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

[alsa-devel] Applied "ASoC: rt5640: Add devicetree-bindings for dmic, jack-detect" to the asoc tree
by Mark Brown 11 May '18

11 May '18

The patch ASoC: rt5640: Add devicetree-bindings for dmic, jack-detect has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e9e7a3bdcdf41efe9137dcf225b021e0b6fd2dc3 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Tue, 8 May 2018 17:35:47 +0200 Subject: [PATCH] ASoC: rt5640: Add devicetree-bindings for dmic, jack-detect Add devicetree-bindings for the dmic, jack-detect source and overcurrent- detect threshold settings. The dmic bindings mirror the existing bindings for the rt5645. The jd-src and ovcd bindings mirror the existing bindings for the rt5651. Cc devicetree(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/rt5640.txt | 35 +++++++++++++++++++ include/dt-bindings/sound/rt5640.h | 25 +++++++++++++ 2 files changed, 60 insertions(+) create mode 100644 include/dt-bindings/sound/rt5640.h diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 57fe64643050..e40e4893eed8 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -22,6 +22,41 @@ Optional properties: - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using IN1P pin as dmic1 data pin + 2: using GPIO3 pin as dmic1 data pin + +- realtek,dmic2-data-pin + 0: dmic2 is not used + 1: using IN1N pin as dmic2 data pin + 2: using GPIO4 pin as dmic2 data pin + +- realtek,jack-detect-source + u32. Valid values: + 0: jack-detect is not used + 1: Use GPIO1 for jack-detect + 2: Use JD1_IN4P for jack-detect + 3: Use JD2_IN4N for jack-detect + 4: Use GPIO2 for jack-detect + 5: Use GPIO3 for jack-detect + 6: Use GPIO4 for jack-detect + +- realtek,jack-detect-not-inverted + bool. Normal jack-detect switches give an inverted signal, set this bool + in the rare case you've a jack-detect switch which is not inverted. + +- realtek,over-current-threshold-microamp + u32, micbias over-current detection threshold in µA, valid values are + 600, 1500 and 2000µA. + +- realtek,over-current-scale-factor + u32, micbias over-current detection scale-factor, valid values are: + 0: Scale current by 0.5 + 1: Scale current by 0.75 + 2: Scale current by 1.0 + 3: Scale current by 1.5 + Pins on the device (for linking into audio routes) for RT5639/RT5640: * DMIC1 diff --git a/include/dt-bindings/sound/rt5640.h b/include/dt-bindings/sound/rt5640.h new file mode 100644 index 000000000000..154c9b4414f2 --- /dev/null +++ b/include/dt-bindings/sound/rt5640.h @@ -0,0 +1,25 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_RT5640_H +#define __DT_RT5640_H + +#define RT5640_DMIC1_DATA_PIN_NONE 0 +#define RT5640_DMIC1_DATA_PIN_IN1P 1 +#define RT5640_DMIC1_DATA_PIN_GPIO3 2 + +#define RT5640_DMIC2_DATA_PIN_NONE 0 +#define RT5640_DMIC2_DATA_PIN_IN1N 1 +#define RT5640_DMIC2_DATA_PIN_GPIO4 2 + +#define RT5640_JD_SRC_GPIO1 1 +#define RT5640_JD_SRC_JD1_IN4P 2 +#define RT5640_JD_SRC_JD2_IN4N 3 +#define RT5640_JD_SRC_GPIO2 4 +#define RT5640_JD_SRC_GPIO3 5 +#define RT5640_JD_SRC_GPIO4 6 + +#define RT5640_OVCD_SF_0P5 0 +#define RT5640_OVCD_SF_0P75 1 +#define RT5640_OVCD_SF_1P0 2 +#define RT5640_OVCD_SF_1P5 3 + +#endif /* __DT_RT5640_H */ -- 2.17.0

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[alsa-devel] [PATCH v5 0/2] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 11 May '18

