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

[alsa-devel] [PATCH] ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS
by Kuninori Morimoto 10 May '16

10 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_dmapp_get_id() returns 0xFF as error code if system used strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is prohibited number. In order not to use prohibited number, this patch indicates error message and returns 0x00 (same as SSI00) in error case. Special thanks to Dung-san. Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8f..6bc93cb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: Intel: Add Broxton Machine
by Vinod Koul 10 May '16

10 May '16

Broxton uses SSP5 for audio on Intel board, so add the 6 SSPs and then machine driver which uses RT298 codec Pardha Saradhi K (1): ASoC: Intel: Skylake: Add more SSP DAIs Ramesh Babu (1): ASoC: Intel: Add Broxton-P machine driver sound/soc/intel/Kconfig | 15 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bxt_rt298.c | 353 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 72 ++++++++ 4 files changed, 442 insertions(+) create mode 100644 sound/soc/intel/boards/bxt_rt298.c -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: da7213: clocking related improvements for codec
by Adam Thomson 10 May '16

10 May '16

This is a collection of updates with regards to clocking on da7213 codec, including power efficiency & stability improvements. Patches are based on v4.6-rc7 kernel. Adam Thomson (5): ASoC: da7213: Add DAI DAPM event to control DAI clocks ASoC: da7213: Add checking of SRM lock status before enabling DAI ASoC: da7213: Default PC counter to free-running when DAI disabled ASoC: da7213: Update PLL ranges to improve locking at frequency boundary ASoC: da7213: Allow PLL disable/bypass when using 32KHz sysclk sound/soc/codecs/da7213.c | 99 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 35 ++++++++++------- 2 files changed, 102 insertions(+), 32 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH 0/7] ALSA: drop reuse of incoming packet parameter for outgoing packet parameter
by Takashi Sakamoto 10 May '16

10 May '16

Hi, This patchset updates my former RFC, and go for merging to Linux 4.7. [alsa-devel] [RFC][PATCH 0/6] ALSA: firewire: drop reusing incoming packet parameters for outgoing packet parameters http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107290.html The aim of this patchset is to remove a feature from firewire-lib module. I call it as 'full duplex streaming with timestamp synchronization'. The feature is based on our understanding of IEC 61883-6. In the specification, a function block should reuse some parameters of incoming packets for outgoing packets. Current ALSA firewire stack partly implement it. However, actual audio and music units on IEEE 1394 bus don't expect corresponding drivers to satisfy the feature. On the other hand, the feature brings inconveniences to this module. When this feature is enabled, a software IRQ context processes packets for both of IR/IT contexts. This consumes more time over local CPUs than handling one context. Furthermore, cycle count for outgoing packets becomes improper because they're for incoming packets. For these reasons, this patchset removes the feature from ALSA firewire stack. Changes: - The last patch is newly added. Regards Takashi Sakamoto (7): ALSA: bebob: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: fireworks: drop reuse of incoming packet parameter for ougoing packet parameter ALSA: firewire-tascam: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: firewire-lib: handle IT/IR contexts in each software interrupt context ALSA: firewire-lib: code cleanup for incoming packet handling ALSA: firewire-lib: code cleanup for outgoing packet handling ALSA: firewire-lib: enable the same feature as CIP_SKIP_INIT_DBC_CHECK flag sound/firewire/amdtp-stream.c | 98 +++++++++------------------ sound/firewire/amdtp-stream.h | 36 ++-------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/bebob/bebob_stream.c | 101 +++++++--------------------- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 84 +++++------------------ sound/firewire/oxfw/oxfw-stream.c | 3 +- sound/firewire/tascam/tascam-stream.c | 26 ++++--- 9 files changed, 94 insertions(+), 258 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: pcm: Bail out when chmap is already present
by Takashi Iwai 10 May '16

10 May '16

When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a chmap has been already assigned, it returns as an error due to the conflicting snd_ctl_add() result. However, this also clears the already assigned chmap_kctl field via pcm_chmap_ctl_private_free(), and becomes inconsistent in the later operation. This patch adds the check of the conflicting chmap kctl before actually trying to allocate / assign. The check failure is treated as a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is basically a driver bug and having the stack trace would help developers to figure out the bad code path. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0aca39762ed0..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; -- 2.8.2

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[alsa-devel] [PATCH 3/4] ASoC: ab8500-codec: remove platform data based probe
by Arnd Bergmann 10 May '16

