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

[alsa-devel] Remove clkdiv and pll setters from pxa dais
by Daniel Mack 02 Jun '18

02 Jun '18

Hi, As discussed in the simple-card thread the other day, the .set_clkdiv and .set_pll callbacks for DAIs are considered legacy. In order to support PXA platforms with DT, we need get rid of those for the DAIs of PXA hardware. On the good side, pxa-ssp is the only problematic one of the PXA dais, the ac97, i2s don't have any special clocking knbos, and the mmp-sspa should be straight forward to fix. I looked into pxa-ssp and it seems the conversion to calculate the audio clock dividers automatically seems doable. There are currently 4 platforms that use this cpu dai in mainline that also use snd_soc_dai_set_pll() or snd_soc_dai_set_clkdiv(): * Zylonite * Brownstone * Magician * Raumfeld I can test modifications on the latter one, but for the others, I need to understand what the machine drivers are doing, and there are some bits I don't grok. For instance, code for Zylonite does this: /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; The commit which introduced these lines dates back to 2009 and was done by you, Mark. Can you remember what the reason for this was? I've never seen sample frames with 17 bits :) This is a setup that we can't generically describe through .hw_params() or .dai_fmt() in the cpu dai, correct? In the magician code, you'll find this: case 22050: acps = 5622000; switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; case 44100: acps = 5622000; switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; So for both 22050 and 44100, the base frequency and all dividers are the same, which can't be right. I assume these rates have never been used. I'll ignore this and implement the table in the datasheet which should do the right thing. Philipp? What we need, however, is a way to describe whether the dai is mclk master or slave. Would we add a DT propery for that? Thanks, Daniel

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[alsa-devel] [PATCH] DMA: OMAP: fix OMAP1510 incorrect residue_granularity
by Janusz Krzysztofik 02 Jun '18

02 Jun '18

Commit 0198d7bb8a0c ("ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm") resulted in broken audio playback on OMAP1510 (discovered on Amstrad Delta). When running on OMAP1510, omap-pcm used to obtain DMA offset from snd_dmaengine_pcm_pointer_no_residue() based on DMA interrupt triggered software calculations instead of snd_dmaengine_pcm_pointer() which depended on residue value calculated from omap_dma_get_src_pos(). Similar code path is still available in now used sound/soc/soc-generic-dmaengine-pcm.c but it is not triggered. It was verified already before that omap_get_dma_src_pos() from arch/arm/plat-omap/dma.c didn't work correctly for OMAP1510 - see commit 1bdd7419910c ("ASoC: OMAP: fix OMAP1510 broken PCM pointer callback") for details. Apparently the same applies to its successor, omap_dma_get_src_pos() from drivers/dma/omap-dma.c. On the other hand, snd_dmaengine_pcm_pointer_no_residue() is described as depreciated and discouraged for use in new drivers because of its unreliable accuracy. However, it seems the only working option for OPAM1510 now, as long as a software calculated residue is not implemented as OMAP1510 fallback in omap-dma. Using snd_dmaengine_pcm_pointer_no_residue() code path instead of snd_dmaengine_pcm_pointer() in sound/soc/soc-generic-dmaengine-pcm.c can be triggered in two ways: - by passing pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE from sound/soc/omap/sdma-pcm.c, - by passing dma_caps.residue_granularity = DMA_RESIDUE_GRANULARITY_DESCRIPTOR from drivers/dma/omap-dma.c. Let's do the latter. Created and tested against next-20180531 tag from linux-next tree. Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- drivers/dma/omap-dma.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/drivers/dma/omap-dma.c b/drivers/dma/omap-dma.c index d21c19822feb..56399bd45179 100644 --- a/drivers/dma/omap-dma.c +++ b/drivers/dma/omap-dma.c @@ -1485,7 +1485,11 @@ static int omap_dma_probe(struct platform_device *pdev) od->ddev.src_addr_widths = OMAP_DMA_BUSWIDTHS; od->ddev.dst_addr_widths = OMAP_DMA_BUSWIDTHS; od->ddev.directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); - od->ddev.residue_granularity = DMA_RESIDUE_GRANULARITY_BURST; + if (__dma_omap15xx(od->plat->dma_attr)) + od->ddev.residue_granularity = + DMA_RESIDUE_GRANULARITY_DESCRIPTOR; + else + od->ddev.residue_granularity = DMA_RESIDUE_GRANULARITY_BURST; od->ddev.max_burst = SZ_16M - 1; /* CCEN: 24bit unsigned */ od->ddev.dev = &pdev->dev; INIT_LIST_HEAD(&od->ddev.channels); -- 2.16.1

