2278,7 +2284,13 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) dpcm_path_put(&list); capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min) + if (!fe->cpu_dai->driver->capture.channels_min + || !fe->codec_dai->driver->capture.channels_min) + continue; + + /* skip if FE doesn't do capture */ + if (!fe->cpu_dai->capture_active + || !fe->codec_dai->capture_active) continue; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); -- 1.7.9.5

2 2

[alsa-devel] Debian Bug#770122: patch as workaround/solution
by Elimar Riesebieter 23 Nov '14

23 Nov '14

Hi, Iam forwarding this Debian bug to alsa-devel due to a not maintained BugTracker at alsa project. Could one please have a look at it? Thanks Elimar -- Alles was viel bedacht wird ist bedenklich!;-) Friedrich Nietzsche

1 1

[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 23 Nov '14

23 Nov '14

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

2 12

[alsa-devel] Debian Bug#770122: patch as workaround/solution
by Elimar Riesebieter 23 Nov '14

23 Nov '14

Hi, I am forwarding this Debian bug [0] to alsa-devel due to a not maintained BugTracker at alsa project. Could one please have a look at it? [0] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=770122 Thanks Elimar -- Alles was viel bedacht wird ist bedenklich!;-) Friedrich Nietzsche

1 0

[alsa-devel] [PATCH 1/2] amixer: Make "dB" case-insensitive in set commands
by Takashi Iwai 23 Nov '14

23 Nov '14

We don't have to be necessarily too strict about case-sensitivity of "dB" suffix used in set commands. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index cf82892bfa2f..6a2fdb96c62f 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -346,7 +346,7 @@ static int set_volume_simple(snd_mixer_elem_t *elem, if (*p == '%') { percent = 1; p++; - } else if (p[0] == 'd' && p[1] == 'B') { + } else if (toupper(p[0]) == 'D' && toupper(p[1]) == 'B') { vol_type = VOL_DB; p += 2; scale = 100; -- 2.1.3

1 1

[alsa-devel] [PATCH 00/10] ALSA: asihpi: updates
by eliot＠blennerhassett.gen.nz 23 Nov '14

23 Nov '14

From: Eliot Blennerhassett <eliot(a)blennerhassett.gen.nz> Major change - Add support for interrupt driven 'low latency' mode PCM streams on some AudioScience adapters. Various other minor updates from AudioScience Eliot Blennerhassett (10): ALSA: asihpi: Minor string and dead code cleanup ALSA: asihpi: New I/O types - AVB & BLUlink, DAB Rf receiver ALSA: asihpi: Logging format improvements ALSA: asihpi: Use CONFIG_64BIT directly ALSA: asihpi: Refactor control cache code. ALSA: asihpi: Add support for stream interrupt. ALSA: asihpi: Turn off msg/resp logging after DSP has crashed. ALSA: asihpi: Use standard printk helpers ALSA: asihpi: don't fail probe if adapter mode read fails ALSA: asihpi: used parts of message/response are zeroed before use sound/pci/asihpi/asihpi.c | 326 +++++++++++++++++++++++++++------------- sound/pci/asihpi/hpi.h | 16 +- sound/pci/asihpi/hpi6205.c | 43 +++++- sound/pci/asihpi/hpi_internal.h | 20 ++- sound/pci/asihpi/hpicmn.c | 107 +++++++------ sound/pci/asihpi/hpicmn.h | 19 ++- sound/pci/asihpi/hpimsginit.c | 30 ++-- sound/pci/asihpi/hpimsgx.c | 15 +- sound/pci/asihpi/hpioctl.c | 123 +++++++++++++-- sound/pci/asihpi/hpios.h | 8 +- 10 files changed, 508 insertions(+), 199 deletions(-) -- 1.9.1

3 12

[alsa-devel] randconfig errors in ts3a227e.c
by devendra.aaru 23 Nov '14

23 Nov '14

Hello All, While testing the build, i have got the following errors: sound/soc/codecs/ts3a227e.c: In function ‘ts3a227e_enable_jack_detect’: sound/soc/codecs/ts3a227e.c:223:2: error: implicit declaration of function ‘snd_jack_set_key’ [-Werror=implicit-function-declaration] snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); ^ cc1: some warnings being treated as errors make[3]: *** [sound/soc/codecs/ts3a227e.o] Error 1 The CONFIG_SND_JACK is not enabled in the .config and the #else portion of include/sound/jack.h doesn't contain snd_jack_set_key stub. This resulting in the build error. I am unsure if adding stub in include/sound/jack.h or adding a Kconfig knob to deselect the ts3a227e.c fixes the issue ? full randconfig attached. (sorry about the 55k file size)

1 1

[alsa-devel] [patch] sound: oss: uart401: remove unneeded NULL check
by Dan Carpenter 22 Nov '14

22 Nov '14

"devc" can't be NULL here so there is no need to check. Also I removed the "devc = NULL" assignment because devc is stored on stack so it's a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 279bc56..dae4d43 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -412,13 +412,10 @@ void unload_uart401(struct address_info *hw_config) if (!devc->share_irq) free_irq(devc->irq, devc); - if (devc) - { - kfree(midi_devs[devc->my_dev]->converter); - kfree(midi_devs[devc->my_dev]); - kfree(devc); - devc = NULL; - } + kfree(midi_devs[devc->my_dev]->converter); + kfree(midi_devs[devc->my_dev]); + kfree(devc); + /* This kills midi_devs[x] */ sound_unload_mididev(hw_config->slots[4]); }

2 1

[alsa-devel] Headset jack and intel based Macmini6,2
by Marco 22 Nov '14

22 Nov '14

/cc David Henningsson that has done something like that for some Dell HW and might have some suggestions Hello, even if poorly advertised, recent mac.* have an headset jack (amplified) along with the non amplified line jack. For example, for MacMini, from: apple.com/mac-mini/specs/ > Support for Apple iPhone headset with microphone Where iPhone headset means a simple headset not preamplified, really. I own a MacMini6,2 (the october 2012 one) and with Mac OSX it works... under linux... well... it is a simple out jack. This jeopardize the ability to use any video conference software (who doesn't today?) on linux if you don't own a USB audio device (or you don't want that extra piece of relatively large hw) The card is a HDA based Cirrus Logic CS4206/4207 and the two specific models do not work (imac27 and mbp55) while auto works (except the headset of course :) I tried to change the pinconfig but on the intel HDA documentation it is not really clear how the code for an headset jack (in/out) would look like. I saw that for card ALC269/270/275/276/28x/29x there are models that support the headset codes but... well.. they don't really seem "standard" intel codes I tried to look if it was possible to extract these codes from OSX but I haven't found anything interesting (but I might have missed something) Any suggestion dear esteemed Alsa developers? :) Any help appreciated, Thanks! Marco

1 0

[alsa-devel] [PATCH] ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
by mengdong.lin＠intel.com 21 Nov '14

21 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..7479ce0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -16,6 +16,9 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_MACHINE + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI @@ -76,3 +79,20 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + select SND_SST_IPC + select SND_SST_MACHINE + default n + + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..4069d3f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,6 +31,7 @@ obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SST_MACHINE) += board/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..9ecc227 --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,2 @@ +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o diff --git a/sound/soc/intel/board/cht_bsw_rt5672.c b/sound/soc/intel/board/cht_bsw_rt5672.c new file mode 100644 index 0000000..dffd8b1 --- /dev/null +++ b/sound/soc/intel/board/cht_bsw_rt5672.c @@ -0,0 +1,286 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Mengdong Lin <mengdong.lin(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + unsigned int fmt; + + if (strncmp(codec_dai->name, "rt5670-aif1", 11)) + return 0; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, + SNDRV_PCM_FORMAT_GSM); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* TDM slave Mode */ + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec DAI fmt %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int cht_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + break; + default: + dev_err(card->dev, "Invalid bias level=%d\n", level); + return -EINVAL; + } + card->dapm.bias_level = level; + return 0; +} + +static int cht_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + + /* Set card bias level */ + cht_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + ret = snd_soc_dapm_sync(&card->dapm); + if (ret) { + dev_err(card->dev, "unable to sync dapm\n"); + return ret; + } + return ret; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .init = cht_init, + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .set_bias_level = cht_set_bias_level, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- 1.9.1

4 24

[alsa-devel] [PATCH 1/1] ALSA: sb: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 21 Nov '14

21 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 21 Nov 2014 19:32:02 +0100 The functions release_and_free_resource() and snd_util_memhdr_free() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/isa/sb/emu8000_synth.c | 3 +-- sound/isa/sb/sb_common.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 4e3fcfb..95b39be 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -105,8 +105,7 @@ static int snd_emu8000_delete_device(struct snd_seq_device *dev) snd_device_free(dev->card, hw->pcm); if (hw->emu) snd_emux_free(hw->emu); - if (hw->memhdr) - snd_util_memhdr_free(hw->memhdr); + snd_util_memhdr_free(hw->memhdr); hw->emu = NULL; hw->memhdr = NULL; return 0; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 3ef9906..f22b448 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -184,8 +184,7 @@ static int snd_sbdsp_probe(struct snd_sb * chip) static int snd_sbdsp_free(struct snd_sb *chip) { - if (chip->res_port) - release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); #ifdef CONFIG_ISA -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: es1688_lib: Deletion of an unnecessary check before the function call "release_and_free_resource"
by SF Markus Elfring 21 Nov '14

21 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 21 Nov 2014 19:05:50 +0100 The release_and_free_resource() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/isa/es1688/es1688_lib.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index b3b4f15..fb259491 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -614,8 +614,7 @@ static int snd_es1688_free(struct snd_es1688 *chip) { if (chip->hardware != ES1688_HW_UNDEF) snd_es1688_init(chip, 0); - if (chip->res_port) - release_and_free_resource(chip->res_port); + release_and_free_resource(chip->res_port); if (chip->irq >= 0) free_irq(chip->irq, (void *) chip); if (chip->dma8 >= 0) { -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: core: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 21 Nov '14

21 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Fri, 21 Nov 2014 18:34:48 +0100 The functions snd_seq_oss_timer_delete() and vunmap() perform also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/core/seq/oss/seq_oss_init.c | 9 +++------ sound/core/sgbuf.c | 3 +-- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b9184d2..b0e32e1 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -403,14 +403,11 @@ free_devinfo(void *private) { struct seq_oss_devinfo *dp = (struct seq_oss_devinfo *)private; - if (dp->timer) - snd_seq_oss_timer_delete(dp->timer); + snd_seq_oss_timer_delete(dp->timer); - if (dp->writeq) - snd_seq_oss_writeq_delete(dp->writeq); + snd_seq_oss_writeq_delete(dp->writeq); - if (dp->readq) - snd_seq_oss_readq_delete(dp->readq); + snd_seq_oss_readq_delete(dp->readq); kfree(dp); } diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 0a41850..84fffab 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -39,8 +39,7 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) if (! sgbuf) return -EINVAL; - if (dmab->area) - vunmap(dmab->area); + vunmap(dmab->area); dmab->area = NULL; tmpb.dev.type = SNDRV_DMA_TYPE_DEV; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l51: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs42l51.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 669c38f..b395152 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -153,15 +153,17 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + switch (event) { case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + snd_soc_update_bits(codec, CS42L51_POWER_CTL1, CS42L51_POWER_CTL1_PDN, CS42L51_POWER_CTL1_PDN); break; default: case SND_SOC_DAPM_POST_PMD: - snd_soc_update_bits(w->codec, CS42L51_POWER_CTL1, + snd_soc_update_bits(codec, CS42L51_POWER_CTL1, CS42L51_POWER_CTL1_PDN, 0); break; } -- 1.8.0

3 4

[alsa-devel] [PATCH v3] ASoC: wm8960: Add device tree support
by Zidan Wang 21 Nov '14

21 Nov '14

Document the device tree binding for the WM8960 codec, and modify the driver to extract the platform data from device tree, if present. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 31 ++++++++++++++++ sound/soc/codecs/wm8960.c | 41 ++++++++++++++++------ 2 files changed, 62 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt new file mode 100644 index 0000000..2deb8a3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -0,0 +1,31 @@ +WM8960 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8960" + + - reg : the I2C address of the device. + +Optional properties: + - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of + R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins + will be disabled only when ADC (Left and Right) and DAC (Left and Right) + are disabled. + When wm8960 works on synchronize mode and DACLRC pin is used to supply + frame clock, it will no frame clock for captrue unless enable DAC to enable + DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue. + + - wlf,capless: This is a boolean property. If present, OUT3 pin will be + enabled and disabled together with HP_L and HP_R pins in response to jack + detect events. + +Example: + +codec: wm8960@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + wlf,shared-lrclk; +}; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..99d6457 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = { static int wm8960_add_widgets(struct snd_soc_codec *codec) { - struct wm8960_data *pdata = codec->dev->platform_data; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + struct wm8960_data *pdata = &wm8960->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -1029,6 +1026,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "wlf,capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "wlf,shared-lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1045,6 +1054,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (pdata && pdata->shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); @@ -1075,10 +1089,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); +static const struct of_device_id wm8960_of_match[] = { + { .compatible = "wlf,wm8960", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8960_of_match); + static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", .owner = THIS_MODULE, + .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau17x1: Mark DSP parameter memory as readable and precious
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

To be able to read back data from the DSP parameter memory the register range needs to be marked as readable. At the same time we do not want them to e.g. appear in debugfs output so mark them as precious as well. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adau1761.c | 1 + sound/soc/codecs/adau1781.c | 1 + sound/soc/codecs/adau17x1.c | 14 ++++++++++++++ sound/soc/codecs/adau17x1.h | 1 + 4 files changed, 17 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 109fd78..a1baeee 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -798,6 +798,7 @@ const struct regmap_config adau1761_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1761_reg_defaults), .readable_reg = adau1761_readable_register, .volatile_reg = adau17x1_volatile_register, + .precious_reg = adau17x1_precious_register, .cache_type = REGCACHE_RBTREE, }; EXPORT_SYMBOL_GPL(adau1761_regmap_config); diff --git a/sound/soc/codecs/adau1781.c b/sound/soc/codecs/adau1781.c index 17966b6..35581f4 100644 --- a/sound/soc/codecs/adau1781.c +++ b/sound/soc/codecs/adau1781.c @@ -472,6 +472,7 @@ const struct regmap_config adau1781_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau1781_reg_defaults), .readable_reg = adau1781_readable_register, .volatile_reg = adau17x1_volatile_register, + .precious_reg = adau17x1_precious_register, .cache_type = REGCACHE_RBTREE, }; EXPORT_SYMBOL_GPL(adau1781_regmap_config); diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index d49bc7e..fa2e690 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -707,8 +707,22 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(adau17x1_set_micbias_voltage); +bool adau17x1_precious_register(struct device *dev, unsigned int reg) +{ + /* SigmaDSP parameter memory */ + if (reg < 0x400) + return true; + + return false; +} +EXPORT_SYMBOL_GPL(adau17x1_precious_register); + bool adau17x1_readable_register(struct device *dev, unsigned int reg) { + /* SigmaDSP parameter memory */ + if (reg < 0x400) + return true; + switch (reg) { case ADAU17X1_CLOCK_CONTROL: case ADAU17X1_PLL_CONTROL: diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 6861aa3..e13583e 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -56,6 +56,7 @@ int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); bool adau17x1_volatile_register(struct device *dev, unsigned int reg); +bool adau17x1_precious_register(struct device *dev, unsigned int reg); int adau17x1_resume(struct snd_soc_codec *codec); extern const struct snd_soc_dai_ops adau17x1_dai_ops; -- 1.8.0

