August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: tas2552: Fix fraction overflow in PLL calculation
by Oskar Schirmer 29 Aug '17

29 Aug '17

Setting the PLL involves the calculation of a fixed point ratio with 4 decimal digits fraction, referred to as "J.D". The fraction "D" is stored separately from the integer part "J" and is limited to 0..9999. The current algorithm uses integer registers to calculate the fraction part, but failed to compensate for rounding errors, resulting in values larger than 9999 for the fraction part occasionally, e.g. for 44.1kHz audio rate and pll_clkin = 3763400 it would set J to 11 and D to 10002, which will at best result in wrong pitch. The critical part is the "pll_clkin / 10000", which would be ok with real numbers, but using integer arithmetic the rounding decreases the divisor, thus increasing the final quotient. The issue is solved by linear interpolation over the reciprocal function between the two adjacent points with integer divisor, i.e. pll_clkin / 10000 and pll_clkin / 10000 + 1, and doing all rounding to the lower result. As a side effect to the bug fix, the approximation to the desired frequency is much better, for the above mentioned example we get 11.9993, while the true ratio is 11.9993623. Signed-off-by: Oskar Schirmer <oskar(a)scara.com> --- sound/soc/codecs/tas2552.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 8840f72..1011616 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -192,7 +192,7 @@ static int tas2552_setup_pll(struct snd_soc_codec *codec, * pll_clk = (.5 * pll_clkin * J.D) / 2^p * Need to fill in J and D here based on incoming freq */ - unsigned int d; + unsigned int d, q, t; u8 j; u8 pll_sel = (tas2552->pll_clk_id << 3) & TAS2552_PLL_SRC_MASK; u8 p = snd_soc_read(codec, TAS2552_PLL_CTRL_1); @@ -200,9 +200,12 @@ static int tas2552_setup_pll(struct snd_soc_codec *codec, p = (p >> 7); recalc: - j = (pll_clk * 2 * (1 << p)) / pll_clkin; - d = (pll_clk * 2 * (1 << p)) % pll_clkin; - d /= (pll_clkin / 10000); + t = (pll_clk * 2) << p; + j = t / pll_clkin; + d = t % pll_clkin; + t = pll_clkin / 10000; + q = d / (t + 1); + d = q + ((9999 - pll_clkin % 10000) * (d / t - q)) / 10000; if (d && (pll_clkin < 512000 || pll_clkin > 9200000)) { if (tas2552->pll_clk_id == TAS2552_PLL_CLKIN_BCLK) { -- 2.7.4

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[alsa-devel] [PATCH 0/2] ASoC: Intel: kbl: Machine driver modifications to support MST Audio
by Naveen M 29 Aug '17

29 Aug '17

Patch modifies dapm routes to enable HDMI Audio and invokes machine pin widget for hdmi port to enable/disable pin, when monitor is connected/disconnected based on the pin status. Naveen M (2): ASoC: Intel: kbl: Add MST route change to kbl machine drivers ASoC: Intel: kbl: Add jack port initialize in kbl machine drivers sound/soc/intel/boards/kbl_rt5663_max98927.c | 15 +++++++++------ sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 14 ++++++++------ 2 files changed, 17 insertions(+), 12 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3] ASoC: fsl_dma: remove dma_object path member
by Rob Herring 29 Aug '17

29 Aug '17

dma_object.path is unused, so rather than fix it to work with DT full_name changes, just remove it. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Timur Tabi <timur(a)tabi.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <fabio.estevam(a)nxp.com> 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 Cc: linuxppc-dev(a)lists.ozlabs.org --- v3: - Update subject: s/sound/ASoC/ v2: - Move printf specifier change to correct patch sound/soc/fsl/fsl_dma.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index c17359648915..0c11f434a374 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -63,7 +63,6 @@ struct dma_object { struct ccsr_dma_channel __iomem *channel; unsigned int irq; bool assigned; - char path[1]; }; /* @@ -903,13 +902,12 @@ static int fsl_soc_dma_probe(struct platform_device *pdev) return ret; } - dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); + dma = kzalloc(sizeof(*dma), GFP_KERNEL); if (!dma) { of_node_put(ssi_np); return -ENOMEM; } - strcpy(dma->path, np->full_name); dma->dai.ops = &fsl_dma_ops; dma->dai.pcm_new = fsl_dma_new; dma->dai.pcm_free = fsl_dma_free_dma_buffers; -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rt5514: Guard Hotword Model bytes loading
by Hsin-Yu Chao 29 Aug '17

