August 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ALSA: usb: move udh01_fb_quirk setting to quirks.c
by Daniel Mack 22 Aug '16

22 Aug '16

That's a quirk, after all, so move it where to all the other quirks live. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/usb/endpoint.c | 4 ---- sound/usb/quirks.c | 5 +++++ 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c07a7ed..4b4ffa0 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -502,10 +502,6 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, ep->syncinterval = 3; ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); - - if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && - ep->syncmaxsize == 4) - ep->udh01_fb_quirk = 1; } list_add_tail(&ep->list, &chip->ep_list); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6cf1f35..299813f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1216,6 +1216,11 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->chip->usb_id == USB_ID(0x0763, 0x2031)) && ep->type == SND_USB_ENDPOINT_TYPE_DATA) ep->skip_packets = 16; + + /* Work around devices that report unreasonable feedback data */ + if (ep->chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && + ep->syncmaxsize == 4) + ep->udh01_fb_quirk = 1; } void snd_usb_set_interface_quirk(struct usb_device *dev) -- 2.5.5

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[alsa-devel] [PATCH] mixer: Fix rounding mode documentation
by David Henningsson 22 Aug '16

22 Aug '16

The documentation specifies three different rounding modes, but this is wrong, because they all end up calling snd_tlv_convert_from_dB, which only has two rounding modes. Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- src/mixer/simple.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/src/mixer/simple.c b/src/mixer/simple.c index fd9ba93..82476d5 100644 --- a/src/mixer/simple.c +++ b/src/mixer/simple.c @@ -376,7 +376,7 @@ int snd_mixer_selem_ask_playback_vol_dB(snd_mixer_elem_t *elem, long value, long * \brief Return corresponding integer playback volume for given dB value for a mixer simple element * \param elem Mixer simple element handle * \param value value to be converted to dB range - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \param dBvalue pointer to returned dB value * \return 0 on success otherwise a negative error code */ @@ -454,7 +454,7 @@ int snd_mixer_selem_set_playback_volume(snd_mixer_elem_t *elem, snd_mixer_selem_ * \param elem Mixer simple element handle * \param channel mixer simple element channel identifier * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_playback_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -491,7 +491,7 @@ int snd_mixer_selem_set_playback_volume_all(snd_mixer_elem_t *elem, long value) * \brief Set value in dB of playback volume control for all channels of a mixer simple element * \param elem Mixer simple element handle * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_playback_dB_all(snd_mixer_elem_t *elem, long value, int dir) @@ -706,7 +706,7 @@ int snd_mixer_selem_ask_capture_vol_dB(snd_mixer_elem_t *elem, long value, long * \param elem Mixer simple element handle * \param dBvalue dB value to be converted to integer range * \param value pointer to returned integer value - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_ask_capture_dB_vol(snd_mixer_elem_t *elem, long dBvalue, int dir, long *value) @@ -777,7 +777,7 @@ int snd_mixer_selem_set_capture_volume(snd_mixer_elem_t *elem, snd_mixer_selem_c * \param elem Mixer simple element handle * \param channel mixer simple element channel identifier * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_capture_dB(snd_mixer_elem_t *elem, snd_mixer_selem_channel_id_t channel, long value, int dir) @@ -814,7 +814,7 @@ int snd_mixer_selem_set_capture_volume_all(snd_mixer_elem_t *elem, long value) * \brief Set value in dB of capture volume control for all channels of a mixer simple element * \param elem Mixer simple element handle * \param value control value in dB * 100 - * \param dir select direction (-1 = accurate or first bellow, 0 = accurate, 1 = accurate or first above) + * \param dir rounding mode - rounds up if dir > 0, otherwise rounds down * \return 0 on success otherwise a negative error code */ int snd_mixer_selem_set_capture_dB_all(snd_mixer_elem_t *elem, long value, int dir) -- 1.9.1

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[alsa-devel] [PATCH 0/2] amidi: fix handling of ignored messages
by Clemens Ladisch 22 Aug '16

22 Aug '16

Clemens Ladisch (2): amidi: ignore not only Active Sensing but also Clock bytes amidi: fix timeout handling amidi/amidi.1 | 19 ++++++++--- amidi/amidi.c | 82 +++++++++++++++++++++++++++++++++++++++----------- 2 files changed, 78 insertions(+), 23 deletions(-)

