August 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] topology: add support to parse private data for pcm
by Keyon Jie 06 Aug '19

06 Aug '19

We have private data section in struct snd_soc_tplg_pcm, but alsatplg doesn't support handling it yet, here add handling in tplg_parse_pcm() to enable it. Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- src/topology/data.c | 3 +++ src/topology/pcm.c | 8 ++++++++ 2 files changed, 11 insertions(+) diff --git a/src/topology/data.c b/src/topology/data.c index b3f4421f..729ce1f4 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -53,6 +53,9 @@ struct snd_soc_tplg_private *get_priv_data(struct tplg_elem *elem) case SND_TPLG_TYPE_BE: priv = &elem->link->priv; break; + case SND_TPLG_TYPE_PCM: + priv = &elem->pcm->priv; + break; default: SNDERR("error: '%s': no support for private data for type %d\n", elem->id, elem->type); diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 5f586dc1..c533ee68 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -732,6 +732,14 @@ int tplg_parse_pcm(snd_tplg_t *tplg, return err; continue; } + + /* private data */ + if (strcmp(id, "data") == 0) { + err = tplg_parse_data_refs(n, elem); + if (err < 0) + return err; + continue; + } } return 0; -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: fix a memory leak bug
by Wenwen Wang 06 Aug '19

06 Aug '19

In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is allocated through kzalloc() before the execution goto 'found_clock'. However, this structure is not deallocated if the memory allocation for 'pd' fails, leading to a memory leak bug. To fix the above issue, free 'fp->chmap' before returning NULL. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 7ee9d17..e852c7f 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1043,6 +1043,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { + kfree(fp->chmap); kfree(fp->rate_table); kfree(fp); return NULL; -- 2.7.4

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[alsa-devel] [PATCH] slimbus: qcom-ngd-ctrl: Add of_node_put() before return
by Nishka Dasgupta 06 Aug '19

06 Aug '19

Each iteration of for_each_available_child_of_node puts the previous node, but in the case of a return from the middle of the loop, there is no put, thus causing a memory leak. Hence add an of_node_put before the return in two places. Issue found with Coccinelle. Signed-off-by: Nishka Dasgupta <nishkadg.linux(a)gmail.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index f3585777324c..29fbab55c3b3 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1338,12 +1338,15 @@ static int of_qcom_slim_ngd_register(struct device *parent, continue; ngd = kzalloc(sizeof(*ngd), GFP_KERNEL); - if (!ngd) + if (!ngd) { + of_node_put(node); return -ENOMEM; + } ngd->pdev = platform_device_alloc(QCOM_SLIM_NGD_DRV_NAME, id); if (!ngd->pdev) { kfree(ngd); + of_node_put(node); return -ENOMEM; } ngd->id = id; -- 2.19.1

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[alsa-devel] [PATCH v1 0/3] Allow reconfiguration of clock rate
by Jiada Wang 06 Aug '19

06 Aug '19

There is use case in Gstreamer ALSA sink, in case of changed caps Gsreatmer reconfigs and it calls snd_pcm_hw_free() before snd_pcm_prepre(). See gstreamer1.0-plugins-base/ gst-libs/gst/audio/gstaudiobasesink.c: gst_audio_base_sink_setcaps(): - gst_audio_ring_buffer_release() - gst_audio_sink_ring_buffer_release() - gst_alsasink_unprepare() - snd_pcm_hw_free() is called before - gst_audio_ring_buffer_acquire() - gst_audio_sink_ring_buffer_acquire() - gst_alsasink_prepare() - set_hwparams() - snd_pcm_hw_params() - snd_pcm_prepare() But with current implementation after clock rate is started in .prepare reconfiguration of clock rate is not allowed, unless the stream is stopped. This patch set by move stop of clock to .hw_free callback, to allow reconfiguration of clock rate. Jiada Wang (1): ASoC: rsnd: call .hw_{params,free} in pair for same stream Timo Wischer (2): ASoC: rsnd: Support hw_free() callback at DAI level ASoC: rsnd: Allow reconfiguration of clock rate sound/soc/sh/rcar/core.c | 22 +++++++++++++-- sound/soc/sh/rcar/rsnd.h | 36 ++++++++++++++++++++---- sound/soc/sh/rcar/ssi.c | 61 +++++++++++++++++++++++++++++----------- sound/soc/sh/rcar/ssiu.c | 3 +- 4 files changed, 96 insertions(+), 26 deletions(-) -- 2.19.2