2 2

[alsa-devel] [PATCH 1/3] ASoC: max98088: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/max98088.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index bb892b3..805b3f8 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -875,7 +875,7 @@ static const struct snd_kcontrol_new max98088_right_ADC_mixer_controls[] = { static int max98088_mic_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); switch (event) { @@ -905,7 +905,7 @@ static int max98088_mic_event(struct snd_soc_dapm_widget *w, static int max98088_line_pga(struct snd_soc_dapm_widget *w, int event, int line, u8 channel) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); u8 *state; -- 1.8.0

2 5

[alsa-devel] [PATCH] ASoC: sirf-audio-codec: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. While we are at it also replace dev_get_drvdata() with snd_soc_codec_get_drvdata(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/sirf-audio-codec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 06ba492..07eea20 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -120,7 +120,8 @@ static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, { #define ATLAS6_CODEC_ENABLE_BITS (1 << 29) #define ATLAS6_CODEC_RESET_BITS (1 << 28) - struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: enable_and_reset_codec(sirf_audio_codec->regmap, @@ -142,7 +143,8 @@ static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, { #define PRIMA2_CODEC_ENABLE_BITS (1 << 27) #define PRIMA2_CODEC_RESET_BITS (1 << 26) - struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_POST_PMU: enable_and_reset_codec(sirf_audio_codec->regmap, -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Fix the issue that the private value cannot be accessed
by Oder Chiou 21 Nov '14

21 Nov '14

Fix the issue that the private value cannot be accessed. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 37 ++++++++++++++++++++++++++++++------- sound/soc/codecs/rt5677.h | 2 +- 2 files changed, 31 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f2211f1..133010d 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4287,6 +4287,7 @@ static int rt5677_probe(struct snd_soc_codec *codec) } mutex_init(&rt5677->dsp_cmd_lock); + mutex_init(&rt5677->dsp_pri_lock); return 0; } @@ -4344,10 +4345,19 @@ static int rt5677_read(void *context, unsigned int reg, unsigned int *val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_read(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_read(rt5677, RT5677_PRIV_DATA, val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_read(rt5677, reg, val); + } + } else { regmap_read(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4357,10 +4367,20 @@ static int rt5677_write(void *context, unsigned int reg, unsigned int val) struct i2c_client *client = context; struct rt5677_priv *rt5677 = i2c_get_clientdata(client); - if (rt5677->is_dsp_mode) - rt5677_dsp_mode_i2c_write(rt5677, reg, val); - else + if (rt5677->is_dsp_mode) { + if (reg > 0xff) { + mutex_lock(&rt5677->dsp_pri_lock); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_INDEX, + reg & 0xff); + rt5677_dsp_mode_i2c_write(rt5677, RT5677_PRIV_DATA, + val); + mutex_unlock(&rt5677->dsp_pri_lock); + } else { + rt5677_dsp_mode_i2c_write(rt5677, reg, val); + } + } else { regmap_write(rt5677->regmap_physical, reg, val); + } return 0; } @@ -4495,10 +4515,13 @@ static const struct regmap_config rt5677_regmap_physical = { .reg_bits = 8, .val_bits = 16, - .max_register = RT5677_VENDOR_ID2 + 1, + .max_register = RT5677_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5677_ranges) * + RT5677_PR_SPACING), .readable_reg = rt5677_readable_register, .cache_type = REGCACHE_NONE, + .ranges = rt5677_ranges, + .num_ranges = ARRAY_SIZE(rt5677_ranges), }; static const struct regmap_config rt5677_regmap = { diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index a02f64c..dbd9ffd 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1670,7 +1670,7 @@ struct rt5677_priv { struct rt5677_platform_data pdata; struct regmap *regmap, *regmap_physical; const struct firmware *fw1, *fw2; - struct mutex dsp_cmd_lock; + struct mutex dsp_cmd_lock, dsp_pri_lock; int sysclk; int sysclk_src; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: samsung: ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk()
by Padmavathi Venna 21 Nov '14

21 Nov '14

In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register in i2s_set_sysclk. It's due to an incorrect mask used for the AND operation which is introduced in commit a5a56871f804edac93a53b5e871c0e9818fb9033 (ASoC: samsung: add support for exynos7 I2S controller) and also adds the missing break statement. Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0df6547..e1ace52 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; @@ -551,10 +551,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } if (clk_id == 0) - mod &= 0 << i2s_regs->rclksrc_off; + mod &= ~(1 << i2s_regs->rclksrc_off); else mod |= 1 << i2s_regs->rclksrc_off; + break; default: dev_err(&i2s->pdev->dev, "We don't serve that!\n"); return -EINVAL; -- 1.7.4.4

3 2

[alsa-devel] [PATCH 1/6] ASoC: sigmadsp: Refuse to load firmware files with a non-supported version
by Lars-Peter Clausen 21 Nov '14

21 Nov '14

Make sure to check the version field of the firmware header to make sure to not accidentally try to parse a firmware file with a different layout. Trying to do so can result in loading invalid firmware code to the device. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/sigmadsp.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index f2de7e0..81a38dd 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -159,6 +159,13 @@ int _process_sigma_firmware(struct device *dev, goto done; } + if (ssfw_head->version != 1) { + dev_err(dev, + "Failed to load firmware: Invalid version %d. Supported firmware versions: 1\n", + ssfw_head->version); + goto done; + } + crc = crc32(0, fw->data + sizeof(*ssfw_head), fw->size - sizeof(*ssfw_head)); pr_debug("%s: crc=%x\n", __func__, crc); -- 1.8.0

2 8

[alsa-devel] [PATCH] alsa-kernel: Add big-endian DSD sample formats and fix XMOS DSD sample format
by Jussi Laako 21 Nov '14

21 Nov '14

This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ sound/core/pcm_misc.c | 8 ++++++++ sound/usb/quirks.c | 4 ++-- 5 files changed, 17 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 40289ec..1e7f74a 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -184,6 +184,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_DSD_U8 _SNDRV_PCM_FMTBIT(DSD_U8) #define SNDRV_PCM_FMTBIT_DSD_U16_LE _SNDRV_PCM_FMTBIT(DSD_U16_LE) #define SNDRV_PCM_FMTBIT_DSD_U32_LE _SNDRV_PCM_FMTBIT(DSD_U32_LE) +#define SNDRV_PCM_FMTBIT_DSD_U16_BE _SNDRV_PCM_FMTBIT(DSD_U16_BE) +#define SNDRV_PCM_FMTBIT_DSD_U32_BE _SNDRV_PCM_FMTBIT(DSD_U32_BE) #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 6ee5867..941d32f 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -220,7 +220,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ #define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE +#define SNDRV_PCM_FORMAT_DSD_U16_BE ((__force snd_pcm_format_t) 51) /* DSD, 2-byte samples DSD (x16), big endian */ +#define SNDRV_PCM_FORMAT_DSD_U32_BE ((__force snd_pcm_format_t) 52) /* DSD, 4-byte samples DSD (x32), big endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_BE #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 8f624b7..cfc56c8 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -216,6 +216,8 @@ static char *snd_pcm_format_names[] = { FORMAT(DSD_U8), FORMAT(DSD_U16_LE), FORMAT(DSD_U32_LE), + FORMAT(DSD_U16_BE), + FORMAT(DSD_U32_BE), }; /** diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index ae7a0fe..ebe8444 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -152,6 +152,14 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 32, .phys = 32, .le = 1, .signd = 0, .silence = { 0x69, 0x69, 0x69, 0x69 }, }, + [SNDRV_PCM_FORMAT_DSD_U16_BE] = { + .width = 16, .phys = 16, .le = 0, .signd = 0, + .silence = { 0x69, 0x69 }, + }, + [SNDRV_PCM_FORMAT_DSD_U32_BE] = { + .width = 32, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x69, 0x69, 0x69, 0x69 }, + }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2c1018e..bc9ebb6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1202,12 +1202,12 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x3008): if (fp->altsetting == 2) - return SNDRV_PCM_FMTBIT_DSD_U32_LE; + return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2009): if (fp->altsetting == 3) - return SNDRV_PCM_FMTBIT_DSD_U32_LE; + return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; default: break; -- 1.9.1

2 3

[alsa-devel] [PATCH] ASoC: samsung: In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register in i2s_set_sysclk. It's due to an incorrect mask used for the AND operation which is introduced in commit a5a56871f804edac93a53b5e871c0e9818fb9033 (ASoC: samsung: add support for exynos7 I2S controller) and also adds the missing break statement.
by Padmavathi Venna 21 Nov '14

21 Nov '14

Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0df6547..e1ace52 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; @@ -551,10 +551,11 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, } if (clk_id == 0) - mod &= 0 << i2s_regs->rclksrc_off; + mod &= ~(1 << i2s_regs->rclksrc_off); else mod |= 1 << i2s_regs->rclksrc_off; + break; default: dev_err(&i2s->pdev->dev, "We don't serve that!\n"); return -EINVAL; -- 1.7.4.4

4 3

Re: [alsa-devel] [PATCH 0/1] Add big-endian DSD sample formats and fix XMOS DSD sample format
by Takashi Iwai 21 Nov '14

21 Nov '14

At Fri, 21 Nov 2014 12:00:26 +0200, Jussi Laako wrote: > > Resend, previous had non matching subject for cover letter vs patch. > > This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds > DSD_U16_BE and DSD_U32_BE sample formats. > > Jussi Laako (1): > Add big-endian DSD sample formats and fix XMOS DSD sample format The changes there look good (supposing you tested with the real device). Since DSD_U32 was added in 3.18-rc1, I think we can put this as a fix for 3.18. Could you submit the patch for alsa-lib as well? thanks, Takashi

2 2

Re: [alsa-devel] [PATCH 0/1] Add big-endian DSD sample formats and fix XMOS DSD sample format
by Jurgen Kramer 21 Nov '14

21 Nov '14

On Fri, 2014-11-21 at 12:00 +0200, Jussi Laako wrote: > Resend, previous had non matching subject for cover letter vs patch. > > This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds > DSD_U16_BE and DSD_U32_BE sample formats. > > Jussi Laako (1): > Add big-endian DSD sample formats and fix XMOS DSD sample format > > include/sound/pcm.h | 2 ++ > include/uapi/sound/asound.h | 4 +++- > sound/core/pcm.c | 2 ++ > sound/core/pcm_misc.c | 8 ++++++++ > sound/usb/quirks.c | 4 ++-- > 5 files changed, 17 insertions(+), 3 deletions(-) > Acked-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> I have tested the complete chain (added DSD_BE to kernel, alsa-lib, alsa-utils) with the change to BE. Everything still works. Jurgen

1 0

[alsa-devel] HP mute led for ALC286
by Kailang 21 Nov '14

21 Nov '14

Hi Takashi, I had sent the discussion mail about the DMI string for HP. But no body response. So, I just send the patch to fix this issue. BR, Kailang

2 5

[alsa-devel] [PATCH] ALSA: hda - One more HP machine needs to change mute led quirk
by Hui Wang 21 Nov '14

21 Nov '14

The machine originally use the quirk ALC269_FIXUP_HP_GPIO_MIC1_LED, but the LED doesn't work at all. After this change, the machine will change to use ALC269_FIXUP_HP_MUTE_LED_MIC1 through pin_fixup_tbl[], and the LED works well. BugLink: https://bugs.launchpad.net/bugs/1389497 Tested-by: TieFu Chen <tienfu.chen(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd08d8a..2ac9e23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4831,7 +4831,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), - SND_PCI_QUIRK(0x103c, 0x2246, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), -- 1.9.1

2 1

[alsa-devel] [PATCH] pcm: add new 32-bit DSD sample format
by Jurgen Kramer 21 Nov '14

21 Nov '14

Add the new DSD_U32_LE sample format to alsa-lib. NB include/pcm.h and include/sound/asound.h are updated so a new sync with the kernel headers is not needed Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- include/pcm.h | 4 +++- include/sound/asound.h | 3 ++- src/pcm/pcm.c | 2 ++ src/pcm/pcm_misc.c | 6 ++++++ 4 files changed, 13 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 11e9f0d..db88ad5 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -211,7 +211,9 @@ typedef enum _snd_pcm_format { SND_PCM_FORMAT_DSD_U8, /* Direct Stream Digital (DSD) in 2-byte samples (x16) */ SND_PCM_FORMAT_DSD_U16_LE, - SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U16_LE, + /* Direct Stream Digital (DSD) in 4-byte samples (x32) */ + SND_PCM_FORMAT_DSD_U32_LE, + SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U32_LE, #if __BYTE_ORDER == __LITTLE_ENDIAN /** Signed 16 bit CPU endian */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 32168f7..6ee5867 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -219,7 +219,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE +#define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1399a5b..2e24338 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1565,6 +1565,7 @@ static const char *const snd_pcm_format_names[] = { FORMAT(G723_40_1B), FORMAT(DSD_U8), FORMAT(DSD_U16_LE), + FORMAT(DSD_U32_LE), }; static const char *const snd_pcm_format_aliases[SND_PCM_FORMAT_LAST+1] = { @@ -1624,6 +1625,7 @@ static const char *const snd_pcm_format_descriptions[] = { FORMATD(G723_40_1B, "G.723 (ADPCM) 40 kbit/s, 1 sample in 1 byte"), FORMATD(DSD_U8, "Direct Stream Digital, 1-byte (x8), oldest bit in MSB"), FORMATD(DSD_U16_LE, "Direct Stream Digital, 2-byte (x16), little endian, oldest bits in MSB"), + FORMATD(DSD_U32_LE, "Direct Stream Digital, 4-byte (x32), little endian, oldest bits in MSB"), }; static const char *const snd_pcm_type_names[] = { diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 46fc771..9272179 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -64,6 +64,7 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -154,6 +155,7 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -232,6 +234,7 @@ int snd_pcm_format_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -292,6 +295,7 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_FLOAT_BE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -348,6 +352,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return samples * 4; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -394,6 +399,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) return 0x8080808080808080ULL; case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0x6969696969696969ULL; #ifdef SNDRV_LITTLE_ENDIAN case SNDRV_PCM_FORMAT_U16_LE: -- 1.9.3

3 7

[alsa-devel] [PATCH 0/1] Add big-endian DSD sample formats and fix XMOS DSD sample format
by Jussi Laako 21 Nov '14

21 Nov '14

Resend, previous had non matching subject for cover letter vs patch. This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Jussi Laako (1): Add big-endian DSD sample formats and fix XMOS DSD sample format include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ sound/core/pcm_misc.c | 8 ++++++++ sound/usb/quirks.c | 4 ++-- 5 files changed, 17 insertions(+), 3 deletions(-) -- 1.9.1

