December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/8] ALSA: fireface: add support for Fireface 800 with MIDI functionality only
by Takashi Sakamoto 11 Dec '18

11 Dec '18

Hi, In 2004, RME GmbH shipped Fireface 800 as a flagship model of its Fireface series. This model consists of some chips: - TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus - TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus - Xilinx Spartan-3 FPGA XC3S400 - Xilinx High-Performance CPLD XC9572XL A remarkable point of this model is that its physical/link layer support S800 (800Mbps) mode of IEEE 1394b. Within devices of audio and music unit on IEEE 1394 bus, this is an unique model to support it, as long as I know. This commit adds support for the model with its MIDI functionality only. The functionality is quite similar to the one of Fireface 400. In patch 01-06, some codes specific to Fireface 400 are arranged to be shared by both models. On the other hand, there're some differences. The largest difference is that the model has no register to switch enabling/disabling tx asynchronous transactions for MIDI messages. Therefore the functionality work well without any assist by userspace. For this work, I use v2.77 firmware and v3.1.24 driver shipped by RME GmbH. If finding some differences between older firmware, please report it with enough information for debug. To help users for test, I prepare a remote branch including these patches. This can be backport to Linux v4.17 or later. https://github.com/takaswie/snd-firewire-improve/tree/topic/ff800-midi Takashi Sakamoto (8): ALSA: fireface: share some registers for status of clock synchronization ALSA: fireface: share status and configuration dump ALSA: fireface: share helper function to get current sampling rate and clock source ALSA: fireface: add support for second optical interface for ADAT stream ALSA: fireface: share register for async transaction of MIDI messages ALSA: fireface: add driver data for register for MIDI high address ALSA: fireface: localize a handler for MIDI messages on tx transaction ALSA: fireface: add support for Fireface 800 with MIDI functionality only sound/firewire/Kconfig | 1 + sound/firewire/fireface/Makefile | 3 +- sound/firewire/fireface/ff-pcm.c | 2 +- sound/firewire/fireface/ff-proc.c | 193 +++++++++++++- sound/firewire/fireface/ff-protocol-ff400.c | 282 ++------------------ sound/firewire/fireface/ff-protocol-ff800.c | 27 ++ sound/firewire/fireface/ff-stream.c | 2 +- sound/firewire/fireface/ff-transaction.c | 157 ++++++----- sound/firewire/fireface/ff.c | 54 +++- sound/firewire/fireface/ff.h | 31 ++- 10 files changed, 406 insertions(+), 346 deletions(-) create mode 100644 sound/firewire/fireface/ff-protocol-ff800.c -- 2.19.1

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[alsa-devel] [PATCH] ALSA: hda/ca0132 - make pci_iounmap() call conditional
by Arnd Bergmann 11 Dec '18

11 Dec '18

When building without CONFIG_PCI, we can (depending on the architecture) get a link failure: ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined! Adding a compile-time check for PCI gets it to work correctly on 32-bit ARM. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index c40cb6336017..e5bdbc245682 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -8451,7 +8451,7 @@ static void ca0132_free(struct hda_codec *codec) ca0132_exit_chip(codec); snd_hda_power_down(codec); - if (spec->mem_base) + if (IS_ENABLED(CONFIG_PCI) && spec->mem_base) pci_iounmap(codec->bus->pci, spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); -- 2.20.0

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[alsa-devel] [PATCH 0/7] ASoC: merge simple-scu-card into simple-card - step1
by Kuninori Morimoto 11 Dec '18

11 Dec '18

Hi Mark I'm posting "merge audio-graph-card" patch-set now. And, I decided to samethings to "simple-card". These are prepare for merging. Kuninori Morimoto (7): ASoC: simple-card: use cpu/codec pointer on simple_dai_props ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing ASoC: simple-scu-card: add dai-link support ASoC: simple-scu-card: care link / dai count ASoC: simple-scu-card: tidyup "convert-rate/channels" parsing ASoC: simple-scu-card: use cpu/codec pointer on graph_dai_props ASoC: simple-scu-card: care multi DPCM codec_conf sound/soc/generic/simple-card.c | 62 +++++---- sound/soc/generic/simple-scu-card.c | 261 +++++++++++++++++++++++++++--------- 2 files changed, 235 insertions(+), 88 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: Intel: bytcr_rt5640: Add quirk for the Prowise PT301 tablet
by Hans de Goede 11 Dec '18

11 Dec '18

Add a quirk for the Prowise PT301 tablet, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. Also it uses IN1 for its MIC and JD2 for jack-detect, rather then the default IN3 and JD1. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 8587bd3d1cc1..6df902e3a380 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -674,6 +674,20 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF2 | BYT_RT5640_MCLK_EN), }, + { + /* Prowise PT301 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Prowise"), + DMI_MATCH(DMI_PRODUCT_NAME, "PT301"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), -- 2.19.2