29 Aug '17

If the rt5514-spi driver is not enabled in kernel, hotword model will not be loaded when "DSP Voice Wake Up" is set to turn on DSP mode, and an error is logged instead. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- sound/soc/codecs/rt5514.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7482df8ff737..250581082e9d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -299,7 +299,6 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); struct snd_soc_codec *codec = rt5514->codec; const struct firmware *fw = NULL; - int ret = 0; if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) return 0; @@ -337,6 +336,9 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, } if (rt5514->model_buf && rt5514->model_len) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) + int ret; + ret = rt5514_spi_burst_write(0x4ff80000, rt5514->model_buf, ((rt5514->model_len / 8) + 1) * 8); @@ -345,13 +347,22 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, "Model load failed %d\n", ret); return ret; } +#else + dev_err(codec->dev, + "No SPI driver for loading firmware\n"); +#endif } else { request_firmware(&fw, RT5514_FIRMWARE3, codec->dev); if (fw) { +#if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_write(0x4ff80000, fw->data, ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, + "No SPI driver to load fw\n"); +#endif release_firmware(fw); fw = NULL; } -- 2.12.2

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[alsa-devel] [PATCH v2] ASoC: Intel: kbl: Add map for Maxim IV Feedback
by Naveen M 29 Aug '17

29 Aug '17

MAX98927 provides IV feedback on the capture widget. Here we are connecting the capture widget to SSP0_RX and SSP0_RX to the algorithm running on host. v2: rebased on newer kernel (d46b182) Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index c144214..8d2a1c0 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -107,6 +107,11 @@ enum { { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "dmic01_hifi", NULL, "DMIC01 Rx" }, { "DMIC01 Rx", NULL, "DMIC AIF" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index c2ef538..8d19da9 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -119,6 +119,11 @@ enum { { "codec1_in", NULL, "ssp0 Rx" }, { "ssp0 Rx", NULL, "AIF1 Capture" }, + /* IV feedback path */ + { "codec0_fb_in", NULL, "ssp0 Rx"}, + { "ssp0 Rx", NULL, "Left HiFi Capture" }, + { "ssp0 Rx", NULL, "Right HiFi Capture" }, + /* DMIC */ { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, -- 1.9.1

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[alsa-devel] [RESENT PATCH v7 0/7] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 29 Aug '17

29 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v7: Rebase on the newest for-next Changes in v6: Add dmic wakeup delay(not used for now). Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (7): ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 255 +++++++++++++++------ 4 files changed, 186 insertions(+), 80 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v3 0/9] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 29 Aug '17

29 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (9): ASoC: rt5514: Switch to snd_soc_register_codec ASoC: rockchip: Remove obsolete dmic-delay ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 9 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/codecs/rt5514-spi.c | 19 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 263 ++++++++++++++------- 5 files changed, 196 insertions(+), 104 deletions(-) -- 2.11.0

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[alsa-devel] [RESENT PATCH v6 0/8] ASoC: rockchip: Parse dai links from dts
by Jeffy Chen 29 Aug '17

