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

[alsa-devel] [PATCH - hwmixvolume v2 7/7] hwmixvolume: mention the new dependencies in the README
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/README b/hwmixvolume/README index 9884ec2..56c7554 100644 --- a/hwmixvolume/README +++ b/hwmixvolume/README @@ -8,7 +8,8 @@ that use hardware mixing, i.e., those based on the following chips: * Yamaha DS-1 (YMF-724/740/744/754) (driver: snd-ymfpci) -This tool requires Python, pygtk, and alsa-pyton 1.0.22 or later. +This tool requires Python, PyGObject, GTK+ 3.0 or later, and alsa-python 1.0.22 +or later. It is recommended to use at least Linux kernel 2.6.32 or alsa-driver 1.0.22; otherwise, the name of the program that is using a stream cannot be shown. -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 6/7] hwmixvolume: add my copyright
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 4804a9b..871c2c5 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -2,6 +2,7 @@ # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch +# Copyright (c) 2018 Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> # # Permission to use, copy, modify, and/or distribute this software for any # purpose with or without fee is hereby granted, provided that the above -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 5/7] hwmixvolume: switch the shebang to python
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This signifies that this code is now compatible with both Python 2 and Python 3, as per PEP-0394[1]. [1] https://www.python.org/dev/peps/pep-0394/ Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 7f8ba8e..4804a9b 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -1,4 +1,4 @@ -#!/usr/bin/python2 +#!/usr/bin/env python # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 4/7] hwmixvolume: use a with context to open files
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This feature has been added in Python 2.5 and automatically closes an open file once the context exits. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 8e0b6b8..7f8ba8e 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -138,26 +138,20 @@ class Stream: subdevice = info.index filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) try: - f = open(filename, "r") + with open(filename, "r") as f: + for line in f: + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) except IOError: return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() return None def get_pid_cmdline(self, pid): try: - f = open("/proc/%d/cmdline" % pid, "r") + with open("/proc/%d/cmdline" % pid, "r") as f: + cmdline = f.read() except IOError: return None - try: - cmdline = f.read() - finally: - f.close() return cmdline.replace('\x00', ' ').strip() class MixerWindow(Gtk.Window): -- 2.19.0

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[alsa-devel] [PATCH - hwmixvolume v2 2/7] hwmixvolume: replace PyGTK with gobject-introspection
by Emmanuel Gil Peyrot 18 Sep '18

18 Sep '18

From: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> This doesn’t work yet, we require Gtk 3.0 rather than 2.0 and the API changed quite a lot, so this is but a preparatory patch. This is done so that we can get rid of GTK+ 2 which has been EOL for many years already, and to add Python 3 support because Python 2 will very soon be EOL as well. Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 038bcb3..64d232c 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -15,7 +15,10 @@ # OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR # PERFORMANCE OF THIS SOFTWARE. -import gobject, gtk +import gi +gi.require_version('GLib', '2.0') +gi.require_version('Gtk', '3.0') +from gi.repository import GLib, Gtk from pyalsa import alsacard, alsahcontrol INTF_PCM = alsahcontrol.interface_id['PCM'] @@ -57,16 +60,16 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = gtk.Label(self.get_label(info)) + self.label = Gtk.Label(self.get_label(info)) self.label.set_single_line_mode(True) self.parent.scales_vbox.pack_start(self.label, expand=False) for i in range(info.count): - adj = gtk.Adjustment(value=values[i], + adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = gtk.HScale(adj) + scale = Gtk.HScale(adj) scale.set_draw_value(False) self.parent.scales_vbox.pack_start(scale, expand=False) self.scales.append(scale) @@ -157,7 +160,7 @@ class Stream: f.close() return cmdline.replace('\x00', ' ').strip() -class MixerWindow(gtk.Window): +class MixerWindow(Gtk.Window): card_numbers = alsacard.card_list() current_card = -1 hcontrol = None @@ -167,21 +170,21 @@ class MixerWindow(gtk.Window): hctl_sources = [] def __init__(self): - gtk.Window.__init__(self) - self.connect('destroy', lambda w: gtk.main_quit()) + Gtk.Window.__init__(self) + self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = gtk.VBox() + vbox = Gtk.VBox() self.add(vbox) - hbox = gtk.HBox() + hbox = Gtk.HBox() vbox.pack_start(hbox, expand=False) - label = gtk.Label("_Sound Card: ") + label = Gtk.Label("_Sound Card: ") label.set_use_underline(True) hbox.pack_start(label, expand=False) - combo = gtk.combo_box_new_text() + combo = Gtk.combo_box_new_text() for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) @@ -191,23 +194,23 @@ class MixerWindow(gtk.Window): hbox.pack_start(combo) label.set_mnemonic_widget(combo) - self.lock_check = gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton(label="_Lock Channels") self.lock_check.set_active(True) vbox.pack_start(self.lock_check, expand=False) - scrollwin = gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=gtk.POLICY_NEVER, vscrollbar_policy=gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(gtk.SHADOW_NONE) + scrollwin = Gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) + scrollwin.set_shadow_type(Gtk.SHADOW_NONE) vbox.pack_start(scrollwin) - self.scales_vbox = gtk.VBox() + self.scales_vbox = Gtk.VBox() scrollwin.add_with_viewport(self.scales_vbox) - label = gtk.Label() + label = Gtk.Label() label.set_single_line_mode(True) line_height = label.size_request()[1] label.destroy() - scale = gtk.HScale() + scale = Gtk.HScale() scale.set_draw_value(False) line_height += scale.size_request()[1] scale.destroy() @@ -223,7 +226,7 @@ class MixerWindow(gtk.Window): def change_card(self, cardnum): for s in self.hctl_sources: - gobject.source_remove(s) + GLib.source_remove(s) self.hctl_sources = [] self.hcontrol = self.open_hcontrol_for_card(cardnum) @@ -249,7 +252,7 @@ class MixerWindow(gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(gobject.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) self.update_msg_label() @@ -267,7 +270,7 @@ class MixerWindow(gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = gtk.Label(msg) + self.msg_label = Gtk.Label(msg) self.scales_vbox.pack_start(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: @@ -280,9 +283,9 @@ class MixerWindow(gtk.Window): mode=alsahcontrol.open_mode['NONBLOCK']) except: # TODO: alsa error msg - dlg = gtk.MessageDialog(self, - gtk.DIALOG_MODAL | gtk.DIALOG_DESTROY_WITH_PARENT, - gtk.MESSAGE_ERROR, gtk.BUTTONS_OK, + dlg = Gtk.MessageDialog(self, + Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, + Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() @@ -304,7 +307,7 @@ class MixerWindow(gtk.Window): def main(): MixerWindow() - gtk.main() + Gtk.main() main() -- 2.19.0

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[alsa-devel] [PATCH] ASoC: es8328: Fix fall-through annotations
by Gustavo A. R. Silva 18 Sep '18

18 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/es8328.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 4b5827d..04a3aa7 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -566,14 +566,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fallthru */ + /* fall through */ case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: hisilicon: fix fall-through annotations
by Gustavo A. R. Silva 18 Sep '18

18 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/hisilicon/hi6210-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 53344a3..a69e5b11 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -269,13 +269,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U16_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S16_LE: bits = HII2S_BITS_16; break; case SNDRV_PCM_FORMAT_U24_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fallthru */ + /* fall through */ case SNDRV_PCM_FORMAT_S24_LE: bits = HII2S_BITS_24; break; -- 2.7.4

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[alsa-devel] Applied "ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c" to the asoc tree
by Mark Brown 18 Sep '18

18 Sep '18

The patch ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c 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 24b7a0aa1abec41b62c9a29a75e511d29f95033b Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Fri, 14 Sep 2018 01:36:04 +0000 Subject: [PATCH] ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c Remove duplicated include. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index c75fab38905d..c3806d7037fc 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -8,7 +8,6 @@ #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/soc.h> -#include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> #include <asm/dma.h> -- 2.19.0

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[alsa-devel] [PATCH 0/2] ASoC: soc-core: adjust platform for new style
by Kuninori Morimoto 18 Sep '18

18 Sep '18

Hi Mark These are v2 of "ASoC: soc-core: use more generic method to find platform" but, renamed patch series naming. v1 patch set had bug under DPCM which can't probe sound card. The cause was "platform name" management and "component" finding loop. These patch set solved this issue. And then, patch set became reduced Now, ALSA SoC is using snd_soc_dai_link_component for platform instead of legacy style (= platform_name/platform_of_node/platform). OTOH, CPU/Codec are finding its DAI by using common snd_soc_find_dai() function which uses snd_soc_dai_link_component. Then, of course, we want to use same style for platform. These patch do it. Kuninori Morimoto (2): ASoC: soc-core: manage platform name under snd_soc_init_platform() ASoC: soc-core: add snd_soc_is_matching_component() sound/soc/soc-core.c | 73 +++++++++++++++++++++++++++------------------------- 1 file changed, 38 insertions(+), 35 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt5514-spi: Get the period_bytes in the copy work to make sure the value correctly
by Oder Chiou 18 Sep '18

18 Sep '18

The value of period_bytes will get the zero before the hw_params() is not run completely. Move the function snd_pcm_lib_period_bytes() to copy work, and make sure that is not zero. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 6478d10..4d46f45 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -91,6 +91,14 @@ static void rt5514_spi_copy_work(struct work_struct *work) runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + if (!period_bytes) { + schedule_delayed_work(&rt5514_dsp->copy_work, 5); + goto done; + } + + if (rt5514_dsp->buf_size % period_bytes) + rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * + period_bytes; if (rt5514_dsp->get_size >= rt5514_dsp->buf_size) { rt5514_spi_burst_read(RT5514_BUFFER_VOICE_WP, (u8 *)&buf, @@ -149,13 +157,11 @@ static void rt5514_spi_copy_work(struct work_struct *work) static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) { - size_t period_bytes; u8 buf[8]; if (!rt5514_dsp->substream) return; - period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); rt5514_dsp->get_size = 0; /** @@ -183,10 +189,6 @@ static void rt5514_schedule_copy(struct rt5514_dsp *rt5514_dsp) rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - if (rt5514_dsp->buf_size % period_bytes) - rt5514_dsp->buf_size = (rt5514_dsp->buf_size / period_bytes) * - period_bytes; - if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && rt5514_dsp->buf_rp && rt5514_dsp->buf_size) schedule_delayed_work(&rt5514_dsp->copy_work, 0); -- 2.7.4