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[alsa-devel] [PATCH v4 0/4] ALSA: seq: obsolete change of address limit
by Takashi Sakamoto 22 Aug '16

22 Aug '16

ALSA sequencer core has two types of clients; application and kernel. For kernel clients, the core should handle data in kernel space, meanwhile this is achieved with a hack of change of address limit. This should be purged. This patchset is a revised version of my previous one: [alsa-devel] obsolete change of address limit http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/110937.html Changes in v2: - Use '_IOC_SIZE' macro to calculate the size of argument of ioctl command. - Keep redundant longer name. - Improve comments. - Merge some patches relevant to the same features. Changes in v3: - Improve series of patch - Use 'dir' field of ioctl command to detect the length of argument Changes in v4: - Update kernel API documentation - Use logical AND to detect ioctl direction Takashi Sakamoto (4): ALSA: seq: add documentation for snd_seq_kernel_client_ctl ALSA: seq: add an alternative way to handle ioctl requests ALSA: seq: change ioctl command operation to get data in kernel space ALSA: seq: obsolete change of address limit sound/core/seq/seq_clientmgr.c | 721 ++++++++++++++++++----------------------- sound/core/seq/seq_compat.c | 7 +- 2 files changed, 317 insertions(+), 411 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH -next] ASoC: sun4i-codec: Fix error return code in sun4i_codec_probe()
by Wei Yongjun 22 Aug '16

22 Aug '16

Fix to return error code -ENOMEM instead of 0 when create card failed, as done elsewhere in this function. Fixes: 45fb6b6f2aa3 ("ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs") Signed-off-by: Wei Yongjun <weiyj.lk(a)gmail.com> --- sound/soc/sunxi/sun4i-codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 0e19c50..e741659 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -846,6 +846,7 @@ static int sun4i_codec_probe(struct platform_device *pdev) card = sun4i_codec_create_card(&pdev->dev); if (!card) { dev_err(&pdev->dev, "Failed to create our card\n"); + ret = -ENOMEM; goto err_unregister_codec; }

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[alsa-devel] [PATCH 000/127] ASoC: codec cleanup - codec probe/remove
by Kuninori Morimoto 22 Aug '16

22 Aug '16

Hi Mark These are codec duplicate functions cleanup patches - v2 We would like to switch to "component" style base ASoC in the future instead of current "CPU/Codec/Platform/Card" style. Current "codec" has its .probe/.remove, and "component" has it too. codec side is just relayed it to component side. This is the time to cleanup. These are very big patch-set, but it doesn't add new features. [001/127] - [126/127] move callback functions from codec to component, and last [127/127] will remove codec side .probe/.remove. We need this kind of cleanup for Platform too. I build-tested these patches with make allyesconfig on x86, and tested on Lager board. Best regards --- Kuninori Morimoto

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Re: [alsa-devel] [PATCH] Try to support analog audio codec at firefly reload
by John Keeping 20 Aug '16

20 Aug '16

On Tue, Aug 16, 2016 at 04:42:06AM +0800, Randy Li wrote: > This patch could enabled the es8328 codec at firefly reload board, > but it doesn't not really work. I disable the mute of channel > Left mix and Right mix in alsamixer, I could hear the noise from > my headphone, but it never be a music. > I just send this patch here to see whether somebody could figure > out what is wrong, I am really not familiar with sound. Have you tried adding an mclk-fs setting to the simple-audio-card? You probably want to set it to 256, which will allow 44.1kHz, 48kHz, 88.2kHz and 96kHz at the codec. This will give more functionality than pinning the clock to a specific rate. For the Rock2 Square, I write a machine driver for the sound because the headphone GPIO setup cannot be made to work with simple-audio-card; I found the Firefly Reload schematic and it looks like the configuration hardware is very similar. It's a bit of a mess but you can find it on Github [1] if you want to take a look. [1] https://github.com/johnkeeping/linux/tree/topic/audio

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[alsa-devel] [PATCH 1/3] ASoC: fsl_asrc: Propagate the real error code
by Fabio Estevam 19 Aug '16