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[alsa-devel] [PATCH] ALSA: pci/hda: Remove unused, broken, header file
by Ben Hutchings 01 Jun '18

01 Jun '18

sound/pci/hda/local.h seems to be an earlier version of sound/hda/local.h; it was added at the same time but doesn't seem to have ever been used (within the git history). Most of its macros depend on a hdac_read_parm() function which is not defined anywhere. Signed-off-by: Ben Hutchings <ben.hutchings(a)codethink.co.uk> --- sound/pci/hda/local.h | 40 ---------------------------------------- 1 file changed, 40 deletions(-) delete mode 100644 sound/pci/hda/local.h diff --git a/sound/pci/hda/local.h b/sound/pci/hda/local.h deleted file mode 100644 index 3b8b7d78f9e0..000000000000 --- a/sound/pci/hda/local.h +++ /dev/null @@ -1,40 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - */ - -#ifndef __HDAC_LOCAL_H -#define __HDAC_LOCAL_H - -int hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); - -#define get_wcaps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) -/* get the widget type from widget capability bits */ -static inline int get_wcaps_type(unsigned int wcaps) -{ - if (!wcaps) - return -1; /* invalid type */ - return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; -} - -#define get_pin_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_PIN_CAP) - -static inline -unsigned int get_pin_cfg(struct hdac_device *codec, hda_nid_t nid) -{ - unsigned int val; - - if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) - return -1; - return val; -} - -#define get_amp_caps(codec, nid, dir) \ - hdac_read_parm(codec, nid, (dir) == HDA_OUTPUT ? \ - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP) - -#define get_power_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_POWER_STATE) - -#endif /* __HDAC_LOCAL_H */ -- Ben Hutchings, Software Developer Codethink Ltd https://www.codethink.co.uk/ Dale House, 35 Dale Street Manchester, M1 2HF, United Kingdom

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[alsa-devel] [PATCH v2] ASoC: simple-card: set cpu dai clk in hw_params
by Daniel Mack 01 Jun '18

01 Jun '18

The simple-card driver currently accepts a clock node in the cpu dai sub-node and only uses it as an alternative to the 'system-clock-frequency' property to get the current frequency. This patch adds another use of the passed clock node. If mclk-fs is specified, the clocks in cpu and codec dai sub-nodes will be set to the calculated rate (stream rate * mclk_fs) in hw_params. This allows platforms to pass tuneable clocks as phandle that will automatically be set to the right rates. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- This is the only patch that's left from the series of three that I posted earlier. v1 → v2: * Apply the setting to both the codec and the cpu dai * Check for errors from clk_set_rate() and avoid unnecessary rate changes. .../devicetree/bindings/sound/simple-card.txt | 5 +++++ sound/soc/generic/simple-card.c | 21 +++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 17c13e74667d..a4c72d09cd45 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -86,6 +86,11 @@ Optional CPU/CODEC subnodes properties: in dai startup() and disabled with clk_disable_unprepare() in dai shutdown(). + If a clock is specified and a + multiplication factor is given with + mclk-fs, the clock will be set to the + calculated mclk frequency when the + stream starts. - system-clock-direction-out : specifies clock direction as 'out' on initialization. It is useful for some aCPUs with fixed clocks. diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6959a74a6f49..4a516c428b3d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -135,6 +135,18 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) asoc_simple_card_clk_disable(&dai_props->codec_dai); } +static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, + unsigned long rate) +{ + if (!simple_dai->clk) + return 0; + + if (clk_get_rate(simple_dai->clk) == rate) + return 0; + + return clk_set_rate(simple_dai->clk, rate); +} + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -154,6 +166,15 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, if (mclk_fs) { mclk = params_rate(params) * mclk_fs; + + ret = asoc_simple_set_clk_rate(&dai_props->codec_dai, mclk); + if (ret < 0) + return ret; + + ret = asoc_simple_set_clk_rate(&dai_props->cpu_dai, mclk); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) -- 2.14.3

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[alsa-devel] [PATCH] ASoC: Intel: kbl: Move codec sysclk config to codec_init function
by mac.chiang＠intel.com 01 Jun '18