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[alsa-devel] [PATCH] ALSA: fireworks: fix memory leak of response buffer at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating memory object for response buffer, ALSA fireworks driver has leak of the memory object at error path. This commit releases the object at the error path. Fixes: 7d3c1d5901aa('ALSA: fireworks: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 71a0613d3da0..f2d073365cf6 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -301,6 +301,8 @@ do_registration(struct work_struct *work) snd_efw_transaction_remove_instance(efw); snd_efw_stream_destroy_duplex(efw); snd_card_free(efw->card); + kfree(efw->resp_buf); + efw->resp_buf = NULL; dev_info(&efw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

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[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak of discovered stream formats at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After finishing discover of stream formats, ALSA OXFW driver has memory leak of allocated memory object at error path. This commit releases the memory object at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 1e5b2c802635..63b3e1ce986e 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -207,6 +207,7 @@ static int detect_quirks(struct snd_oxfw *oxfw) static void do_registration(struct work_struct *work) { struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); + int i; int err; if (oxfw->registered) @@ -269,6 +270,12 @@ static void do_registration(struct work_struct *work) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); if (oxfw->has_output) snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; ++i) { + kfree(oxfw->tx_stream_formats[i]); + oxfw->tx_stream_formats[i] = NULL; + kfree(oxfw->rx_stream_formats[i]); + oxfw->rx_stream_formats[i] = NULL; + } snd_card_free(oxfw->card); dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); -- 2.17.1

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[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak for model-dependent data at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data, ALSA OXFW driver has memory leak of the data at error path. This commit releases the data at the error path. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 63b3e1ce986e..9c8c527ebff9 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -277,6 +277,8 @@ static void do_registration(struct work_struct *work) oxfw->rx_stream_formats[i] = NULL; } snd_card_free(oxfw->card); + kfree(oxfw->spec); + oxfw->spec = NULL; dev_info(&oxfw->unit->device, "Sound card registration failed: %d\n", err); } -- 2.17.1

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[alsa-devel] [PATCH] ALSA: bebob: fix memory leak for M-Audio FW1814 and ProjectMix I/O at error path
by Takashi Sakamoto 17 Sep '18

17 Sep '18

After allocating model-dependent data for M-Audio FW1814 and ProjectMix I/O, ALSA bebob driver has memory leak at error path. This commit releases the allocated data at the error path. Fixes: 04a2c73c97eb('ALSA: bebob: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 2 ++ sound/firewire/bebob/bebob_maudio.c | 4 ---- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 730ea91d9be8..93676354f87f 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -263,6 +263,8 @@ do_registration(struct work_struct *work) error: mutex_unlock(&devices_mutex); snd_bebob_stream_destroy_duplex(bebob); + kfree(bebob->maudio_special_quirk); + bebob->maudio_special_quirk = NULL; snd_card_free(bebob->card); dev_info(&bebob->unit->device, "Sound card registration failed: %d\n", err); diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index bd55620c6a47..164f2126a676 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -290,10 +290,6 @@ snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814) bebob->midi_output_ports = 2; } end: - if (err < 0) { - kfree(params); - bebob->maudio_special_quirk = NULL; - } mutex_unlock(&bebob->mutex); return err; } -- 2.17.1

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[alsa-devel] [GIT PULL] ASoC fixes for v4.19
by Mark Brown 17 Sep '18

17 Sep '18

The following changes since commit f2cf0ef7c0ce141bb38f315c34c56e6ef5667a27: ASoC: adav80x: mark expected switch fall-through (2018-08-09 14:41:50 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.19-rc4 for you to fetch changes up to 75383f8d39d4c0fb96083dd460b7b139fbdac492: sound: don't call skl_init_chip() to reset intel skl soc (2018-09-12 11:22:09 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.19 This is the usual set of small fixes scatterd around various drivers, plus one fix for DAPM and a UAPI build fix. There's not a huge amount that stands out here relative to anything else. ---------------------------------------------------------------- Akshu Agrawal (1): ASoC: AMD: Ensure reset bit is cleared before configuring Andrew F. Davis (1): ASoC: tas6424: Save last fault register even when clear Charles Keepax (1): ASoC: dapm: Fix NULL pointer deference on CODEC to CODEC DAIs Danny Smith (1): ASoC: sigmadsp: safeload should not have lower byte limit Dmitry V. Levin (1): ASoC: uapi: fix sound/skl-tplg-interface.h userspace compilation errors Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirks for 2 more devices Jiada Wang (1): ASoC: rsnd: fixup not to call clk_get/set under non-atomic Katsuhiro Suzuki (1): ASoC: uniphier: change status to orphan Kuninori Morimoto (2): ASoC: rsnd: adg: care clock-frequency size ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER Marcel Ziswiler (1): ASoC: wm9712: fix replace codec to component Oder Chiou (1): ASoC: rt5514: Fix the issue of the delay volume applied again Pierre-Louis Bossart (1): ASoC: wm8804: Add ACPI support Ryan Lee (2): ASoC: max98373: Added speaker FS gain cotnrol register to volatile. ASoC: max98373: Added 10ms sleep after amp software reset Shuming Fan (1): ASoC: rt5682: Change DAC/ADC volume scale Srinivas Kandagatla (1): ASoC: q6routing: initialize data correctly Sébastien Szymanski (1): ASoC: cs4265: fix MMTLR Data switch control Yong Zhi (1): ASoC: Intel: Skylake: Acquire irq after RIRB allocation Yu Zhao (3): Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" sound: enable interrupt after dma buffer initialization sound: don't call skl_init_chip() to reset intel skl soc MAINTAINERS | 3 +- include/sound/hdaudio.h | 1 + include/sound/soc-dapm.h | 1 + include/uapi/sound/skl-tplg-interface.h | 106 ++++++++++++++++---------------- sound/hda/hdac_controller.c | 15 +++-- sound/soc/amd/acp-pcm-dma.c | 21 +++++++ sound/soc/codecs/cs4265.c | 4 +- sound/soc/codecs/max98373.c | 3 + sound/soc/codecs/rt5514.c | 8 +-- sound/soc/codecs/rt5682.c | 8 +-- sound/soc/codecs/sigmadsp.c | 3 +- sound/soc/codecs/tas6424.c | 12 +++- sound/soc/codecs/wm8804-i2c.c | 15 ++++- sound/soc/codecs/wm9712.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++++++ sound/soc/intel/skylake/skl.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 4 +- sound/soc/sh/rcar/adg.c | 5 ++ sound/soc/sh/rcar/core.c | 21 ++++++- sound/soc/sh/rcar/dma.c | 4 ++ sound/soc/sh/rcar/rsnd.h | 7 +++ sound/soc/sh/rcar/ssi.c | 16 +++-- sound/soc/soc-core.c | 4 +- sound/soc/soc-dapm.c | 4 ++ 24 files changed, 208 insertions(+), 87 deletions(-)

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[alsa-devel] Bug#908938: alsa-utils: amixer does not unmute pulseaudio
by Elimar Riesebieter 17 Sep '18

17 Sep '18

Control: forwarded -1 alsa-devel(a)alsa-project.org Hi alsa devs, please notice Debian Bug https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=908938 and let us know whether a fix is possible. Thanks in advance -- Elimar * Łukasz Stelmach <steelman(a)post.pl> [2018-09-16 11:23 +0200]: > Package: alsa-utils > Version: 1.1.3-1 > Severity: normal > > Dear Maintainer, > > I've got PulseAudio running, so amixer (and alsamixer) controls the > default PA output. I use the following command to control muting of > audio. > > amixer -q -c 0 sset Master toggle > > It mutes the audio but can't unmute. Neither can > > amixer -q -c 0 sset Master unmute > > Only alsamixer and pavucontrol can unmute. -- Do you smell something burning or is it me?

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[alsa-devel] [PATCH] Documentation: soundwire: fix stream.rst markup warnings
by Randy Dunlap 17 Sep '18

17 Sep '18

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc markup warnings in soundwire/stream.rst: rc4/Documentation/driver-api/soundwire/stream.rst:177: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:203: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:248: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:277: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:304: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:328: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:352: WARNING: Explicit markup ends without a blank line; unexpected unindent. rc4/Documentation/driver-api/soundwire/stream.rst:364: WARNING: Explicit markup ends without a blank line; unexpected unindent. Fixes: 89634f99a83e ("Documentation: soundwire: Add more documentation") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: Shreyas NC <shreyas.nc(a)intel.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-doc(a)vger.kernel.org --- Documentation/driver-api/soundwire/stream.rst | 8 ++++++++ 1 file changed, 8 insertions(+) --- lnx-419-rc4.orig/Documentation/driver-api/soundwire/stream.rst +++ lnx-419-rc4/Documentation/driver-api/soundwire/stream.rst @@ -174,6 +174,7 @@ per stream. From ASoC DPCM framework, th .startup() operation. .. code-block:: c + int sdw_alloc_stream(char * stream_name); @@ -200,6 +201,7 @@ only be invoked once by respective Maste framework, this stream state is linked to .hw_params() operation. .. code-block:: c + int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, struct sdw_ports_config * ports_config, @@ -245,6 +247,7 @@ stream. From ASoC DPCM framework, this s .prepare() operation. .. code-block:: c + int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -274,6 +277,7 @@ stream. From ASoC DPCM framework, this s .trigger() start operation. .. code-block:: c + int sdw_enable_stream(struct sdw_stream_runtime * stream); SDW_STREAM_DISABLED @@ -301,6 +305,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -325,6 +330,7 @@ per stream. From ASoC DPCM framework, th .trigger() stop operation. .. code-block:: c + int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -349,6 +355,7 @@ all the Master(s) and Slave(s) associate framework, this stream state is linked to .hw_free() operation. .. code-block:: c + int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); int sdw_stream_remove_slave(struct sdw_slave * slave, @@ -361,6 +368,7 @@ stream assigned as part of ALLOCATED sta In .shutdown() the data structure maintaining stream state are freed up. .. code-block:: c + void sdw_release_stream(struct sdw_stream_runtime * stream); Not Supported

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[alsa-devel] [PATCH] ASoC: AMD: Fix capture unstable in beginning for some runs
by Akshu Agrawal 17 Sep '18

17 Sep '18

alsa_conformance_test -C hw:0,4 -p 1024 --debug would sometime show: TIME_DIFF(s) HW_LEVEL READ RATE 0.000095970 1024 1024 10670001.041992 0.042609555 1024 2048 24032.168372 0.021330364 1024 3072 48006.681930 0.021339559 1024 4096 47985.996337 The issue is that in dma pointer function we can have stale value of the register for current descriptor of channel. The register retains the number of the last descriptor that was transferred. Fix ensures that we report position, 0, till the one period worth of data is transferred. After one period of data, in handler of period completion interrupt we update the config and correct value of descriptor starts reflecting. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index c7e972b..f27e20c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1050,6 +1050,8 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) bytescount -= rtd->bytescount; delay = do_div(bytescount, period_bytes); runtime->delay = bytes_to_frames(runtime, delay); + if (bytescount < period_bytes) + pos = 0; } else { buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd); -- 1.9.1

