February 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hdsp: remove redundant assignment to variable err
by Colin King 10 Feb '20

10 Feb '20

From: Colin Ian King <colin.king(a)canonical.com> Variable err is being assigned with a value that is never read, it is assigned a new value in the next statement. The assignment is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cc06f0a1a7e4..227aece17e39 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3353,7 +3353,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) return; } } else { - int err = -EINVAL; + int err; + err = hdsp_request_fw_loader(hdsp); if (err < 0) { snd_iprintf(buffer, -- 2.25.0

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[alsa-devel] [PATCH 0/4] ALSA: hda/hdmi: Clean up jack handling
by Takashi Iwai 10 Feb '20

10 Feb '20

Hi, this is a series of cleanup of the jack handling in HD-audio HDMI codec driver, which I realized after the recent regression fix. Basically this changes the jack handling from hda_jack to the own jack objects as Intel HDMI did but applying now to all (generic) codecs. It resulted in a good amount of code reduction in the end. As this is no real fixes, and I'd like to rather share the code between 5.5 and 5.6, it's targeted for 5.7. Only lightly tested on Intel and nouveau. Takashi === Takashi Iwai (4): ALSA: hda/hdmi: Reduce hda_jack_tbl lookup at unsol event handling ALSA: hda/hdmi: Don't use standard hda_jack for generic HDMI jacks ALSA: hda/hdmi: Move runtime PM resume into hdmi_present_sense_via_verbs() ALSA: hda/hdmi: Move ELD parse and jack reporting into update_eld() sound/pci/hda/patch_hdmi.c | 313 +++++++++++++-------------------------------- 1 file changed, 90 insertions(+), 223 deletions(-) -- 2.16.4

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[alsa-devel] broadwell-rt286: more conformance and other fixes
by GitHub pull_request - opened 09 Feb '20

09 Feb '20

alsa-project/alsa-ucm-conf pull request #12 was opened from dpward: Remove JackHWMute, because the hardware does not forcibly shut off the internal speakers or microphone when the jack is in use. Set either PlaybackPriority or CapturePriority for each device. For capture devices, the "ADC0 Capture Switch" was never enabled. CaptureMixerElem and CaptureMasterElem were swapped. Correct "Handset" to "Headset". Apply maximum gain to the headset microphone ("AMIC Volume") when in use, based on testing. Signed-off-by: David Ward <david.ward(a)ll.mit.edu> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/12 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/12.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[alsa-devel] [PATCH v2 0/2] Update FE/BE trigger order based on the command
by Ranjani Sridharan 09 Feb '20

09 Feb '20

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. The above change has been tested with the Skylake driver on the Kabylake based chromebook and legacy HDA driver on Kabylake based Dell XPS13. It has also been tested on all platforms with SOF CI without any regressions for the past 3 weeks. More details on the issue can be found here: https://github.com/thesofproject/linux/issues/1160 Changes in v2: Fixed comments on function name and consolidating the switch cases. Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 95 ++++++++++++++++++++++++++++------------ sound/soc/sof/topology.c | 4 ++ 2 files changed, 72 insertions(+), 27 deletions(-) -- 2.17.1

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[alsa-devel] alsa-gobject: fix workflow of Github Actions to clone requested repository
by GitHub pull_request - opened 09 Feb '20

09 Feb '20

alsa-project/alsa-gobject pull request #7 was opened from takaswie: Even if any pull request is triggered, in 'build_in_ubuntu_on_lxd' job of 'Build test' workflow, upstream repository is cloned and requested git hash is checked out. However this fails because the hash is in requested repository. This commit fixes the workflow so that requested repository is cloned. Fixes: 8cb6715aab65 ('Add support workflow of Github Actions') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/7 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/7.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[alsa-devel] alsa-gobject Add shared library to operate ALSA Timer character device
by GitHub pull_request - opened 09 Feb '20

09 Feb '20

alsa-project/alsa-gobject pull request #6 was opened from takaswie: This request is to add a new library, libalsatimer, to operate ALSA Timer character device. In design of ALSA Timer core, one character device is added to Linux system. Usually a special file, '/dev/snd/timer' is used for the character device. Userspace applications can enumerate actual timer device to operate the character device. For enumerations of the timer devices, the library includes a global functions, 'alsatimer_get_device_id_list()'. The timer devices are identified by 'ALSATimerDeviceId' which is a GObject boxed object to handle 'struct snd_timer_id'. The other global functions are available to the timer device, as a wrapper of some ioctl commands below: * alsatimer_get_device_info() = SNDRV_TIMER_IOCTL_GINFO * alsatimer_get_device_status() = SNDRV_TIMER_IOCTL_GPARAMS * alsatimer_set_device_params() = SNDRV_TIMER_IOCTL_GSTATUS Userspace applications can allocate timer instances and attach them to the timer device to receive timer events. The library includes 'ALSATimerUserInstance' which is an GObject-derived object to maintain file descriptor of ALSA Timer character device. Usually, the series of below calls are expected to use in the order to start emission of timer events: * alsatimer_user_instance_new() * alsatimer_user_instance_open() * alsatimer_user_instance_attach() * alsatimer_user_instance_set_params() * alsatimer_user_instance_start() * alsatimer_user_instance_stop() Against the attachment to timer device, there's a way to associate the instance to the another instance as slave. This is achieved by a call of 'alsatimer_user_instance_attach_as_slave()'. The 'ALSATimerDeviceId' is the lack of properties relevant to slave association since the information relevant to slave association in kernel space is not available for userspace application at all and the properties are just used to transfer the information from userspace by one way. ALSA Timer core supports two types of events, which is switched by 'tread operation'. To handle these two, the library has 'ALSATimerEventData' which is an GObject-derived abstract class. There are two specialized classes from it; 'ALSATimerEventDataTick' and 'ALSATimerEventDataTimestamp'. The former is just for tick event and the latter is for events with its type and timestamp. The selection is done by a call of attach function with the type of event data. Request URL : https://github.com/alsa-project/alsa-gobject/pull/6 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[alsa-devel] [PATCH][next] ASoC: wcd934x: remove redundant null checks on rx_chs and tx_chs
by Colin King 08 Feb '20