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[alsa-devel] [PATCH] ALSA: usb-audio: fix a memory leak bug
by Wenwen Wang 06 Aug '19

06 Aug '19

In add_new_ctl(), a mixer element structure is allocated through kzalloc() and the pointer is saved to 'elem'. Later on, a new alsa control element is created and added to this structure. In case the add process fails, i.e., the return value of snd_usb_mixer_add_control() is less than 0, the allocated structure is not freed, leading to a memory leak. To fix the above issue, free 'elem' before returning the error. Signed-off-by: Wenwen Wang <wenwen(a)cs.uga.edu> --- sound/usb/mixer_scarlett.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 83715fd..a6c028a 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -562,8 +562,10 @@ static int add_new_ctl(struct usb_mixer_interface *mixer, strlcpy(kctl->id.name, name, sizeof(kctl->id.name)); err = snd_usb_mixer_add_control(&elem->head, kctl); - if (err < 0) + if (err < 0) { + kfree(elem); return err; + } if (elem_ret) *elem_ret = elem; -- 2.7.4

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[alsa-devel] [PATCH v2 00/29] ASoC: add soc-component.c
by Kuninori Morimoto 06 Aug '19

06 Aug '19

Hi Mark These are v2 of soc-component.c patch-set. I want to add multi CPU DAI support to ALSA SoC. But I noticed that we want to cleanup ALSA SoC before that. These patches will do nothing from "technical" point of view, but, will makes code readable. These are focusing to "component". Actually, soc-io.c is also for component. I'm thinking that we can merge soc-io.c into soc-component.c, but do nothing by these patch-set. 1) - 4) : new patch. it removes un-used component functions 5) - 19) : small bug fixed from v1 20) - 22) : more cleanuped 23) - 29) : cares for_each_rtdcom() loop Kuninori Morimoto (29): 1) ASoC: Intel: skl-pcm: disable skl_get_time_info 2) ASoC: soc-pcm: remove soc_rtdcom_ack() 3) ASoC: soc-pcm: remove soc_rtdcom_copy_kernel() 4) ASoC: soc-pcm: remove soc_fill_silence() 5) ASoC: add soc-component.c 6) ASoC: soc-component: add snd_soc_component_get/put() 7) ASoC: soc-component: add snd_soc_component_open() 8) ASoC: soc-component: add snd_soc_component_close() 9) ASoC: soc-component: add snd_soc_component_prepare() 10) ASoC: soc-component: add snd_soc_component_hw_params() 11) ASoC: soc-component: add snd_soc_component_hw_free() 12) ASoC: soc-component: add snd_soc_component_trigger() 13) ASoC: soc-component: add snd_soc_component_suspend() 14) ASoC: soc-component: add snd_soc_component_resume() 15) ASoC: soc-component: add snd_soc_component_is_suspended() 16) ASoC: soc-component: add snd_soc_component_probe() 17) ASoC: soc-component: add snd_soc_component_remove() 18) ASoC: soc-component: add snd_soc_component_of_xlate_dai_id() 19) ASoC: soc-component: add snd_soc_component_of_xlate_dai_name() 20) ASoC: soc-component: move snd_soc_component_seq_notifier() 21) ASoC: soc-component: move snd_soc_component_stream_event() 22) ASoC: soc-component: move snd_soc_component_set_bias_level() 23) ASoC: soc-component: add snd_soc_pcm_component_pointer() 24) ASoC: soc-component: add snd_soc_pcm_component_ioctrl() 25) ASoC: soc-component: add snd_soc_pcm_component_copy_user() 26) ASoC: soc-component: add snd_soc_pcm_component_page() 27) ASoC: soc-component: add snd_soc_pcm_component_mmap() 28) ASoC: soc-component: add snd_soc_pcm_component_pcm_new() 29) ASoC: soc-component: add snd_soc_pcm_component_pcm_free() include/sound/soc-component.h | 387 ++++++++++++++++++++++++++ include/sound/soc-dapm.h | 6 - include/sound/soc.h | 306 +-------------------- sound/soc/Makefile | 2 +- sound/soc/intel/skylake/skl-pcm.c | 3 +- sound/soc/soc-component.c | 561 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 123 ++------- sound/soc/soc-dapm.c | 28 +- sound/soc/soc-jack.c | 18 -- sound/soc/soc-pcm.c | 260 ++---------------- sound/soc/soc-utils.c | 199 -------------- 11 files changed, 1012 insertions(+), 881 deletions(-) create mode 100644 include/sound/soc-component.h create mode 100644 sound/soc/soc-component.c -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC updates for v5.3
by Mark Brown 05 Aug '19