1 0

[alsa-devel] [PATCH v3 00/13] ASoC: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

Thankyou for reviewing v2 patchset, here is v3 addressing comments from v2. Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback , capture paths and MBHC. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Changes since v2 (https://lwn.net/Articles/761107/) - Updated dt bindings according to Robs Comments. - Added MBHC support with new bindings - Moved mfd irq in to single file as suggested by Lee Jones. - Addressed various minor comments on mfd driver by Lee Jones. - Rebased on top of 4.19-rc1 - renamed clsh as clsh-v2 as its v2 block Srinivas Kandagatla (13): ASoC: dt-bindings: update wcd9335 bindings. mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: dt-bindings: Add WCD9335 MBHC specific properties ASoC: wcd9335: add mbhc support ASoC: apq8096: add slim support ASoC: apq8096: add headset JACK support .../bindings/sound/qcom,wcd9335.txt | 30 +- drivers/mfd/Kconfig | 14 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 369 ++ include/linux/mfd/wcd9335/registers.h | 627 ++ include/linux/mfd/wcd9335/wcd9335.h | 65 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh-v2.c | 577 ++ sound/soc/codecs/wcd-clsh-v2.h | 49 + sound/soc/codecs/wcd9335.c | 5234 +++++++++++++++++ sound/soc/qcom/apq8096.c | 122 +- 12 files changed, 7092 insertions(+), 6 deletions(-) create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh-v2.c create mode 100644 sound/soc/codecs/wcd-clsh-v2.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.18.0

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[alsa-devel] [PATCH 0/7] slimbus: fix module loading
by srinivas.kandagatla＠linaro.org 17 Sep '18

17 Sep '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Hi Greg, Here are some of the fixes for issues found while testing module loading on DB820c debian setup. First 3 core fixes are to do with generating uevents, calling status callback in probe sequence if the device is up and one fix is able to match device id from device tree compatible strings. Second 4 ngd patches fixes few issues firstly with registration, secondly checking the validity of logical address assigned by remote. Lastly there are few typo fixes. Thanks, Srini Srinivas Kandagatla (7): slimbus: core: add support to uevent slimbus: core: update device status in probe slimbus: core: match device tree based devices correctly slimbus: ngd: validate logical address assigned by remote silmbus: ngd: register controller after power up. slimbus: ngd: return proper error code instead of zero slimbus: ngd: register ngd driver only once. drivers/slimbus/core.c | 37 +++++++++++++++++++++++++++++++++++-- drivers/slimbus/qcom-ngd-ctrl.c | 28 +++++++++++++++++----------- 2 files changed, 52 insertions(+), 13 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH 0/3] ASoC: qdsp6: add compress offload support
by Srinivas Kandagatla 17 Sep '18

17 Sep '18

This patchset adds support to very basic MP3 playback compress offload support via QDSP ASM module. Tested this patchset on DB410c with APQ8016 and DB820c with APQ8096 using tinycompress library. Adding other codec support should be trivial w.r.t qdsp6 side, however there are other dependencies like compress UAPI header changes and FastRPC which are being worked in parallel. Once ready will post them! thanks, srini Srinivas Kandagatla (3): ASoC: q6asm-dai: dt-bindings: Add support to compress dais ASoC: qdsp6: q6asm: add support to MP3 format ASoC: qdsp6: q6asm-dai: Add support to compress offload .../devicetree/bindings/sound/qcom,q6asm.txt | 27 ++ sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 418 +++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.c | 5 + 4 files changed, 450 insertions(+), 1 deletion(-) -- 2.18.0

3 7

[alsa-devel] [RFC v2 - AAF PCM plugin 0/5] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 15 Sep '18

15 Sep '18

Hi all, This is the version 2 of AAF plugin RFC series. For general context information about the AAF plugin, please refer to the cover letter from version 1 [1] and the documentation in doc/aaf.txt. Code-wise, the main changes in this new version are: 1) The media clock was generating timestamps in UTC coordinate. However, AVTP timestamps are in TAI coordinate system so the media clock was fixed to use TAI instead. 2) All plugin resources (e.g. buffers, fds) are now allocated/freed in hw_params/hw_free ioplug callbacks. 3) The plugin constraints are applied at entry point instead of hw_params callback. Also, in this new version, the commit message from patch "aaf: Implement Playback mode support" was improved so it discusses the AVTP protocol restrictions in terms of formats and rates as well as why the plugin uses the system clock to implement a media clock. For further information about what has been discussed in the previous version, please refer to the RFC v1 patch series which starts in [1]. All versions of this series can be also found in my alsa-plugins tree in github [2]. Best regards, Andre [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-August/139494.Even [2] https://github.com/aguedes/alsa-plugins Andre Guedes (5): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1214 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 142 +++++++ 5 files changed, 1377 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

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[alsa-devel] Wrong UCM profile for Dell TB16 / WD15
by KillerRex 14 Sep '18

14 Sep '18

Hello, I have a Dell TB16 (same sound hardware as the WD15) and since the update to kernel 4.18 pulseaudio does not recognise any more the audio ports in the dock. After some work tracking the problem I have found that pulseaudio reports a failure when using the UCM profile from Dell-WD15-Dock: ... ( 0.027| 0.000) I: [pulseaudio] module-alsa-card.c: Found UCM profiles ( 0.027| 0.000) D: [pulseaudio] alsa-ucm.c: UCM mapping: Mapping HiFi: hw:WD15Dock,0: source dev Mic ( 0.027| 0.000) D: [pulseaudio] alsa-ucm.c: UCM mapping: Mapping HiFi: hw:WD15Dock,1: sink dev LineOut ( 0.027| 0.000) D: [pulseaudio] alsa-ucm.c: UCM mapping: Mapping HiFi: hw:WD15Dock,0: sink dev Headphone ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Profile HiFi (Default), input=(null), output=(null) priority=8000, supported=yes n_input_mappings=1, n_output_mappings=2 ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Input HiFi: hw:WD15Dock,0: source ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Output HiFi: hw:WD15Dock,1: sink ( 0.027| 0.000) D: [pulseaudio] alsa-mixer.c: Output HiFi: hw:WD15Dock,0: sink ( 0.027| 0.000) I: [pulseaudio] alsa-ucm.c: Set ucm verb to HiFi ( 0.027| 0.000) D: [pulseaudio] alsa-util.c: Trying hw:WD15Dock,1 with SND_PCM_NO_AUTO_FORMAT ... ( 0.028| 0.000) I: [pulseaudio] (alsa-lib)pcm_hw.c: Invalid value for card ( 0.028| 0.000) I: [pulseaudio] alsa-util.c: Error opening PCM device hw:WD15Dock,1: No such device ... The problem is that the device is called "Dock" and not "WD15Dock" so changing the file "alsa/ucm/Dell-WD15-Dock/HiFi.conf" with the right device name corrects the problem: --- alsa/ucm/Dell-WD15-Dock/HiFi.conf_ori 2018-09-14 17:54:35.022035850 +0200 +++ alsa/ucm/Dell-WD15-Dock/HiFi.conf 2018-09-14 17:54:55.722248500 +0200 @@ -3,7 +3,7 @@ Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,0" + PlaybackPCM "hw:Dock,0" } } @@ -12,7 +12,7 @@ Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,1" + PlaybackPCM "hw:Dock,1" } } @@ -21,6 +21,6 @@ Value { CaptureChannels "2" - CapturePCM "hw:WD15Dock,0" + CapturePCM "hw:Dock,0" } } Is this the right change or the real problem is that the device name is "Dock" instead "WD15Dock"? Removing the "WD15" from the name is better, in that way it is also valid for the TB16. Thanks Andres Ayala

1 0

[alsa-devel] [PATCH] ALSA: atiixp: fix fall-through annotations
by Gustavo A. R. Silva 14 Sep '18

14 Sep '18

Replace "fallthru" with a proper "fall through" annotation. This fix is part of the ongoing efforts to enabling -Wimplicit-fallthrough Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/atiixp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index a1e4944..1a41f8c 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -903,15 +903,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fallthru */ + /* fall through */ case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fallthru */ + /* fall through */ case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fallthru */ + /* fall through */ default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); -- 2.7.4

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Re: [alsa-devel] [PATCH -next] ASoC: qdsp6: q6asm-dai: remove duplicated include from q6asm-dai.c
by Srinivas Kandagatla 14 Sep '18

14 Sep '18

On 14/09/18 02:36, YueHaibing wrote: > Remove duplicated include. > > Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > --- > sound/soc/qcom/qdsp6/q6asm-dai.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c > index c75fab3..c3806d7 100644 > --- a/sound/soc/qcom/qdsp6/q6asm-dai.c > +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c > @@ -8,7 +8,6 @@ > #include <linux/platform_device.h> > #include <linux/slab.h> > #include <sound/soc.h> > -#include <sound/soc.h> > #include <sound/soc-dapm.h> > #include <sound/pcm.h> > #include <asm/dma.h> > > >

1 0

Re: [alsa-devel] of_dma_request_slave_channel() failed ?
by Geert Uytterhoeven 14 Sep '18

