September 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/13] Add DSP topology management for SKL
by Subhransu S. Prusty 07 Oct '15

07 Oct '15

The SKL driver does not code DSP topology in driver. With this series the soc topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, add building blocks to instanatiate modules and pipes. It uses SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules, create pipeline and manage them. Last patch in this series removes the unused dais. Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Jeeja KP (12): ASoC: Intel: Skylake: Add helpers for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove CPU dai that is not used sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-messages.c | 751 +++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 173 ++-- sound/soc/intel/skylake/skl-topology.c | 1264 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 307 +++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 166 ++++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 8 files changed, 2636 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h -- 2.4.3

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[alsa-devel] [PATCH v2 0/8] ASoC: topology: ABI - Add support for PCM, BE & CC link
by mengdong.lin＠intel.com 07 Oct '15

07 Oct '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series includes topology ABI update to support PCM (FE DAI/DAI links), BE and CC DAI links. Bump the ABI version to 4. Mengdong Lin (2): ASoC: topology: ABI - Change stream formats to a bitwise flag ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream Vedang Patel (6): ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget ASoC: topology: ABI - Add name element to snd_soc_tplg_stream ASoC: topology: ABI - Add the type for BE DAI link ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links include/uapi/sound/asoc.h | 76 +++++++++++++++++++++++++---------------------- sound/soc/soc-topology.c | 4 +-- 2 files changed, 43 insertions(+), 37 deletions(-) -- 1.9.1

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[alsa-devel] Zoom R16/24 playback slient
by Ricard Wanderlof 07 Oct '15

07 Oct '15

Bringing up an old thread from 2014 since I've gotten access to a Zoom R16 and am keen on getting it to work in full duplex under Linux. Apparently (not yet verified here yet, rebuilding a kernel with the patch applied at the moment), 8 channel capture works with the quirk, but playback doesn't (well, it appears to work, but no sound comes out) and neither does the mixer. > At Sun, 30 Nov 2014 12:08:38 +0200, > Panu Matilainen wrote: > > > > On 11/29/2014 09:39 PM, Takashi Iwai wrote: > > > At Sat, 29 Nov 2014 21:35:04 +0200, > > > Panu Matilainen wrote: > > >> > > >> 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. > > > > [ ... ] > > > > In the meanwhile I started looking at the mixer - this is just guessing > > but maybe its starting with outputs muted which could explain the > > playback stream running but without sound. > > > > The descriptors for interface 0 are garbage but what is there would seem > > to map to logical elements by subtype and terminaltype, after those the> > data doesn't seem to add up (eg number of input/output channels > > missing/wrongly placed etc): > > > > ** UNRECOGNIZED: 0b 24 01 00 01 35 00 03 01 02 03 > > -> Header? > > ** UNRECOGNIZED: 0c 24 02 05 01 01 00 02 03 00 00 00 > > -> USB streaming input terminal? > > ** UNRECOGNIZED: 09 24 03 08 01 03 00 05 00 > > -> Speaker output terminal? > > ** UNRECOGNIZED: 0c 24 02 09 01 02 00 08 00 00 00 00 > > -> Microphone input terminal? > > ** UNRECOGNIZED: 09 24 03 0c 01 01 00 09 00 > > -> USB streaming output terminal? > > > > Anything there that might ring a bell, or other ideas where to start > > poking at this thing? > > Yeah, most of these look almost sane, so any off-by-one firmware > issue? Searching the archives, I could find no further posts on the subject of the R16. Panu, did you do any more work on this? Where does the above output come from, and where could I find information on what type of data is expected? I assume what's expected is something one would get from an USB mixer device? In an earlier incarnation of a patch for the R16 (on the 10th of March 2014), there's a discussion between Jason Mancine and Takashi regarding the .format field in the quirks table (http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074117.html) which didn't seem to reach a conclusion; the problem then seemed to be that even when setting the .format to SNDRV_PCM_FMTBIT_S24_3LE, ALSA still setup up a 32 bit transfer. However, Panu's patch also added a small delay in snd_usb_ctl_msg_quirk(), so that perhaps was the real solution to the alleged format problem? (The final patch is much smaller than in this particular thread, possibly due to the use of QUIRK_ANY_INTERFACE since kernel version >3.11 ?) /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH 0/5] Add ASoC support for AMD APUs [v4]
by Alex Deucher 06 Oct '15