05 Aug '19

The following changes since commit 87a6fe80d56be85bda093c4dd8a3e6535bda7e4a: ASoC: SOF: Intel: implement runtime idle for CNL/APL (2019-07-06 12:23:47 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.3-rc3 for you to fetch changes up to 30c21734d853dae99d05a5295a59b7e26ccd5135: ASoC: amd: acp3x: use dma address for acp3x dma driver (2019-08-02 19:04:43 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.3 A relatively large batch of mostly unremarkable fixes here, a couple of small core fixes for fairly obscure issues, more comment/email updates with no code impact than usual and a bunch of small driver fixes. The support for new sample rates in the max98373 driver is a fix for the fact that the driver declared support for those rates but would in fact return an error if these rates were selected. ---------------------------------------------------------------- Cezary Rojewski (1): MAINTAINERS: Update Intel ASoC drivers maintainers Charles Keepax (1): ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks Cheng-Yi Chiang (1): ASoC: rockchip: Fix mono capture Christophe JAILLET (1): ASoC: Intel: Fix some acpi vs apci typo in somme comments Enric Balletbo i Serra (1): SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection Hans de Goede (1): ASoC: Intel: bytcht_es8316: Add quirk for Irbis NB41 netbook Joe Perches (1): ASoC: rt1308: Remove executable attribute from source files Kai Vehmanen (1): ASoC: SOF: Intel: hda: remove misleading error trace from IRQ thread Kirill Marinushkin (1): ASoC: Relocate my e-mail to .com domain zone Kuninori Morimoto (3): ASoC: simple_card_utils.h: care NULL dai at asoc_simple_debug_dai() ASoC: simple-card-utils: care no Platform for DPCM ASoC: audio-graph-card: add missing const at graph_get_dai_id() Marcus Cooper (1): ASoC: sun4i-i2s: Incorrect SR and WSS computation Masahiro Yamada (1): ASoC: SOF: use __u32 instead of uint32_t in uapi headers Peter Ujfalusi (2): ASoC: ti: davinci-mcasp: Fix clk PDIR handling for i2s master mode ASoC: ti: davinci-mcasp: Correct slot_width posed constraint Ricard Wanderlof (1): ASoC: Fail card instantiation if DAI format setup fails Shengjiu Wang (1): ASoC: cs42xx8: Fix MFREQ selection issue for async mode Shuming Fan (1): ASoC: rt1011: fix DC calibration offset not applying Stephan Gerhold (1): ASoC: qcom: apq8016_sbc: Fix oops with multiple DAI links Tzung-Bi Shih (1): ASoC: max98357a: use mdelay for sdmode-delay Vijendar Mukunda (2): ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver ASoC: amd: acp3x: use dma address for acp3x dma driver Wen Yang (6): ASoC: simple-card: fix an use-after-free in simple_dai_link_of_dpcm() ASoC: simple-card: fix an use-after-free in simple_for_each_link() ASoC: audio-graph-card: fix use-after-free in graph_dai_link_of_dpcm() ASoC: audio-graph-card: fix an use-after-free in graph_get_dai_id() ASoC: samsung: odroid: fix an use-after-free issue for codec ASoC: samsung: odroid: fix a double-free issue for cpu_dai Wenwen Wang (1): ASoC: dapm: fix a memory leak bug fengchunguo (1): ASoC: max98373: add 88200 and 96000 sampling rate support MAINTAINERS | 3 +- include/sound/simple_card_utils.h | 4 + include/uapi/sound/sof/fw.h | 16 +-- include/uapi/sound/sof/header.h | 14 +-- sound/soc/amd/raven/acp3x-pcm-dma.c | 20 ++-- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++---- sound/soc/codecs/max98357a.c | 25 +---- sound/soc/codecs/max98373.c | 6 ++ sound/soc/codecs/max98373.h | 2 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt1011.c | 4 +- sound/soc/codecs/rt1308.c | 0 sound/soc/codecs/rt1308.h | 0 sound/soc/generic/audio-graph-card.c | 30 +++--- sound/soc/generic/simple-card-utils.c | 7 ++ sound/soc/generic/simple-card.c | 26 ++--- sound/soc/intel/boards/bytcht_es8316.c | 8 ++ sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 2 +- .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 16 +-- sound/soc/rockchip/rockchip_i2s.c | 5 +- sound/soc/rockchip/rockchip_max98090.c | 32 ++++++ sound/soc/samsung/odroid.c | 8 +- sound/soc/soc-core.c | 7 +- sound/soc/soc-dapm.c | 10 +- sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sunxi/sun4i-i2s.c | 4 +- sound/soc/ti/davinci-mcasp.c | 46 ++++++-- 40 files changed, 304 insertions(+), 147 deletions(-) mode change 100644 => 100755 sound/soc/codecs/max98373.c mode change 100644 => 100755 sound/soc/codecs/max98373.h mode change 100755 => 100644 sound/soc/codecs/rt1308.c mode change 100755 => 100644 sound/soc/codecs/rt1308.h