11 May '18

Hi, Changes since v4: - Really rebased on a clean tree - Dropped the already applied patches Changes since v3: - Rebased to make sure it applies cleanly (I had one local Kconfig patch) Changes since v2: - Return -ENODEV from sdma_pcm_platform_register() in case !CONFIG_SND_SDMA_SOC Changes since v1: - Rebased on the n810 patches (Kconfig conflict) - Delete the omap-pcm as the last step instead of keeping the .c/.h file. If there is a need it can be resurrected with a simple revert. - I left out the Acked-by from Jarkko for the last patch since I have changed it omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap2 n810, omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 for various reasons, but mainly because the Nokia 770 audio is not supported in mainline... Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Delete the obsolete omap-pcm include/sound/omap-pcm.h | 30 ---- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 +- sound/soc/omap/Kconfig | 3 +- sound/soc/omap/Makefile | 2 - sound/soc/omap/omap-pcm.c | 262 ------------------------------ 6 files changed, 6 insertions(+), 301 deletions(-) delete mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v4 0/7] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 11 May '18

11 May '18

Hi, Changes since v3: - Rebased to make sure it applies cleanly (I had one local Kconfig patch) Changes since v2: - Return -ENODEV from sdma_pcm_platform_register() in case !CONFIG_SND_SDMA_SOC Changes since v1: - Rebased on the n810 patches (Kconfig conflict) - Delete the omap-pcm as the last step instead of keeping the .c/.h file. If there is a need it can be resurrected with a simple revert. - I left out the Acked-by from Jarkko for the last patch since I have changed it omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap2 n810, omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 for various reasons, but mainly because the Nokia 770 audio is not supported in mainline... Regards, Peter --- Peter Ujfalusi (7): ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-hdmi-audio: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-dmic: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcpdm: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Delete the obsolete omap-pcm include/sound/omap-pcm.h | 30 ---- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 +- sound/soc/omap/Kconfig | 27 +-- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 5 +- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/omap-pcm.c | 262 ------------------------------ sound/soc/omap/sdma-pcm.c | 68 ++++++++ sound/soc/omap/sdma-pcm.h | 21 +++ 12 files changed, 121 insertions(+), 318 deletions(-) delete mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.h -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add platform-data for Lenovo Ideapad Mixx 310
by Hans de Goede 11 May '18

11 May '18

The Lenovo Ideapad Mixx 310 has a differential internal analog mic, add platform-data for this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc8d829ce45b..70f351c61d41 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3652,6 +3652,11 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { + .jd_mode = 3, + .in2_diff = true, +}; + static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; @@ -3735,6 +3740,15 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&jd_mode3_platform_data, }, + { + .ident = "Lenovo Ideapad Miix 310", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "80SG"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "MIIX 310-10ICR"), + }, + .driver_data = (void *)&lenovo_ideapad_miix_310_pdata, + }, { } }; -- 2.17.0

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[alsa-devel] [PATCH 0/5] ASoC: cirrus: Series of fixes for ep93xx-i2s
by Alexander Sverdlin 11 May '18

11 May '18

This series is a result of long-running debugging of rarely screwed I2S stream out of EP9302 chip. Logic analyser shows that the stream is shifted by one byte (out of 32 bits per sample) which leads to noise intead of original stream. DMA is verified to work fine (no underrun). The issue is only reproducible under high load (CPU or AMBA bus -- not yet clear). This stream corruption happens at the same time I2S controller reports internal FIFO underrun (even though according to documentation, it should handle an underrun gracefully). First two patches are fixing unrelated issues in I2S driver found during debugging of the stream shift. Third one is simplification of the driver to prepare for workaround. Both controller configuration options for TX FIFO underrun handling were tested, both do not work reliable. Hence, this ugly watchdog in the last patch. The whole series is tested on a board similar to EDB9302 (identical in sound part). Alexander Sverdlin (5): ASoC: cirrus: i2s: Fix LRCLK configuration ASoC: cirrus: i2s: Fix {TX|RX}LinCtrlData setup ASoC: cirrus: i2s: Stop enabling I2S2 and I2S3 FIFOs ARM: ep93xx: i2s: Add IRQ to platform device resources ASoC: cirrus: i2s: IRQ-based stream watchdog arch/arm/mach-ep93xx/core.c | 1 + sound/soc/cirrus/Kconfig | 17 +++++++ sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 106 +++++++++++++++++++++++++++++++++-------- sound/soc/cirrus/snappercl15.c | 2 +- 5 files changed, 106 insertions(+), 22 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH 1/2] ASoC: nau8810: change input PGA mixer stage
by John Hsu 11 May '18