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[alsa-devel] Applied "ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7f63196eaa83e033de523602324ff91288390a67 Mon Sep 17 00:00:00 2001 From: Zhuohao Lee <zhuohao(a)chromium.org> Date: Mon, 10 Dec 2018 12:32:13 +0800 Subject: [PATCH] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping >From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND. Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 58eb0fe69978..723a4935ed76 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.19.0.rc2

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[alsa-devel] [PATCH] ASoC: sdm845: add rt5663 codec select
by Arnd Bergmann 11 Dec '18

11 Dec '18

Calling into the codec driver adds a dependency on that being reachable from the module: ERROR: "rt5663_sel_asrc_clk_src" [sound/soc/qcom/snd-soc-sdm845.ko] undefined! Add the corresponding select statement, as it is done in the other user (Intel). Fixes: f7485875a687 ("ASoC: sdm845: Add configuration for headset codec") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 2a4c912d1e48..1b9fc0665792 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -100,6 +100,7 @@ config SND_SOC_SDM845 depends on QCOM_APR select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON + select SND_SOC_RT5663 help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.20.0

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[alsa-devel] [PATCH] ASoC: simple-card-utils: fix build warning without CONFIG_OF
by Arnd Bergmann 11 Dec '18

11 Dec '18

When CONFIG_OF is disabled, of_graph_parse_endpoint() does not initialize 'info', and gcc can see that: sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_card_parse_graph_dai': sound/soc/generic/simple-card-utils.c:284:13: error: 'info.port' may be used uninitialized in this function [-Werror=maybe-uninitialized] It's probably best to check the return code anyway, and that also takes care of the warning. Fixes: b6f3fc005a2c ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/generic/simple-card-utils.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 6a31d07976b9..17d8aee43835 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -280,7 +280,10 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep) * Non HDMI sound case, counting port/endpoint on its DT * is enough. Let's count it. */ - of_graph_parse_endpoint(ep, &info); + ret = of_graph_parse_endpoint(ep, &info); + if (ret) + return -ENXIO; + return info.port; } -- 2.20.0

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[alsa-devel] Applied "ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing" to the asoc tree
by Mark Brown 11 Dec '18

11 Dec '18

The patch ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From dbd08fe59cf8c2f4bad0a72e912913c20db4774e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Tue, 4 Dec 2018 08:20:13 +0000 Subject: [PATCH] ASoC: simple-scu-card: tidyup asoc_simple_card_parse_daifmt() timing Current simple-scu-card driver is parsing codec position for DPCM and consider DAI format. But, current operation is doing totally pointless, because it should be called for each CPU/Codec pair. Let's tidyup asoc_simple_card_parse_daifmt() timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-scu-card.c | 37 +++++++++++++++++------------ 1 file changed, 22 insertions(+), 15 deletions(-) diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 656abe2015e1..24099e61d1b1 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -92,17 +92,24 @@ static int asoc_simple_card_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int asoc_simple_card_dai_link_of(struct device_node *np, +static int asoc_simple_card_dai_link_of(struct device_node *link, + struct device_node *np, + struct device_node *codec, struct simple_card_data *priv, - unsigned int daifmt, - int idx, bool is_fe) + int idx, bool is_fe, + bool is_top_level_node) { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, idx); struct simple_dai_props *dai_props = simple_priv_to_props(priv, idx); struct snd_soc_card *card = simple_priv_to_card(priv); + char *prefix = ""; int ret; + /* For single DAI link & old style of DT node */ + if (is_top_level_node) + prefix = PREFIX; + if (is_fe) { int is_single_links = 0; struct snd_soc_dai_link_component *codecs; @@ -178,7 +185,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - dai_link->dai_fmt = daifmt; + ret = asoc_simple_card_parse_daifmt(dev, link, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + return ret; + dai_link->dpcm_playback = 1; dai_link->dpcm_capture = 1; dai_link->ops = &asoc_simple_card_ops; @@ -191,10 +202,10 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) { struct device *dev = simple_priv_to_dev(priv); + struct device_node *node = dev->of_node; struct device_node *np; + struct device_node *codec; struct snd_soc_card *card = simple_priv_to_card(priv); - struct device_node *node = dev->of_node; - unsigned int daifmt = 0; bool is_fe; int ret, i; @@ -211,22 +222,18 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) asoc_simple_card_parse_convert(dev, node, PREFIX, &priv->adata); - /* find 1st codec */ - np = of_get_child_by_name(node, PREFIX "codec"); - if (!np) + i = 0; + codec = of_get_child_by_name(node, PREFIX "codec"); + if (!codec) return -ENODEV; - ret = asoc_simple_card_parse_daifmt(dev, node, np, PREFIX, &daifmt); - if (ret < 0) - return ret; - - i = 0; for_each_child_of_node(node, np) { is_fe = false; if (of_node_name_eq(np, PREFIX "cpu")) is_fe = true; - ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); + ret = asoc_simple_card_dai_link_of(node, np, codec, priv, + i, is_fe, true); if (ret < 0) return ret; i++; -- 2.19.0.rc2