06 Oct '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL V4 changes: - get the master vs slave mode for the designware codec from registers - copy designware quirks from platform data in local struct Patch 4 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream4 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (4): ASoC : dwc : support dw i2s in slave mode ASoC : dwc : support dw i2s in AMD platform ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 269 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 5 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 276 ++-- 24 files changed, 6146 insertions(+), 102 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

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[alsa-devel] [PATCH RFC] ASoC: simple-card: Update clocks binding for simple-card DAI subnodes
by Jyri Sarha 06 Oct '15

06 Oct '15

The updated binding provides a way to set clock-ID and direction parameters for DAI drivers set_sysclk() call back. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I proposed something similar about a year ago, but Mark rejected that at the time. This RFC is to start that discussion again. This time before I do any code changes. Best regards, Jyri Documentation/devicetree/bindings/sound/simple-card.txt | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index cf3979e..d10bf2d 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -76,6 +76,7 @@ Optional CPU/CODEC subnodes properties: - dai-tdm-slot-num : Please refer to tdm-slot.txt. - dai-tdm-slot-width : Please refer to tdm-slot.txt. - clocks / system-clock-frequency : specify subnode's clock if needed. + it can be specified via "clocks" if system has clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) @@ -83,7 +84,21 @@ Optional CPU/CODEC subnodes properties: enabled with clk_prepare_enable() in dai startup() and disabled with clk_disable_unprepare() in dai - shutdown(). + shutdown(). "system-clock-frequency" + can also be an array if more than one + clock is described. +- clock-ids : An array of clock ID integers, + preferrably defined in DT header. + Each entry corresponds to the same + index postion first in "clocks" and + after the end of clocks array to + "system-clock-frequency" array. +- clock-dirs : An array of integers describing + clock directions: CLK_IN (= 0) or + OUT (= 1). Entries in the array + refer to clocks in the same way as + in clock-ids property. + Example 1 - single DAI link: -- 1.9.1

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[alsa-devel] [PATCH 1/4] ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver
by Zidan Wang 05 Oct '15

05 Oct '15

Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume be called in suspend/resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/imx-spdif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 33da26a..a407e83 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -89,6 +89,7 @@ MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); static struct platform_driver imx_spdif_driver = { .driver = { .name = "imx-spdif", + .pm = &snd_soc_pm_ops, .of_match_table = imx_spdif_dt_ids, }, .probe = imx_spdif_audio_probe, -- 1.9.1

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[alsa-devel] [PATCH v3 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 05 Oct '15

05 Oct '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

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[alsa-devel] [PATCH linux-next v3 1/1] ASoC: ad193x: add support to ad1934
by Cyrille Pitchen 05 Oct '15