1 1

[alsa-devel] [PATCH 0/1] Fix DSD sample format endianess and add big-endian format
by Jussi Laako 21 Nov '14

21 Nov '14

This patch adds big-endian DSD sample format and fixes return value of DSD formats for snd_pcm_format_little_endian(). Jussi Laako (1): Fix DSD sample format endianess and add big-endian format include/pcm.h | 6 +++++- include/sound/asound.h | 4 +++- src/pcm/pcm.c | 4 ++++ src/pcm/pcm_misc.c | 17 ++++++++++++++--- 4 files changed, 26 insertions(+), 5 deletions(-) -- 1.9.1

1 1

[alsa-devel] [PATCH 0/1] Fix XMOS DSD suppport and add big-endian DSD sample formats
by Jussi Laako 21 Nov '14

21 Nov '14

This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Jussi Laako (1): Add big-endian DSD sample formats and fix XMOS DSD sample format include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ sound/core/pcm_misc.c | 8 ++++++++ sound/usb/quirks.c | 4 ++-- 5 files changed, 17 insertions(+), 3 deletions(-) -- 1.9.1

1 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: cleanup runtime_pm initialization
by Vinod Koul 21 Nov '14

21 Nov '14

For ACPI we missed to pm_runtime_enable() call which is required to tell PM core that runtime on this device is enabled now. Since this is common to both PCI and APCI move it out. Also for ACPI we do not require pm_runtime_allow() call, so remove that Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/sst/sst.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b2b5604..9e68a7c 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -378,13 +378,13 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) * initially active. So change the state to active before * enabling the pm */ - if (acpi_disabled) { + pm_runtime_enable(ctx->dev); + + if (acpi_disabled) pm_runtime_set_active(ctx->dev); - pm_runtime_enable(ctx->dev); - } else { - pm_runtime_allow(ctx->dev); + else pm_runtime_put_noidle(ctx->dev); - } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); -- 1.7.0.4

3 7

[alsa-devel] [PATCH] ALSA: usb-audio: Use snd_usb_ctl_msg() for Native Instruments quirk
by Takashi Iwai 20 Nov '14

20 Nov '14

snd_nativeinstruments_control_get() uses a stack as a buffer for usb_control_msg(), but it's basically not allowed. Replace the call with a safer helper, snd_usb_ctl_msg(), instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 03405b93b438..8b55c0667f60 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -702,10 +702,10 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, if (mixer->chip->shutdown) ret = -ENODEV; else - ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0, wIndex, - &tmp, sizeof(tmp), 1000); + &tmp, sizeof(tmp)); up_read(&mixer->chip->shutdown_rwsem); if (ret < 0) { -- 2.1.3

1 0

[alsa-devel] [PATCH RFC 00/10] Add proper resume support to USB-audio quirks
by Takashi Iwai 20 Nov '14

20 Nov '14

Hi, here is a patch series to add proper resume support to USB-audio quirks. So far, the control elements created directly by quirk functions aren't resumed restore_mixer_value(). This patch series try to address it. [PATCH RFC 01/10] ALSA: usb-audio: Refactor ignore_ctl_error checks [PATCH RFC 02/10] ALSA: usb-audio: Allow quirks to handle own resume [PATCH RFC 03/10] ALSA: usb-audio: Add audigy2nx resume support [PATCH RFC 04/10] ALSA: usb-audio: Add Emu0204 channel switch resume [PATCH RFC 05/10] ALSA: usb-audio: Add Xonar U1 resume support [PATCH RFC 06/10] ALSA: usb-audio: Add Digidesign Mbox 1 resume [PATCH RFC 07/10] ALSA: usb-audio: Add resume support for Native [PATCH RFC 08/10] ALSA: usb-audio: Add resume support for FTU [PATCH RFC 09/10] ALSA: usb-audio: Add resume support for MicroII [PATCH RFC 10/10] ALSA: usb-audio: Add resume support for Scarlett I have only compile-tested this, so far. It'd be appreciated if anyone with such devices can test with real devices. thanks, Takashi

1 10

[alsa-devel] [PATCH] ASoC: samsung: Fix IISMOD setting in i2s_set_sysclk()
by Sylwester Nawrocki 20 Nov '14

20 Nov '14

In the i2s_set_sysclk() callback we are currently clearing all bits of the IISMOD register when clk_id is SAMSUNG_I2S_CDCLK and dir is SND_SOC_CLOCK_OUT. It's due to an incorrect mask used for the AND operation and breaks the I2S interface operation when we attempt to use the CDCLK output clock. It fixes commit a5a56871f804edac93a53b5e871c0e9818fb9033 ("ASoC: samsung: add support for exynos7 I2S controller") and restores sound support for Odroid X2/U3. Cc: Padmavathi Venna <padma.v(a)samsung.com> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 947352d..8db8c66 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -494,7 +494,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) mod |= 1 << i2s_regs->cdclkcon_off; else - mod &= 0 << i2s_regs->cdclkcon_off; + mod &= ~(1 << i2s_regs->cdclkcon_off); i2s->rfs = rfs; break; -- 1.7.9.5

2 3

[alsa-devel] [PATCH v2 1/2] ASoC: wm8960: Add device tree support
by Zidan Wang 20 Nov '14

20 Nov '14

Document the device tree binding for the WM8960 codec, and modify the driver to extract the platform data from device tree, if present. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 27 ++++++++++++++ sound/soc/codecs/wm8960.c | 41 ++++++++++++++++------ 2 files changed, 58 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt new file mode 100644 index 0000000..1a9ad07 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -0,0 +1,27 @@ +WM8960 audio CODEC + +This device supports I2C only. + +Required properties: + + - compatible : "wlf,wm8960" + + - reg : the I2C address of the device. + +Optional properties: + - shared_lrclk: This is a boolean property. If present, the LRCM bit of + R24 (Additional control 2 ) gets set, indicating that ADCLRC and DACLRC + will be disabled only when ADC (Left and Right) and DAC (Left and Right) + are disabled. + + - capless: This is a boolean property. If present, OUT3 will be enabled + and disabled together with HP_L and HP_R in response to jack detect events. + +Example: + +codec: wm8960@1a { + compatible = "wlf,wm8960"; + reg = <0x1a>; + + shared_lrclk; +}; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..e5d89a2 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = { static int wm8960_add_widgets(struct snd_soc_codec *codec) { - struct wm8960_data *pdata = codec->dev->platform_data; struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + struct wm8960_data *pdata = &wm8960->pdata; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dapm_widget *w; @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -1029,6 +1026,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "shared_lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1045,6 +1054,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); + if (pdata && pdata->shared_lrclk) { ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 0x4, 0x4); @@ -1075,10 +1089,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); +static const struct of_device_id wm8960_of_match[] = { + { .compatible = "wlf,wm8960", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8960_of_match); + static struct i2c_driver wm8960_i2c_driver = { .driver = { .name = "wm8960", .owner = THIS_MODULE, + .of_match_table = wm8960_of_match, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

3 3

[alsa-devel] AKAI EIE PRO
by Martin Svensson 20 Nov '14

20 Nov '14

Dont know if its ok to ask a question here so please let me know if its not ok then i wont ask again ;) i just bought an external usb AKAI EIE PRO sound card, is there any future plans to implement it in Alsa? Sincerally Martin aka "ztealmax"

4 7

[alsa-devel] Baytrail init file for alsa-utils
by Bastien Nocera 20 Nov '14

20 Nov '14

Hey, I've done some work on the community generated "alsa state" file for Baytrail tablets, and shrunk it massively: https://github.com/AdamWill/baytrail-m/blob/master/alsa/t100_B.state Are there any tools, or hints as to how I might convert that to an "init" file that alsactl could run to set the default state of the mixer? Or have any Intel folks already done that work? I'll have a few more sound-related questions for that tablet, for which I could do with a few hints as to where to direct the questions/bug reports: - no jack sensing support (driver problem?) - left/right headphones output is inverted (another driver bug?) - HDMI audio is not detected (Intel video driver bug?) Cheers

3 5

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Minor coding style and typo fix
by Oder Chiou 20 Nov '14

20 Nov '14

Minor coding style and typo fix Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677-spi.h | 4 ++-- sound/soc/codecs/rt5677.c | 14 +++++++------- 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index 7528bfd..ec41b2b 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -9,8 +9,8 @@ * published by the Free Software Foundation. */ -#ifndef __RT5671_SPI_H__ -#define __RT5671_SPI_H__ +#ifndef __RT5677_SPI_H__ +#define __RT5677_SPI_H__ #define RT5677_SPI_BUF_LEN 240 #define RT5677_SPI_CMD_BURST_WRITE 0x05 diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ca264f8..0d24dc4 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -1353,7 +1353,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_ib45_bypass_src_mux = SOC_DAPM_ENUM("IB45 Bypass Source", rt5677_ib45_bypass_src_enum); -/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ +/* Stereo ADC Source 2 */ /* MX-27 MX26 MX25 [11:10] */ static const char * const rt5677_stereo_adc2_src[] = { "DD MIX1", "DMIC", "Stereo DAC MIX" }; @@ -1438,7 +1438,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_sto2_adc_lr_mux = SOC_DAPM_ENUM("Stereo2 ADC LR Source", rt5677_stereo2_adc_lr_enum); -/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ +/* Stereo1 ADC Source 1 */ /* MX-27 MX26 MX25 [13:12] */ static const char * const rt5677_stereo_adc1_src[] = { "DD MIX1", "ADC1/2", "Stereo DAC MIX" }; @@ -1710,7 +1710,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_pdm2_r_mux = SOC_DAPM_ENUM("PDM2 Source", rt5677_pdm2_r_enum); -/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0]*/ +/* TDM IF1/2 SLB ADC1 Data Selection */ /* MX-3C MX-41 [5:4] MX-08 [1:0] */ static const char * const rt5677_if12_adc1_src[] = { "STO1 ADC MIX", "OB01", "VAD ADC" }; @@ -1788,7 +1788,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_slb_adc3_mux = SOC_DAPM_ENUM("SLB ADC3 Source", rt5677_slb_adc3_enum); -/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ +/* TDM IF1/2 SLB ADC4 Data Selection */ /* MX-3C MX-41 [11:10] MX-08 [7:6] */ static const char * const rt5677_if12_adc4_src[] = { "STO4 ADC MIX", "OB67", "OB01" }; @@ -1814,7 +1814,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_slb_adc4_mux = SOC_DAPM_ENUM("SLB ADC4 Source", rt5677_slb_adc4_enum); -/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4]*/ +/* Interface3/4 ADC Data Input */ /* MX-2F [3:0] MX-30 [7:4] */ static const char * const rt5677_if34_adc_src[] = { "STO1 ADC MIX", "STO2 ADC MIX", "STO3 ADC MIX", "STO4 ADC MIX", "MONO ADC MIX", "OB01", "OB23", "VAD ADC" @@ -1895,7 +1895,7 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5677_if2_adc4_swap_mux = SOC_DAPM_ENUM("IF2 ADC4 Swap Source", rt5677_if2_adc4_swap_enum); -/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ +/* TDM IF1 ADC Data Selection */ /* MX-3C [2:0] */ static const char * const rt5677_if1_adc_tdm_swap_src[] = { "1/2/3/4", "2/1/3/4", "2/3/1/4", "4/1/2/3", "1/3/2/4", "1/4/2/3", "3/1/2/4", "3/4/1/2" @@ -2442,7 +2442,7 @@ static const struct snd_soc_dapm_widget rt5677_dapm_widgets[] = { rt5677_ob_7_mix, ARRAY_SIZE(rt5677_ob_7_mix)), /* Output Side */ - /* DAC mixer before sound effect */ + /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, rt5677_dac_l_mix, ARRAY_SIZE(rt5677_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, -- 1.8.1.1.439.g50a6b54

3 4

[alsa-devel] [PATCH] ASoC: wm8960: Add shared_lrclk and capless to device tree
by Zidan Wang 19 Nov '14

19 Nov '14

wm8960 codec has a wm8960_data struct which has two fields shared_lrclk and capless. The wm8960_data is get from platform_data and it is reasonable to set it from device tree when platform_data is null. And when shared_lrclk is set, LRCM will be enabled. But the following software reset in wm8960_probe will reset it to the default state. So LRCM operation should after software reset. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 49 ++++++++++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 4dc4e85..1d56c90 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -125,6 +125,7 @@ struct wm8960_priv { struct snd_soc_dapm_widget *out3; bool deemph; int playback_fs; + struct wm8960_data pdata; }; #define wm8960_reset(c) snd_soc_write(c, WM8960_RESET, 0) @@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec) static int wm8960_probe(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - struct wm8960_data *pdata = dev_get_platdata(codec->dev); + struct wm8960_data *pdata = &wm8960->pdata; int ret; - wm8960->set_bias_level = wm8960_set_bias_level_out3; - - if (!pdata) { - dev_warn(codec->dev, "No platform data supplied\n"); - } else { - if (pdata->capless) - wm8960->set_bias_level = wm8960_set_bias_level_capless; - } + if (pdata->capless) + wm8960->set_bias_level = wm8960_set_bias_level_capless; + else + wm8960->set_bias_level = wm8960_set_bias_level_out3; ret = wm8960_reset(codec); if (ret < 0) { @@ -979,6 +976,14 @@ static int wm8960_probe(struct snd_soc_codec *codec) return ret; } + if (pdata->shared_lrclk) { + ret = snd_soc_update_bits(codec, WM8960_ADDCTL2, 0x4, 0x4); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable LRCM: %d\n", ret); + return ret; + } + } + wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ @@ -1029,6 +1034,18 @@ static const struct regmap_config wm8960_regmap = { .volatile_reg = wm8960_volatile, }; +static void wm8960_set_pdata_from_of(struct i2c_client *i2c, + struct wm8960_data *pdata) +{ + const struct device_node *np = i2c->dev.of_node; + + if (of_property_read_bool(np, "capless")) + pdata->capless = true; + + if (of_property_read_bool(np, "shared_lrclk")) + pdata->shared_lrclk = true; +} + static int wm8960_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1038,6 +1055,7 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, wm8960 = devm_kzalloc(&i2c->dev, sizeof(struct wm8960_priv), GFP_KERNEL); + if (wm8960 == NULL) return -ENOMEM; @@ -1045,15 +1063,10 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); - if (pdata && pdata->shared_lrclk) { - ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, - 0x4, 0x4); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to enable LRCM: %d\n", - ret); - return ret; - } - } + if (pdata) + memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data)); + else if (i2c->dev.of_node) + wm8960_set_pdata_from_of(i2c, &wm8960->pdata); i2c_set_clientdata(i2c, wm8960); -- 1.9.1

4 3

[alsa-devel] [PATCH 0/3] Driver for Cirrus Logic CS47L24 codec
by Richard Fitzgerald 19 Nov '14