14 Sep '18

Hi Morimoto-san, CC asoc, dma, iommu On Tue, Sep 4, 2018 at 8:06 AM Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: > Hi Renesas ML > Cc Rob > > I noticed that Sound driver is using PIO mode > on latest kernel. > > ... > rcar_sound ec500000.sound: src[0] : probe error -11 > rcar_sound ec500000.sound: ssi[0] fallback to PIO mode > rcar_sound ec500000.sound: src[1] : probe error -11 > rcar_sound ec500000.sound: ssi[1] fallback to PIO mode > rcar_sound ec500000.sound: ssi[2] : probe error -11 > rcar_sound ec500000.sound: ssi[2] fallback to PIO mode > rcar_sound ec500000.sound: ssi[3] : probe error -11 > rcar_sound ec500000.sound: ssi[3] fallback to PIO mode > rcar_sound ec500000.sound: probed > ... > > Sound can use DMA mode and PIO mode. > It automatically fallbacks to PIO mode if it can't use DMA. > > I bisected this issue, and found the 1st bad commit > > ----------------------------------------------------- > commit ac6bbf0cdf4206c517ac9789814c23e372ebce4d > Author: Rob Herring <robh(a)kernel.org> > Date: Mon Jul 9 09:41:52 2018 -0600 > > iommu: Remove IOMMU_OF_DECLARE > > Now that we use the driver core to stop deferred probe for missing > drivers, IOMMU_OF_DECLARE can be removed. > > This is slightly less optimal than having a list of built-in drivers in > that we'll now defer probe twice before giving up. This shouldn't have a > significant impact on boot times as past discussions about deferred > probe have given no evidence of deferred probe having a substantial > impact. > ... > ----------------------------------------------------- > > In more detail, it seems it can't find DMA controller, > and of_dma_request_slave_channel() returns > -EPROBE_DEFER from this commit. > > struct dma_chan *of_dma_request_slave_channel(struct device_node *np, > const char *name) > { > ... > ofdma = of_dma_find_controller(&dma_spec); > > if (ofdma) { > chan = ofdma->of_dma_xlate(&dma_spec, ofdma); > } else { > => ret_no_channel = -EPROBE_DEFER; > chan = NULL; > } > ... > => return ERR_PTR(ret_no_channel); > } > > Is this known issue ? > Or sound shouldn't fallbacks to PIO mode ? I assume you wrote commit 6c92d5a2744e2761 ("ASoC: rsnd: don't fallback to PIO mode when -EPROBE_DEFER") to work around this? While this got rid of the error messages, and postpones sound initialization until the audio DMAC is available, it does mean that the driver will _never_ fall back to PIO. I.e. if CONFIG_RCAR_DMAC=n, audio will fail to probe instead of falling back to PIO. With CONFIG_RCAR_DMAC=y: rcar-dmac e6700000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac e7300000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac e7310000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac ec700000.dma-controller: ignoring dependency for device, assuming no driver rcar-dmac ec720000.dma-controller: ignoring dependency for device, assuming no driver So it seems the audio DMAC is deferred a second time, before the iommu driver probed. subsys_initcall(ipmmu_init); calls platform_driver_register module_platform_driver(rcar_dmac_driver); Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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Re: [alsa-devel] i915, HDMI/DP audio with multiple monitors
by Ville Syrjälä 13 Sep '18

13 Sep '18

On Tue, Sep 11, 2018 at 03:50:13PM +0200, Bruno Prémont wrote: > Hi, > > I have a system with multiple monitors and would like to send > notification sounds to the monitor on which corresponding > window is visible. > > For a workstation and a tiny computer things look different: > - workstation (Intel(R) Core(TM) i7-4770 CPU @ 3.40GHz): > 00:02.0 VGA compatible controller [0300]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller [8086:0412] (rev 06) > 00:03.0 Audio device [0403]: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller [8086:0c0c] (rev 06) > 00:1b.0 Audio device [0403]: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller [8086:8c20] (rev 04) > > Here alsa show me two cards: > - HDA Intel PCH (Realtek ALC671) > - HDA Intel HDMI (Intel Generic) > > **** List of PLAYBACK Hardware Devices **** > card 0: HDMI [HDA Intel HDMI], device 3: Generic Digital [Generic Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 There should normally be multiple HDMI devices (one for each HDMI/DP connector more or less). Eg. my hsw shows: card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 Looking at the hda_codec.c we see: static int audio_idx[HDA_PCM_NTYPES][5] = { ... [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 }, So you always get those device numbers, but I don't see an immediate relationship between those and the pin numbers (which do have some kind of relationship with the HDMI/DP port). I guess if they always get registered in order of the pin numbers then those would translate to 3 == port B, 7 == port C, etc. And after that the problem is figuring out which port is related to which connector, for which we have nothing at the moment. I suppose the ideal solution might be to have a sysfs symlink (or something) to connect the two together. But then there's MST where I think the pcm device correlates with the crtc rather than the connector. Maybe. I can't remember anymore, and I'm not sure it even works (I've recently heard people saying it doesn't). > card 1: PCH [HDA Intel PCH], device 0: ALC671 Analog [ALC671 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > - tiny computer (Intel(R) Core(TM) i5-6500T CPU @ 2.50GHz): > 00:02.0 VGA compatible controller [0300]: Intel Corporation HD Graphics 530 [8086:1912] (rev 06) > 00:1f.3 Audio device [0403]: Intel Corporation Sunrise Point-H HD Audio [8086:a170] (rev 31) > > Here alsa shows a single card: > - HDA Intel PCH (Realtek ALC671) > > **** List of PLAYBACK Hardware Devices **** > card 0: PCH [HDA Intel PCH], device 0: ALC671 Analog [ALC671 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: Generic Digital [Generic Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > How can I determine/set to which monitor the sound should go, and preferably send > different sounds to both monitors at same time? > > > Cheers, > Bruno > _______________________________________________ > dri-devel mailing list > dri-devel(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/dri-devel -- Ville Syrjälä Intel

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[alsa-devel] [PATCH v2] ALSA: hda - Enable runtime PM only for discrete GPU
by Takashi Iwai 13 Sep '18

13 Sep '18

The recent change of vga_switcheroo allowed the runtime PM for HD-audio on AMD GPUs, but this also resulted in a regression. When the HD-audio controller driver gets runtime-suspended, HD-audio link is turned off, and the hotplug notification is ignored. This leads to the inconsistent audio state (the connection isn't notified and ELD is ignored). The best fix would be to implement the proper ELD notification via the audio component, but it's still not ready. As a quick workaround, this patch adds the check of runtime_idle and allows the runtime suspend only when the vga_switcheroo is bound with discrete GPU. That is, a system with a single GPU and APU would be again without runtime PM to keep the HD-audio link for the hotplug notification and ELD read out. Also, the codec->auto_runtime_pm flag is set only for the discrete GPU at the time GPU gets bound via vga_switcheroo (i.e. only dGPU is forcibly runtime-PM enabled), so that APU can still get the ELD notification. For identifying which GPU is bound, a new vga_switcheroo client callback, gpu_bound, is implemented. The vga_switcheroo simply calls this when GPU is bound, and tells whether it's dGPU or APU. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200945 Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 2 + include/linux/vga_switcheroo.h | 3 ++ sound/pci/hda/hda_intel.c | 86 +++++++++++++++++++++++--------- sound/pci/hda/hda_intel.h | 1 + 4 files changed, 69 insertions(+), 23 deletions(-) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index a96bf46bc483..cf2a18571d48 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -215,6 +215,8 @@ static void vga_switcheroo_enable(void) return; client->id = ret | ID_BIT_AUDIO; + if (client->ops->gpu_bound) + client->ops->gpu_bound(client->pdev, ret); } vga_switcheroo_debugfs_init(&vgasr_priv); diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index a34539b7f750..7e6ac0114d55 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -133,15 +133,18 @@ struct vga_switcheroo_handler { * @can_switch: check if the device is in a position to switch now. * Mandatory. The client should return false if a user space process * has one of its device files open + * @gpu_bound: notify the client id to audio client when the GPU is bound. * * Client callbacks. A client can be either a GPU or an audio device on a GPU. * The @set_gpu_state and @can_switch methods are mandatory, @reprobe may be * set to NULL. For audio clients, the @reprobe member is bogus. + * OTOH, @gpu_bound is only for audio clients, and not used for GPU clients. */ struct vga_switcheroo_client_ops { void (*set_gpu_state)(struct pci_dev *dev, enum vga_switcheroo_state); void (*reprobe)(struct pci_dev *dev); bool (*can_switch)(struct pci_dev *dev); + void (*gpu_bound)(struct pci_dev *dev, enum vga_switcheroo_client_id); }; #if defined(CONFIG_VGA_SWITCHEROO) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..aa4c672dbaf7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -365,8 +365,10 @@ enum { */ #ifdef SUPPORT_VGA_SWITCHEROO #define use_vga_switcheroo(chip) ((chip)->use_vga_switcheroo) +#define needs_eld_notify_link(chip) ((chip)->need_eld_notify_link) #else #define use_vga_switcheroo(chip) 0 +#define needs_eld_notify_link(chip) false #endif #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ @@ -453,6 +455,7 @@ static inline void mark_runtime_wc(struct azx *chip, struct azx_dev *azx_dev, #endif static int azx_acquire_irq(struct azx *chip, int do_disconnect); +static void set_default_power_save(struct azx *chip); /* * initialize the PCI registers @@ -1201,6 +1204,10 @@ static int azx_runtime_idle(struct device *dev) azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; + /* ELD notification gets broken when HD-audio bus is off */ + if (needs_eld_notify_link(hda)) + return -EBUSY; + return 0; } @@ -1298,6 +1305,36 @@ static bool azx_vs_can_switch(struct pci_dev *pci) return true; } +/* + * The discrete GPU cannot power down unless the HDA controller runtime + * suspends, so activate runtime PM on codecs even if power_save == 0. + */ +static void setup_vga_switcheroo_runtime_pm(struct azx *chip) +{ + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + struct hda_codec *codec; + + if (hda->use_vga_switcheroo && !hda->need_eld_notify_link) { + list_for_each_codec(codec, &chip->bus) + codec->auto_runtime_pm = 1; + /* reset the power save setup */ + if (chip->running) + set_default_power_save(chip); + } +} + +static void azx_vs_gpu_bound(struct pci_dev *pci, + enum vga_switcheroo_client_id client_id) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip = card->private_data; + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); + + if (client_id == VGA_SWITCHEROO_DIS) + hda->need_eld_notify_link = 0; + setup_vga_switcheroo_runtime_pm(chip); +} + static void init_vga_switcheroo(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -1306,6 +1343,7 @@ static void init_vga_switcheroo(struct azx *chip) dev_info(chip->card->dev, "Handle vga_switcheroo audio client\n"); hda->use_vga_switcheroo = 1; + hda->need_eld_notify_link = 1; /* cleared in gpu_bound op */ chip->driver_caps |= AZX_DCAPS_PM_RUNTIME; pci_dev_put(p); } @@ -1314,6 +1352,7 @@ static void init_vga_switcheroo(struct azx *chip) static const struct vga_switcheroo_client_ops azx_vs_ops = { .set_gpu_state = azx_vs_set_state, .can_switch = azx_vs_can_switch, + .gpu_bound = azx_vs_gpu_bound, }; static int register_vga_switcheroo(struct azx *chip) @@ -1339,6 +1378,7 @@ static int register_vga_switcheroo(struct azx *chip) #define init_vga_switcheroo(chip) /* NOP */ #define register_vga_switcheroo(chip) 0 #define check_hdmi_disabled(pci) false +#define setup_vga_switcheroo_runtime_pm(chip) /* NOP */ #endif /* SUPPORT_VGA_SWITCHER */ /* @@ -1352,6 +1392,7 @@ static int azx_free(struct azx *chip) if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); + chip->running = 0; azx_del_card_list(chip); @@ -2230,6 +2271,25 @@ static struct snd_pci_quirk power_save_blacklist[] = { }; #endif /* CONFIG_PM */ +static void set_default_power_save(struct azx *chip) +{ + int val = power_save; + +#ifdef CONFIG_PM + if (pm_blacklist) { + const struct snd_pci_quirk *q; + + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } +#endif /* CONFIG_PM */ + snd_hda_set_power_save(&chip->bus, val * 1000); +} + /* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] = { [AZX_DRIVER_NVIDIA] = 8, @@ -2241,9 +2301,7 @@ static int azx_probe_continue(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; - struct hda_codec *codec; int dev = chip->dev_index; - int val; int err; hda->probe_continued = 1; @@ -2322,31 +2380,13 @@ static int azx_probe_continue(struct azx *chip) if (err < 0) goto out_free; + setup_vga_switcheroo_runtime_pm(chip); + chip->running = 1; azx_add_card_list(chip); - val = power_save; -#ifdef CONFIG_PM - if (pm_blacklist) { - const struct snd_pci_quirk *q; - - q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); - if (q && val) { - dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", - q->subvendor, q->subdevice); - val = 0; - } - } -#endif /* CONFIG_PM */ - /* - * The discrete GPU cannot power down unless the HDA controller runtime - * suspends, so activate runtime PM on codecs even if power_save == 0. - */ - if (use_vga_switcheroo(hda)) - list_for_each_codec(codec, &chip->bus) - codec->auto_runtime_pm = 1; + set_default_power_save(chip); - snd_hda_set_power_save(&chip->bus, val * 1000); if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index e3a3d318d2e5..f59719e06b91 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -37,6 +37,7 @@ struct hda_intel { /* vga_switcheroo setup */ unsigned int use_vga_switcheroo:1; + unsigned int need_eld_notify_link:1; unsigned int vga_switcheroo_registered:1; unsigned int init_failed:1; /* delayed init failed */ -- 2.18.0