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[alsa-devel] [PATCH - AAF PCM plugin 0/7] Follow-up improvements
by Andre Guedes 10 Dec '18

10 Dec '18

Hi all, This patch series provides some follow-up improvements to the AVTP Audio Format (AAF) plugin. The highlight of this series is the implementation of a transmission offload mechanism which improves the plugin performance considerably. Details are provided in the following paragraphs. Currently, the AVTPDU transmission interval is controlled by the AAF plugin, in software. At every timer expiration, the plugin transmits one AVTPDU. This has some implications in terms of task scheduling. For instance, in class A streams, AVTPDUs are transmitted at every 125 us which means the application task should be scheduled-in at every 125 us. In a general purpose Linux system, such scheduling interval can be hard to cope with in the long-run. To mitigate that issue, this series introduces a transmission offload mechanism that leverages the SO_TXTIME sockopt and ETF qdisc features recently introduced to kernel 4.19. Instead of sending one AVTPDU at every timer expiration, the plugin sends several AVTPDUs at once to the kernel, configuring their Tx time so the transmission interval is maintained. The kernel can then offload packet transmission to the hardware (if the network controller supports it), providing more transmission time accuracy. This offloading mechanism enables the application to sleep for longer times, easing on task scheduling. To illustrate the improvements provided by this series, I ran a before-after experiment. The experiment setup consisted in 2 PCs with Intel i210 card connected back-to-back running an up-to-date Archlinux with kernel 4.19.4. I ran 'aplay' for 5 minutes on one PC and captured the AVTPDUs on the other PC. The metric under evaluation is the transmission interval and it is measured by checking the 'time_delta' information from ethernet frames captured at the receiving side. If you're interested in reproducing the experiments, let me know and I can share my helper scripts. The table below shows the experiment outcome for a Class A, stereo, 16-bit sample, 48 kHz stream. The unit is nanoseconds. | Mean | Stdev | Min | Max | Range | -------+--------+---------+---------+---------+---------+ Before | 125000 | 1154 | 75311 | 172144 | 96833 | After | 125000 | 18 | 124960 | 125048 | 88 | Before this patchset, the transmission interval mean is equal to the optimal value (Class A stream -> 125 us interval), and it is kept the same after the patchset. However, the dispersion measurements had improved considerably, meaning the system is consistently transmitting AVTPDUs at the correct interval. Finally, to help the review process, here follows a quick summary of the patches within this series: * PATCH 1: fixes a type in the plugin documentation. * PATCH 2: enables the user to configure the presentation time tolerance. * PATCH 3-5: refactor the code in order to land the offload mechanism code smoothly. * PATCH 6-7: implement the transmission offload mechanism. This series can also be found in my alsa-plugins tree in github [1]. Regards, Andre [1] https://github.com/aguedes/alsa-plugins Andre Guedes (7): doc: Fix typo in AAF doc aaf: Add presentation time tolerance aaf: Refactor AVTPDU transmission routines aaf: Refactor AVTPDU reception routines aaf: Refactor timeout routines aaf: Tx multiple AVTPDUs per media clock tick aaf: AVTPDU transmission periodicity aaf/pcm_aaf.c | 600 +++++++++++++++++++++++++++++++++----------------- configure.ac | 2 +- doc/aaf.txt | 40 +++- 3 files changed, 429 insertions(+), 213 deletions(-) -- 2.19.1

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[alsa-devel] [PATCH v1] ASoC: Intel: kbl_da7219_max98927: fix the audio jack button remapping
by Zhuohao Lee 10 Dec '18

10 Dec '18

>From the da7219 spec, the button A, B, C and D are remapped to 0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE, B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND Change-Id: I67f929f226883bd289cb7beb0a7b23f40e7121e2 Signed-off-by: Zhuohao Lee <zhuohao(a)chromium.org> Signed-off-by: Max Chang <changmax(a)chromium.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 3fa1c3ca6d37..d3531f0eb357 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -262,9 +262,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) jack = &ctx->kabylake_headset; snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); da7219_aad_jack_det(component, &ctx->kabylake_headset); -- 2.20.0.rc2.403.gdbc3b29805-goog

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