19 Nov '14

This set of patches add support for the Cirrus Logic CS47L24 codec. The patches depend on two open patches from Charles Keepax which must be applied first: [PATCH] mfd: wm5110: Add missing registers for AIF2 channels 3-6 [PATCH] mfd: arizona: Document HP_CTRL_1L and HP_CTRL_1R registers Richard Fitzgerald (3): mfd: arizona: Add support for CS47L24 gpio: arizona: Add support for CS47L24 ASoC: arizona: Add initial CS47L24 codec driver drivers/gpio/gpio-arizona.c | 5 + drivers/mfd/Kconfig | 6 + drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 91 ++- drivers/mfd/arizona-irq.c | 42 +- drivers/mfd/arizona-spi.c | 9 + drivers/mfd/arizona.h | 5 + drivers/mfd/cs47l24-tables.c | 1632 ++++++++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/core.h | 4 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 5 + sound/soc/codecs/cs47l24.c | 1133 ++++++++++++++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 14 files changed, 2941 insertions(+), 27 deletions(-) create mode 100644 drivers/mfd/cs47l24-tables.c create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h -- 1.7.2.5

5 22

[alsa-devel] [PATCH 0/4] ASoC: AC'97 regmap support and conversion
by Lars-Peter Clausen 19 Nov '14

19 Nov '14

This series contains the final pieces for the AC'97 conversion to regmap. The first patch adds a regmap backend for AC'97 devices that will use the AC'97 bus of the device to perform reads and writes. The second patch adds a bit of plumbing to the ASoC framework that it makes it possible to assign a regmap instance to ASoC CODEC (or component) after it has been registered. This is necessary since for AC'97 drivers we allocate the AC'97 device in the CODEC drivers probe callback. For now the series only converts the ad1980 driver, others might follow later. - Lars Lars-Peter Clausen (3): ASoC: Add helper functions for deferred regmap setup ASoC: ad1980: Convert to regmap ASoC: ad1980: Remove ac97_read/ac97_write wrappers Mark Brown (1): regmap: ac97: Add generic AC'97 callbacks drivers/base/regmap/Kconfig | 5 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-ac97.c | 114 +++++++++++++++++++++++++ include/linux/regmap.h | 7 ++ include/sound/soc.h | 35 ++++++++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/ad1980.c | 173 +++++++++++++++++++++----------------- sound/soc/soc-core.c | 58 ++++++++++--- 8 files changed, 306 insertions(+), 88 deletions(-) create mode 100644 drivers/base/regmap/regmap-ac97.c -- 1.8.0

2 9

[alsa-devel] [PATCH 0/2] ASoC: rockchip: i2s: add support for grabbing output clock to codec
by Jianqun Xu 19 Nov '14

19 Nov '14

Reference to Chapter 3 Clock & Reset Unit (CRU) of RK3288 TRM, i2s0_clkout has two source clock, XIN24M_DIV2 for 12M and clk_i2s0 for more kinds of frequencies, I2S0(i2s_clk) has only single source clock clk_i2s0. i2s0_clkout: generate to output to outside of chip, generally for codec I2S0(i2s_clk): generate for i2s controller inside of chip Jianqun Xu (2): ASoC: rockchip-i2s: dt: add i2s_clkout to list of clocks ASoC: rockchip: i2s: add support for grabbing output clock to codec Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 5 +++-- sound/soc/rockchip/rockchip_i2s.c | 11 +++++++++++ 2 files changed, 14 insertions(+), 2 deletions(-) -- 1.9.1

6 9

[alsa-devel] [PATCH 1/3] ASoC: rt5677: Align the reg_default table with tab character
by Oder Chiou 19 Nov '14

19 Nov '14

Align the reg_default table with tab character Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 9ae2e84..b2d88bb 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,13 +55,13 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_ASRC_12, 0x0018}, - {RT5677_PR_BASE + 0x3d, 0x364d}, - {RT5677_PR_BASE + 0x17, 0x4fc0}, - {RT5677_PR_BASE + 0x13, 0x0312}, - {RT5677_PR_BASE + 0x1e, 0x0000}, - {RT5677_PR_BASE + 0x12, 0x0eaa}, - {RT5677_PR_BASE + 0x14, 0x018a}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_PR_BASE + 0x17, 0x4fc0}, + {RT5677_PR_BASE + 0x13, 0x0312}, + {RT5677_PR_BASE + 0x1e, 0x0000}, + {RT5677_PR_BASE + 0x12, 0x0eaa}, + {RT5677_PR_BASE + 0x14, 0x018a}, }; #define RT5677_INIT_REG_LEN ARRAY_SIZE(init_list) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v2 0/6] media token resource framework
by Shuah Khan 18 Nov '14

18 Nov '14

Add media token device resource framework to allow sharing resources such as tuner, dma, audio etc. across media drivers and non-media sound drivers that control media hardware. The Media token resource is created at the main struct device that is common to all drivers that claim various pieces of the main media device, which allows them to find the resource using the main struct device. As an example, digital, analog, and snd-usb-audio drivers can use the media token resource API using the main struct device for the interface the media device is attached to. This patch series consists of media token resource framework and changes to use it in dvb-core, v4l2-core, au0828 driver, and snd-usb-audio driver. With these changes dvb and v4l2 can share the tuner without disrupting each other. Used tvtime, xawtv, kaffeine, and vlc, vlc audio capture option, arecord/aplay during development to identify v4l2 vb2 and vb1 ioctls and file operations that disrupt the digital stream and would require changes to check tuner ownership prior to changing the tuner configuration. vb2 changes are made in the v4l2-core and vb1 changes are made in the au0828 driver to encourage porting drivers to vb2 to advantage of the new media token resource framework with changes in the core. In this patch v2 series, fixed problems identified in the patch v1 series. Important ones are changing snd-usb-audio to use media tokens, holding tuner lock in VIDIOC_ENUMINPUT, and VIDIOC_QUERYSTD. Shuah Khan (6): media: add media token device resource framework media: v4l2-core changes to use media token api media: au0828-video changes to use media token api media: dvb-core changes to use media token api sound/usb: pcm changes to use media token api media: au0828-core changes to create and destroy media MAINTAINERS | 2 + drivers/media/dvb-core/dvb_frontend.c | 14 +- drivers/media/usb/au0828/au0828-core.c | 23 +++ drivers/media/usb/au0828/au0828-video.c | 42 +++++- drivers/media/v4l2-core/v4l2-fh.c | 7 + drivers/media/v4l2-core/v4l2-ioctl.c | 61 ++++++++ include/linux/media_tknres.h | 50 +++++++ lib/Makefile | 2 + lib/media_tknres.c | 237 +++++++++++++++++++++++++++++++ sound/usb/pcm.c | 9 ++ 10 files changed, 445 insertions(+), 2 deletions(-) create mode 100644 include/linux/media_tknres.h create mode 100644 lib/media_tknres.c -- 1.7.10.4

11 56

[alsa-devel] [PATCH] ASoC: wm_adsp: Move core_ena to be co-located with start bit
by Charles Keepax 18 Nov '14

18 Nov '14

Many firmwares do not wait for the start bit before they begin processing audio, whilst this is a bug on the firmware side there are too many such firmwares in the wild to ignore the situation. This patch moves the core enable to happen at same time as the start, the firmware looses the ability to overlap its own startup with the audio path bring up but we ensure that all firmwares behave. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index e54d7b5..720d6e8 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1593,13 +1593,6 @@ static void wm_adsp2_boot_work(struct work_struct *work) if (ret != 0) goto err; - ret = regmap_update_bits_async(dsp->regmap, - dsp->base + ADSP2_CONTROL, - ADSP2_CORE_ENA, - ADSP2_CORE_ENA); - if (ret != 0) - goto err; - dsp->running = true; return; @@ -1649,8 +1642,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret = regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_START, - ADSP2_START); + ADSP2_CORE_ENA | ADSP2_START, + ADSP2_CORE_ENA | ADSP2_START); if (ret != 0) goto err; break; -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2] ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs
by Charles Keepax 18 Nov '14

18 Nov '14

From: JS Park <aitdark.park(a)samsung.com> If we fail to allocate memory for one of the memory regions whilst parsing the DSP firmware, we will leak the buffer holding the algorithm information. This patch ensures that this buffer is freed on all error paths. Signed-off-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 ++++++++++++++++++++---------- 1 files changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 08d8ec3..e54d7b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1053,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); @@ -1071,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); @@ -1101,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); @@ -1119,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); @@ -1137,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: fix kerneldoc errors
by Takashi Sakamoto 18 Nov '14

18 Nov '14

Complete missing parameters, correct wrong reference, and add an explaination about the differences between the latest specification and our implementation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp.h | 24 ++++++++++++++++++++++-- sound/firewire/cmp.c | 2 ++ 2 files changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index 4823c08..e6e8926 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -23,7 +23,7 @@ * corresponds to the end of event in the packet. Out of IEC 61883. * @CIP_WRONG_DBS: Only for in-stream. The value of dbs is wrong in in-packets. * The value of data_block_quadlets is used instead of reported value. - * @SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is + * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first * packet is not continuous from an initial value. @@ -43,7 +43,27 @@ enum cip_flags { }; /** - * enum cip_sfc - a stream's sample rate + * enum cip_sfc - supported Sampling Frequency Codes (SFCs) + * @CIP_SFC_32000: 32,000 data blocks + * @CIP_SFC_44100: 44,100 data blocks + * @CIP_SFC_48000: 48,000 data blocks + * @CIP_SFC_88200: 88,200 data blocks + * @CIP_SFC_96000: 96,000 data blocks + * @CIP_SFC_176400: 176,400 data blocks + * @CIP_SFC_192000: 192,000 data blocks + * @CIP_SFC_COUNT: the number of supported SFCs + * + * These values are used to show nominal Sampling Frequency Code in + * Format Dependent Field (FDF) of AMDTP packet header. In IEC 61883-6:2002, + * this code means the number of events per second. Actually the code + * represents the number of data blocks transferred per second in an AMDTP + * stream. + * + * In IEC 61883-6:2005, some extensions were added to support more types of + * data such as 'One Bit LInear Audio', therefore the meaning of SFC became + * different depending on the types. + * + * Currently our implementation is compatible with IEC 61883-6:2002. */ enum cip_sfc { CIP_SFC_32000 = 0, diff --git a/sound/firewire/cmp.c b/sound/firewire/cmp.c index ba8df5a..ae3bc19 100644 --- a/sound/firewire/cmp.c +++ b/sound/firewire/cmp.c @@ -114,6 +114,7 @@ static int pcr_modify(struct cmp_connection *c, * cmp_connection_init - initializes a connection manager * @c: the connection manager to initialize * @unit: a unit of the target device + * @direction: input or output * @pcr_index: the index of the iPCR/oPCR on the target device */ int cmp_connection_init(struct cmp_connection *c, @@ -154,6 +155,7 @@ EXPORT_SYMBOL(cmp_connection_init); /** * cmp_connection_check_used - check connection is already esablished or not * @c: the connection manager to be checked + * @used: the pointer to store the result of checking the connection */ int cmp_connection_check_used(struct cmp_connection *c, bool *used) { -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Fix memory leak in wm_adsp_setup_algs
by Charles Keepax 18 Nov '14

18 Nov '14

From: JS Park <aitdark.park(a)samsung.com> Signed-off-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 30 ++++++++++++++++++++---------- 1 files changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 08d8ec3..e54d7b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1053,8 +1053,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_DM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].dm); @@ -1071,8 +1073,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP1_ZM; region->alg = be32_to_cpu(adsp1_alg[i].alg.id); region->base = be32_to_cpu(adsp1_alg[i].zm); @@ -1101,8 +1105,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_alg[i].zm)); region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_XM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].xm); @@ -1119,8 +1125,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_YM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].ym); @@ -1137,8 +1145,10 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) } region = kzalloc(sizeof(*region), GFP_KERNEL); - if (!region) - return -ENOMEM; + if (!region) { + ret = -ENOMEM; + goto out; + } region->type = WMFW_ADSP2_ZM; region->alg = be32_to_cpu(adsp2_alg[i].alg.id); region->base = be32_to_cpu(adsp2_alg[i].zm); -- 1.7.2.5

2 1

[alsa-devel] [PATCH 0/6] Add support for the WM8280 and WM8281 codecs
by Richard Fitzgerald 18 Nov '14

18 Nov '14

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. Richard Fitzgerald (6): mfd: arizona: add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 10 ++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 14 ++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 32 insertions(+), 8 deletions(-) -- 1.7.2.5

5 13

[alsa-devel] [PATCH] ASoC: rt286: build warning of section mismatch
by Sudip Mukherjee 18 Nov '14

18 Nov '14

From: Sudip Mukherjee <sudip(a)vectorindia.org> while building we were getting the following build warning: Section mismatch in reference from the function rt286_i2c_probe() to the variable .init.data:force_combo_jack_table The function rt286_i2c_probe() references the variable __initdata force_combo_jack_table. This is often because rt286_i2c_probe lacks a __initdata annotation or the annotation of force_combo_jack_table is wrong. we were getting the warning as force_combo_jack_table was marked with __initdata Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2e818aa..2cd4fe4 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static const struct acpi_device_id rt286_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); -static struct dmi_system_id force_combo_jack_table[] __initdata = { +static struct dmi_system_id force_combo_jack_table[] = { { .ident = "Intel Wilson Beach", .matches = { -- 1.8.1.2

2 1

[alsa-devel] [PATCH 00/11] ASoC: Untangle AC'97 support form the core
by Lars-Peter Clausen 18 Nov '14

18 Nov '14

The AC'97 support has had a special role within the ASoC framework with it being (in contrast to other audio buses) tightly integrated into the core. This patch series creates a clean separation between the core and the AC'97 support making AC'97 from the point of view of the core just another bus. Some of the changes in this series are required for the upcoming conversion of the AC'97 drivers to regmap, the other changes just fit nicely with the idea of componentization and having a generic ASoC core. The series starts with moving AC'97 support to its own file and doing a few general cleanups. After that it drops support of two features which were mainly responsible for the tight integration but are seemingly unused. This is a) Delaying the initialization and registration of the AC'97 device until after the card has been registered. There doesn't seem to be a good reason to delay to registration, all the information we need to register the device is available at the time when the device is created. Furthermore this will be needed for regmap. b) Setting platform data for the AC'97 via the CPU DAI. This hasn't been used in the last 5 years and doesn't seem to be that useful in general. If we should ever have a need for those features again they can still be implemented on top of the current code while keeping the clean separation. But for the time being it is simpler to just drop unused code rather than restructuring it. Finally the series pushes the pointer to the AC'97 device from the snd_soc_codec struct out to the driver private structs. Depends on asoc/topic/ad1980, asoc/topic/wm9705 and asoc/topic/wm971x - Lars Lars-Peter Clausen (11): ASoC: mpc5200_psc_ac97: Remove unused on-stack snd_ac97 device ASoC: mpc5200_dma: Don't overwrite ac97 device private_data ASoC: Properly handle AC'97 device lifetime management ASoC: Move AC'97 support to its own file ASoC: ac97: Use static ac97_bus ASoC: ac97: Merge soc_ac97_dev_{un,}register()/soc_{un,}register_ac97_codec() ASoC: ac97: Drop support for setting platform data via the CPU DAI ASoC: ac97: Drop delayed device registration ASoC: Drop ac97_control initialization from CODEC driver DAIs ASoC: Rename snd_soc_dai_driver struct ac97_control field to bus_control ASoC: ac97: Push snd_ac97 pointer to the driver level include/sound/soc-dai.h | 4 +- include/sound/soc.h | 25 ++- sound/soc/Makefile | 4 + sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/cirrus/ep93xx-ac97.c | 2 +- sound/soc/codecs/ac97.c | 18 +- sound/soc/codecs/ad1980.c | 28 +-- sound/soc/codecs/stac9766.c | 40 +++-- sound/soc/codecs/wm9705.c | 32 ++-- sound/soc/codecs/wm9712.c | 33 ++-- sound/soc/codecs/wm9713.c | 32 ++-- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 3 - sound/soc/fsl/mpc5200_psc_ac97.c | 6 +- sound/soc/nuc900/nuc900-ac97.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 6 +- sound/soc/samsung/ac97.c | 4 +- sound/soc/sh/hac.c | 2 +- sound/soc/soc-ac97.c | 256 +++++++++++++++++++++++++++ sound/soc/soc-core.c | 372 ++------------------------------------- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- 25 files changed, 428 insertions(+), 455 deletions(-) create mode 100644 sound/soc/soc-ac97.c -- 1.8.0