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[alsa-devel] [PATCH v2 10/17] compat_ioctl: remove translation for sound ioctls
by Arnd Bergmann 13 Sep '18

13 Sep '18

The SNDCTL_* and SOUND_* commands are the old OSS user interface. I checked all the sound ioctl commands listed in fs/compat_ioctl.c to see if we still need the translation handlers. Here is what I found: - sound/oss/ is (almost) gone from the kernel, this is what actually needed all the translations - The ALSA emulation for OSS correctly handles all compat_ioctl commands already. - sound/oss/dmasound/ is the last holdout of the original OSS code, this is only used on arch/m68k, which has no 64-bit mode and hence needs no compat handlers - arch/um/drivers/hostaudio_kern.c may run in 64-bit mode with 32-bit x86 user space underneath it. This rare corner case is the only one that still needs the compat handlers. By adding a simple redirect of .compat_ioctl to .unlocked_ioctl in the UML driver, we can remove all the COMPATIBLE_IOCTL() annotations without a change in functionality. For completeness, I'm adding the same thing to the dmasound file, knowing that it makes no difference. The compat_ioctl list contains one comment about SNDCTL_DSP_MAPINBUF and SNDCTL_DSP_MAPOUTBUF, which actually would need a translation handler if implemented. However, the native implementation just returns -EINVAL, so we don't care. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/um/drivers/hostaudio_kern.c | 1 + fs/compat_ioctl.c | 157 ----------------------------- sound/core/oss/pcm_oss.c | 4 + sound/oss/dmasound/dmasound_core.c | 2 + 4 files changed, 7 insertions(+), 157 deletions(-) diff --git a/arch/um/drivers/hostaudio_kern.c b/arch/um/drivers/hostaudio_kern.c index 7f9dbdbc4eb7..9dfd209ed316 100644 --- a/arch/um/drivers/hostaudio_kern.c +++ b/arch/um/drivers/hostaudio_kern.c @@ -298,6 +298,7 @@ static const struct file_operations hostaudio_fops = { .write = hostaudio_write, .poll = hostaudio_poll, .unlocked_ioctl = hostaudio_ioctl, + .compat_ioctl = generic_compat_ioctl_ptrarg, .mmap = NULL, .open = hostaudio_open, .release = hostaudio_release, diff --git a/fs/compat_ioctl.c b/fs/compat_ioctl.c index 001c9fd89d30..c28fd9f6b1e8 100644 --- a/fs/compat_ioctl.c +++ b/fs/compat_ioctl.c @@ -77,7 +77,6 @@ #include <linux/if_bonding.h> #include <linux/watchdog.h> -#include <linux/soundcard.h> #include <linux/lp.h> #include <linux/ppdev.h> @@ -775,162 +774,6 @@ COMPATIBLE_IOCTL(PPPIOCGL2TPSTATS) /* PPPOX */ COMPATIBLE_IOCTL(PPPOEIOCSFWD) COMPATIBLE_IOCTL(PPPOEIOCDFWD) -/* Big A */ -/* sparc only */ -/* Big Q for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_SEQ_RESET) -COMPATIBLE_IOCTL(SNDCTL_SEQ_SYNC) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_INFO) -COMPATIBLE_IOCTL(SNDCTL_SEQ_CTRLRATE) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETOUTCOUNT) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETINCOUNT) -COMPATIBLE_IOCTL(SNDCTL_SEQ_PERCMODE) -COMPATIBLE_IOCTL(SNDCTL_FM_LOAD_INSTR) -COMPATIBLE_IOCTL(SNDCTL_SEQ_TESTMIDI) -COMPATIBLE_IOCTL(SNDCTL_SEQ_RESETSAMPLES) -COMPATIBLE_IOCTL(SNDCTL_SEQ_NRSYNTHS) -COMPATIBLE_IOCTL(SNDCTL_SEQ_NRMIDIS) -COMPATIBLE_IOCTL(SNDCTL_MIDI_INFO) -COMPATIBLE_IOCTL(SNDCTL_SEQ_THRESHOLD) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_MEMAVL) -COMPATIBLE_IOCTL(SNDCTL_FM_4OP_ENABLE) -COMPATIBLE_IOCTL(SNDCTL_SEQ_PANIC) -COMPATIBLE_IOCTL(SNDCTL_SEQ_OUTOFBAND) -COMPATIBLE_IOCTL(SNDCTL_SEQ_GETTIME) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_ID) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_CONTROL) -COMPATIBLE_IOCTL(SNDCTL_SYNTH_REMOVESAMPLE) -/* Big T for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_TMR_TIMEBASE) -COMPATIBLE_IOCTL(SNDCTL_TMR_START) -COMPATIBLE_IOCTL(SNDCTL_TMR_STOP) -COMPATIBLE_IOCTL(SNDCTL_TMR_CONTINUE) -COMPATIBLE_IOCTL(SNDCTL_TMR_TEMPO) -COMPATIBLE_IOCTL(SNDCTL_TMR_SOURCE) -COMPATIBLE_IOCTL(SNDCTL_TMR_METRONOME) -COMPATIBLE_IOCTL(SNDCTL_TMR_SELECT) -/* Little m for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_MIDI_PRETIME) -COMPATIBLE_IOCTL(SNDCTL_MIDI_MPUMODE) -COMPATIBLE_IOCTL(SNDCTL_MIDI_MPUCMD) -/* Big P for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_DSP_RESET) -COMPATIBLE_IOCTL(SNDCTL_DSP_SYNC) -COMPATIBLE_IOCTL(SNDCTL_DSP_SPEED) -COMPATIBLE_IOCTL(SNDCTL_DSP_STEREO) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETBLKSIZE) -COMPATIBLE_IOCTL(SNDCTL_DSP_CHANNELS) -COMPATIBLE_IOCTL(SOUND_PCM_WRITE_FILTER) -COMPATIBLE_IOCTL(SNDCTL_DSP_POST) -COMPATIBLE_IOCTL(SNDCTL_DSP_SUBDIVIDE) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETFRAGMENT) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETFMTS) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETFMT) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETOSPACE) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETISPACE) -COMPATIBLE_IOCTL(SNDCTL_DSP_NONBLOCK) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETCAPS) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETTRIGGER) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETTRIGGER) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETIPTR) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETOPTR) -/* SNDCTL_DSP_MAPINBUF, XXX needs translation */ -/* SNDCTL_DSP_MAPOUTBUF, XXX needs translation */ -COMPATIBLE_IOCTL(SNDCTL_DSP_SETSYNCRO) -COMPATIBLE_IOCTL(SNDCTL_DSP_SETDUPLEX) -COMPATIBLE_IOCTL(SNDCTL_DSP_GETODELAY) -COMPATIBLE_IOCTL(SNDCTL_DSP_PROFILE) -COMPATIBLE_IOCTL(SOUND_PCM_READ_RATE) -COMPATIBLE_IOCTL(SOUND_PCM_READ_CHANNELS) -COMPATIBLE_IOCTL(SOUND_PCM_READ_BITS) -COMPATIBLE_IOCTL(SOUND_PCM_READ_FILTER) -/* Big C for sound/OSS */ -COMPATIBLE_IOCTL(SNDCTL_COPR_RESET) -COMPATIBLE_IOCTL(SNDCTL_COPR_LOAD) -COMPATIBLE_IOCTL(SNDCTL_COPR_RDATA) -COMPATIBLE_IOCTL(SNDCTL_COPR_RCODE) -COMPATIBLE_IOCTL(SNDCTL_COPR_WDATA) -COMPATIBLE_IOCTL(SNDCTL_COPR_WCODE) -COMPATIBLE_IOCTL(SNDCTL_COPR_RUN) -COMPATIBLE_IOCTL(SNDCTL_COPR_HALT) -COMPATIBLE_IOCTL(SNDCTL_COPR_SENDMSG) -COMPATIBLE_IOCTL(SNDCTL_COPR_RCVMSG) -/* Big M for sound/OSS */ -COMPATIBLE_IOCTL(SOUND_MIXER_READ_VOLUME) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_BASS) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_TREBLE) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_SYNTH) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_PCM) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_SPEAKER) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_MIC) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_CD) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_IMIX) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_ALTPCM) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECLEV) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_IGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_OGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE1) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE2) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_LINE3) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL1)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL2)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_DIGITAL3)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_PHONEIN)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_PHONEOUT)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_VIDEO)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_RADIO)) -COMPATIBLE_IOCTL(MIXER_READ(SOUND_MIXER_MONITOR)) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_MUTE) -/* SOUND_MIXER_READ_ENHANCE, same value as READ_MUTE */ -/* SOUND_MIXER_READ_LOUD, same value as READ_MUTE */ -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECSRC) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_DEVMASK) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_RECMASK) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_STEREODEVS) -COMPATIBLE_IOCTL(SOUND_MIXER_READ_CAPS) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_VOLUME) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_BASS) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_TREBLE) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_SYNTH) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_PCM) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_SPEAKER) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_MIC) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_CD) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_IMIX) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_ALTPCM) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_RECLEV) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_IGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_OGAIN) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE1) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE2) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_LINE3) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL1)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL2)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_DIGITAL3)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_PHONEIN)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_PHONEOUT)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_VIDEO)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_RADIO)) -COMPATIBLE_IOCTL(MIXER_WRITE(SOUND_MIXER_MONITOR)) -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_MUTE) -/* SOUND_MIXER_WRITE_ENHANCE, same value as WRITE_MUTE */ -/* SOUND_MIXER_WRITE_LOUD, same value as WRITE_MUTE */ -COMPATIBLE_IOCTL(SOUND_MIXER_WRITE_RECSRC) -COMPATIBLE_IOCTL(SOUND_MIXER_INFO) -COMPATIBLE_IOCTL(SOUND_OLD_MIXER_INFO) -COMPATIBLE_IOCTL(SOUND_MIXER_ACCESS) -COMPATIBLE_IOCTL(SOUND_MIXER_AGC) -COMPATIBLE_IOCTL(SOUND_MIXER_3DSE) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE1) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE2) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE3) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE4) -COMPATIBLE_IOCTL(SOUND_MIXER_PRIVATE5) -COMPATIBLE_IOCTL(SOUND_MIXER_GETLEVELS) -COMPATIBLE_IOCTL(SOUND_MIXER_SETLEVELS) -COMPATIBLE_IOCTL(OSS_GETVERSION) /* Raw devices */ COMPATIBLE_IOCTL(RAW_SETBIND) COMPATIBLE_IOCTL(RAW_GETBIND) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index f8d4a419f3af..3f25c9d7b7df 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2732,6 +2732,10 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long static long snd_pcm_oss_ioctl_compat(struct file *file, unsigned int cmd, unsigned long arg) { + /* + * Everything is compatbile except SNDCTL_DSP_MAPINBUF/SNDCTL_DSP_MAPOUTBUF, + * which are not implemented for the native case either + */ return snd_pcm_oss_ioctl(file, cmd, (unsigned long)compat_ptr(arg)); } #else diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index fc9bcd47d6a4..cda6bbfedd22 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -384,6 +384,7 @@ static const struct file_operations mixer_fops = .owner = THIS_MODULE, .llseek = no_llseek, .unlocked_ioctl = mixer_unlocked_ioctl, + .compat_ioctl = generic_compat_ioctl_ptrarg, .open = mixer_open, .release = mixer_release, }; @@ -1167,6 +1168,7 @@ static const struct file_operations sq_fops = .write = sq_write, .poll = sq_poll, .unlocked_ioctl = sq_unlocked_ioctl, + .compat_ioctl = generic_compat_ioctl_ptrarg, .open = sq_open, .release = sq_release, }; -- 2.18.0

