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[95.84.228.163]) by smtp.gmail.com with ESMTPSA id p9sm358906lfu.83.2022.01.13.10.26.00 (version=TLS1 cipher=ECDHE-ECDSA-AES128-SHA bits=128/128); Thu, 13 Jan 2022 10:26:01 -0800 (PST) Date: Thu, 13 Jan 2022 21:31:41 +0300 From: Alexander Sergeyev To: Takashi Iwai Subject: Re: [PATCH 1/4] ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8 Message-ID: <20220113183141.kla37mbqmo4x6wxp@localhost.localdomain> User-Agent: mtt References: <20210519170357.58410-1-jeremy.szu@canonical.com> <20220111195229.a77wrpjclqwrx4bx@localhost.localdomain> <20220112101249.ya73jvpmqmeh4ggg@localhost.localdomain> <20220112104827.4aymoth7ua65nwge@localhost.localdomain> <20220112201824.qmphnz2hx4frda6e@localhost.localdomain> MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii; format=flowed Content-Disposition: inline In-Reply-To: Cc: "moderated list:SOUND" , Kailang Yang , Jeremy Szu , Huacai Chen , open list , tiwai@suse.com, Hui Wang , PeiSen Hou , Jian-Hong Pan X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" On Thu, Jan 13, 2022 at 08:14:29AM +0100, Takashi Iwai wrote: >> First, about unbind and bind via sysfs -- attempts to unbind the >> HD-audio controller immediately trigger BUGs: >> Is it normal/expected? > >A sort of. The sysfs unbind is little tested and may be still buggy >if done during the stream operation. > >To be sure, could you check with my latest sound.git tree for-linus >branch? There are a few fixes that harden the dynamic unbind. I assume that the referred repository is the one at [1]. I've tried 081c73701ef0 "ALSA: hda: intel-dsp-config: reorder the config table". It crashed with nearly identical logs. >> 1) Coeff 0x0b is flapping between 0x8003 and 0x7770 and does not seem >> to have any effect in both non-working and working versions. Not sure >> about this, maybe microphone is not operational since I haven't >> checked it yet. I got some time to poke the internal microphone. It works, but oddities are there as well. Initially I get "Mic Boost", "Capture" and "Internal Mic Boost" controls in alsamixer. When I run arecord, "Digital" control appears, but it cannot be changed until arecord is stopped. Subsequent arecord calls do not lock "Digital" control. This control affects sensitivity of the microphone and seems useful. /proc/asound/card1/codec#0: Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC285 Analog", type="Audio", device=0 Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 Amp-In vals: [0x27 0x27] - Converter: stream=0, channel=0 + Converter: stream=1, channel=0 This is the only change in /proc/asound after the first arecord run. Overall, seems like a small annoyance, but I'm curious -- is it how it's supposed to work? [1]https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/?h=for-linus