01 Jun '18

From: Mac Chiang <mac.chiang(a)intel.com> unexpected error: PLL input clock 0 below valid range backport from broonie's tree commit fd0f237572ad ("ASoC: Intel: bxt: Move codec sysclk config to codec_init function") The MCLK for DA7219 does not change in this platform, but is currently being configured everytime as part of the platform_clock event handler for DAPM. The upshot of this is that we have unnecessary calls to this function, and it also means that if a stream hasn't yet been started, DA7219 driver does not have the correct MCLK rates programmed and so the HP detection feature does not operate as expected. This patch rectifies this issue by moving the sysclk call to codec_init function so it's only called once at initialisation. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Acked-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Singed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 60e739f..88fbc6a 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -65,14 +65,6 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(card->dev, "can't set codec sysclk configuration\n"); - return ret; - } - if (SND_SOC_DAPM_EVENT_OFF(event)) { ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); @@ -169,9 +161,18 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_jack *jack; int ret; + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + /* * Headset buttons map to the google Reference headset. * These can be configured by userspace. -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] slimbus: Add QCOM SLIMBus NGD driver
by Srinivas Kandagatla 01 Jun '18

01 Jun '18

This patchset adds support to basic version of Qualcomm NGD SLIMBus controller driver found SoCs from B family. This controller is light-weight SLIMBus controller driver responsible for communicating with slave HW directly over the bus using messaging interface, and communicating with master component residing on ADSP for bandwidth and data-channel management. Some parts fot the work is derived from Qualcomm downstream kernels and intial work from Karthikeyan Ramasubramanian <kramasub(a)codeaurora.org> and Sagar Dharia <sdharia(a)codeaurora.org> Tested this patchset on DB820c with WCD9335 codec and on msm8974 with wcd9320 codec. I have pushed my working branch to [1] incase someone want to try. This patch has dependency on https://lkml.org/lkml/2018/5/17/251 Changes sinc v1 (https://lkml.org/lkml/2018/5/25/296): - Added Tested-by and Reviewed-by Tags - Added Credits to Qualcomm engineers in commit log - Fixed a spelling mistake for prepare. Thanks, srini [1] https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimbus-… Srinivas Kandagatla (2): slimbus: ngd: dt-bindings: Add slim ngd dt bindings slimbus: ngd: Add qcom SLIMBus NGD driver .../bindings/slimbus/slim-ngd-qcom-ctrl.txt | 84 ++ drivers/slimbus/Kconfig | 11 + drivers/slimbus/Makefile | 3 + drivers/slimbus/qcom-ngd-ctrl.c | 1342 ++++++++++++++++++++ drivers/slimbus/slimbus.h | 8 + 5 files changed, 1448 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt create mode 100644 drivers/slimbus/qcom-ngd-ctrl.c -- 2.16.2

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[alsa-devel] [PATCH] ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
by Bo Chen 01 Jun '18

01 Jun '18

When 'kzalloc()' fails in 'snd_hda_attach_pcm_stream()', a new pcm instance is created without setting its operators via 'snd_pcm_set_ops()'. Following operations on the new pcm instance can trigger kernel null pointer dereferences and cause kernel oops. This bug was found with my work on building a gray-box fault-injection tool for linux-kernel-module binaries. A kernel null pointer dereference was confirmed from line 'substream->ops->open()' in function 'snd_pcm_open_substream()' in file 'sound/core/pcm_native.c'. This patch fixes the bug by calling 'snd_device_free()' in the error handling path of 'kzalloc()', which removes the new pcm instance from the snd card before returns with an error code. Signed-off-by: Bo Chen <chenbo(a)pdx.edu> --- sound/pci/hda/hda_controller.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index d1eb14842340..a12e594d4e3b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -748,8 +748,10 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, return err; strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (apcm == NULL) + if (apcm == NULL) { + snd_device_free(chip->card, pcm); return -ENOMEM; + } apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; -- 2.17.0

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[alsa-devel] [PATCH] ASoC: AMD: Configure channel 1 or channel 0 for capture
by Akshu Agrawal 01 Jun '18