11 May '18

Organize the paths of the mixer, "Input Boost Stage", including the routes of the mixer. The control is not used correctly before. Besides, the driver changes the name of the mixer controls. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index ca2ba1c..bfd74b8 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -373,9 +373,11 @@ static const struct snd_kcontrol_new nau8810_mono_mixer_controls[] = { }; /* PGA Mute */ -static const struct snd_kcontrol_new nau8810_inpga_mute[] = { +static const struct snd_kcontrol_new nau8810_pgaboost_mixer_controls[] = { SOC_DAPM_SINGLE("PGA Mute Switch", NAU8810_REG_PGAGAIN, - NAU8810_PGAMT_SFT, 1, 0), + NAU8810_PGAMT_SFT, 1, 1), + SOC_DAPM_SINGLE("PMIC PGA Switch", NAU8810_REG_ADCBOOST, + NAU8810_PMICBSTGAIN_SFT, 0x7, 0), }; /* Input PGA */ @@ -386,11 +388,6 @@ static const struct snd_kcontrol_new nau8810_inpga[] = { NAU8810_PMICPGA_SFT, 1, 0), }; -/* Mic Input boost vol */ -static const struct snd_kcontrol_new nau8810_mic_boost_controls = - SOC_DAPM_SINGLE("Mic Volume", NAU8810_REG_ADCBOOST, - NAU8810_PMICBSTGAIN_SFT, 0x7, 0); - /* Loopback Switch */ static const struct snd_kcontrol_new nau8810_loopback = SOC_DAPM_SINGLE("Switch", NAU8810_REG_COMP, @@ -429,8 +426,8 @@ static const struct snd_soc_dapm_widget nau8810_dapm_widgets[] = { NAU8810_PGA_EN_SFT, 0, nau8810_inpga, ARRAY_SIZE(nau8810_inpga)), SND_SOC_DAPM_MIXER("Input Boost Stage", NAU8810_REG_POWER2, - NAU8810_BST_EN_SFT, 0, nau8810_inpga_mute, - ARRAY_SIZE(nau8810_inpga_mute)), + NAU8810_BST_EN_SFT, 0, nau8810_pgaboost_mixer_controls, + ARRAY_SIZE(nau8810_pgaboost_mixer_controls)), SND_SOC_DAPM_SUPPLY("Mic Bias", NAU8810_REG_POWER1, NAU8810_MICBIAS_EN_SFT, 0, NULL, 0), @@ -469,8 +466,8 @@ static const struct snd_soc_dapm_route nau8810_dapm_routes[] = { /* Input Boost Stage */ {"ADC", NULL, "Input Boost Stage"}, {"ADC", NULL, "PLL", check_mclk_select_pll}, - {"Input Boost Stage", NULL, "Input PGA"}, - {"Input Boost Stage", NULL, "MICP"}, + {"Input Boost Stage", "PGA Mute Switch", "Input PGA"}, + {"Input Boost Stage", "PMIC PGA Switch", "MICP"}, /* Input PGA */ {"Input PGA", NULL, "Mic Bias"}, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: zx-i2s: fix spelling mistake: "timeing" -> "timing"
by Colin King 10 May '18

10 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err message text Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/zte/zx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 9a0565937d1f..b6b056c607fa 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -197,7 +197,7 @@ static int zx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) val |= (ZX_I2S_TIMING_I2S | ZX_I2S_TIMING_LSB_JUSTIF); break; default: - dev_err(cpu_dai->dev, "Unknown i2s timeing\n"); + dev_err(cpu_dai->dev, "Unknown i2s timing\n"); return -EINVAL; } -- 2.17.0