05 Oct '15

The AD1934 codec has no DAC feature. Hence it register mapping is slightly different from the register mapping of other members of the AD193x family. Some ASoC controls and widgets are related to the DAC feature so are not relevant in the case of an AD1934 codec. Signed-off-by: Cyrille Pitchen <cyrille.pitchen(a)atmel.com> --- Hi all, This patch adds support to the AD1934 codec, which is DAC only. Best Regards, Cyrille ChangeLog v3: - rebase patch for linux-next: don't use the CODEC's dapm field, which no longer exists, but call snd_soc_codec_get_dapm() instead. v2: - replace AD193x by AD193X in ad193x-i2c.c - add ad193x_ prefix to adc_audio_paths --- sound/soc/codecs/ad193x-i2c.c | 8 +-- sound/soc/codecs/ad193x-spi.c | 16 +++++- sound/soc/codecs/ad193x.c | 122 ++++++++++++++++++++++++++++++++---------- sound/soc/codecs/ad193x.h | 9 +++- 4 files changed, 121 insertions(+), 34 deletions(-) diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c index df3a1a415825..171313664bc8 100644 --- a/sound/soc/codecs/ad193x-i2c.c +++ b/sound/soc/codecs/ad193x-i2c.c @@ -15,8 +15,8 @@ #include "ad193x.h" static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, { } }; MODULE_DEVICE_TABLE(i2c, ad193x_id); @@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client, config.val_bits = 8; config.reg_bits = 8; - return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); + return ad193x_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index 390cef9b9dc2..364e82112e8c 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -16,6 +16,7 @@ static int ad193x_spi_probe(struct spi_device *spi) { + const struct spi_device_id *id = spi_get_device_id(spi); struct regmap_config config; config = ad193x_regmap_config; @@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi) config.read_flag_mask = 0x09; config.write_flag_mask = 0x08; - return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config), + (enum ad193x_type)id->driver_data); } static int ad193x_spi_remove(struct spi_device *spi) @@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi) return 0; } +static const struct spi_device_id ad193x_spi_id[] = { + { "ad193x", AD193X }, + { "ad1933", AD1933 }, + { "ad1934", AD1934 }, + { "ad1936", AD193X }, + { "ad1937", AD193X }, + { "ad1938", AD193X }, + { } +}; +MODULE_DEVICE_TABLE(spi, ad193x_spi_id); + static struct spi_driver ad193x_spi_driver = { .driver = { .name = "ad193x", @@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = { }, .probe = ad193x_spi_probe, .remove = ad193x_spi_remove, + .id_table = ad193x_spi_id, }; module_spi_driver(ad193x_spi_driver); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 17c953595660..76d7966bd72c 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -23,6 +23,7 @@ /* codec private data */ struct ad193x_priv { struct regmap *regmap; + enum ad193x_type type; int sysclk; }; @@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL, AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv), - /* ADC switch control */ - SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, - AD193X_ADCR1_MUTE, 1, 1), - SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, - AD193X_ADCR2_MUTE, 1, 1), - /* DAC switch control */ SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE, AD193X_DACR1_MUTE, 1, 1), @@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = { SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE, AD193X_DACR4_MUTE, 1, 1), + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), +}; + +static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = { + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE, + AD193X_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE, + AD193X_ADCR2_MUTE, 1, 1), + /* ADC high-pass filter */ SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0, AD193X_ADC_HIGHPASS_FILTER, 1, 0), - - /* DAC de-emphasis */ - SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum), }; static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0), SND_SOC_DAPM_VMID("VMID"), SND_SOC_DAPM_OUTPUT("DAC1OUT"), SND_SOC_DAPM_OUTPUT("DAC2OUT"), SND_SOC_DAPM_OUTPUT("DAC3OUT"), SND_SOC_DAPM_OUTPUT("DAC4OUT"), +}; + +static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0), SND_SOC_DAPM_INPUT("ADC1IN"), SND_SOC_DAPM_INPUT("ADC2IN"), }; @@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "SYSCLK" }, { "DAC