19 Aug '16

Instead of returning -EINVAL on error, return the real error code. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c1a0e01..15f8ba5 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -892,7 +892,7 @@ static int fsl_asrc_probe(struct platform_device *pdev) ret = fsl_asrc_init(asrc_priv); if (ret) { dev_err(&pdev->dev, "failed to init asrc %d\n", ret); - return -EINVAL; + return ret; } asrc_priv->channel_avail = 10; @@ -901,14 +901,14 @@ static int fsl_asrc_probe(struct platform_device *pdev) &asrc_priv->asrc_rate); if (ret) { dev_err(&pdev->dev, "failed to get output rate\n"); - return -EINVAL; + return ret; } ret = of_property_read_u32(np, "fsl,asrc-width", &asrc_priv->asrc_width); if (ret) { dev_err(&pdev->dev, "failed to get output width\n"); - return -EINVAL; + return ret; } if (asrc_priv->asrc_width != 16 && asrc_priv->asrc_width != 24) { -- 1.9.1

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[alsa-devel] [PATCH for 4.8] ASoC: omap-abe-twl6040: Correct dmic-codec device registration
by Peter Ujfalusi 19 Aug '16

19 Aug '16

The dmic-codec was registered within the platform_driver's probe function, which can cause deferred probe to run in loops as reported and analyzed by Russell King. Use module_init/exit in the driver and handle the dmic-codec device registration and removal at that level instead of the platform_driver probe/remove. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Russell King <linux(a)arm.linux.org.uk> --- sound/soc/omap/omap-abe-twl6040.c | 61 ++++++++++++++++++++------------------- 1 file changed, 32 insertions(+), 29 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 0843a68f277c..f61b3b58083b 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -38,10 +38,10 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ - - struct platform_device *dmic_codec_dev; }; +struct platform_device *dmic_codec_dev; + static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -258,8 +258,6 @@ static int omap_abe_probe(struct platform_device *pdev) if (priv == NULL) return -ENOMEM; - priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -284,13 +282,6 @@ static int omap_abe_probe(struct platform_device *pdev) num_links = 2; abe_twl6040_dai_links[1].cpu_of_node = dai_node; abe_twl6040_dai_links[1].platform_of_node = dai_node; - - priv->dmic_codec_dev = platform_device_register_simple( - "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } } else { num_links = 1; } @@ -299,16 +290,14 @@ static int omap_abe_probe(struct platform_device *pdev) of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + return -EINVAL; } card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - ret = -ENODEV; - goto err_unregister; + return -ENODEV; } card->dai_link = abe_twl6040_dai_links; @@ -317,17 +306,9 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err_unregister; - } - - return 0; - -err_unregister: - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -335,13 +316,9 @@ err_unregister: static int omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); - return 0; } @@ -361,7 +338,33 @@ static struct platform_driver omap_abe_driver = { .remove = omap_abe_remove, }; -module_platform_driver(omap_abe_driver); +static int __init omap_abe_init(void) +{ + int ret; + + dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL, + 0); + if (IS_ERR(dmic_codec_dev)) { + pr_err("%s: dmic-codec device registration failed\n", __func__); + return PTR_ERR(dmic_codec_dev); + } + + ret = platform_driver_register(&omap_abe_driver); + if (ret) { + pr_err("%s: platform driver registration failed\n", __func__); + platform_device_unregister(dmic_codec_dev); + } + + return ret; +} +module_init(omap_abe_init); + +static void __exit omap_abe_exit(void) +{ + platform_driver_unregister(&omap_abe_driver); + platform_device_unregister(dmic_codec_dev); +} +module_exit(omap_abe_exit); MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec"); -- 2.9.2

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Re: [alsa-devel] [PATCH] Try to support analog audio codec at firefly reload
by Heiko Stuebner 19 Aug '16

19 Aug '16

Am Dienstag, 16. August 2016, 04:42:06 CEST schrieb Randy Li: > Hello: > This patch could enabled the es8328 codec at firefly reload board, > but it doesn't not really work. I disable the mute of channel > Left mix and Right mix in alsamixer, I could hear the noise from > my headphone, but it never be a music. > I just send this patch here to see whether somebody could figure > out what is wrong, I am really not familiar with sound. I can't really help here as I have neither the board nor that much sound experience, but it's definitly customary to label patches as rfc when they're expected to need more work, so make that [PATCH RFC] ... for stuff that requires more work before it can actually be merged. Heiko

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