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

[alsa-devel] [PATCH] ASoC: ssm4567: Add sense support
by Lars-Peter Clausen 15 Aug '15

15 Aug '15

The ssm4567 has sensing circuitry that can be used to monitor the current and voltage on the speaker amplifier output has well as the VBAT input. This data can be output over the I2S interface so it can be processed by a DSP or similar. This patch adds the sense capture output stream to the CODEC DAI as well as DAPM widgets that ensure that the sensing circuitry is powered up when the capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm4567.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index ac242b2..e619d56 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -174,6 +174,12 @@ static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, &ssm4567_amplifier_boost_control), + SND_SOC_DAPM_SIGGEN("Sense"), + + SND_SOC_DAPM_PGA("Current Sense", SSM4567_REG_POWER_CTRL, 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("Voltage Sense", SSM4567_REG_POWER_CTRL, 5, 1, NULL, 0), + SND_SOC_DAPM_PGA("VBAT Sense", SSM4567_REG_POWER_CTRL, 6, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -181,6 +187,13 @@ static const struct snd_soc_dapm_route ssm4567_routes[] = { { "OUT", NULL, "Amplifier Boost" }, { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, + + { "Current Sense", NULL, "Sense" }, + { "Voltage Sense", NULL, "Sense" }, + { "VBAT Sense", NULL, "Sense" }, + { "Capture Sense", NULL, "Current Sense" }, + { "Capture Sense", NULL, "Voltage Sense" }, + { "Capture Sense", NULL, "VBAT Sense" }, }; static int ssm4567_hw_params(struct snd_pcm_substream *substream, @@ -388,6 +401,14 @@ static struct snd_soc_dai_driver ssm4567_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32, }, + .capture = { + .stream_name = "Capture Sense", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, .ops = &ssm4567_dai_ops, }; -- 2.1.4

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[alsa-devel] [PATCH 1/4] ASoC: cs35l32: Use case range at appropriate places
by Axel Lin 15 Aug '15

15 Aug '15

Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. Below are the summary of readable/volatile/precious registers. The readable registers: 0x01 ~ 0x0D, 0x0F ~ 0x1C The volatile registers: 0x01 ~ 0x05, 0x15 ~ 0x18 The precious registers: 0x15 ~ 0x18 No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.c | 45 +++++---------------------------------------- 1 file changed, 5 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 203c720..44c30fe 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -74,33 +74,8 @@ static const struct reg_default cs35l32_reg_defaults[] = { static bool cs35l32_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_PWRCTL1: - case CS35L32_PWRCTL2: - case CS35L32_CLK_CTL: - case CS35L32_BATT_THRESHOLD: - case CS35L32_VMON: - case CS35L32_BST_CPCP_CTL: - case CS35L32_IMON_SCALING: - case CS35L32_AUDIO_LED_MNGR: - case CS35L32_ADSP_CTL: - case CS35L32_CLASSD_CTL: - case CS35L32_PROTECT_CTL: - case CS35L32_INT_MASK_1: - case CS35L32_INT_MASK_2: - case CS35L32_INT_MASK_3: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: - case CS35L32_FLASH_MODE: - case CS35L32_MOVIE_MODE: - case CS35L32_FLASH_TIMER: - case CS35L32_FLASH_INHIBIT: + case CS35L32_DEVID_AB ... CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL ... CS35L32_FLASH_INHIBIT: return true; default: return false; @@ -110,15 +85,8 @@ static bool cs35l32_readable_register(struct device *dev, unsigned int reg) static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_DEVID_AB ... CS35L32_REV_ID: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; @@ -128,10 +96,7 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) static bool cs35l32_precious_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; -- 2.1.0

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[alsa-devel] Applied "ASoC: cs42l56: Use case range at appropriate place" to the asoc tree
by Mark Brown 15 Aug '15