01 Jun '18

ST/CZ SoC have 2 channels for capture in the I2SSP path. The DMA though these channels is done using the same dma descriptors. We configure the channel and enable it on the basis of channel selected by machine driver. Machine driver knows which codec sits on which channel and thus sends the information to dma driver. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- This patch is dependent on ASoC: AMD: Change codec to channel link as per hardware redesign https://patchwork.kernel.org/patch/10388099/ sound/soc/amd/acp-da7219-max98357a.c | 43 +++++++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 79 +++++++++++++++++++++++++++++++++++- sound/soc/amd/acp.h | 4 ++ 3 files changed, 119 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 566bd26..f42606e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -149,6 +149,7 @@ static int cz_da7219_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL1; return da7219_clk_enable(substream); } @@ -172,7 +173,7 @@ static void cz_max_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } -static int cz_dmic_startup(struct snd_pcm_substream *substream) +static int cz_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; @@ -182,6 +183,17 @@ static int cz_dmic_startup(struct snd_pcm_substream *substream) return da7219_clk_enable(substream); } +static int cz_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->i2s_instance = I2S_SP_INSTANCE; + machine->capture_channel = CAP_CHANNEL0; + return da7219_clk_enable(substream); +} + static void cz_dmic_shutdown(struct snd_pcm_substream *substream) { da7219_clk_disable(); @@ -197,8 +209,13 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .shutdown = cz_max_shutdown, }; -static const struct snd_soc_ops cz_dmic_cap_ops = { - .startup = cz_dmic_startup, +static const struct snd_soc_ops cz_dmic0_cap_ops = { + .startup = cz_dmic0_startup, + .shutdown = cz_dmic_shutdown, +}; + +static const struct snd_soc_ops cz_dmic1_cap_ops = { + .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, }; @@ -241,8 +258,9 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .ops = &cz_max_play_ops, }, { - .name = "dmic", - .stream_name = "DMIC Capture", + /* C panel DMIC */ + .name = "dmic0", + .stream_name = "DMIC0 Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "adau7002-hifi", @@ -250,7 +268,20 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, - .ops = &cz_dmic_cap_ops, + .ops = &cz_dmic0_cap_ops, + }, + { + /* A/B panel DMIC */ + .name = "dmic1", + .stream_name = "DMIC1 Capture", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "adau7002-hifi", + .codec_name = "ADAU7002:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_capture = 1, + .ops = &cz_dmic1_cap_ops, }, }; diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 7720384..ad85354 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -336,6 +336,68 @@ static void config_acp_dma(void __iomem *acp_mmio, rtd->dma_dscr_idx_2, asic_type); } +static void acp_dma_cap_channel(void __iomem *acp_mmio, u16 cap_channel, + bool enable) +{ + u32 val; + + if (enable) { + switch (cap_channel) { + case CAP_CHANNEL1: + val = acp_reg_read(acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); + if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { + acp_reg_write(0x0, acp_mmio, + mmACP_I2SMICSP_RER1); + /* Set 16bit resolution on capture */ + acp_reg_write(0x2, acp_mmio, + mmACP_I2SMICSP_RCR1); + } + val = acp_reg_read(acp_mmio, mmACP_I2SMICSP_IMR1); + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val &= ~ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, mmACP_I2SMICSP_IMR1); + acp_reg_write(0x1, acp_mmio, mmACP_I2SMICSP_RER1); + break; + case CAP_CHANNEL0: + default: + val = acp_reg_read(acp_mmio, + mmACP_I2S_16BIT_RESOLUTION_EN); + if (val & ACP_I2S_MIC_16BIT_RESOLUTION_EN) { + acp_reg_write(0x0, acp_mmio, + mmACP_I2SMICSP_RER0); + /* Set 16bit resolution on capture */ + acp_reg_write(0x2, acp_mmio, + mmACP_I2SMICSP_RCR0); + } + val = acp_reg_read(acp_mmio, mmACP_I2SMICSP_IMR0); + val &= ~ACP_I2SMICSP_IMR0__I2SMICSP_RXDAM_MASK; + val &= ~ACP_I2SMICSP_IMR0__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, mmACP_I2SMICSP_IMR0); + acp_reg_write(0x1, acp_mmio, mmACP_I2SMICSP_RER0); + break; + } + } else { + switch (cap_channel) { + case CAP_CHANNEL1: + val = acp_reg_read(acp_mmio, mmACP_I2SMICSP_IMR1); + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXDAM_MASK; + val |= ACP_I2SMICSP_IMR1__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, mmACP_I2SMICSP_IMR1); + acp_reg_write(0x0, acp_mmio, mmACP_I2SMICSP_RER1); + break; + case CAP_CHANNEL0: + default: + val = acp_reg_read(acp_mmio, mmACP_I2SMICSP_IMR0); + val |= ACP_I2SMICSP_IMR0__I2SMICSP_RXDAM_MASK; + val |= ACP_I2SMICSP_IMR0__I2SMICSP_RXFOM_MASK; + acp_reg_write(val, acp_mmio, mmACP_I2SMICSP_IMR0); + acp_reg_write(0x0, acp_mmio, mmACP_I2SMICSP_RER0); + break; + } + } +} + /* Start a given DMA channel transfer */ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) { @@ -773,7 +835,10 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; - rtd->i2s_instance = pinfo->i2s_instance; + if (pinfo) { + rtd->i2s_instance = pinfo->i2s_instance; + rtd->capture_channel = pinfo->capture_channel; + } if (adata->asic_type == CHIP_STONEY) { val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); @@ -989,6 +1054,18 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_start(rtd->acp_mmio, rtd->ch1); acp_dma_start(rtd->acp_mmio, rtd->ch2); } else { + if (rtd->capture_channel == CAP_CHANNEL0) { + acp_dma_cap_channel(rtd->acp_mmio, + CAP_CHANNEL1, false); + acp_dma_cap_channel(rtd->acp_mmio, + CAP_CHANNEL0, true); + } + if (rtd->capture_channel == CAP_CHANNEL1) { + acp_dma_cap_channel(rtd->acp_mmio, + CAP_CHANNEL0, false); + acp_dma_cap_channel(rtd->acp_mmio, + CAP_CHANNEL1, true); + } acp_dma_start(rtd->acp_mmio, rtd->ch2); acp_dma_start(rtd->acp_mmio, rtd->ch1); } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96..3190fdc 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -55,6 +55,8 @@ #define I2S_SP_INSTANCE 0x01 #define I2S_BT_INSTANCE 0x02 +#define CAP_CHANNEL0 0x00 +#define CAP_CHANNEL1 0x01 #define ACP_TILE_ON_MASK 0x03 #define ACP_TILE_OFF_MASK 0x02 @@ -125,6 +127,7 @@ struct audio_substream_data { unsigned int order; u16 num_of_pages; u16 i2s_instance; + u16 capture_channel; u16 direction; u16 ch1; u16 ch2; @@ -155,6 +158,7 @@ struct audio_drv_data { */ struct acp_platform_info { u16 i2s_instance; + u16 capture_channel; }; union acp_dma_count { -- 1.9.1