3 13

[alsa-devel] [PATCH 1/2] ALSA: hda - move DELL_WMI_MIC_MUTE_LED to the tail in the quirk chain
by Hui Wang 18 Nov '14

18 Nov '14

We have one more Dell machine needs DELL_WMI_MIC_MUTE_LED quirk, but the machine uses alc293 instead of alc255. So if DELL_WMI_MIC_MUTE_LED still chain ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, the machine can't use this quirk. To change this situation, let the DELL_WMI_MIC_MUTE_LED to be a standalone quirk, and let other quirks chain it. After this change, this quirk can be chained to any existing quirks, and as a result, it is possible that this quirk is applied to a non-Dell machine or a Dell machine without mic mute led on it, but it is still safe since alc_fixup_dell_wmi() will return an error in these situations. And remove the quirk for machine with subsystem id 0x6010 and 0x601f, these two machines will fall back to the quirk ALC255_FIXUP_DELL1_MIC_NO_PRESENCE-->ALC255_FIXUP_HEADSET_MODE--> ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED through pin_fixup_tbl[]. BugLink: https://bugs.launchpad.net/bugs/1381856 Reported-and-Tested-by: Po-Hsu Lin <po-hsu.lin(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1723954..9da5798 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4709,6 +4709,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC255_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc255, + .chained = true, + .chain_id = ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED }, [ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC] = { .type = HDA_FIXUP_FUNC, @@ -4744,8 +4746,6 @@ static const struct hda_fixup alc269_fixups[] = { [ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_wmi, - .chained_before = true, - .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, [ALC282_FIXUP_ASPIRE_V5_PINS] = { .type = HDA_FIXUP_PINS, @@ -4783,10 +4783,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), - SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: hda: separate stream tag for playback and capture
by Vinod Koul 18 Nov '14

18 Nov '14

From: Rafal Redzimski <rafal.f.redzimski(a)intel.com> According to hda specification stream tag must be unique throughout the input streams group, however an output stream might use a stream tag which is already in use by an input stream. Newer Intel HW has the capability to provides a total of more than 15 stream DMA engines which with the current implementation causes an overflow on STRM field (and the whole register) which results in usage of reserved value 0 in the STRM field which confuses HDA controller. So make stream_tag assignment separate for input and output streams Signed-off-by: Rafal Redzimski <rafal.f.redzimski(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_controller.c | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 8337645..703ac6e 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1921,11 +1921,17 @@ int azx_mixer_create(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_mixer_create); +static bool is_input_stream(struct azx *chip, unsigned char index) +{ + return index < chip->playback_index_offset; +} /* initialize SD streams */ int azx_init_stream(struct azx *chip) { int i; + int in_stream_tag = 0; + int out_stream_tag = 0; /* initialize each stream (aka device) * assign the starting bdl address to each stream (device) @@ -1938,9 +1944,13 @@ int azx_init_stream(struct azx *chip) azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80); /* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */ azx_dev->sd_int_sta_mask = 1 << i; - /* stream tag: must be non-zero and unique */ azx_dev->index = i; - azx_dev->stream_tag = i + 1; + + /* stream tag must be unique throughout the stream direction + * group, valid values 1...15 + */ + azx_dev->stream_tag = is_input_stream(chip, i) ? + ++in_stream_tag : ++out_stream_tag; } return 0; -- 1.7.0.4

2 1

[alsa-devel] ASoC: omap-mcbsp: Deletion of an unnecessary check before the function call "kfree"
by SF Markus Elfring 18 Nov '14

18 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 14:05:27 +0100 The kfree() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/omap/mcbsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 86c7538..68a1252 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -621,8 +621,7 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) mcbsp->reg_cache = NULL; spin_unlock(&mcbsp->lock); - if (reg_cache) - kfree(reg_cache); + kfree(reg_cache); } /* -- 2.1.3

3 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Remove comment about SSI running only in slave mode
by Fabio Estevam 18 Nov '14

18 Nov '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Current driver can also run in I2S master mode, so remove the old comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ab05262..1ef0c84 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,8 +67,6 @@ /** * FSLSSI_I2S_FORMATS: audio formats supported by the SSI * - * This driver currently only supports the SSI running in I2S slave mode. - * * The SSI has a limitation in that the samples must be in the same byte * order as the host CPU. This is because when multiple bytes are written * to the STX register, the bytes and bits must be written in the same -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5677: Modify the default value of the MX-8E[4] for ASRC function
by Oder Chiou 18 Nov '14

18 Nov '14

Modify the default value of the MX-8E[4] to 1 for ASRC function. It could prevent the pop noise with ASRC function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index d659237..49ac9ba 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -55,7 +55,8 @@ static const struct regmap_range_cfg rt5677_ranges[] = { }; static const struct reg_default init_list[] = { - {RT5677_PR_BASE + 0x3d, 0x364d}, + {RT5677_ASRC_12, 0x0018}, + {RT5677_PR_BASE + 0x3d, 0x364d}, {RT5677_PR_BASE + 0x17, 0x4fc0}, {RT5677_PR_BASE + 0x13, 0x0312}, {RT5677_PR_BASE + 0x1e, 0x0000}, @@ -173,7 +174,7 @@ static const struct reg_default rt5677_reg[] = { {RT5677_ASRC_9 , 0x0000}, {RT5677_ASRC_10 , 0x0000}, {RT5677_ASRC_11 , 0x0000}, - {RT5677_ASRC_12 , 0x0008}, + {RT5677_ASRC_12 , 0x0018}, {RT5677_ASRC_13 , 0x0000}, {RT5677_ASRC_14 , 0x0000}, {RT5677_ASRC_15 , 0x0000}, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] ASoC fixes for v3.18
by Mark Brown 17 Nov '14

17 Nov '14

The following changes since commit fc14f9c1272f62c3e8d01300f52467c0d9af50f9: Linux 3.18-rc5 (2014-11-16 16:36:20 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.18-rc5 for you to fetch changes up to cf9a7f7823c67243da44da2ac47ca944a3108282: Merge remote-tracking branches 'asoc/fix/rt5670', 'asoc/fix/samsung' and 'asoc/fix/sgtl5000' into asoc-linus (2014-11-17 16:41:11 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.18 As well as the usual driver fixes there's a few other things here: One is a fix for a race in DPCM which is unfortuantely a rather large diffstat, this is the result of growing usage of the mainline code and hence more detailed testing so I'm relatively happy. The other is a fix for non-DT machine driver matching following some of the componentization work which is much more focused. Both have had a while to cook in -next. ---------------------------------------------------------------- Andreas Färber (1): ASoC: samsung: Add MODULE_DEVICE_TABLE for Snow Axel Lin (1): ASoC: es8328-i2c: Fix i2c_device_id name field in es8328_id Bard Liao (2): ASoC: rt5670: correct the incorrect default values ASoC: rt5670: change dapm routes of PLL connection Charles Keepax (1): ASoC: wm_adsp: Avoid attempt to free buffers that might still be in use Dylan Reid (1): ASoC: max98090: Correct pclk divisor settings Fabio Estevam (1): ASoC: sgtl5000: Fix SMALL_POP bit definition Fang, Yang A (1): ASoC: rt5645: Mark RT5645_TDM_CTRL_3 as readable Jianqun (1): ASoC: rockchip-i2s: fix infinite loop in rockchip_snd_rxctrl Kuninori Morimoto (2): ASoC: fsi: remove unsupported PAUSE flag ASoC: rsnd: remove unsupported PAUSE flag Lars-Peter Clausen (1): ASoC: Fix snd_soc_find_dai() matching component by name Mark Brown (3): Merge remote-tracking branch 'asoc/fix/core' into asoc-linus Merge remote-tracking branches 'asoc/fix/adsp', 'asoc/fix/cs41l51', 'asoc/fix/dpcm', 'asoc/fix/es8328', 'asoc/fix/fsl-asrc', 'asoc/fix/max98090', 'asoc/fix/rcar', 'asoc/fix/rockchip' and 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5670', 'asoc/fix/samsung' and 'asoc/fix/sgtl5000' into asoc-linus Nicolin Chen (1): ASoC: fsl_asrc: Add reg_defaults for regmap to fix kernel dump Takashi Iwai (1): ASoC: dpcm: Fix race between FE/BE updates and trigger Thomas Petazzoni (1): ASoC: cs42l51: re-hook of_match_table pointer include/sound/soc-dpcm.h | 2 ++ sound/soc/codecs/cs42l51-i2c.c | 1 + sound/soc/codecs/cs42l51.c | 4 ++- sound/soc/codecs/cs42l51.h | 1 + sound/soc/codecs/es8328-i2c.c | 2 +- sound/soc/codecs/max98090.c | 6 ++-- sound/soc/codecs/rt5645.c | 2 ++ sound/soc/codecs/rt5670.c | 36 ++++++++++---------- sound/soc/codecs/sgtl5000.c | 3 +- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/wm_adsp.c | 1 + sound/soc/fsl/fsl_asrc.c | 26 ++++++++++++++ sound/soc/rockchip/rockchip_i2s.c | 4 ++- sound/soc/samsung/snow.c | 1 + sound/soc/sh/fsi.c | 3 +- sound/soc/sh/rcar/core.c | 3 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-pcm.c | 72 ++++++++++++++++++++++++++++++--------- 18 files changed, 123 insertions(+), 48 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: samsung-i2s: Add 192kHz config option
by Richard Fitzgerald 17 Nov '14

17 Nov '14

Adds QUIRK_SUPPORTS_192KHZ to allow 192kHz rate to be selected for hardware that supports it. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 2 ++ include/linux/platform_data/asoc-s3c.h | 1 + sound/soc/samsung/i2s.c | 8 ++++++++ 3 files changed, 11 insertions(+), 0 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt index d188296..c3f3950 100644 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt @@ -40,6 +40,8 @@ Optional SoC Specific Properties: sub system(used in secondary sound source). - pinctrl-0: Should specify pin control groups used for this controller. - pinctrl-names: Should contain only one value - "default". +- samsung,supports-192khz: specify this (without a value) if you want to allow + 192000 sample rate over the I2S link. Example: diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h index 5e0bc77..6d0702d 100644 --- a/include/linux/platform_data/asoc-s3c.h +++ b/include/linux/platform_data/asoc-s3c.h @@ -28,6 +28,7 @@ struct samsung_i2s { #define QUIRK_NEED_RSTCLR (1 << 3) #define QUIRK_SUPPORTS_TDM (1 << 4) #define QUIRK_SUPPORTS_IDMA (1 << 5) +#define QUIRK_SUPPORTS_192KHZ (1 << 6) /* Quirks of the I2S controller */ u32 quirks; dma_addr_t idma_addr; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 947352d..f688e89 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1219,6 +1219,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) "specified"); } } + + if (of_find_property(np, "samsung,supports-192khz", NULL)) + quirks |= QUIRK_SUPPORTS_192KHZ; } res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1247,6 +1250,11 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (quirks & QUIRK_PRI_6CHAN) pri_dai->i2s_dai_drv.playback.channels_max = 6; + if (quirks & QUIRK_SUPPORTS_192KHZ) { + pri_dai->i2s_dai_drv.playback.rates |= SNDRV_PCM_RATE_192000; + pri_dai->i2s_dai_drv.capture.rates |= SNDRV_PCM_RATE_192000; + } + if (quirks & QUIRK_SEC_DAI) { sec_dai = i2s_alloc_dai(pdev, true); if (!sec_dai) { -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Avoid attempt to free buffers that might still be in use
by Charles Keepax 17 Nov '14

17 Nov '14

We should not free any buffers associated with writing out coefficients to the DSP until all the async writes have completed. This patch updates the out of memory path when allocating a new buffer to include a call to regmap_async_complete. Reported-by: JS Park <aitdark.park(a)samsung.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0a08ef5e..08d8ec3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1343,6 +1343,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) file, blocks, pos - firmware->size); out_fw: + regmap_async_complete(regmap); release_firmware(firmware); wm_adsp_buf_free(&buf_list); out: -- 1.7.2.5

2 1

[alsa-devel] [PATCH] usb-audio: Set the Control Selector to SU_SELECTOR_CONTROL for UAC2
by Johan Rastén 17 Nov '14

17 Nov '14

Specified in section 5.2.5.6.1 of the USB Audio Class 2.0 definition. Solves the following error for C-Media 6632A (Asus Xonar U7): [ 8219.676164] cannot get ctl value: req = 0x81, wValue = 0x0, wIndex = 0x1400, type = 3 Signed-off-by: Johan Rastén <johan(a)oljud.se> --- sound/usb/mixer.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 14e1455..c6209ce 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2032,10 +2032,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, cval->res = 1; cval->initialized = 1; - if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) - cval->control = UAC2_CX_CLOCK_SELECTOR; - else + if (state->mixer->protocol == UAC_VERSION_1) cval->control = 0; + else /* UAC_VERSION_2 */ + cval->control = (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR) ? + UAC2_CX_CLOCK_SELECTOR : UAC2_SU_SELECTOR; namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL); if (!namelist) { -- 2.1.0

2 1

[alsa-devel] [PATCH (alsa-lib)] USB-audio: Add five more cards to IEC958 blacklist
by David Henningsson 17 Nov '14

17 Nov '14

These devices do not have any IEC958 outputs, so prevent them from being opened. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/conf/cards/USB-Audio.conf | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index a6754d9..05e5d09 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -38,10 +38,14 @@ USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 # The below don't have digital in/out, so prevent them from being opened. + "Andrea PureAudio USB-SA Headset" 999 "Blue Snowball" 999 "HP Digital Stereo Headset" 999 + "GN 9330" 999 "Logitech Speaker Lapdesk N700" 999 + "Logitech G35 Headset" 999 "Logitech USB Headset" 999 + "Logitech USB Headset H540" 999 "Logitech Wireless Headset" 999 "Plantronics GameCom 780" 999 "Plantronics USB Headset" 999 @@ -50,6 +54,7 @@ USB-Audio.pcm.iec958_device { "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 + "SWTOR Gaming Headset by Razer" 999 "USB Device 0x46d:0x992" 999 } -- 1.9.1

