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

[alsa-devel] [PATCH 00/28] ASoC: cleanup patches for soc-core
by Kuninori Morimoto 07 Aug '19

07 Aug '19

Hi Mark 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. Just for cleanup Kuninori Morimoto (28): ASoC: soc-core: use device_register() ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: don't check widget for snd_soc_dapm_new_controls() ASoC: soc-core: don't check controls for snd_soc_add_component_controls() ASoC: soc-core: don't check routes for snd_soc_dapm_add_routes() ASoC: soc-core: don't check controls for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: remove verbose debug message from soc_bind_dai_link() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: initialize rtd->list ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: don't call snd_soc_component_set_jack() ASoC: soc-core: don't alloc memory carelessly when creating debugfs ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 ++- include/sound/soc.h | 15 +--- sound/soc/soc-core.c | 196 ++++++++++++++++++++++------------------------- sound/soc/soc-pcm.c | 3 + sound/soc/soc-topology.c | 2 +- 6 files changed, 108 insertions(+), 120 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: Intel: boards: Add Cometlake Dialog Maxim machine driver
by mac.chiang＠intel.com 06 Aug '19

06 Aug '19

From: Mac Chiang <mac.chiang(a)intel.com> enable Cometlake support with: SSP0 for DA7219 headphone codec SSP1 for MAX98357a speaker amp codec Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/Kconfig | 31 +++++++------ sound/soc/intel/boards/kbl_da7219_max98357a.c | 55 ++++++++++++++++++++--- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 +++++ 3 files changed, 80 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149..545aef8 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -322,19 +322,6 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH - tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for DA7219 + MAX98357A I2S audio codec. - Say Y if you have such a device. - config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH tristate "KBL with DA7219 and MAX98927 in I2S Mode" depends on I2C && ACPI @@ -363,6 +350,24 @@ config SND_SOC_INTEL_KBL_RT5660_MACH endif ## SND_SOC_INTEL_KBL +if SND_SOC_INTEL_KBL || SND_SOC_SOF_COMETLAKE_LP + +config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH + tristate "KBL with DA7219 and MAX98357A in I2S Mode" + depends on I2C && ACPI + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for DA7219 + MAX98357A I2S audio codec for + Kabylake/Cometlake platforms. + Say Y if you have such a device. + +endif ## SND_SOC_INTEL_KBL || SND_SOC_SOF_COMETLAKE_LP + if SND_SOC_INTEL_GLK || (SND_SOC_SOF_GEMINILAKE && SND_SOC_SOF_HDA_LINK) config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a889..fe3ac70 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -9,6 +9,7 @@ * RT5663 codecs */ +#include <asm/cpu_device_id.h> #include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> @@ -17,6 +18,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include "../../codecs/da7219.h" #include "../../codecs/hdac_hdmi.h" #include "../../codecs/da7219-aad.h" @@ -210,7 +212,11 @@ static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) if (!pcm) return -ENOMEM; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + pcm->device = dai->id; +#else pcm->device = device; +#endif pcm->codec_dai = dai; list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); @@ -587,9 +593,17 @@ static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { .late_probe = kabylake_card_late_probe, }; +static const struct x86_cpu_id cml_ids[] = { + { X86_VENDOR_INTEL, 6, 0x8E }, /* Cometlake CPU_ID */ + {} +}; + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; + struct snd_soc_acpi_mach *mach; + const char *platform_name; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -600,17 +614,46 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card = (struct snd_soc_card *)pdev->id_entry->driver_data; + kabylake_audio_card = &kabylake_audio_card_da7219_m98357a; + kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); + + if (x86_match_cpu(cml_ids)) { + unsigned int i; + + kabylake_audio_card->name = "cmlda7219max"; + + for (i = 0; i < ARRAY_SIZE(kabylake_dais); i++) { + /* MAXIM_CODEC is connected to SSP1. */ + if (!strcmp(kabylake_dais[i].cpus->dai_name, + KBL_MAXIM_CODEC_DAI)) { + kabylake_dais[i].name = "SSP1-Codec"; + kabylake_dais[i].cpus->dai_name = "SSP1 Pin"; + } + /* DIALOG_CODEC is connected to SSP0 */ + else if (!strcmp(kabylake_dais[i].cpus->dai_name, + KBL_DIALOG_CODEC_DAI)) { + kabylake_dais[i].name = "SSP0-Codec"; + kabylake_dais[i].cpus->dai_name = "SSP0 Pin"; + } + } + } + + mach = (&pdev->dev)->platform_data; + platform_name = mach->mach_params.platform; + + ret = snd_soc_fixup_dai_links_platform_name(kabylake_audio_card, + platform_name); + if (ret) + return ret; + return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); } static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_da7219_max98357a", - .driver_data = - (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, - }, + {.name = "kbl_da7219_max98357a",}, + {.name = "cml_da7219_max98357a",}, { } }; @@ -628,5 +671,7 @@ module_platform_driver(kabylake_audio) /* Module information */ MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:kbl_da7219_max98357a"); +MODULE_ALIAS("platform:cml_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index c36c0aa..4ea32b2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs cml_codecs = { .codecs = {"10EC5682"} }; +static struct snd_soc_acpi_codecs cml_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98357A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", @@ -29,6 +34,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .quirk_data = &cml_spk_codecs, + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, + { .id = "MX98357A", .drv_name = "sof_rt5682", .quirk_data = &cml_codecs, -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda - Don't override global PCM hw info flag
by Takashi Iwai 06 Aug '19