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[alsa-devel] [PATCH v2 0/2] slimbus: Add QCOM SLIMBus NGD driver
by Srinivas Kandagatla 01 Jun '18

01 Jun '18

This patchset adds support to basic version of Qualcomm NGD SLIMBus controller driver found SoCs from B family. This controller is light-weight SLIMBus controller driver responsible for communicating with slave HW directly over the bus using messaging interface, and communicating with master component residing on ADSP for bandwidth and data-channel management. Tested this patchset on DB820c with WCD9335 codec. I have pushed my working branch to [1] incase someone want to try. This patch has dependency on https://lkml.org/lkml/2018/5/17/251 Changes sinc v1 (https://lwn.net/Articles/754679): - Fixed DT binding making NGD slave of the slim controller. - Fixed few typos spotted by Vinod. - Removed unnessary dmaengine_slave_config suggested by Vinod - Removed few unnessary initializations. - fixed commit log to remove reference to B family suggested by Trilok Thanks, srini [1] https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimbus-… Srinivas Kandagatla (2): slimbus: ngd: dt-bindings: Add slim ngd dt bindings slimbus: ngd: Add qcom SLIMBus NGD driver .../bindings/slimbus/slim-ngd-qcom-ctrl.txt | 84 ++ drivers/slimbus/Kconfig | 11 + drivers/slimbus/Makefile | 3 + drivers/slimbus/qcom-ngd-ctrl.c | 1342 ++++++++++++++++++++ drivers/slimbus/slimbus.h | 8 + 5 files changed, 1448 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt create mode 100644 drivers/slimbus/qcom-ngd-ctrl.c -- 2.16.2

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[alsa-devel] [PATCH v2 16/21] ASoC: max98088: use match_string() helper
by Yisheng Xie 01 Jun '18

01 Jun '18

match_string() returns the index of an array for a matching string, which can be used instead of open coded variant. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- v2: - split ret=xxx and move assignment to the line before 'if' - fix error return value - both per Andy sound/soc/codecs/max98088.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 865f64c..fb515aa 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1382,15 +1382,12 @@ static int max98088_set_bias_level(struct snd_soc_component *component, static int max98088_get_channel(struct snd_soc_component *component, const char *name) { - int i; + int ret; - for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++) - if (strcmp(name, eq_mode_name[i]) == 0) - return i; - - /* Shouldn't happen */ - dev_err(component->dev, "Bad EQ channel name '%s'\n", name); - return -EINVAL; + ret = match_string(eq_mode_name, ARRAY_SIZE(eq_mode_name), name); + if (ret < 0) + dev_err(component->dev, "Bad EQ channel name '%s'\n", name); + return ret; } static void max98088_setup_eq1(struct snd_soc_component *component) -- 1.7.12.4

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