2 1

[alsa-devel] [PATCH] ALSA: oxfw: fix memory leak of private data
by Takashi Sakamoto 13 Sep '18

13 Sep '18

Although private data of sound card instance is usually allocated in the tail of the instance, drivers in ALSA firewire stack allocate the private data before allocating the instance. In this case, the private data should be released explicitly at .private_free callback of the instance. This commit fixes memory leak following to the above design. Fixes: 6c29230e2a5f ('ALSA: oxfw: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 1e5b2c802635..fd34ef2ac679 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -130,6 +130,7 @@ static void oxfw_free(struct snd_oxfw *oxfw) kfree(oxfw->spec); mutex_destroy(&oxfw->mutex); + kfree(oxfw); } /* -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: firewire-tascam: fix memory leak of private data
by Takashi Sakamoto 13 Sep '18

13 Sep '18

Although private data of sound card instance is usually allocated in the tail of the instance, drivers in ALSA firewire stack allocate the private data before allocating the instance. In this case, the private data should be released explicitly at .private_free callback of the instance. This commit fixes memory leak following to the above design. Fixes: b610386c8afb ('ALSA: firewire-tascam: deleyed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/tascam/tascam.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 44ad41fb7374..d3fdc463a884 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -93,6 +93,7 @@ static void tscm_free(struct snd_tscm *tscm) fw_unit_put(tscm->unit); mutex_destroy(&tscm->mutex); + kfree(tscm); } static void tscm_card_free(struct snd_card *card) -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: firewire-digi00x: fix memory leak of private data
by Takashi Sakamoto 13 Sep '18

13 Sep '18

Although private data of sound card instance is usually allocated in the tail of the instance, drivers in ALSA firewire stack allocate the private data before allocating the instance. In this case, the private data should be released explicitly at .private_free callback of the instance. This commit fixes memory leak following to the above design. Fixes: 86c8dd7f4da3 ('ALSA: firewire-digi00x: delayed registration of sound card') Cc: <stable(a)vger.kernel.org> # v4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/digi00x/digi00x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f5e1d23f31a..ef689997d6a5 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -49,6 +49,7 @@ static void dg00x_free(struct snd_dg00x *dg00x) fw_unit_put(dg00x->unit); mutex_destroy(&dg00x->mutex); + kfree(dg00x); } static void dg00x_card_free(struct snd_card *card) -- 2.17.1

2 1

[alsa-devel] Applied "ASoC: remove unneeded static set .owner field in platform_driver" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch ASoC: remove unneeded static set .owner field in platform_driver 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 e1e38ea14ea3294546e6350d05a1376197a73589 Mon Sep 17 00:00:00 2001 From: zhong jiang <zhongjiang(a)huawei.com> Date: Wed, 12 Sep 2018 11:41:49 +0800 Subject: [PATCH] ASoC: remove unneeded static set .owner field in platform_driver platform_driver_register will set the .owner field. So it is safe to remove the redundant assignment. The issue is detected with the help of Coccinelle. Signed-off-by: zhong jiang <zhongjiang(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 1 - sound/soc/mediatek/mt6797/mt6797-mt6351.c | 1 - sound/soc/rockchip/rk3288_hdmi_analog.c | 1 - 3 files changed, 3 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index 89f34efd9747..e5d49e6e2f99 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -150,7 +150,6 @@ static const struct of_device_id mt2701_wm8960_machine_dt_match[] = { static struct platform_driver mt2701_wm8960_machine = { .driver = { .name = "mt2701-wm8960", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt2701_wm8960_machine_dt_match, #endif diff --git a/sound/soc/mediatek/mt6797/mt6797-mt6351.c b/sound/soc/mediatek/mt6797/mt6797-mt6351.c index b1558c57b9ca..6e578e830e42 100644 --- a/sound/soc/mediatek/mt6797/mt6797-mt6351.c +++ b/sound/soc/mediatek/mt6797/mt6797-mt6351.c @@ -205,7 +205,6 @@ static const struct of_device_id mt6797_mt6351_dt_match[] = { static struct platform_driver mt6797_mt6351_driver = { .driver = { .name = "mt6797-mt6351", - .owner = THIS_MODULE, #ifdef CONFIG_OF .of_match_table = mt6797_mt6351_dt_match, #endif diff --git a/sound/soc/rockchip/rk3288_hdmi_analog.c b/sound/soc/rockchip/rk3288_hdmi_analog.c index 929b3fe289b0..a472d5eb2950 100644 --- a/sound/soc/rockchip/rk3288_hdmi_analog.c +++ b/sound/soc/rockchip/rk3288_hdmi_analog.c @@ -286,7 +286,6 @@ static struct platform_driver rockchip_sound_driver = { .probe = snd_rk_mc_probe, .driver = { .name = DRV_NAME, - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = rockchip_sound_of_match, }, -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2] ASoC: core: add support to card rebind
by Srinivas Kandagatla 13 Sep '18

13 Sep '18

Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - None, just rebased on top of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/?h=for-ne… include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 85 ++++++++++++++++++++++++++++++-------------- 2 files changed, 60 insertions(+), 27 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 821bf99992b8..46521b416aa1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1098,6 +1098,7 @@ struct snd_soc_card { /* lists of probed devices belonging to this card */ struct list_head component_dev_list; + struct list_head list; struct list_head widgets; struct list_head paths; @@ -1374,6 +1375,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) INIT_LIST_HEAD(&card->dapm_list); INIT_LIST_HEAD(&card->aux_comp_list); INIT_LIST_HEAD(&card->component_dev_list); + INIT_LIST_HEAD(&card->list); } static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e9367aacc0c..bc99bd5b2065 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -52,6 +52,7 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); static DEFINE_MUTEX(client_mutex); static LIST_HEAD(component_list); +static LIST_HEAD(unbind_card_list); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -2681,6 +2682,33 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); +static int snd_soc_bind_card(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + int ret; + + ret = snd_soc_instantiate_card(card); + if (ret != 0) + return ret; + + /* deactivate pins to sleep state */ + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dai(rtd, j, codec_dai) { + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + } + + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + + return ret; +} + /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2690,7 +2718,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); int snd_soc_register_card(struct snd_soc_card *card) { int i, ret; - struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2722,28 +2749,23 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); - ret = snd_soc_instantiate_card(card); - if (ret != 0) - return ret; - - /* deactivate pins to sleep state */ - list_for_each_entry(rtd, &card->rtd_list, list) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; - int j; - - for_each_rtd_codec_dai(rtd, j, codec_dai) { - if (!codec_dai->active) - pinctrl_pm_select_sleep_state(codec_dai->dev); - } + return snd_soc_bind_card(card); +} +EXPORT_SYMBOL_GPL(snd_soc_register_card); - if (!cpu_dai->active) - pinctrl_pm_select_sleep_state(cpu_dai->dev); +static void snd_soc_unbind_card(struct snd_soc_card *card, bool unregister) +{ + if (card->instantiated) { + card->instantiated = false; + snd_soc_dapm_shutdown(card); + soc_cleanup_card_resources(card); + if (!unregister) + list_add(&card->list, &unbind_card_list); + } else { + if (unregister) + list_del(&card->list); } - - return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); /** * snd_soc_unregister_card - Unregister a card with the ASoC core @@ -2753,12 +2775,8 @@ EXPORT_SYMBOL_GPL(snd_soc_register_card); */ int snd_soc_unregister_card(struct snd_soc_card *card) { - if (card->instantiated) { - card->instantiated = false; - snd_soc_dapm_shutdown(card); - soc_cleanup_card_resources(card); - dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); - } + snd_soc_unbind_card(card, true); + dev_dbg(card->dev, "ASoC: Unregistered card '%s'\n", card->name); return 0; } @@ -3102,7 +3120,7 @@ static void snd_soc_component_del_unlocked(struct snd_soc_component *component) struct snd_soc_card *card = component->card; if (card) - snd_soc_unregister_card(card); + snd_soc_unbind_card(card, false); list_del(&component->list); } @@ -3142,6 +3160,18 @@ static void convert_endianness_formats(struct snd_soc_pcm_stream *stream) stream->formats |= endianness_format_map[i]; } +static void snd_soc_try_rebind_card(void) +{ + struct snd_soc_card *card, *c; + + if (!list_empty(&unbind_card_list)) { + list_for_each_entry_safe(card, c, &unbind_card_list, list) { + if (!snd_soc_bind_card(card)) + list_del(&card->list); + } + } +} + int snd_soc_add_component(struct device *dev, struct snd_soc_component *component, const struct snd_soc_component_driver *component_driver, @@ -3169,6 +3199,7 @@ int snd_soc_add_component(struct device *dev, } snd_soc_component_add(component); + snd_soc_try_rebind_card(); return 0; -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: soc-core: remove dai->driver NULL check
by Kuninori Morimoto 13 Sep '18