06 Aug '19

The commit bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") introduced the conditional PCM hw info setup, but it overwrites the global azx_pcm_hw object. This will cause a problem if any other HD-audio controller, as it'll inherit the same bit flag although another controller doesn't support that feature. Fix the bug by setting the PCM hw info flag locally. Fixes: bfcba288b97f ("ALSA - hda: Add support for link audio time reporting") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index c8d1b4316245..2fbdde239936 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -598,11 +598,9 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) } runtime->private_data = azx_dev; - if (chip->gts_present) - azx_pcm_hw.info = azx_pcm_hw.info | - SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; - runtime->hw = azx_pcm_hw; + if (chip->gts_present) + runtime->hw.info |= SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME; runtime->hw.channels_min = hinfo->channels_min; runtime->hw.channels_max = hinfo->channels_max; runtime->hw.formats = hinfo->formats; -- 2.16.4

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[alsa-devel] [GIT PULL] ASoC fixes for v5.3
by Mark Brown 06 Aug '19

06 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 asoc-fix-v5.3-rc3-2 for you to fetch changes up to 7c0767643f3b6b0dd2cda923ae37a18590d431cf: ASoC: max98373: Remove executable bits (2019-08-06 11:15:06 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.3 Incremental fix removing executable bits added in a prior patch accidentally. ---------------------------------------------------------------- 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 Mark Brown (1): ASoC: max98373: Remove executable bits 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 100755 => 100644 sound/soc/codecs/rt1308.c mode change 100755 => 100644 sound/soc/codecs/rt1308.h

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[alsa-devel] Possible bug report
by Harry Bloomberg 06 Aug '19

06 Aug '19

While running an amateur radio application on a Raspberry Pi 3B+ with the stretch version of Raspberrian, I see the following in my syslog: Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Aug 1 17:31:06 raspberrypi pulseaudio[853]: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. So...I am reporting this to the developers as instructed by the error message. What additional information do you need? I'm not exactly sure what I was doing to trigger this error. I may have been exiting the application. Thanks, Harry Bloomberg

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[alsa-devel] [PATCH 0/2] ALSA snd-usb-audio: Add Pioneer DDJ-SX3 PCM support
by Ard van Breemen 06 Aug '19

06 Aug '19

The Pioneer lacks the UAC descriptors (added to quircks). It also needs a fix in synchronisation handling for those cases that bSynchAddress == 0 Ard van Breemen (2): Skip bSynchAddress endpoint check if it is invalid. Add Pioneer DDJ-SX3 PCM quirck. sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) -- 2.17.1

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[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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