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

[alsa-devel] [GIT PULL] ASoC fixes for v4.4
by Mark Brown 23 Dec '15

23 Dec '15

The following changes since commit 4ef7675344d687a0ef5b0d7c0cee12da005870c0: Linux 4.4-rc6 (2015-12-20 16:06:09 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v4.4-rc6 for you to fetch changes up to 3dd5fc0eeb5d7cc07b8e3e383037261799b38897: Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus (2015-12-23 00:23:27 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.4 A collection of small driver specific fixes here, nothing that'll affect users who don't have the devices concerned. At least the wm8974 bug indicates that there's not too many users of some of these devices. ---------------------------------------------------------------- Jean-Michel Hautbois (1): ASoC: sgtl5000: fix VAG power up timing John Keeping (2): ASoC: es8328: Fix deemphasis values ASoC: es8328: Fix shifts for mixer switches Mans Rullgard (1): ASoC: wm8974: set cache type for regmap Mark Brown (1): Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx Sjoerd Simons (1): ASoC: rockchip: spdif: Set transmit data level to 16 samples Zidan Wang (1): ASoC: fsl_sai: fix no frame clk in master mode sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + sound/soc/codecs/sgtl5000.c | 1 + sound/soc/codecs/wm8974.c | 1 + sound/soc/davinci/davinci-mcasp.c | 4 ++-- sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.c | 6 ++++-- sound/soc/rockchip/rockchip_spdif.h | 2 +- 8 files changed, 45 insertions(+), 13 deletions(-)

2 2

[alsa-devel] [PATCH 0/4] add NULL test
by Julia Lawall 23 Dec '15

23 Dec '15

4 10

[alsa-devel] [PATCH] ASoC: rt5677: set PLL_CTRL2 non-volatile
by Bard Liao 23 Dec '15

23 Dec '15

There is a status bit on RT5677_PLL1_CTRL2 and RT5677_PLL2_CTRL2. That's why those registers are set volatile. However, the status bit is currently not used by codec driver. So, it should be no problem if we set them non-volatile. The purpose of setting them non-volatile is to restore the setting after a syspend/resume cycle. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, Please revert the "ASoC: rt5677: Reconfigure PLL1 after resume" patch before applying this patch. Thanks. --- sound/soc/codecs/rt5677.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f73fd12..13fef00 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -297,8 +297,6 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_HAP_GENE_CTRL2: case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: - case RT5677_PLL1_CTRL2: - case RT5677_PLL2_CTRL2: case RT5677_ASRC_22: case RT5677_ASRC_23: case RT5677_VAD_CTRL5: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 0/8] Add support for voice control on Arizona ADSP
by Charles Keepax 23 Dec '15

23 Dec '15

This series add support for voice control on wm8280/5110. This is done by opening a compressed record channel, upon which data will be available once the voice control functionality has been triggered on the DSP. Changes since v1: - Merged selecting of SND_SOC_COMPRESS into patch adding basic hookup as suggested by Vinod. - Fixed a minor typo in usleep_delay, which was using a longer delay than required. - Use kcalloc to allocate an array rather than caculating size manually. Thanks, Charles Charles Keepax (8): ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together ASoC: wm_adsp: Add a handler for the compressed IRQ ASoC: wm_adsp: Pull data through compressed read include/uapi/sound/compress_params.h | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 95 +++- sound/soc/codecs/wm_adsp.c | 907 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 21 + 6 files changed, 1010 insertions(+), 21 deletions(-) -- 2.1.4

2 10

[alsa-devel] [PATCH 01/13] ASoC: wm_adsp: Fixup some minor formatting and checkpatch errors
by Charles Keepax 23 Dec '15

23 Dec '15

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++-------------- sound/soc/codecs/wm_adsp.h | 4 ++-- 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 905ae99..3808190 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -544,10 +544,10 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } -static int wm_coeff_info(struct snd_kcontrol *kcontrol, +static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -593,10 +593,10 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_put(struct snd_kcontrol *kcontrol, +static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; memcpy(ctl->cache, p, ctl->len); @@ -647,10 +647,10 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_get(struct snd_kcontrol *kcontrol, +static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { @@ -829,8 +829,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, break; } - list_for_each_entry(ctl, &dsp->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; @@ -1109,7 +1108,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) goto out_fw; } - header = (void*)&firmware->data[0]; + header = (void *)&firmware->data[0]; if (memcmp(&header->magic[0], "WMFW", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); @@ -1189,7 +1188,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) offset = le32_to_cpu(region->offset) & 0xffffff; type = be32_to_cpu(region->type) & 0xff; mem = wm_adsp_find_region(dsp, type); - + switch (type) { case WMFW_NAME_TEXT: region_name = "Firmware name"; @@ -1646,7 +1645,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) goto out_fw; } - hdr = (void*)&firmware->data[0]; + hdr = (void *)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); goto out_fw; @@ -1672,7 +1671,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && pos - firmware->size > sizeof(*blk)) { - blk = (void*)(&firmware->data[pos]); + blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); offset = le16_to_cpu(blk->offset); @@ -1816,7 +1815,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; - int val; + unsigned int val; dsp->card = codec->component.card; @@ -1829,7 +1828,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, * For simplicity set the DSP clock rate to be the * SYSCLK rate rather than making it configurable. */ - if(dsp->sysclk_reg) { + if (dsp->sysclk_reg) { ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); if (ret != 0) { adsp_err(dsp, "Failed to read SYSCLK state: %d\n", diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 2d117cf..89b7d71 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -45,8 +45,8 @@ struct wm_adsp { struct list_head alg_regions; - int fw_id; - int fw_id_version; + unsigned int fw_id; + unsigned int fw_id_version; const struct wm_adsp_region *mem; int num_mems; -- 2.1.4