15 Aug '15

The patch ASoC: cs42l56: Use case range at appropriate place has been applied to the asoc tree at git://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 7f325bfc933d140337e55507821d6bb021321059 Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)ingics.com> Date: Wed, 12 Aug 2015 11:09:39 +0800 Subject: [PATCH] ASoC: cs42l56: Use case range at appropriate place The readable registers are in continuous range: 0x01 ~ 0x2e. Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 47 +--------------------------------------------- 1 file changed, 1 insertion(+), 46 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba4..98a68c6 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -115,52 +115,7 @@ static const struct reg_default cs42l56_reg_defaults[] = { static bool cs42l56_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L56_CHIP_ID_1: - case CS42L56_CHIP_ID_2: - case CS42L56_PWRCTL_1: - case CS42L56_PWRCTL_2: - case CS42L56_CLKCTL_1: - case CS42L56_CLKCTL_2: - case CS42L56_SERIAL_FMT: - case CS42L56_CLASSH_CTL: - case CS42L56_MISC_CTL: - case CS42L56_INT_STATUS: - case CS42L56_PLAYBACK_CTL: - case CS42L56_DSP_MUTE_CTL: - case CS42L56_ADCA_MIX_VOLUME: - case CS42L56_ADCB_MIX_VOLUME: - case CS42L56_PCMA_MIX_VOLUME: - case CS42L56_PCMB_MIX_VOLUME: - case CS42L56_ANAINPUT_ADV_VOLUME: - case CS42L56_DIGINPUT_ADV_VOLUME: - case CS42L56_MASTER_A_VOLUME: - case CS42L56_MASTER_B_VOLUME: - case CS42L56_BEEP_FREQ_ONTIME: - case CS42L56_BEEP_FREQ_OFFTIME: - case CS42L56_BEEP_TONE_CFG: - case CS42L56_TONE_CTL: - case CS42L56_CHAN_MIX_SWAP: - case CS42L56_AIN_REFCFG_ADC_MUX: - case CS42L56_HPF_CTL: - case CS42L56_MISC_ADC_CTL: - case CS42L56_GAIN_BIAS_CTL: - case CS42L56_PGAA_MUX_VOLUME: - case CS42L56_PGAB_MUX_VOLUME: - case CS42L56_ADCA_ATTENUATOR: - case CS42L56_ADCB_ATTENUATOR: - case CS42L56_ALC_EN_ATTACK_RATE: - case CS42L56_ALC_RELEASE_RATE: - case CS42L56_ALC_THRESHOLD: - case CS42L56_NOISE_GATE_CTL: - case CS42L56_ALC_LIM_SFT_ZC: - case CS42L56_AMUTE_HPLO_MUX: - case CS42L56_HPA_VOLUME: - case CS42L56_HPB_VOLUME: - case CS42L56_LOA_VOLUME: - case CS42L56_LOB_VOLUME: - case CS42L56_LIM_THRESHOLD_CTL: - case CS42L56_LIM_CTL_RELEASE_RATE: - case CS42L56_LIM_ATTACK_RATE: + case CS42L56_CHIP_ID_1 ... CS42L56_LIM_ATTACK_RATE: return true; default: return false; -- 2.5.0

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[alsa-devel] [PATCH] ASoC: Intel: fix runtime pm imbalance on error
by Junjie Mao 15 Aug '15

15 Aug '15

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Junjie Mao <junjie.mao(a)enight.me> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d1b9e3..27a9653aa3d9 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -152,6 +152,7 @@ static int sst_power_control(struct device *dev, bool state) dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { + pm_runtime_put_sync(dev); dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } @@ -204,8 +205,10 @@ static int sst_cdev_open(struct device *dev, struct intel_sst_drv *ctx = dev_get_drvdata(dev); retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) + if (retval < 0) { + pm_runtime_put_sync(ctx->dev); return retval; + } str_id = sst_get_stream(ctx, str_params); if (str_id > 0) { @@ -672,8 +675,10 @@ static int sst_send_byte_stream(struct device *dev, if (NULL == bytes) return -EINVAL; ret_val = pm_runtime_get_sync(ctx->dev); - if (ret_val < 0) + if (ret_val < 0) { + pm_runtime_put_sync(ctx->dev); return ret_val; + } ret_val = sst_send_byte_stream_mrfld(ctx, bytes); sst_pm_runtime_put(ctx); -- 1.9.3

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[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9c46c7d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

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[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

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[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2ce9e1d..f3034b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

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[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f3034b9..0b4fcd9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

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[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0b4fcd9..e79dc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

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[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8e5ff5e..37aabe3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

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