2 1

[alsa-devel] [PATCH 1/1] ALSA: powermac: Deletion of an unnecessary check before the function call "pci_dev_put"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 13:35:54 +0100 The pci_dev_put() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/ppc/pmac.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 8a431bc..5a13b22 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -887,8 +887,7 @@ static int snd_pmac_free(struct snd_pmac *chip) } } - if (chip->pdev) - pci_dev_put(chip->pdev); + pci_dev_put(chip->pdev); of_node_put(chip->node); kfree(chip); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: hdsp: Deletion of an unnecessary check before the function call "release_firmware"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 13:04:14 +0100 The release_firmware() function tests whether its argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/rme9652/hdsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7646ba1..0ae568d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5368,8 +5368,7 @@ static int snd_hdsp_free(struct hdsp *hdsp) snd_hdsp_free_buffers(hdsp); - if (hdsp->firmware) - release_firmware(hdsp->firmware); + release_firmware(hdsp->firmware); vfree(hdsp->fw_uploaded); if (hdsp->iobase) -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: lola: Deletion of an unnecessary check before the function call "vfree"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 12:42:16 +0100 The vfree() function performs also input parameter validation. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/lola/lola_mixer.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index 782f4d8..e7fe15d 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -108,8 +108,7 @@ int lola_init_pins(struct lola *chip, int dir, int *nidp) void lola_free_mixer(struct lola *chip) { - if (chip->mixer.array_saved) - vfree(chip->mixer.array_saved); + vfree(chip->mixer.array_saved); } int lola_init_mixer_widget(struct lola *chip, int nid) -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: ice17xx: Deletion of unnecessary checks before the function call "snd_ac97_resume"
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 11:28:02 +0100 The snd_ac97_resume() function tests whether its argument is NULL and then returns immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/ice1712/ice1712.c | 3 +-- sound/pci/ice1712/ice1724.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 87f7fc4..e1e18b5 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2905,8 +2905,7 @@ static int snd_ice1712_resume(struct device *dev) outw(ice->pm_saved_spdif_ctrl, ICEMT(ice, ROUTE_SPDOUT)); outw(ice->pm_saved_route, ICEMT(ice, ROUTE_PSDOUT03)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 08cb08a..0e56835 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2884,8 +2884,7 @@ static int snd_vt1724_resume(struct device *dev) outb(ice->pm_saved_spdif_cfg, ICEREG1724(ice, SPDIF_CFG)); outl(ice->pm_saved_route, ICEMT1724(ice, ROUTE_PLAYBACK)); - if (ice->ac97) - snd_ac97_resume(ice->ac97); + snd_ac97_resume(ice->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; -- 2.1.3

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda: Deletion of unnecessary checks before two function calls
by SF Markus Elfring 17 Nov '14

17 Nov '14

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Mon, 17 Nov 2014 10:44:33 +0100 The functions kfree() and release_firmware() test whether their argument is NULL and then return immediately. Thus the test around the call is not needed. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/pci/hda/hda_codec.c | 3 +-- sound/pci/hda/hda_intel.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec6a7d0..3fe8859 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -827,8 +827,7 @@ static void snd_hda_bus_free(struct hda_bus *bus) WARN_ON(!list_empty(&bus->codec_list)); if (bus->workq) flush_workqueue(bus->workq); - if (bus->unsol) - kfree(bus->unsol); + kfree(bus->unsol); if (bus->ops.private_free) bus->ops.private_free(bus); if (bus->workq) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0a7f848..b4ec4e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1129,8 +1129,7 @@ static int azx_free(struct azx *chip) pci_disable_device(chip->pci); kfree(chip->azx_dev); #ifdef CONFIG_SND_HDA_PATCH_LOADER - if (chip->fw) - release_firmware(chip->fw); + release_firmware(chip->fw); #endif if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { hda_display_power(false); -- 2.1.3

2 1

[alsa-devel] [PATCH] ASoC: pxa: prepare/unprepare clocks in pxa-ssp
by Dmitry Eremin-Solenikov 17 Nov '14

17 Nov '14

Change clk_enable/disable() calls to clk_prepare_enable() and clk_disable_unrepapre(). Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> --- sound/soc/pxa/pxa-ssp.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a8e0974..cbba063 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -97,7 +97,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); pxa_ssp_disable(ssp); } @@ -121,7 +121,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, if (!cpu_dai->active) { pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); } kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); @@ -136,7 +136,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; if (!cpu_dai->active) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); priv->cr0 = __raw_readl(ssp->mmio_base + SSCR0); priv->cr1 = __raw_readl(ssp->mmio_base + SSCR1); @@ -144,7 +144,7 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) priv->psp = __raw_readl(ssp->mmio_base + SSPSP); pxa_ssp_disable(ssp); - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -154,7 +154,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) struct ssp_device *ssp = priv->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); __raw_writel(sssr, ssp->mmio_base + SSSR); __raw_writel(priv->cr0 & ~SSCR0_SSE, ssp->mmio_base + SSCR0); @@ -165,7 +165,7 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (cpu_dai->active) pxa_ssp_enable(ssp); else - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); return 0; } @@ -256,11 +256,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (ssp->type != PXA3xx_SSP) - clk_disable(ssp->clk); + clk_disable_unprepare(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); if (ssp->type != PXA3xx_SSP) - clk_enable(ssp->clk); + clk_prepare_enable(ssp->clk); return 0; } -- 2.1.1

3 2

[alsa-devel] [PATCH] ASoC: Intel: add missing ACPI device table
by mengdong.lin＠intel.com 17 Nov '14

17 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> The ACPI device table will generate the driver module alias for Intel audio devices enumerated from ACPI. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c index 2b1c5d9..b261821 100644 --- a/sound/soc/intel/sst/sst_acpi.c +++ b/sound/soc/intel/sst/sst_acpi.c @@ -342,6 +342,8 @@ static const struct acpi_device_id sst_acpi_ids[] = { { }, }; +MODULE_DEVICE_TABLE(acpi, sst_acpi_ids); + static struct platform_driver sst_acpi_driver = { .driver = { .name = "intel_sst_acpi", -- 1.9.1

3 2

[alsa-devel] [PATCH 1/3] ASoC: adav80x: Replace w->codec with snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 17 Nov '14

17 Nov '14

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index ce3cdca..b67480f 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -212,7 +212,7 @@ static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); const char *clk; @@ -236,7 +236,7 @@ static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { - struct snd_soc_codec *codec = source->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; -- 1.8.0

2 5

[alsa-devel] TLV320AIC3106 + DM3730 soc doens't capture audio (no BLCK, no WLCK)
by Sergio Serrano 17 Nov '14

17 Nov '14

Dear members, Please, let me explain my setup: I'm using DM3730 soc <https://www.isee.biz/products/igep-processor-boards/igepv2-dm3730> with McBSP3 and I2C2 to drive audio and control to the TLV320AIC3106 codec (which it is acting as master). Currently I'm using 2.6.37 kernel version <http://git.isee.biz/?p=pub/scm/linux-omap-2.6.git;a=tree>. For reference I've used BBB Audio Cape <http://elinux.org/CircuitCo:Audio_Cape_RevA> as base design. Right now I've got playback working as expected (codec generates BCLK and WCLK). Whenever I try to record using "arecor,d -f cd -Dhw:1,0" it's only showing wave header but no audio is captured at all. I can not see any clocks running on the codec. How can I debug or go further with this issue? Any thoughts will be appreciated!! BTW this is the init code: static int TLV320AIC3106_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int ret; printk(KERN_ERR "+tlv_hw_params(%p,%p) %p, %p, %p\n", substream, params, rtd, codec_dai, cpu_dai); // Set codec DAI configuration ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec DAI configuration\n"); return ret; } // Set cpu DAI configuration ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set cpu DAI configuration\n"); return ret; } // Set the codec system clock for DAC and ADC ret = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "tlv_hw_params: can't set codec system clock\n"); return ret; } printk(KERN_ERR "-tlv_hw_params\n"); return 0; } BR, Sergio

1 0

[alsa-devel] Pop and offsets at start of audio playback/record for SGTL5000 on i.MX28
by Craig McQueen 17 Nov '14

17 Nov '14

I'm testing the 3.14.19 kernel for i.MX28 EVK, which has an SGTL5000 CODEC. I've also tested on the 3.18-rc4 kernel and confirmed this issue still occurs. Playback When doing audio playback, I notice that the audio "fades in" over the first (approximately) 500 ms of playback. This is very noticeable and not ideal in certain applications (e.g. audio notifications that are of short duration). However, if audio playback starts while audio recording is already in-progress, then there is no fade-in; instead there is a loud pop. I assume the fade-in is there to mask the pop. The pop is very unpleasant through headphones. (However, if I do audio playback a _second_ time while audio recording is still in-progress, then there is no loud pop the second time.) I didn't think either the pop or the fade-in occurred on the 2.6.35 kernel. How feasible is it to improve the driver to avoid or reduce the pop, and remove the necessity for a fade-in? Record When recording, I notice there is a low-frequency ramping offset at the start of a recording (after the 400-500 ms of previous audio) of approx 600-700 ms duration. This also reduces the audio quality at the start of the recording. It could reduce the quality of captured audio depending on the application usage. I am getting distortion (clipping I assume) of my audio input during this time. It could cause a pop on playback (local playback or e.g. VoIP transmission to a remote device). It could reduce the audio quality through an audio CODEC. How feasible is it to improve the driver to avoid a ramping offset at the start of recording? -- Craig McQueen

3 6

[alsa-devel] What tree to base patches on?
by Johan Aires Rastén 16 Nov '14

16 Nov '14

If I want to send you a patch for snd_usb_audio, is there a git repo + branch or something else which I should base it on?

2 1

[alsa-devel] [PATCH v2] ASoC: Fix kerneldoc errors
by Takashi Iwai 16 Nov '14

16 Nov '14

Lots of typos and missing parameter descriptions. Some private struct fields are moved to the end of struct and put markers to make kerneldoc happy (and it make even easier for human readers). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc.h | 16 +++++++++++----- sound/soc/soc-core.c | 21 ++++++++++++++------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-devres.c | 2 +- sound/soc/soc-jack.c | 2 +- 5 files changed, 28 insertions(+), 15 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7ba7130037a0..10129be3cbb1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -588,10 +588,11 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @invert: if non-zero then pin is enabled when status is not reported */ struct snd_soc_jack_pin { - struct list_head list; const char *pin; int mask; bool invert; + /* private: */ + struct list_head list; }; /** @@ -602,13 +603,13 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container */ struct snd_soc_jack_zone { unsigned int min_mv; unsigned int max_mv; unsigned int jack_type; unsigned int debounce_time; + /* private: */ struct list_head list; }; @@ -618,16 +619,19 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debouce time in ms * @wake: enable as wake source + * @jack: assigned jack + * @desc: GPIO descriptor * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an * ADC). + * @data: arbitrary data passed to jack_status_check callback */ struct snd_soc_jack_gpio { unsigned int gpio; @@ -640,11 +644,13 @@ struct snd_soc_jack_gpio { bool wake; struct snd_soc_jack *jack; - struct delayed_work work; struct gpio_desc *desc; void *data; int (*jack_status_check)(void *data); + + /* private: */ + struct delayed_work work; }; struct snd_soc_jack { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4c8f8a23a0e9..74a8a80e2d9a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2113,6 +2113,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: AC97 opts to set + * @pdev: platform to deal with * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. @@ -2685,7 +2687,7 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx); /** * snd_soc_put_volsw_sx - double mixer set callback * @kcontrol: mixer control - * @uinfo: control element information + * @ucontrol: control element information * * Callback to set the value of a double mixer control that spans 2 registers. * @@ -3458,7 +3460,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -4082,6 +4084,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core + * @dev: device to unregister * */ void snd_soc_unregister_component(struct device *dev) @@ -4119,7 +4122,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -4158,7 +4161,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: platform to register + * @platform_drv: the driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -4215,7 +4219,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -4306,7 +4310,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: codec to register + * @codec_drv: the codec driver to register + * @dai_drv: the array of DAI drivers to register + * @num_dai: the number of DAIs to register */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -4406,7 +4413,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c61cb9cedbcd..7050ec1d5b72 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2680,7 +2680,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card object containing widgets * * Checks the codec for any new dapm widgets and creates them if found. * diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 057e5ef7dcce..a57921eeee81 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -60,7 +60,7 @@ static void devm_platform_release(struct device *dev, void *res) /** * devm_snd_soc_register_platform - resource managed platform registration * @dev: Device used to manage platform - * @platform: platform to register + * @platform_drv: platform to register * * Register a platform driver with automatic unregistration when the device is * unregistered. diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ab47fea997a3..ef1d42d7c6f6 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -116,7 +116,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_report); * * @jack: ASoC jack * @count: Number of zones - * @zone: Array of zones + * @zones: Array of zones * * After this function has been called the zones specified in the * array will be associated with the jack. -- 2.1.3

2 7

[alsa-devel] [PATCH] usb-audio: Add ctrl message delay quirk for Marantz/Denon devices
by Jurgen Kramer 16 Nov '14

16 Nov '14

This patch adds a USB control message delay quirk for a few specific Marantz/Denon devices. Without the delay the DACs will not work properly and produces the following type of messages: Nov 15 10:09:21 orwell kernel: [ 91.342880] usb 3-13: clock source 41 is not valid, cannot use Nov 15 10:09:21 orwell kernel: [ 91.343775] usb 3-13: clock source 41 is not valid, cannot use There are likely other Marantz/Denon devices using the same USB module which exhibit the same problems. But as this cannot be verified I limited the patch to the devices I could test. The following two devices are covered by this path: - Marantz SA-14S1 - Marantz HD-DAC1 Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d2aa45a..a5941f8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1146,6 +1146,20 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); + + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + + switch (le16_to_cpu(dev->descriptor.idProduct)) { + case 0x3005: /* Marantz HD-DAC1 */ + case 0x3006: /* Marantz SA-14S1 */ + mdelay(20); + break; + } + } } /* -- 1.9.3

2 1

[alsa-devel] [PATCH] snd_dlsym: only do versioning checks if built --with-versioning
by John Spencer 16 Nov '14