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[alsa-devel] [PATCH 00/19] ALSA: firewire: introduce AMDTP domain
by Takashi Sakamoto 05 Aug '19

05 Aug '19

Hi, This patchset is to implement AMDTP domain I addressed in my previous patchset: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/152430.html My work for libhinoko[1][2] allows me to sniff actual packet transmission between devices and drivers in Windows/Mac OS for long period (e.g. 1 hour). As a result, some supported devices don't follow packet sequences transferred by the drivers. They expect drivers to parse isoc packets in tx stream and recover frequency of data block then transfer isoc packets as rx stream. (I note that no specification describes this mechanism as long as I know. In the specification, clock recovery is one-way from transmitter to receivers, thus recovered clock is not necessarily used for transmission from the receiver to the transmitter.) For the clock recovery in driver side, several isoc contexts should be handled in one time to parse packets in incoming stream and to build packets in outgoing stream. The AMDTP domain is designed for this purpose. In this time, the AMDTP domain implements the functionalities to start/stop a couple of isoc contexts for some AMDTP streams. Each context still runs in own handler for scheduled hardware IRQs. I'll post further work after merged. [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2019-April/147862.html [2] https://github.com/takaswie/libhinoko Takashi Sakamoto (19): ALSA: firewire-lib: add AMDTP domain structure to handle several isoc context in one interrupt callback ALSA: firewire-lib: add a kernel API to stop a couple of AMDTP streams in AMDTP domain ALSA: firewire-lib: add a kernel API to add AMDTP stream into AMDTP domain ALSA: firewire-lib: add a kernel API to start AMDTP streams in AMDTP domain ALSA: fireworks: code refactoring for initialization/destruction of AMDTP streams ALSA: fireworks: code refactoring for bus reset handler ALSA: firewire-digi00x: code refactoring for initialization/destruction of AMDTP stream ALSA: firewire-tascam: code refactoring for initialization/destruction of AMDTP stream ALSA: firewire-motu: code refactoring for initialization/destruction of AMDTP stream ALSA: fireface: code refactoring for initialization/destruction of AMDTP stream ALSA: bebob: support AMDTP domain ALSA: fireworks: support AMDTP domain ALSA: oxfw: support AMDTP domain ALSA: dice: support AMDTP domain ALSA: firewire-digi00x: support AMDTP domain ALSA: firewire-tascam: support AMDTP domain ALSA: firewire-motu: support AMDTP domain ALSA: fireface: support AMDTP domain ALSA: firewire-lib: localize kernel APIs to start/stop each AMDTP stream sound/firewire/amdtp-stream.c | 94 ++++++++++- sound/firewire/amdtp-stream.h | 20 ++- sound/firewire/bebob/bebob.h | 2 + sound/firewire/bebob/bebob_stream.c | 62 +++---- sound/firewire/dice/dice-stream.c | 32 +++- sound/firewire/dice/dice.h | 2 + sound/firewire/digi00x/digi00x-stream.c | 106 +++++++----- sound/firewire/digi00x/digi00x.h | 2 + sound/firewire/fireface/ff-stream.c | 96 ++++++----- sound/firewire/fireface/ff.h | 2 + sound/firewire/fireworks/fireworks.h | 2 + sound/firewire/fireworks/fireworks_stream.c | 176 ++++++++++---------- sound/firewire/motu/motu-stream.c | 131 +++++++-------- sound/firewire/motu/motu.h | 2 + sound/firewire/oxfw/oxfw-stream.c | 72 ++++---- sound/firewire/oxfw/oxfw.h | 2 + sound/firewire/tascam/tascam-stream.c | 123 ++++++++------ sound/firewire/tascam/tascam.h | 2 + 18 files changed, 558 insertions(+), 370 deletions(-) -- 2.20.1

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Re: [alsa-devel] [Sound-open-firmware] Signed firmware availability for kbl/cnl
by Jaroslav Kysela 05 Aug '19