08 Feb '20

From: Colin Ian King <colin.king(a)canonical.com> The null checks on wcd->rx_chs and wcd->tx_chs are redundant as these objects are arrays and not pointers and hence can never be null. Remove the redundant null checks. Addresses-Coverity: ("Array compared against 0") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wcd934x.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 158e878abd6c..e780ecd554d2 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1883,20 +1883,16 @@ static int wcd934x_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; -- 2.24.0

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[alsa-devel] USB Audio Interface / Denon MC7000 and MC8000 controller
by Tobias 07 Feb '20

07 Feb '20

Hello dear ALSA developers. I have purchased a MC7000 controller in order to control MIXXX on Ubuntu 16.04. According to the Denon specification the controller should have been class compliant but there is an issue with the Audio interface to work properly giving following message ... $ dmesg ... |[ 74.522831] usb 1-1.3: new high-speed USB device number 6 using xhci_hcd [ 74.623784] usb 1-1.3: New USB device found, idVendor=15e4, idProduct=8004 [ 74.623789] usb 1-1.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 74.623793] usb 1-1.3: Product: DENON DJ MC7000 [ 74.623796] usb 1-1.3: Manufacturer: DENON DJ [ 74.623798] usb 1-1.3: SerialNumber: 201603 [ 74.625134] usb 1-1.3: clock source 65 is not valid, cannot use "|||clock source 65 is not valid, cannot use" is repeated uncountable times then |I found a discussion here for the MC8000 Audio interface which was not working and showing the same error message: https://alsa-user.narkive.com/2tDAO87f/troubleshooting-new-usb-audio-device… The solution was to change sound/usb/clock.c and recompile the kernel. This is not practical for daily use case so I would like to ask if the ALSA team could possibly find a permanent fix for ordinary users. To get my outputs for aplay -l aplay -L |||aplay -D plughw:CARD=MC7000,DEV=0 Grimmaldika-MakeBelieve.wav |lsusb -v mixxx --controllerDebug|| JackServer error message| you may have a look at the Mixxx Community Forums here(just on page, you don't need to find and read on multiple pages) https://www.mixxx.org/forums/viewtopic.php?f=7&t=12962&start=10 If you need any more information then please let me know. Thanks a lot for your support on this matter. Cheers! OsZ ||

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[alsa-devel] [PATCH] ALSA: uapi: Replace __packed with __attribute__((packed))
by Amadeusz Sławiński 07 Feb '20

07 Feb '20

Userspace doesn't know what __packed is, change it to __attribute__((packed)), like in the rest of a header. Fixes: 348f48220b97 (ASoC: topology: Move v4 manifest header data structures to uapi) Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- include/uapi/sound/asoc.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6048553c119d..1ae8b06691cb 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -586,7 +586,7 @@ struct snd_soc_tplg_manifest_v4 { __le32 pcm_elems; /* number of PCM elements */ __le32 dai_link_elems; /* number of DAI link elements */ struct snd_soc_tplg_private priv; -} __packed; +} __attribute((packed)); /* Stream Capabilities v4 */ struct snd_soc_tplg_stream_caps_v4 { @@ -604,7 +604,7 @@ struct snd_soc_tplg_stream_caps_v4 { __le32 period_size_max; /* max period size bytes */ __le32 buffer_size_min; /* min buffer size bytes */ __le32 buffer_size_max; /* max buffer size bytes */ -} __packed; +} __attribute((packed)); /* PCM v4 */ struct snd_soc_tplg_pcm_v4 { @@ -619,7 +619,7 @@ struct snd_soc_tplg_pcm_v4 { struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */ __le32 num_streams; /* number of streams */ struct snd_soc_tplg_stream_caps_v4 caps[2]; /* playback and capture for DAI */ -} __packed; +} __attribute((packed)); /* Physical link config v4 */ struct snd_soc_tplg_link_config_v4 { @@ -627,6 +627,6 @@ struct snd_soc_tplg_link_config_v4 { __le32 id; /* unique ID - used to match */ struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */ __le32 num_streams; /* number of streams */ -} __packed; +} __attribute((packed)); #endif -- 2.17.1

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[alsa-devel] [PATCH] ASoC: soc-core: fix an uninitialized use
by Jian Cai 07 Feb '20

07 Feb '20

Fixed the uninitialized use of a signed integer variable ret in soc_probe_component when all its definitions are not executed. This caused -ftrivial-auto-var-init=pattern to initialize the variable to repeated 0xAA (i.e. a negative value) and triggered the following code unintentionally. err_probe: if (ret < 0) soc_cleanup_component(component); Signed-off-by: Jian Cai <caij2003(a)gmail.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a..bfb813ba34f3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1180,7 +1180,7 @@ static int soc_probe_component(struct snd_soc_card *card, snd_soc_component_get_dapm(component); struct snd_soc_dai *dai; int probed = 0; - int ret; + int ret = 0; if (!strcmp(component->name, "snd-soc-dummy")) return 0; -- 2.25.0.341.g760bfbb309-goog

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