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[alsa-devel] [PATCH] ASoC: amd: modified dma pointer callback implementation
by Vijendar Mukunda 10 May '18

10 May '18

Modified dma pointer callback implementation. hw ptr calculated based on System Memory to ACP SRAM dma channel transfer count in case of playback. Increased min period size for playback to 4096 to avoid underruns. In case of capture, hw ptr calculated based on dma transfer count for ACP SRAM to System Memory Dma channel. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- This patch is dependent on [V4,10/10] ASoC: amd: dma driver changes for bt i2s instance https://patchwork.kernel.org/patch/10385381/ sound/soc/amd/acp-pcm-dma.c | 94 ++++++++++++++++++++++++++------------------- sound/soc/amd/acp.h | 11 ------ 2 files changed, 55 insertions(+), 50 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 906eb77..969c46c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -28,7 +28,7 @@ #define PLAYBACK_MIN_NUM_PERIODS 2 #define PLAYBACK_MAX_NUM_PERIODS 2 #define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 1024 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 #define CAPTURE_MIN_NUM_PERIODS 2 #define CAPTURE_MAX_NUM_PERIODS 2 #define CAPTURE_MAX_PERIOD_SIZE 16384 @@ -854,10 +854,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_BLUETOOTH; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH8; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH9; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_TRANSMIT_BYTE_CNT_LOW; adata->play_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -875,10 +871,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = TO_ACP_I2S_1; rtd->dma_dscr_idx_1 = PLAYBACK_START_DMA_DESCR_CH12; rtd->dma_dscr_idx_2 = PLAYBACK_START_DMA_DESCR_CH13; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_TRANSMIT_BYTE_CNT_LOW; adata->play_i2ssp_stream = substream; } } else { @@ -891,10 +883,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_BLUETOOTH; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH10; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH11; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_BT_RECEIVE_BYTE_CNT_LOW; adata->capture_i2sbt_stream = substream; break; case I2S_SP_INSTANCE: @@ -914,10 +902,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->destination = FROM_ACP_I2S_1; rtd->dma_dscr_idx_1 = CAPTURE_START_DMA_DESCR_CH14; rtd->dma_dscr_idx_2 = CAPTURE_START_DMA_DESCR_CH15; - rtd->byte_cnt_high_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_HIGH; - rtd->byte_cnt_low_reg_offset = - mmACP_I2S_RECEIVED_BYTE_CNT_LOW; adata->capture_i2ssp_stream = substream; } } @@ -955,22 +939,14 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } -static u64 acp_get_byte_count(struct audio_substream_data *rtd) -{ - union acp_dma_count byte_count; - - byte_count.bcount.high = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_high_reg_offset); - byte_count.bcount.low = acp_reg_read(rtd->acp_mmio, - rtd->byte_cnt_low_reg_offset); - return byte_count.bytescount; -} - static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) { u32 buffersize; u32 pos = 0; - u64 bytescount = 0; + u32 dma_count = 0; + u32 mul = 0; + u16 dscr; + u32 dma_config, period_bytes; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; @@ -979,11 +955,56 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) return -EINVAL; buffersize = frames_to_bytes(runtime, runtime->buffer_size); - bytescount = acp_get_byte_count(rtd); - - if (bytescount > rtd->bytescount) - bytescount -= rtd->bytescount; - pos = do_div(bytescount, buffersize); + period_bytes = frames_to_bytes(runtime, runtime->period_size); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_8); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_8); + if (dscr == PLAYBACK_END_DMA_DESCR_CH8) + mul = 1; + break; + case I2S_SP_INSTANCE: + default: + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_12); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_12); + if (dscr == PLAYBACK_END_DMA_DESCR_CH12) + mul = 1; + break; + } + } else { + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + dma_config = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CNTL_10); + if (dma_config != 0) { + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_10); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_10); + if (dscr == CAPTURE_END_DMA_DESCR_CH10) + mul = 1; + } + break; + case I2S_SP_INSTANCE: + default: + dma_config = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CNTL_14); + if (dma_config != 0) { + dscr = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_DSCR_14); + dma_count = acp_reg_read(rtd->acp_mmio, + mmACP_DMA_CUR_TRANS_CNT_14); + if (dscr == CAPTURE_END_DMA_DESCR_CH14) + mul = 1; + } + } + } + pos = ((mul * period_bytes) + dma_count) % buffersize; return bytes_to_frames(runtime, pos); } @@ -1016,7 +1037,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; u32 loops = 4000; - u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -1030,9 +1050,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - bytescount = acp_get_byte_count(rtd); - if (rtd->bytescount == 0) - rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { acp_dma_start(rtd->acp_mmio, rtd->ch1, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & @@ -1066,7 +1083,6 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_stop(rtd->acp_mmio, rtd->ch2); ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); } - rtd->bytescount = 0; break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 9cd3e96..b6d52ef 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -133,10 +133,7 @@ struct audio_substream_data { u16 dma_dscr_idx_2; u32 pte_offset; u32 sram_bank; - u32 byte_cnt_high_reg_offset; - u32 byte_cnt_low_reg_offset; uint64_t size; - u64 bytescount; void __iomem *acp_mmio; }; @@ -157,14 +154,6 @@ struct acp_platform_info { u16 i2s_instance; }; -union acp_dma_count { - struct { - u32 low; - u32 high; - } bcount; - u64 bytescount; -}; - enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- 2.7.4