05 Aug '19

Dne 24. 07. 19 v 18:09 Liam Girdwood napsal(a): > On Wed, 2019-07-24 at 18:00 +0200, Jaroslav Kysela wrote: >> Dne 24. 07. 19 v 16:41 Pierre-Louis Bossart napsal(a): >>> On 7/24/19 8:59 AM, Jaroslav Kysela wrote: >>>> Dne 23. 07. 19 v 16:22 Liam Girdwood napsal(a): >>>>> On Thu, 2019-07-18 at 16:43 +0800, Daniel Drake >>>>> wrote:_______________________________________________ >>>>>> Sound-open-firmware mailing list >>>>>> Sound-open-firmware(a)alsa-project.org >>>>>> >>>>> https://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware >>>>> >>>>>> On Wed, Jul 17, 2019 at 8:09 AM Pierre-Louis Bossart >>>>>> <pierre-louis.bossart(a)linux.intel.com> >>>>>> wrote:_______________________________________________ >>>>>>> Sound-open-firmware mailing list >>>>>>> Sound-open-firmware(a)alsa-project.org >>>>>>> >>>>>> https://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware >>>>>> >>>>>>> I was indeed told a while ago that there was a limited >>>>>>> number of >>>>>>> KBL-based devices with DMIC, but mistakenly assumed we >>>>>>> could avoid >>>>>>> dealing with this configuration and Murphy's Law applied of >>>>>>> course. >>>>>>> we'll have to huddle with our Intel colleagues to figure >>>>>>> this one >>>>>>> out. >>>>>> >>>>>> OK, thanks. >>>>>> >>>>>> Is there any update on the release of signed firmware files >>>>>> for the >>>>>> other platforms? We are under pressure to return the other >>>>>> unit we >>>>>> have to the vendor (which needs the cnl files), but we would >>>>>> like to >>>>>> try SOF first. >>>>>> >>>>> >>>>> Apologies for the delay, I hurt my back and was off work for a >>>>> few >>>>> weeks. Signed binaries now on v1.3 github release tag. Will now >>>>> be >>>>> upstreaming into Linux FW repo. >>>> >>>> Liam, the sizes of signed firmware binaries are a lot different >>>> than the >>>> unsigned ones (v1.3 tag) which I can build in docker: >>>> >>>> -rw-rw-r--. 1 perex perex 270336 Jul 24 15:44 sof-apl-signed- >>>> intel.ri >>>> -rw-r--r--. 1 perex perex 167936 Jul 24 15:44 sof-apl.ri >>>> -rw-rw-r--. 1 perex perex 278528 Jul 24 15:44 sof-cnl-signed- >>>> intel.ri >>>> -rw-r--r--. 1 perex perex 172032 Jul 24 15:44 sof-cnl.ri >>>> -rw-rw-r--. 1 perex perex 278528 Jul 24 15:44 sof-icl-signed- >>>> intel.ri >>>> -rw-r--r--. 