13 Sep '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is strange if it has "dai" but doesn't have "dai->driver". And more over "dai->driver->xxx" is used everywhere without "dai->driver" pointer NULL checking. It got Oops already if "dai->driver" was NULL. Let's remove un-needed "dai->driver" NULL check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 41fef33..1f859cc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2524,8 +2524,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->driver == NULL) - return -EINVAL; if (dai->driver->ops->set_fmt == NULL) return -ENOTSUPP; return dai->driver->ops->set_fmt(dai, fmt); @@ -2672,9 +2670,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction) { - if (!dai->driver) - return -ENOTSUPP; - if (dai->driver->ops->mute_stream) return dai->driver->ops->mute_stream(dai, mute, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && -- 2.7.4

2 1

[alsa-devel] Applied "Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation"" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" 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 542cedec53c9e8b73f3f05bf8468823598c50489 Mon Sep 17 00:00:00 2001 From: Yu Zhao <yuzhao(a)google.com> Date: Tue, 11 Sep 2018 15:12:46 -0600 Subject: [PATCH] Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation" This reverts commit 12eeeb4f4733bbc4481d01df35933fc15beb8b19. The patch doesn't fix accessing memory with null pointer in skl_interrupt(). There are two problems: 1) skl_init_chip() is called twice, before and after dma buffer is allocate. The first call sets bus->chip_init which prevents the second from initializing bus->corb.buf and rirb.buf from bus->rb.area. 2) snd_hdac_bus_init_chip() enables interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers. There is a small window which skl_interrupt() can be called if irq has been acquired. If so, it crashes when using null dma buffer pointers. Will fix the problems in the following patches. Also attaching the crash for future reference. [ 16.949148] general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI <snipped> [ 16.950903] Call Trace: [ 16.950906] <IRQ> [ 16.950918] skl_interrupt+0x19e/0x2d6 [snd_soc_skl] [ 16.950926] ? dma_supported+0xb5/0xb5 [snd_soc_skl] [ 16.950933] __handle_irq_event_percpu+0x27a/0x6c8 [ 16.950937] ? __irq_wake_thread+0x1d1/0x1d1 [ 16.950942] ? __do_softirq+0x57a/0x69e [ 16.950944] handle_irq_event_percpu+0x95/0x1ba [ 16.950948] ? _raw_spin_unlock+0x65/0xdc [ 16.950951] ? __handle_irq_event_percpu+0x6c8/0x6c8 [ 16.950953] ? _raw_spin_unlock+0x65/0xdc [ 16.950957] ? time_cpufreq_notifier+0x483/0x483 [ 16.950959] handle_irq_event+0x89/0x123 [ 16.950962] handle_fasteoi_irq+0x16f/0x425 [ 16.950965] handle_irq+0x1fe/0x28e [ 16.950969] do_IRQ+0x6e/0x12e [ 16.950972] common_interrupt+0x7a/0x7a [ 16.950974] </IRQ> <snipped> [ 16.951031] RIP: snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] RSP: ffff88015c807c08 [ 16.951036] ---[ end trace 58bf9ece1775bc92 ]--- Fixes: 2eeeb4f4733b ("ASoC: Intel: Skylake: Acquire irq after RIRB allocation") Signed-off-by: Yu Zhao <yuzhao(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cf09721ca13e..dce649485649 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -838,7 +838,11 @@ static int skl_first_init(struct hdac_bus *bus) snd_hdac_bus_parse_capabilities(bus); + if (skl_acquire_irq(bus, 0) < 0) + return -EBUSY; + pci_set_master(pci); + synchronize_irq(bus->irq); gcap = snd_hdac_chip_readw(bus, GCAP); dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); @@ -871,12 +875,6 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - err = skl_acquire_irq(bus, 0); - if (err < 0) - return err; - - synchronize_irq(bus->irq); - /* initialize chip */ skl_init_pci(skl); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "sound: enable interrupt after dma buffer initialization" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch sound: enable interrupt after dma buffer initialization 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 b61749a89f826eb61fc59794d9e4697bd246eb61 Mon Sep 17 00:00:00 2001 From: Yu Zhao <yuzhao(a)google.com> Date: Tue, 11 Sep 2018 15:14:04 -0600 Subject: [PATCH] sound: enable interrupt after dma buffer initialization In snd_hdac_bus_init_chip(), we enable interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers. If irq has been acquired and irq handler uses the dma buffer, kernel may crash when interrupt comes in. Fix the problem by postponing enabling irq after dma buffer initialization. And warn once on null dma buffer pointer during the initialization. Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Yu Zhao <yuzhao(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/hda/hdac_controller.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 560ec0986e1a..11057d9f84ec 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -40,6 +40,8 @@ static void azx_clear_corbrp(struct hdac_bus *bus) */ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) { + WARN_ON_ONCE(!bus->rb.area); + spin_lock_irq(&bus->reg_lock); /* CORB set up */ bus->corb.addr = bus->rb.addr; @@ -479,13 +481,15 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) /* reset controller */ azx_reset(bus, full_reset); - /* initialize interrupts */ + /* clear interrupts */ azx_int_clear(bus); - azx_int_enable(bus); /* initialize the codec command I/O */ snd_hdac_bus_init_cmd_io(bus); + /* enable interrupts after CORB/RIRB buffers are initialized above */ + azx_int_enable(bus); + /* program the position buffer */ if (bus->use_posbuf && bus->posbuf.addr) { snd_hdac_chip_writel(bus, DPLBASE, (u32)bus->posbuf.addr); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "sound: don't call skl_init_chip() to reset intel skl soc" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch sound: don't call skl_init_chip() to reset intel skl soc 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 75383f8d39d4c0fb96083dd460b7b139fbdac492 Mon Sep 17 00:00:00 2001 From: Yu Zhao <yuzhao(a)google.com> Date: Tue, 11 Sep 2018 15:15:16 -0600 Subject: [PATCH] sound: don't call skl_init_chip() to reset intel skl soc Internally, skl_init_chip() calls snd_hdac_bus_init_chip() which 1) sets bus->chip_init to prevent multiple entrances before device is stopped; 2) enables interrupt. We shouldn't use it for the purpose of resetting device only because 1) when we really want to initialize device, we won't be able to do so; 2) we are ready to handle interrupt yet, and kernel crashes when interrupt comes in. Rename azx_reset() to snd_hdac_bus_reset_link(), and use it to reset device properly. Fixes: 60767abcea3d ("ASoC: Intel: Skylake: Reset the controller in probe") Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Yu Zhao <yuzhao(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/hdaudio.h | 1 + sound/hda/hdac_controller.c | 7 ++++--- sound/soc/intel/skylake/skl.c | 2 +- 3 files changed, 6 insertions(+), 4 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ab5ee3ef2198..207e816ce6e1 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -384,6 +384,7 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus); void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus); void snd_hdac_bus_enter_link_reset(struct hdac_bus *bus); void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus); +int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset); void snd_hdac_bus_update_rirb(struct hdac_bus *bus); int snd_hdac_bus_handle_stream_irq(struct hdac_bus *bus, unsigned int status, diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 11057d9f84ec..74244d8e2909 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -385,7 +385,7 @@ void snd_hdac_bus_exit_link_reset(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_bus_exit_link_reset); /* reset codec link */ -static int azx_reset(struct hdac_bus *bus, bool full_reset) +int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset) { if (!full_reset) goto skip_reset; @@ -410,7 +410,7 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) skip_reset: /* check to see if controller is ready */ if (!snd_hdac_chip_readb(bus, GCTL)) { - dev_dbg(bus->dev, "azx_reset: controller not ready!\n"); + dev_dbg(bus->dev, "controller not ready!\n"); return -EBUSY; } @@ -425,6 +425,7 @@ static int azx_reset(struct hdac_bus *bus, bool full_reset) return 0; } +EXPORT_SYMBOL_GPL(snd_hdac_bus_reset_link); /* enable interrupts */ static void azx_int_enable(struct hdac_bus *bus) @@ -479,7 +480,7 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) return false; /* reset controller */ - azx_reset(bus, full_reset); + snd_hdac_bus_reset_link(bus, full_reset); /* clear interrupts */ azx_int_clear(bus); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index dce649485649..1d17be0f78a0 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -834,7 +834,7 @@ static int skl_first_init(struct hdac_bus *bus) return -ENXIO; } - skl_init_chip(bus, true); + snd_hdac_bus_reset_link(bus, true); snd_hdac_bus_parse_capabilities(bus); -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ALSA: hda: Fix implicit definition of pci_iomap() on SH" to the asoc tree
by Mark Brown 13 Sep '18