Output", NULL, "DAC" }, { "DAC Output", NULL, "VMID" }, - { "ADC", NULL, "SYSCLK" }, - { "DAC", NULL, "ADC_PWR" }, - { "ADC", NULL, "ADC_PWR" }, { "DAC1OUT", NULL, "DAC Output" }, { "DAC2OUT", NULL, "DAC Output" }, { "DAC3OUT", NULL, "DAC Output" }, { "DAC4OUT", NULL, "DAC Output" }, + { "SYSCLK", NULL, "PLL_PWR" }, +}; + +static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = { + { "ADC", NULL, "SYSCLK" }, + { "ADC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC1IN" }, { "ADC", NULL, "ADC2IN" }, - { "SYSCLK", NULL, "PLL_PWR" }, }; +static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x) +{ + switch (ad193x->type) { + case AD1933: + case AD1934: + return false; + default: + break; + } + + return true; +} + /* * DAI ops entries */ @@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_CHAN_MASK, + channels << AD193X_ADC_CHAN_SHFT); return 0; } @@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, adc_serfmt |= AD193X_ADC_SERFMT_AUX; break; default: - return -EINVAL; + if (ad193x_has_adc(ad193x)) + return -EINVAL; + break; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_SERFMT_MASK, adc_serfmt); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, - AD193X_ADC_FMT_MASK, adc_fmt); + if (ad193x_has_adc(ad193x)) { + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_SERFMT_MASK, adc_serfmt); + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2, + AD193X_ADC_FMT_MASK, adc_fmt); + } regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1, AD193X_DAC_FMT_MASK, dac_fmt); @@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, AD193X_DAC_WORD_LEN_MASK, word_len << AD193X_DAC_WORD_LEN_SHFT); - regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, - AD193X_ADC_WORD_LEN_MASK, word_len); + if (ad193x_has_adc(ad193x)) + regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1, + AD193X_ADC_WORD_LEN_MASK, word_len); return 0; } @@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + int num, ret; /* default setting for ad193x */ @@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); /* dac in tdm mode */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); - /* high-pass filter enable */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); - /* sata delay=1, adc aux mode */ - regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* high-pass filter enable */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); + /* sata delay=1, adc aux mode */ + regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43); + } + /* pll input: mclki/xi */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); + /* adc only */ + if (ad193x_has_adc(ad193x)) { + /* add adc controls */ + num = ARRAY_SIZE(ad193x_adc_snd_controls); + ret = snd_soc_add_codec_controls(codec, + ad193x_adc_snd_controls, + num); + if (ret) + return ret; + + /* add adc widgets */ + num = ARRAY_SIZE(ad193x_adc_widgets); + ret = snd_soc_dapm_new_controls(dapm, + ad193x_adc_widgets, + num); + if (ret) + return ret; + + /* add adc routes */ + num = ARRAY_SIZE(ad193x_adc_audio_paths); + ret = snd_soc_dapm_add_routes(dapm, + ad193x_adc_audio_paths, + num); + if (ret) + return ret; + } + return 0; } @@ -367,7 +429,8 @@ const struct regmap_config ad193x_regmap_config = { }; EXPORT_SYMBOL_GPL(ad193x_regmap_config); -int ad193x_probe(struct device *dev, struct regmap *regmap) +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type) { struct ad193x_priv *ad193x; @@ -379,6 +442,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap) return -ENOMEM; ad193x->regmap = regmap; + ad193x->type = type; dev_set_drvdata(dev, ad193x); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index ab9a998f15be..8b1e65f928d2 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -13,8 +13,15 @@ struct device; +enum ad193x_type { + AD193X, + AD1933, + AD1934, +}; + extern const struct regmap_config ad193x_regmap_config; -int ad193x_probe(struct device *dev, struct regmap *regmap); +int ad193x_probe(struct device *dev, struct regmap *regmap, + enum ad193x_type type); #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 -- 1.8.2.2