16 Nov '14

From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) From 291553a0132936611d48f41e924601dff8941537 Mon Sep 17 00:00:00 2001 From: John Spencer <maillist-alsa(a)barfooze.de> Date: Sat, 15 Nov 2014 13:41:00 +0100 Subject: [PATCH] snd_dlsym: only do versioning checks if built --with-versioning a combination of some of the following garbage collecting LD/CFLAGS -Os -g0 -fdata-sections -ffunction-sections -s -Wl,--gc-sections -fno-unwind-tables -fno-asynchronous-unwind-tables causes the symbol versioning marker symbols to be removed from BSS since they're otherwise unreferenced. this causes dlsym failing to find them which results in runtime breakage: $ alsamixer ALSA lib dlmisc.c:142:(snd_dlsym_verify) unable to verify version for symbol snd_config_hook_load ALSA lib conf.c:3328:(snd_config_hooks_call) symbol snd_config_hook_load is not defined inside (null) ALSA lib conf.c:3788:(snd_config_update_r) hooks failed, removing configuration cannot open mixer: No such device or address correct DSO: $ objdump -T libasound.so.2.strip | grep _snd_config_hook_load_dlsym_config_hook_001 001196bc g DO .bss 00000001 _snd_config_hook_load_dlsym_config_hook_001 incorrect DSO: $ objdump -T libs/libasound.so.2 | grep _snd_config_hook_load_dlsym_config_hook_001 00000000 g DO *ABS* 00000001 _snd_config_hook_load_dlsym_config_hook_001 since alsa was built with --without-versioning, doing those versioning checks at all is unnecessary and harmful and thus now disabled in this case. Signed-off-by: John Spencer <maillist-alsa(a)barfooze.de> --- src/dlmisc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/dlmisc.c b/src/dlmisc.c index 4b8a02c..62f7047 100644 --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -177,11 +177,13 @@ void *snd_dlsym(void *handle, const char *name, const char *version) } #endif #ifdef HAVE_LIBDL +#ifdef VERSIONED_SYMBOLS if (version) { err = snd_dlsym_verify(handle, name, version); if (err < 0) return NULL; } +#endif return dlsym(handle, name); #else return NULL; -- 1.8.4

2 1

[alsa-devel] DTS-HD over HDMI
by Kip Warner 15 Nov '14

15 Nov '14

Hey list, I apologize if this ought to have gone to alsa-user. I wasn't sure as a developer using API, but not developing ALSA itself, if it have been better there than here. I'd like to be able to stream DTS-HD Master Audio bits out the HDMI port in their compressed form, and not decompressed as LPCM. From what I can gather, this may be possible with the ALSA API, provided the HDMI port is >= 1.3 and the audio card supports high bit-rate (HBR). Is my thinking correct? -- Kip Warner -- Senior Software Engineer OpenPGP encrypted/signed mail preferred http://www.thevertigo.com

2 2

[alsa-devel] usb-audio: check_input_term (mixer.c) iterating to clock selectors?
by Johan Aires Rastén 15 Nov '14

15 Nov '14

Hi everybody! I've just started looking at ALSA because I want to get the mic working with my Asus Xonar U7. Some of you might have seen me on IRC as johanar. I'm wondering about check_input_term(...) in sound/usb/mixer.c, which for case UAC_INPUT_TERMINAL and UAC_VERSION_2 will recursively go into the clock source. Is this correct behaviour? For my card both mic and line will show up as "Unit 22" when I call "amixer contents" since the two inputs have the same unnamed clock source (they also share physical connector). However, if I comment out the recursive call they show up as "mic" and "line" as they're supposed to. Btw, if this is correct, couldn't we set id = d->bCSourceID and break; in the same way that case UAC_FEATURE_UNIT does? //Johan Rastén

1 0

[alsa-devel] [PATCH 1/2] spi: Fix mapping from vmalloc-ed buffer to scatter list
by Charles Keepax 14 Nov '14

14 Nov '14

We can only use page_address on memory that has been mapped using kmap, when the buffer passed to the SPI has been allocated by vmalloc the page has not necessarily been mapped through kmap. This means sometimes page_address will return NULL causing the pointer we pass to sg_set_buf to be invalid. As we only call page_address such that we can pass a virtual address to sg_set_buf which will then immediately call virt_to_page on it, fix this by calling sg_set_page directly rather then relying on the sg_set_buf helper. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/spi/spi.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/spi/spi.c b/drivers/spi/spi.c index ebcb33d..50f20f2 100644 --- a/drivers/spi/spi.c +++ b/drivers/spi/spi.c @@ -615,13 +615,13 @@ static int spi_map_buf(struct spi_master *master, struct device *dev, sg_free_table(sgt); return -ENOMEM; } - sg_buf = page_address(vm_page) + - ((size_t)buf & ~PAGE_MASK); + sg_set_page(&sgt->sgl[i], vm_page, + min, offset_in_page(buf)); } else { sg_buf = buf; + sg_set_buf(&sgt->sgl[i], sg_buf, min); } - sg_set_buf(&sgt->sgl[i], sg_buf, min); buf += min; len -= min; -- 1.7.2.5

2 7

[alsa-devel] Question on Compressed offload session
by gsantosh＠codeaurora.org 14 Nov '14

14 Nov '14

Hi All, The Question is for the compressed offload session. For a generic codec driver during the startup function it will set some of the hw_constraints rule similarly like this. snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_12_24); pcm_lib.c will try to add the rule to the runtime structure by accessing the pointers which will be initialized during opening of the session, as The Constraints added by the codec driver will be updated in the struct snd_pcm_hw_constraints of runtime structure which will be part of substream handle. But for the compressed offload I do not see the initialization done for HW constraints, as done in pcm session 2092int snd_pcm_open_substream(struct snd_pcm *pcm, int stream, 2093 struct file *file, 2094 struct snd_pcm_substream **rsubstream) most of the existing drivers which has the hw_constraint_list code will not be applicable for compress offload session, how to solve this? Regards, Santosh M G. Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,a Linux Foundation Collaborative Project

3 4

[alsa-devel] (no subject)
by Angelo Dureghello 14 Nov '14

14 Nov '14

unsubscribe alsa-devel angelo70(a)gmail.com -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 00/15] new locomo driver
by Dmitry Eremin-Solenikov 14 Nov '14

14 Nov '14

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. Current drivers present in Linux kernel has some problems: * It uses custom bus instead of platform bus/mfd core. * Device drivers are not well layered/separated. * It uses custom gpio accessors instead of using GPIOLIB. With this patchset I tried to modernise and restructure the LoCoMo driver. Also I added an experimental SPI driver that will be used by mmc-spi on Sharp SL-5500 collie devices. ---------------------------------------------------------------- Dmitry Eremin-Solenikov (15): mfd: add new driver for Sharp LoCoMo GPIO: port LoCoMo gpio support from old driver leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver video: backlight: drop old locomo bl/lcd driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ARM: pxa: poodle: use new LoCoMo driver sound: soc: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: don't preallocate IRQ space for locomo ARM: drop old LoCoMo driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 --------------------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 16 +- arch/arm/mach-pxa/poodle.c | 23 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 112 +-- arch/arm/mach-sa1100/include/mach/collie.h | 25 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 7 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 386 +++++++++ drivers/input/keyboard/Kconfig | 1 - drivers/input/keyboard/locomokbd.c | 165 ++-- drivers/leds/Kconfig | 1 - drivers/leds/leds-locomo.c | 104 ++- drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 644 +++++++++++++++ drivers/spi/Kconfig | 8 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 370 +++++++++ drivers/video/backlight/Kconfig | 14 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 171 ++++ drivers/video/backlight/locomo_lcd.c | 224 +++++ drivers/video/backlight/locomolcd.c | 255 ------ .../asm/hardware => include/linux/mfd}/locomo.h | 148 ++-- sound/soc/pxa/poodle.c | 51 +- 30 files changed, 2185 insertions(+), 1493 deletions(-) delete mode 100644 arch/arm/common/locomo.c create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c rename {arch/arm/include/asm/hardware => include/linux/mfd}/locomo.h (55%)

9 53

[alsa-devel] [PATCH 1/4] ASoC: rt5670: make bias level more reasonable
by bardliao＠realtek.com 14 Nov '14

14 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patah separate bias level off to standby and off. The standby level will provide the necessary power for JD and push button functions. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index c4c42d5..85dc210 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2309,6 +2309,8 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2330,16 +2332,27 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - snd_soc_write(codec, RT5670_PWR_DIG1, 0x0000); - snd_soc_write(codec, RT5670_PWR_DIG2, 0x0001); - snd_soc_write(codec, RT5670_PWR_VOL, 0x0000); - snd_soc_write(codec, RT5670_PWR_MIXER, 0x0001); - snd_soc_write(codec, RT5670_PWR_ANLG1, 0x2800); - snd_soc_write(codec, RT5670_PWR_ANLG2, 0x0004); - snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); snd_soc_update_bits(codec, RT5670_PWR_ANLG1, RT5670_LDO_SEL_MASK, 0x1); break; + case SND_SOC_BIAS_OFF: + if (rt5670->pdata.jd_mode) + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, + RT5670_PWR_MB | RT5670_PWR_BG); + else + snd_soc_update_bits(codec, RT5670_PWR_ANLG1, + RT5670_PWR_VREF1 | RT5670_PWR_MB | + RT5670_PWR_BG | RT5670_PWR_VREF2 | + RT5670_PWR_FV1 | RT5670_PWR_FV2, 0); + + snd_soc_update_bits(codec, RT5670_DIG_MISC, 0x1, 0x0); + break; default: break; -- 1.8.1.1.439.g50a6b54

4 7

[alsa-devel] [PATCH v4] ALSA: ice1712: consider error value
by Sudip Mukherjee 14 Nov '14

14 Nov '14

earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v3: spec->ak4114 is now dereferenced if we dont get an error value from snd_ak4114_create. change in v4: error value returned according to coding style. sound/pci/ice1712/revo.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..1d81ae6 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -494,11 +494,13 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ap192_ak4114_write, ak4114_init_vals, ak4114_init_txcsb, ice, &spec->ak4114); + if (err < 0) + return err; /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return 0; } static int revo_init(struct snd_ice1712 *ice) -- 1.8.1.2

2 4

[alsa-devel] [PATCH v3] ALSA: ice1712: consider error value
by Sudip Mukherjee 14 Nov '14

14 Nov '14

earlier we were ignoring the return value of snd_ak4114_create and always returning 0. now we are returning the actual status. revo_init is calling this function, and revo_init is checking the return value. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v3: spec->ak4114 is now dereferenced if we dont get an error value from snd_ak4114_create. sound/pci/ice1712/revo.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 1112ec1..24a7c6a 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -496,9 +496,10 @@ static int ap192_ak4114_init(struct snd_ice1712 *ice) ice, &spec->ak4114); /* AK4114 in Revo cannot detect external rate correctly. * No reason to stop capture stream due to incorrect checks */ - spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; + if (err == 0) + spec->ak4114->check_flags = AK4114_CHECK_NO_RATE; - return 0; /* error ignored; it's no fatal error */ + return err; } static int revo_init(struct snd_ice1712 *ice) -- 1.8.1.2

2 1

[alsa-devel] [PATCH v4] ASoC: Intel: mrfld- add ACPI module
by Vinod Koul 14 Nov '14