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[alsa-devel] [PATCH v3] ALSA: core api: define offsets for TLV items
by Ranjani Sridharan 10 May '18

10 May '18

Currently, there are no pre-defined accessors for the elements in topology TLV data. In the absence of such offsets, the tlv data will have to be decoded using hardwired offset numbers 0-N depending on the type of TLV. This patch defines accessor offsets for the type, length, min and mute/step items in TLV data for DB_SCALE type tlv's. These will be used by drivers to decode the TLV data while loading topology thereby improving code readability. The type and len offsets are common for all TLV types. The min and step/mute offsets are specific to DB_SCALE tlv type. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Notes: v3: fix prefix for offset definitions include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..13f425c2abbd 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -98,4 +98,12 @@ #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 +/* Accessor offsets for TLV data items */ +#define SNDRV_CTL_TLVO_TYPE 0 +#define SNDRV_CTL_TLVO_LEN 1 + +/* Accessor offsets for min, mute and step items in dB scale type TLV */ +#define SNDRV_CTL_TLVO_DB_SCALE_MIN 2 +#define SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP 3 + #endif -- 2.14.1

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[alsa-devel] [PATCH 1/3] ASoC: pcm_native: Use 'snd_pcm_format_t' type
by Fabio Estevam 09 May '18

09 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> 'snd_pcm_format_t' type is better suited for iterating through the SNDRV_PCM_FORMAT members. Also, use SNDRV_PCM_FORMAT_FIRST for the first element. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/core/pcm_native.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7585444..5d62b7c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2051,13 +2051,13 @@ static int snd_pcm_hw_rule_mulkdiv(struct snd_pcm_hw_params *params, static int snd_pcm_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - unsigned int k; + snd_pcm_format_t k; const struct snd_interval *i = hw_param_interval_c(params, rule->deps[0]); struct snd_mask m; struct snd_mask *mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_any(&m); - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + for (k = SNDRV_PCM_FORMAT_FIRST; k <= SNDRV_PCM_FORMAT_LAST; ++k) { int bits; if (! snd_mask_test(mask, k)) continue; @@ -2074,12 +2074,12 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { struct snd_interval t; - unsigned int k; + snd_pcm_format_t k; t.min = UINT_MAX; t.max = 0; t.openmin = 0; t.openmax = 0; - for (k = 0; k <= SNDRV_PCM_FORMAT_LAST; ++k) { + for (k = SNDRV_PCM_FORMAT_FIRST; k <= SNDRV_PCM_FORMAT_LAST; ++k) { int bits; if (! snd_mask_test(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), k)) continue; -- 2.7.4

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