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[alsa-devel] [PATCH v2 0/2] ASoC: add driver for Atmel I2S controller
by Cyrille Pitchen 05 Oct '15

05 Oct '15

This series of patches adds support to the new Atmel I2S controller embedded on sama5d2 SoCs. ChangeLog v2: - initialize dev->dev before calling dev->caps->mck_init(). Cyrille Pitchen (2): ASoC: atmel-i2s: add DT bindings for I2S controller ASoC: atmel-i2s: add driver for the new Atmel I2S controller .../devicetree/bindings/sound/atmel-i2s.txt | 43 ++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-i2s.c | 760 +++++++++++++++++++++ 4 files changed, 815 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt create mode 100644 sound/soc/atmel/atmel-i2s.c -- 1.8.2.2

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[alsa-devel] Missing return check of of_property_read_*()
by Takashi Iwai 05 Oct '15

05 Oct '15

Hi, I hoped that I wouldn't need to write this kind of post, but since nothing happened so far, so here we go: The following compile warnings are present for quite some time. They are due to the lack of return check from of_property_read*(). sound/soc/codecs/cs35l32.c: In function ‘cs35l32_i2c_probe’: sound/soc/codecs/cs35l32.c:278:2: warning: ‘val’ may be used uninitialized in this function [-Wmaybe-uninitialized] switch (val) { ^ sound/soc/codecs/cs35l32.c:272:15: note: ‘val’ was declared here unsigned int val; ^ sound/soc/sti/uniperif_player.c: In function ‘uni_player_init’: sound/soc/sti/uniperif_player.c:1006:6: warning: ‘mode’ may be used uninitialized in this function [-Wmaybe-uninitialized] if (strcasecmp(mode, "hdmi") == 0) ^ sound/soc/sti/uniperif_player.c:985:14: note: ‘mode’ was declared here const char *mode; ^ Both should be trivial to fix, but I won't bother to fix these things by myself, as it results always in a matter of taste how to write. So, please fix them appropriately. Although not shown as compile warning like the above, lots of codec drivers have no return value checks from of_property_read*(), too. A quick grep reveals: % git grep -B1 '^[[:space:]]*of_property_read_' sound/soc sound/soc/codecs/adau1701.c- sound/soc/codecs/adau1701.c: of_property_read_u32(dev->of_node, "adi,pll-clkdiv", -- sound/soc/codecs/adau1701.c- sound/soc/codecs/adau1701.c: of_property_read_u8_array(dev->of_node, "adi,pin-config", -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,boost-manager", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,sdout-datacfg", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,battery-threshold", &val); -- sound/soc/codecs/cs35l32.c- sound/soc/codecs/cs35l32.c: of_property_read_u32(np, "cirrus,battery-recovery", &val); -- sound/soc/codecs/sta32x.c- sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,output-conf", sound/soc/codecs/sta32x.c- &pdata->output_conf); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch1-output-mapping", sound/soc/codecs/sta32x.c- &pdata->ch1_output_mapping); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch2-output-mapping", sound/soc/codecs/sta32x.c- &pdata->ch2_output_mapping); sound/soc/codecs/sta32x.c: of_property_read_u8(np, "st,ch3-output-mapping", -- sound/soc/codecs/sta32x.c- tmp = 140; sound/soc/codecs/sta32x.c: of_property_read_u16(np, "st,drop-compensation-ns", &tmp); -- sound/soc/codecs/sta350.c- sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,output-conf", sound/soc/codecs/sta350.c- &pdata->output_conf); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch1-output-mapping", sound/soc/codecs/sta350.c- &pdata->ch1_output_mapping); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch2-output-mapping", sound/soc/codecs/sta350.c- &pdata->ch2_output_mapping); sound/soc/codecs/sta350.c: of_property_read_u8(np, "st,ch3-output-mapping", -- sound/soc/codecs/sta350.c- tmp = 140; sound/soc/codecs/sta350.c: of_property_read_u16(np, "st,drop-compensation-ns", &tmp); -- sound/soc/codecs/tas5086.c- sound/soc/codecs/tas5086.c: of_property_read_u32(of_node, "ti,charge-period", -- sound/soc/codecs/tlv320aic31xx.c- sound/soc/codecs/tlv320aic31xx.c: of_property_read_u32(np, "ai31xx-micbias-vg", &value); -- sound/soc/codecs/twl4030.c- sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,digimic_delay", sound/soc/codecs/twl4030.c- &pdata->digimic_delay); sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,ramp_delay_value", sound/soc/codecs/twl4030.c- &pdata->ramp_delay_value); sound/soc/codecs/twl4030.c: of_property_read_u32(node, "ti,offset_cncl_path", -- sound/soc/fsl/fsl_esai.c- esai_priv->synchronous = sound/soc/fsl/fsl_esai.c: of_property_read_bool(np, "fsl,esai-synchronous"); -- sound/soc/omap/omap-abe-twl6040.c- priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); sound/soc/omap/omap-abe-twl6040.c: of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); -- sound/soc/samsung/i2s.c- } sound/soc/samsung/i2s.c: of_property_read_string_index(dev->of_node, -- sound/soc/sh/rcar/rsrc-card.c- /* sampling rate convert */ sound/soc/sh/rcar/rsrc-card.c: of_property_read_u32(node, "convert-rate", &priv->convert_rate); -- sound/soc/sti/uniperif_player.c- sound/soc/sti/uniperif_player.c: of_property_read_u32(pnode, "version", &player->ver); -- sound/soc/sti/uniperif_player.c- sound/soc/sti/uniperif_player.c: of_property_read_u32(pnode, "uniperiph-id", &info->id); -- sound/soc/sti/uniperif_player.c- /* Read the device mode property */ sound/soc/sti/uniperif_player.c: of_property_read_string(pnode, "mode", &mode); -- sound/soc/sti/uniperif_reader.c- sound/soc/sti/uniperif_reader.c: of_property_read_u32(node, "version", &reader->ver); Many of them look really buggy (while some have the default value to fall back). And many codes have even no value checks. These should be fixed as well. thanks, Takashi

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