1 perex perex 172032 Jul 24 15:44 sof-icl.ri >>>> >>>> Is that ok? >>> >>> The firmware used for production is typically built with the >>> Cadence >>> tools, which unfortunately are not available publicly (but can be >>> made >>> available to Intel partners). It wouldn't be surprising if the code >>> size >>> was different due to the use of intrinsics (though 100K seems like >>> a lot >>> indeed). >>> >>> Liam, I think we ought to release binaries with the community key >>> as >>> well so that people can use them as is, e.g. on the Up2 board which >>> does >>> not require the Intel production key. Same for GLK Chromebooks. >> >> It would be probably more nice to create a tar ball with all firmware >> and >> topology files bundled with the proper (usual) filesystem location >> (/lib/firmware/intel/sof/... and /lib/firmware/intel/sof-tplg/...). >> So we >> (users/distribution packagers) can just use it. > > Yeah, been thinking about this atm. It may be better to package the > binaries (firmware and topologies) as part of Linux firmware repo > (since the driver expects to load them all from lib/firmware) and > package the sources (firmware and topology) via sof tarball ? It looks good in my eyes, because topology files are another pieces of the driver from the user space perspective. The unanswered question is the UCM configuration which is linked to the topology configuration (if I understand this correctly). I proposed to place an unique identifier/version to the topology file and propagate this identifier to the user space, so the alsa-lib can pick the right UCM configuration when topology changes. The component string (snd_component_add function / struct snd_ctl_card_info -> components) can be used for this identification. Jaroslav > > Would this be OK for the distros ? > > Liam > >> >> BTW: Do we need UCM config files for SOF, too? >> >> Jaroslav >> > -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH 1/2] ASoC: amd: acp3x: use dma_ops of parent device for acp3x dma driver
by Vijendar Mukunda 02 Aug '19

02 Aug '19

AMD platform device acp3x_rv_i2s created by parent PCI device driver. Pass struct device of the parent to snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use correct dma_ops. Otherwise, it will use default dma_ops which is nommu_dma_ops on x86_64 even when IOMMU is enabled and set to non passthrough mode. Signed-off-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index a4ade6b..905ed2f 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -385,9 +385,11 @@ static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream) static int acp3x_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, + DRV_NAME); + struct device *parent = component->dev->parent; snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, - rtd->pcm->card->dev, - MIN_BUFFER, MAX_BUFFER); + parent, MIN_BUFFER, MAX_BUFFER); return 0; } -- 2.7.4

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