3 22

[alsa-devel] [PATCH v2] ALSA: compress: Add SND_AUDIOCODEC_BESPOKE
by Charles Keepax 23 Dec '15

23 Dec '15

When working with the compressed framework occasionally vendors will use esoteric internal audio formats. For such formats it doesn't really make sense to add an new define to the kernel as their use is not sufficiently general. This patch adds a new define SND_AUDIOCODEC_BESPOKE that vendors can use in such situations. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Added a comment to note that users of BESPOKE are free to use the reserved space in snd_enc_generic. Thanks, Charles include/uapi/sound/compress_params.h | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index d9bd9ca..9625484 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -73,7 +73,8 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE /* * Profile and modes are listed with bit masks. This allows for a @@ -312,7 +313,7 @@ struct snd_enc_flac { struct snd_enc_generic { __u32 bw; /* encoder bandwidth */ - __s32 reserved[15]; + __s32 reserved[15]; /* Can be used for SND_AUDIOCODEC_BESPOKE */ } __attribute__((packed, aligned(4))); union snd_codec_options { -- 1.7.2.5

3 2

[alsa-devel] [PATCH 53/54] sound: soc: ac97: Be sure to clamp return value
by Linus Walleij 23 Dec '15

23 Dec '15

As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Mark: as mentioned in 00/54: either apply this directly or ACK it and I will take it into the GPIO tree. --- sound/soc/soc-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index a2012652f212..7e0acd83b0e6 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -92,7 +92,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) dev_dbg(codec->dev, "get gpio %d : %d\n", offset, ret < 0 ? ret : ret & (1 << offset)); - return ret < 0 ? ret : ret & (1 << offset); + return ret < 0 ? ret : !!(ret & (1 << offset)); } static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, -- 2.4.3

2 1

[alsa-devel] [PATCH] ASoC: rt5616: Return error if device ID mismatch
by Axel Lin 23 Dec '15

23 Dec '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5616.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..0e9414a 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1314,7 +1314,7 @@ static int rt5616_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Device with ID register %#x is not rt5616\n", val); - ret = -ENODEV; + return -ENODEV; } regmap_write(rt5616->regmap, RT5616_RESET, 0); regmap_update_bits(rt5616->regmap, RT5616_PWR_ANLG1, -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: mediatek: Turn AFE on/off in runtime resume/suspend
by Koro Chen 23 Dec '15

23 Dec '15

AFE is actually allowed to be turn on before configuration of DAIs since each DAI has its own enabling control. Turn on/off AFE in runtime resume/suspend to avoid AFE being shut down when closing a DAI while other DAIs are still active. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 24 ++++++------------------ 1 file changed, 6 insertions(+), 18 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 5399a0e..08af9f5 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -382,9 +382,6 @@ static void mtk_afe_i2s_shutdown(struct snd_pcm_substream *substream, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M); mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); - - /* disable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); } static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, @@ -433,9 +430,6 @@ static void mtk_afe_hdmi_shutdown(struct snd_pcm_substream *substream, mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S3_M], afe->clocks[MTK_CLK_I2S3_B]); - - /* disable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); } static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, @@ -679,17 +673,6 @@ static int mtk_afe_dais_hw_free(struct snd_pcm_substream *substream, return snd_pcm_lib_free_pages(substream); } -static int mtk_afe_dais_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mtk_afe *afe = snd_soc_platform_get_drvdata(rtd->platform); - - /* enable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); - return 0; -} - static int mtk_afe_dais_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -757,7 +740,6 @@ static const struct snd_soc_dai_ops mtk_afe_dai_ops = { .shutdown = mtk_afe_dais_shutdown, .hw_params = mtk_afe_dais_hw_params, .hw_free = mtk_afe_dais_hw_free, - .prepare = mtk_afe_dais_prepare, .trigger = mtk_afe_dais_trigger, }; @@ -1118,6 +1100,9 @@ static int mtk_afe_runtime_suspend(struct device *dev) { struct mtk_afe *afe = dev_get_drvdata(dev); + /* disable AFE */ + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); + /* disable AFE clk */ regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_AFE, AUD_TCON0_PDN_AFE); @@ -1164,6 +1149,9 @@ static int mtk_afe_runtime_resume(struct device *dev) /* unmask all IRQs */ regmap_update_bits(afe->regmap, AFE_IRQ_MCU_EN, 0xff, 0xff); + + /* enable AFE */ + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); return 0; err_bck0: -- 1.7.9.5

2 1

[alsa-devel] Applied "ASoC: rt5616: add devicetree document for rt5616" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: rt5616: add devicetree document for rt5616 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 27af4e488c6a2a8c4b6102e777c29fff467a3895 Mon Sep 17 00:00:00 2001 From: Caesar Wang <wxt(a)rock-chips.com> Date: Tue, 22 Dec 2015 13:45:03 +0800 Subject: [PATCH] ASoC: rt5616: add devicetree document for rt5616 Add the description for rt5616 codec. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rt5616.txt | 26 ++++++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt new file mode 100644 index 000000000000..efc48c65198d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -0,0 +1,26 @@ +RT5616 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5616". + +- reg : The I2C address of the device. + +Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +Example: + +codec: rt5616@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; +}; -- 2.6.2

1 0