14 Nov '14

Add the last ACPI module support which also uses core module like the PCI part Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 5 + sound/soc/intel/sst/Makefile | 4 +- sound/soc/intel/sst/sst.c | 22 ++- sound/soc/intel/sst/sst.h | 1 + sound/soc/intel/sst/sst_acpi.c | 362 ++++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst_pvt.c | 2 + 6 files changed, 391 insertions(+), 5 deletions(-) create mode 100644 sound/soc/intel/sst/sst_acpi.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c963a5d..a992e85 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -20,6 +20,11 @@ config SND_SST_IPC_PCI tristate select SND_SST_IPC +config SND_SST_IPC_ACPI + tristate + select SND_SST_IPC + depends on ACPI + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI diff --git a/sound/soc/intel/sst/Makefile b/sound/soc/intel/sst/Makefile index b8aa1d3..fd21726 100644 --- a/sound/soc/intel/sst/Makefile +++ b/sound/soc/intel/sst/Makefile @@ -1,7 +1,7 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_loader.o sst_pvt.o snd-intel-sst-pci-objs += sst_pci.o - +snd-intel-sst-acpi-objs += sst_acpi.o obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o - +obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index b97c231..b2b5604 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -26,6 +26,7 @@ #include <linux/pm_runtime.h> #include <linux/pm_qos.h> #include <linux/async.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/soc.h> #include <asm/platform_sst_audio.h> @@ -181,6 +182,7 @@ int sst_driver_ops(struct intel_sst_drv *sst) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst->tstamp = SST_TIME_STAMP_MRFLD; sst->ops = &mrfld_ops; return 0; @@ -323,7 +325,7 @@ EXPORT_SYMBOL_GPL(sst_context_init); void sst_context_cleanup(struct intel_sst_drv *ctx) { pm_runtime_get_noresume(ctx->dev); - pm_runtime_forbid(ctx->dev); + pm_runtime_disable(ctx->dev); sst_unregister(ctx->dev); sst_set_fw_state_locked(ctx, SST_SHUTDOWN); flush_scheduled_work(); @@ -371,8 +373,19 @@ void sst_configure_runtime_pm(struct intel_sst_drv *ctx) { pm_runtime_set_autosuspend_delay(ctx->dev, SST_SUSPEND_DELAY); pm_runtime_use_autosuspend(ctx->dev); - pm_runtime_allow(ctx->dev); - pm_runtime_put_noidle(ctx->dev); + /* + * For acpi devices, the actual physical device state is + * initially active. So change the state to active before + * enabling the pm + */ + if (acpi_disabled) { + pm_runtime_set_active(ctx->dev); + pm_runtime_enable(ctx->dev); + } else { + pm_runtime_allow(ctx->dev); + pm_runtime_put_noidle(ctx->dev); + } + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); } EXPORT_SYMBOL_GPL(sst_configure_runtime_pm); @@ -395,6 +408,9 @@ static int intel_sst_runtime_suspend(struct device *dev) synchronize_irq(ctx->irq_num); flush_workqueue(ctx->post_msg_wq); + /* save the shim registers because PMC doesn't save state */ + sst_save_shim64(ctx, ctx->shim, ctx->shim_regs64); + return ret; } diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 2dcbf47..683dc71 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -29,6 +29,7 @@ /* driver names */ #define SST_DRV_NAME "intel_sst_driver" #define SST_MRFLD_PCI_ID 0x119A +#define SST_BYT_ACPI_ID 0x80860F28 #define SST_SUSPEND_DELAY 2000 #define FW_CONTEXT_MEM (64*1024) diff --git a/sound/soc/intel/sst/sst_acpi.c b/sound/soc/intel/sst/sst_acpi.c new file mode 100644 index 0000000..2b1c5d9 --- /dev/null +++ b/sound/soc/intel/sst/sst_acpi.c @@ -0,0 +1,362 @@ +/* + * sst_acpi.c - SST (LPE) driver init file for ACPI enumeration. + * + * Copyright (c) 2013, Intel Corporation. + * + * Authors: Ramesh Babu K V <Ramesh.Babu(a)intel.com> + * Authors: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * + */ + +#include <linux/module.h> +#include <linux/fs.h> +#include <linux/interrupt.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <linux/miscdevice.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/pm_runtime.h> +#include <linux/pm_qos.h> +#include <linux/acpi.h> +#include <asm/platform_sst_audio.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/compress_driver.h> +#include <acpi/acbuffer.h> +#include <acpi/platform/acenv.h> +#include <acpi/platform/aclinux.h> +#include <acpi/actypes.h> +#include <acpi/acpi_bus.h> +#include "../sst-mfld-platform.h" +#include "../sst-dsp.h" +#include "sst.h" + +struct sst_machines { + char codec_id[32]; + char board[32]; + char machine[32]; + void (*machine_quirk)(void); + char firmware[32]; + struct sst_platform_info *pdata; + +}; + +/* LPE viewpoint addresses */ +#define SST_BYT_IRAM_PHY_START 0xff2c0000 +#define SST_BYT_IRAM_PHY_END 0xff2d4000 +#define SST_BYT_DRAM_PHY_START 0xff300000 +#define SST_BYT_DRAM_PHY_END 0xff320000 +#define SST_BYT_IMR_VIRT_START 0xc0000000 /* virtual addr in LPE */ +#define SST_BYT_IMR_VIRT_END 0xc01fffff +#define SST_BYT_SHIM_PHY_ADDR 0xff340000 +#define SST_BYT_MBOX_PHY_ADDR 0xff344000 +#define SST_BYT_DMA0_PHY_ADDR 0xff298000 +#define SST_BYT_DMA1_PHY_ADDR 0xff29c000 +#define SST_BYT_SSP0_PHY_ADDR 0xff2a0000 +#define SST_BYT_SSP2_PHY_ADDR 0xff2a2000 + +#define BYT_FW_MOD_TABLE_OFFSET 0x80000 +#define BYT_FW_MOD_TABLE_SIZE 0x100 +#define BYT_FW_MOD_OFFSET (BYT_FW_MOD_TABLE_OFFSET + BYT_FW_MOD_TABLE_SIZE) + +static const struct sst_info byt_fwparse_info = { + .use_elf = false, + .max_streams = 25, + .iram_start = SST_BYT_IRAM_PHY_START, + .iram_end = SST_BYT_IRAM_PHY_END, + .iram_use = true, + .dram_start = SST_BYT_DRAM_PHY_START, + .dram_end = SST_BYT_DRAM_PHY_END, + .dram_use = true, + .imr_start = SST_BYT_IMR_VIRT_START, + .imr_end = SST_BYT_IMR_VIRT_END, + .imr_use = true, + .mailbox_start = SST_BYT_MBOX_PHY_ADDR, + .num_probes = 0, + .lpe_viewpt_rqd = true, +}; + +static const struct sst_ipc_info byt_ipc_info = { + .ipc_offset = 0, + .mbox_recv_off = 0x400, +}; + +static const struct sst_lib_dnld_info byt_lib_dnld_info = { + .mod_base = SST_BYT_IMR_VIRT_START, + .mod_end = SST_BYT_IMR_VIRT_END, + .mod_table_offset = BYT_FW_MOD_TABLE_OFFSET, + .mod_table_size = BYT_FW_MOD_TABLE_SIZE, + .mod_ddr_dnld = false, +}; + +static const struct sst_res_info byt_rvp_res_info = { + .shim_offset = 0x140000, + .shim_size = 0x000100, + .shim_phy_addr = SST_BYT_SHIM_PHY_ADDR, + .ssp0_offset = 0xa0000, + .ssp0_size = 0x1000, + .dma0_offset = 0x98000, + .dma0_size = 0x4000, + .dma1_offset = 0x9c000, + .dma1_size = 0x4000, + .iram_offset = 0x0c0000, + .iram_size = 0x14000, + .dram_offset = 0x100000, + .dram_size = 0x28000, + .mbox_offset = 0x144000, + .mbox_size = 0x1000, + .acpi_lpe_res_index = 0, + .acpi_ddr_index = 2, + .acpi_ipc_irq_index = 5, +}; + +struct sst_platform_info byt_rvp_platform_data = { + .probe_data = &byt_fwparse_info, + .ipc_info = &byt_ipc_info, + .lib_info = &byt_lib_dnld_info, + .res_info = &byt_rvp_res_info, + .platform = "sst-mfld-platform", +}; + +static int sst_platform_get_resources(struct intel_sst_drv *ctx) +{ + struct resource *rsrc; + struct platform_device *pdev = to_platform_device(ctx->dev); + + /* All ACPI resource request here */ + /* Get Shim addr */ + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_lpe_res_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid SHIM base from IFWI"); + return -EIO; + } + dev_info(ctx->dev, "LPE base: %#x size:%#x", (unsigned int) rsrc->start, + (unsigned int)resource_size(rsrc)); + + ctx->iram_base = rsrc->start + ctx->pdata->res_info->iram_offset; + ctx->iram_end = ctx->iram_base + ctx->pdata->res_info->iram_size - 1; + dev_info(ctx->dev, "IRAM base: %#x", ctx->iram_base); + ctx->iram = devm_ioremap_nocache(ctx->dev, ctx->iram_base, + ctx->pdata->res_info->iram_size); + if (!ctx->iram) { + dev_err(ctx->dev, "unable to map IRAM"); + return -EIO; + } + + ctx->dram_base = rsrc->start + ctx->pdata->res_info->dram_offset; + ctx->dram_end = ctx->dram_base + ctx->pdata->res_info->dram_size - 1; + dev_info(ctx->dev, "DRAM base: %#x", ctx->dram_base); + ctx->dram = devm_ioremap_nocache(ctx->dev, ctx->dram_base, + ctx->pdata->res_info->dram_size); + if (!ctx->dram) { + dev_err(ctx->dev, "unable to map DRAM"); + return -EIO; + } + + ctx->shim_phy_add = rsrc->start + ctx->pdata->res_info->shim_offset; + dev_info(ctx->dev, "SHIM base: %#x", ctx->shim_phy_add); + ctx->shim = devm_ioremap_nocache(ctx->dev, ctx->shim_phy_add, + ctx->pdata->res_info->shim_size); + if (!ctx->shim) { + dev_err(ctx->dev, "unable to map SHIM"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->shim_phy_add = ctx->pdata->res_info->shim_phy_addr; + + /* Get mailbox addr */ + ctx->mailbox_add = rsrc->start + ctx->pdata->res_info->mbox_offset; + dev_info(ctx->dev, "Mailbox base: %#x", ctx->mailbox_add); + ctx->mailbox = devm_ioremap_nocache(ctx->dev, ctx->mailbox_add, + ctx->pdata->res_info->mbox_size); + if (!ctx->mailbox) { + dev_err(ctx->dev, "unable to map mailbox"); + return -EIO; + } + + /* reassign physical address to LPE viewpoint address */ + ctx->mailbox_add = ctx->info.mailbox_start; + + rsrc = platform_get_resource(pdev, IORESOURCE_MEM, + ctx->pdata->res_info->acpi_ddr_index); + if (!rsrc) { + dev_err(ctx->dev, "Invalid DDR base from IFWI"); + return -EIO; + } + ctx->ddr_base = rsrc->start; + ctx->ddr_end = rsrc->end; + dev_info(ctx->dev, "DDR base: %#x", ctx->ddr_base); + ctx->ddr = devm_ioremap_nocache(ctx->dev, ctx->ddr_base, + resource_size(rsrc)); + if (!ctx->ddr) { + dev_err(ctx->dev, "unable to map DDR"); + return -EIO; + } + + /* Find the IRQ */ + ctx->irq_num = platform_get_irq(pdev, + ctx->pdata->res_info->acpi_ipc_irq_index); + return 0; +} + +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_machines *sst_acpi_find_machine( + struct sst_machines *machines) +{ + struct sst_machines *mach; + bool found = false; + + for (mach = machines; mach->codec_id; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->codec_id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + +int sst_acpi_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + int ret = 0; + struct intel_sst_drv *ctx; + const struct acpi_device_id *id; + struct sst_machines *mach; + struct platform_device *mdev; + struct platform_device *plat_dev; + unsigned int dev_id; + + id = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!id) + return -ENODEV; + dev_dbg(dev, "for %s", id->id); + + mach = (struct sst_machines *)id->driver_data; + mach = sst_acpi_find_machine(mach); + if (mach == NULL) { + dev_err(dev, "No matching machine driver found\n"); + return -ENODEV; + } + + ret = kstrtouint(id->id, 16, &dev_id); + if (ret < 0) { + dev_err(dev, "Unique device id conversion error: %d\n", ret); + return ret; + } + + dev_dbg(dev, "ACPI device id: %x\n", dev_id); + + plat_dev = platform_device_register_data(dev, mach->pdata->platform, -1, NULL, 0); + if (plat_dev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->pdata->platform); + return -ENODEV; + } + + /* Create platform device for sst machine driver */ + mdev = platform_device_register_data(dev, mach->machine, -1, NULL, 0); + if (mdev == NULL) { + dev_err(dev, "Failed to create machine device: %s\n", mach->machine); + return -ENODEV; + } + + ret = sst_alloc_drv_context(&ctx, dev, dev_id); + if (ret < 0) + return ret; + + /* Fill sst platform data */ + ctx->pdata = mach->pdata; + strcpy(ctx->firmware_name, mach->firmware); + + ret = sst_platform_get_resources(ctx); + if (ret) + return ret; + + ret = sst_context_init(ctx); + if (ret < 0) + return ret; + + /* need to save shim registers in BYT */ + ctx->shim_regs64 = devm_kzalloc(ctx->dev, sizeof(*ctx->shim_regs64), + GFP_KERNEL); + if (!ctx->shim_regs64) { + return -ENOMEM; + goto do_sst_cleanup; + } + + sst_configure_runtime_pm(ctx); + platform_set_drvdata(pdev, ctx); + return ret; + +do_sst_cleanup: + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + dev_err(ctx->dev, "failed with %d\n", ret); + return ret; +} + +/** +* intel_sst_remove - remove function +* +* @pdev: platform device structure +* +* This function is called by OS when a device is unloaded +* This frees the interrupt etc +*/ +int sst_acpi_remove(struct platform_device *pdev) +{ + struct intel_sst_drv *ctx; + + ctx = platform_get_drvdata(pdev); + sst_context_cleanup(ctx); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct sst_machines sst_acpi_bytcr[] = { + {"10EC5640", "T100", "bytt100_rt5640", NULL, "fw_sst_0f28.bin", + &byt_rvp_platform_data }, + {}, +}; + +static const struct acpi_device_id sst_acpi_ids[] = { + { "80860F28", (unsigned long)&sst_acpi_bytcr}, + { }, +}; + +static struct platform_driver sst_acpi_driver = { + .driver = { + .name = "intel_sst_acpi", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(sst_acpi_ids), + .pm = &intel_sst_pm, + }, + .probe = sst_acpi_probe, + .remove = sst_acpi_remove, +}; + +module_platform_driver(sst_acpi_driver); + +MODULE_DESCRIPTION("Intel (R) SST(R) Audio Engine ACPI Driver"); +MODULE_AUTHOR("Ramesh Babu K V"); +MODULE_AUTHOR("Omair Mohammed Abdullah"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("sst"); diff --git a/sound/soc/intel/sst/sst_pvt.c b/sound/soc/intel/sst/sst_pvt.c index 9a5df19..4b77208 100644 --- a/sound/soc/intel/sst/sst_pvt.c +++ b/sound/soc/intel/sst/sst_pvt.c @@ -117,6 +117,7 @@ unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: val = sst_shim_read64(sst->shim, addr); break; } @@ -128,6 +129,7 @@ void write_shim_data(struct intel_sst_drv *sst, int addr, { switch (sst->dev_id) { case SST_MRFLD_PCI_ID: + case SST_BYT_ACPI_ID: sst_shim_write64(sst->shim, addr, (u64) data); break; } -- 1.7.0.4

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[alsa-devel] [PATCH v2 1/3] ALSA: ice1712: remove unneeded return statement
by Sudip Mukherjee 14 Nov '14

14 Nov '14

the functions: snd_ice1712_akm4xxx_build_controls snd_ice1712_build_pro_mixer snd_ctl_add snd_ak4114_build prodigy192_ak4114_init snd_ak4113_build are all returning either 0 or a negetive error value. so we can easily remove the check for a negative value and return the value instead. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- change in v2: returning the return value of the function directly. sound/pci/ice1712/hoontech.c | 6 +----- sound/pci/ice1712/ice1712.c | 19 ++++++------------- sound/pci/ice1712/ice1724.c | 7 ++----- sound/pci/ice1712/juli.c | 5 +---- sound/pci/ice1712/prodigy192.c | 4 +--- sound/pci/ice1712/quartet.c | 5 +---- 6 files changed, 12 insertions(+), 34 deletions(-) diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 59e37c5..a40001c 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -309,11 +309,7 @@ static int snd_ice1712_value_init(struct snd_ice1712 *ice) return err; /* ak4524 controls */ - err = snd_ice1712_akm4xxx_build_controls(ice); - if (err < 0) - return err; - - return 0; + return snd_ice1712_akm4xxx_build_controls(ice); } static int snd_ice1712_ez8_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 48a0c33..5975334 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1295,10 +1295,7 @@ static int snd_ice1712_pcm_profi(struct snd_ice1712 *ice, int device, struct snd return err; } - err = snd_ice1712_build_pro_mixer(ice); - if (err < 0) - return err; - return 0; + return snd_ice1712_build_pro_mixer(ice); } /* @@ -1545,10 +1542,9 @@ static int snd_ice1712_ac97_mixer(struct snd_ice1712 *ice) dev_warn(ice->card->dev, "cannot initialize ac97 for consumer, skipped\n"); else { - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, ice)); - if (err < 0) - return err; - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_digmix_route_ac97, + ice)); } } @@ -2497,11 +2493,8 @@ static int snd_ice1712_build_controls(struct snd_ice1712 *ice) err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_volume_rate, ice)); if (err < 0) return err; - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_ice1712_mixer_pro_peak, ice)); } static int snd_ice1712_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index f633e3b..ea53167 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2497,11 +2497,8 @@ static int snd_vt1724_build_controls(struct snd_ice1712 *ice) return err; } - err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); - if (err < 0) - return err; - - return 0; + return snd_ctl_add(ice->card, + snd_ctl_new1(&snd_vt1724_mixer_pro_peak, ice)); } static int snd_vt1724_free(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index 7a6c078..a1536c1 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -475,11 +475,8 @@ static int juli_add_controls(struct snd_ice1712 *ice) return err; /* only capture SPDIF over AK4114 */ - err = snd_ak4114_build(spec->ak4114, NULL, + return snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } /* diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 1eb151aa..3919aed 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -758,10 +758,8 @@ static int prodigy192_init(struct snd_ice1712 *ice) "AK4114 initialized with status %d\n", err); } else dev_dbg(ice->card->dev, "AK4114 not found\n"); - if (err < 0) - return err; - return 0; + return err; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index d4caf9d..6f55e02 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -833,11 +833,8 @@ static int qtet_add_controls(struct snd_ice1712 *ice) if (err < 0) return err; /* only capture SPDIF over AK4113 */ - err = snd_ak4113_build(spec->ak4113, + return snd_ak4113_build(spec->ak4113, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - if (err < 0) - return err; - return 0; } static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) -- 1.8.1.2

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