29 Aug '17

Currently we are using a fixed list of dai links in the driver. This serial of patches would let the driver parse dai links from dts, so that we can make some of them optional for future boards. Tested on my chromebook bob(with cros 4.4 kernel), it still works after disabled rt5514 codecs in the dts. Changes in v6: Add dmic wakeup delay(not used for now). Changes in v5: Keep the dmic-delay property, but correct it with the name in dt-binding. -- Suggested-by Mark Brown <broonie(a)kernel.org> Changes in v3: Use compatible to match audio codecs -- Suggested-by Matthias Kaehlcke <mka(a)chromium.org> Changes in v2: Let rockchip,codec-names be a required property, because we plan to add more supported codecs to the fixed dai link list in the driver. Jeffy Chen (8): ASoC: rockchip: Correct 'dmic-delay' property name ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp arm64: dts: rockchip: Add rt5514 dsp for Gru arm64: dts: rockchip: Update rt5514 devices' compatible for Gru ASoC: rockchip: Parse dai links from dts ASoC: rockchip: Add support for DP codec ASoC: rockchip: Add support for DMIC codec dt-bindings: ASoC: rockchip: Update description of rockchip,codec .../bindings/sound/rockchip,rk3399-gru-sound.txt | 2 +- arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 7 +- sound/soc/rockchip/Kconfig | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 270 ++++++++++++++------- 4 files changed, 193 insertions(+), 88 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2 00/10] Remove unnecessary function
by Peng Donglin 29 Aug '17

29 Aug '17

When calling the function devm_snd_soc_register_card to register a sound card,the address of snd_soc_card will be assigned to dev->driver_data automatically,so there is no extra need to call platform_set_drvdata. Peng Donglin (10): ASoC: s3c24xx_uda134x: Remove unnecessary function call ASoC: mediatek: Remove unnecessary function call ASoC: Intel: Remove unnecessary function call ASoC: atmel: Remove unnecessary function call ASoC: rockchip: Remove unnecessary function call ASoC: mxs-sgtl5000: Remove unnecessary function call ASoC: qcom: Remove useless function call ASoC: sun4i-codec: Remove unnecessary function call ASoC: tegra: Remove unnecessary function call ASoC: ux500: Remove unnecessary function call sound/soc/atmel/atmel-classd.c | 1 - sound/soc/atmel/atmel-pdmic.c | 1 - sound/soc/intel/boards/bytcht_da7213.c | 6 ++---- sound/soc/intel/boards/bytcht_es8316.c | 6 ++---- sound/soc/intel/boards/bytcht_nocodec.c | 6 ++---- sound/soc/intel/boards/bytcr_rt5640.c | 7 ++----- sound/soc/intel/boards/bytcr_rt5651.c | 7 ++----- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 ++---- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 ++---- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - sound/soc/mxs/mxs-sgtl5000.c | 7 ++----- sound/soc/qcom/apq8016_sbc.c | 1 - sound/soc/qcom/storm.c | 1 - sound/soc/rockchip/rk3288_hdmi_analog.c | 6 +----- sound/soc/samsung/s3c24xx_uda134x.c | 1 - sound/soc/sunxi/sun4i-codec.c | 1 - sound/soc/tegra/tegra_alc5632.c | 1 - sound/soc/tegra/tegra_max98090.c | 1 - sound/soc/tegra/tegra_rt5640.c | 1 - sound/soc/tegra/tegra_rt5677.c | 1 - sound/soc/tegra/tegra_sgtl5000.c | 1 - sound/soc/tegra/tegra_wm8753.c | 1 - sound/soc/tegra/tegra_wm8903.c | 1 - sound/soc/tegra/tegra_wm9712.c | 1 - sound/soc/tegra/trimslice.c | 1 - sound/soc/ux500/mop500.c | 1 - 29 files changed, 19 insertions(+), 63 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH v3] sound: aoa: Convert to using %pOF instead of full_name
by Rob Herring 29 Aug '17

29 Aug '17

Now that we have a custom printf format specifier, convert users of full_name to use %pOF instead. This is preparation to remove storing of the full path string for each node. Signed-off-by: Rob Herring <robh(a)kernel.org> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org --- v3: Split aoa to separate patch sound/aoa/codecs/tas.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 733b6365dad6..15c05755d270 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -905,8 +905,8 @@ static int tas_i2c_probe(struct i2c_client *client, goto fail; } printk(KERN_DEBUG - "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", - (unsigned int)client->addr, node->full_name); + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); return 0; fail: mutex_destroy(&tas->mtx); -- 2.11.0

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