10 May '16

The ux500 platform hasn't used board files for a long time, and nothing defines a ab8500_codec_platform_data, so we can just remove the probing based on that and always use device tree properties directly. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/ab8500-codec.c | 33 ++++----------------------------- 1 file changed, 4 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 8b1d0c1a7839..2fc89155f14a 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2464,45 +2464,20 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) struct device *dev = codec->dev; struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); - struct ab8500_platform_data *pdata; + struct ab8500_codec_platform_data codec_pdata; struct filter_control *fc; int status; dev_dbg(dev, "%s: Enter.\n", __func__); - /* Setup AB8500 according to board-settings */ - pdata = dev_get_platdata(dev->parent); + ab8500_codec_of_probe(dev, np, &codec_pdata); - if (np) { - if (!pdata) - pdata = devm_kzalloc(dev, - sizeof(struct ab8500_platform_data), - GFP_KERNEL); - - if (pdata && !pdata->codec) - pdata->codec - = devm_kzalloc(dev, - sizeof(struct ab8500_codec_platform_data), - GFP_KERNEL); - - if (!(pdata && pdata->codec)) - return -ENOMEM; - - ab8500_codec_of_probe(dev, np, pdata->codec); - - } else { - if (!(pdata && pdata->codec)) { - dev_err(dev, "No codec platform data or DT found\n"); - return -EINVAL; - } - } - - status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + status = ab8500_audio_setup_mics(codec, &codec_pdata.amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); return status; } - status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + status = ab8500_audio_set_ear_cmv(codec, codec_pdata.ear_cmv); if (status < 0) { pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", __func__, status); -- 2.7.0

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[alsa-devel] bug: reconfig broken on kernel 4.5
by Jochen Henneberg 10 May '16

10 May '16

Hi, it looks as if 'reconfig' feature is broken since the sysfs attributes have been moved to device attributes. In hda_sysfs.c:reconfig_codec() the device is cleared with snd_hda_codec_reset() which calls snd_hdac_device_unregister() which removes the sysfs entries (including the currently in-use 'reconfig'), the re-registration fails afterwards in snd_hda_codec_build_controls(). I am not sure when things got broken, but snd_hda_codec_reset() called from reconfig_codec() unregisters the device and removes the sysfs entries where in the past it only stripped all configurations (pcms, controls ...) from the device. Regards, Jochen

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[alsa-devel] [PATCH] ALSA: Fix subwoofer pin on ASUS N751 and N551
by Takashi Iwai 10 May '16

10 May '16

From: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Subwoofer does not work out of the box on ASUS N751/N551 laptops. This patch fixes it. Patch tested on N751 laptop. N551 part is not tested, but according to [1] and [2] this laptop requires similar changes, so I included them in the patch. 1. https://github.com/honsiorovskyi/asus-n551-hda-fix 2. https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/1405691 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117781 Signed-off-by: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b5cd19876fc..4918ffa5ba68 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6426,6 +6426,7 @@ enum { ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51, }; static const struct hda_fixup alc662_fixups[] = { @@ -6672,6 +6673,15 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_BASS_1A }, + [ALC668_FIXUP_ASUS_Nx51] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x90170151}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6700,6 +6710,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), + SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 2.8.2

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[alsa-devel] Kernel config issue: SND_HDA_RECONFIG depends on SND_HDA_HWDEP
by Jochen Henneberg 09 May '16

09 May '16

Hi, with kernel 4.5 (and it looks as if this has not changed since then) the SND_HDA_RECONFIG sysfs entries still show up in the hwdep devices. Enabling SND_HDA_RECONFIG without SND_HDA_HWDEP is worth nothing, but the dependency has been removed from Kconfig. The commits from 13aeaf68 / 'Manage each codec instance individually' read as if the sysfs entries should show up at the codec driver but they do not. Regards, Jochen

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[alsa-devel] [PATCH 0/5] Fix poll error returns
by Charles Keepax 09 May '16

09 May '16

We return negative values from the file_operations poll callback in a few places, this callback returns an unsigned int and is expected to only return the poll specific flags. This series fixes up these issues and makes a couple of small tidy ups to the code around the edges of those functions. I noticed this issue whilst doing some additional testing on my propagation of compressed stream error series, but I decided to push these up separately as it is worth getting the fixes in now and not tying them up with that patch chain which is taking longer to get merged. Also I included the first patch of that chain (Replace complex if statement with switch) because it is a trivial tidy up and might as well get merged now as well. Thanks, Charles Charles Keepax (5): ALSA: pcm: Fix poll error return codes ALSA: compress: Use snd_compr_get_poll on error path ALSA: compress: Remove pointless NULL check ALSA: compress: Fix poll error return codes ALSA: compress: Replace complex if statement with switch sound/core/compress_offload.c | 25 ++++++++++++------------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 14 insertions(+), 15 deletions(-) -- 2.1.4

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