13 Sep '18

The patch ALSA: hda: Fix implicit definition of pci_iomap() on SH 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 d9b84a15892c02334ac8a5c28865ae54168d9b22 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 12 Sep 2018 12:31:32 +0100 Subject: [PATCH] ALSA: hda: Fix implicit definition of pci_iomap() on SH Include asm/io.h directly so we've got a definition of pci_iomap(), the current set of includes do this implicitly on most architectures but not on SH. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/pci/hda/patch_ca0132.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index a585d0ec6d77..6ea04f889809 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -31,6 +31,7 @@ #include <linux/types.h> #include <linux/io.h> #include <linux/pci.h> +#include <asm/io.h> #include <sound/core.h> #include <sound/hda_codec.h> #include "hda_local.h" -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH 1/3] Revert "ASoC: Intel: Skylake: Acquire irq after RIRB allocation"
by Yu Zhao 13 Sep '18

13 Sep '18

This reverts commit 12eeeb4f4733bbc4481d01df35933fc15beb8b19. The patch doesn't fix accessing memory with null pointer in skl_interrupt(). There are two problems: 1) skl_init_chip() is called twice, before and after dma buffer is allocate. The first call sets bus->chip_init which prevents the second from initializing bus->corb.buf and rirb.buf from bus->rb.area. 2) snd_hdac_bus_init_chip() enables interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers. There is a small window which skl_interrupt() can be called if irq has been acquired. If so, it crashes when using null dma buffer pointers. Will fix the problems in the following patches. Also attaching the crash for future reference. [ 16.949148] general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI [ 16.950829] gsmi: Log Shutdown Reason 0x03 [ 16.950830] Modules linked in: uvcvideo(+) videobuf2_vmalloc snd_soc_skl(+) videobuf2_memops videobuf2_v4l2 videobuf2_core snd_soc_skl_ipc snd_soc_sst_ipc snd_soc_sst_dsp snd_soc_acpi snd_hda_ext_core snd_hda_core snd_soc_max98357a acpi_als snd_soc_da7219 lzo lzo_compress zram snd_seq_dummy snd_seq snd_seq_device bridge stp llc ipt_MASQUERADE nf_nat_masquerade_ipv4 xt_mark fuse cfg80211 iio_trig_sysfs cros_ec_sensors cros_ec_sensors_ring cros_ec_sensors_core industrialio_triggered_buffer kfifo_buf industrialio smsc95xx usbnet mii joydev [ 16.950874] CPU: 0 PID: 1083 Comm: chrome Not tainted 4.14.64 #14 [ 16.950875] Hardware name: Google Yorp/Yorp, BIOS Google_Yorp.10985.0.2018_08_20_1648 08/17/2018 [ 16.950878] task: ffff88015b1c2b80 task.stack: ffff880155f30000 [ 16.950887] RIP: 0010:snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] [ 16.950889] RSP: 0000:ffff88015c807c08 EFLAGS: 00010003 [ 16.950891] RAX: 0000000000000101 RBX: 000000000000080c RCX: 1ffff10026822185 [ 16.950893] RDX: dffffc0000000000 RSI: ffff88015b1c2b80 RDI: ffffc90000514058 [ 16.950894] RBP: ffff88015c807c68 R08: 0000000000000000 R09: 0000000000000000 [ 16.950895] R10: 0000000000000000 R11: ffffffffc043074f R12: 0000000000000800 [ 16.950897] R13: 0000000000000001 R14: 0000000000000002 R15: 1ffff10026822119 [ 16.950899] FS: 00007d85924cc740(0000) GS:ffff88015c800000(0000) knlGS:0000000000000000 [ 16.950900] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 16.950902] CR2: 000058a54df16038 CR3: 00000001551c0000 CR4: 00000000003406f0 [ 16.950903] Call Trace: [ 16.950906] <IRQ> [ 16.950918] skl_interrupt+0x19e/0x2d6 [snd_soc_skl] [ 16.950926] ? dma_supported+0xb5/0xb5 [snd_soc_skl] [ 16.950933] __handle_irq_event_percpu+0x27a/0x6c8 [ 16.950937] ? __irq_wake_thread+0x1d1/0x1d1 [ 16.950942] ? __do_softirq+0x57a/0x69e [ 16.950944] handle_irq_event_percpu+0x95/0x1ba [ 16.950948] ? _raw_spin_unlock+0x65/0xdc [ 16.950951] ? __handle_irq_event_percpu+0x6c8/0x6c8 [ 16.950953] ? _raw_spin_unlock+0x65/0xdc [ 16.950957] ? time_cpufreq_notifier+0x483/0x483 [ 16.950959] handle_irq_event+0x89/0x123 [ 16.950962] handle_fasteoi_irq+0x16f/0x425 [ 16.950965] handle_irq+0x1fe/0x28e [ 16.950969] do_IRQ+0x6e/0x12e [ 16.950972] common_interrupt+0x7a/0x7a [ 16.950974] </IRQ> [ 16.950976] RIP: 0033:0x58097f61a5c0 [ 16.950978] RSP: 002b:00007ffe95c971a8 EFLAGS: 00000206 ORIG_RAX: ffffffffffffffbc [ 16.950980] RAX: 000058097f61a5c0 RBX: 00000e4ac5220560 RCX: 0000000000004e10 [ 16.950982] RDX: 000058098563df20 RSI: 00007ffe95c97250 RDI: 00000e4ac5220500 [ 16.950983] RBP: 00007ffe95c97410 R08: 0000000000000000 R09: 00007ffe95c97250 [ 16.950984] R10: 0000000000000000 R11: 0000000000000000 R12: 00000e4ac5220560 [ 16.950986] R13: 00000e4ac5220560 R14: 00000e4ac47c9650 R15: 0000580987646350 [ 16.950988] Code: 74 12 48 89 df e8 eb 2d 8e cd 48 ba 00 00 00 00 00 fc ff df 4c 8b 23 44 89 f0 83 c8 01 0f b7 c0 49 8d 1c 84 48 89 d8 48 c1 e8 03 <8a> 04 10 84 c0 0f 85 da 01 00 00 44 8b 3b 41 0f b7 c6 49 8d 1c [ 16.951031] RIP: snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] RSP: ffff88015c807c08 [ 16.951036] ---[ end trace 58bf9ece1775bc92 ]--- [ 16.956871] Kernel panic - not syncing: Fatal exception in interrupt [ 16.956888] Kernel Offset: 0xc800000 from 0xffffffff81000000 (relocation range: 0xffffffff80000000-0xffffffffbfffffff) Signed-off-by: Yu Zhao <yuzhao(a)google.com> --- sound/soc/intel/skylake/skl.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cf09721ca13e..dce649485649 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -838,7 +838,11 @@ static int skl_first_init(struct hdac_bus *bus) snd_hdac_bus_parse_capabilities(bus); + if (skl_acquire_irq(bus, 0) < 0) + return -EBUSY; + pci_set_master(pci); + synchronize_irq(bus->irq); gcap = snd_hdac_chip_readw(bus, GCAP); dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); @@ -871,12 +875,6 @@ static int skl_first_init(struct hdac_bus *bus) if (err < 0) return err; - err = skl_acquire_irq(bus, 0); - if (err < 0) - return err; - - synchronize_irq(bus->irq); - /* initialize chip */ skl_init_pci(skl); -- 2.19.0.rc2.392.g5ba43deb5a-goog

4 19

[alsa-devel] [PATCH 0/2][RFC] ALSA: firewire-lib: change packet processing timing in process context
by Takashi Sakamoto 13 Sep '18

13 Sep '18

In IEC 61883-1/6 engine in ALSA firewire stack, a work to process packet is done in three places: 1. bottom-half of handler for interrupts scheduled on 1394OHCI controller 2. a callback of struct 'snd_pcm_ops.pointer()' 3. a callback of struct 'snd_pcm_ops.ack()' This patchset adds optimization for 3., and remove 2. for efficiency. In Linux kernel v2.6.39, 1. and 2. was committed in initial implementation of this engine at a commit 31ef9134eb52('ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver'). This engine schedules hardware interrupts 16 packets (=INTERRUPT_INTERVAL) thus 1. appears in software IRQ context every 2 msec (= 16packets * 1sec / 8,000cycle). In Linux kernel v3.5, an idea of 2. was introduced at a commit e9148dddc3c7('ALSA: firewire-lib: flush completed packets when reading PCM position') to reduce timing gap between queueing/dequeueing PCM frames and scheduling packet transmission. Additionally, it has an effect to reduce load of the software IRQ context. It's done in a call of ioctl(2) with SNDRV_PCM_IOCTL_HWSYNC. As a result, in mmap programming scenario, a call of snd_pcm_hwsync() can process packets for recent isochronous cycle. while (1): snd_pcm_hwsync() ->ioctl(HWSYNC) ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() In Linux kernel v4.13, ALSA PCM core was improved in an aspect of updating appl_ptr at a commit 9027c4639ef1('ALSA: pcm: Call ack() whenever appl_ptr is updated'). This commit ensures callback of 'struct snd_pcm_ops.ack()' when updating appl_ptr. This callback can be done in below ioctl(2) requests: - SNDRV_PCM_IOCTL_REWIND - SNDRV_PCM_IOCTL_FORWARD - SNDRV_PCM_IOCTL_SYNC_PTR - SNDRV_PCM_IOCTL_READ[I|N]_FRAMES - SNDRV_PCM_IOCTL_WRITE[I|N]_FRAMES As I notes in a commit 875becf8412c('ALSA: firewire: process packets in 'struct snd_pcm_ops.ack' callback'), 3. works less efficiently because in usual PCM operation the above ioctl(2) requests are not used in event loop. At the same time of the appl_ptr improvement, a new flag of PCM hardware definition was added at a commit 42f945970af9('ALSA: pcm: Add the explicit appl_ptr sync support'), and relevant changes were added to ALSA PCM core and alsa-lib v1.1.5. This flag expects userspace to call ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR request per iteration of PCM frame queueing/dequeueing in mmap programming scenario below: while (1): snd_pcm_hwsync() ->ioctl(HWSYNC) ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() ->ioctl(SYNC_PTR) This patchset aims to support the new flag, SNDRV_PCM_INFO_SYNC_APPLPTR, to tell userspace to call ioctl(2) with SNDRV_PCM_IOCTL_SYNC_PTR after queueing/dequeueing PCM frames in each iteration of event dispatcher. As a result: while (1): snd_pcm_hwsync() ->ioctl(HWSYNC) ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() ->ioctl(SYNC_PTR) ->struct snd_pcm_ops.ack() = amdtp_stream_pcm_ack() ->fw_iso_context_flush_completions() This patchset also adds optimization for timing to process packets in process context. As the above call graph shows, support of the flag brings closest two calls to process packet against intervals of isochronous cycle and it is not enough efficient. This patchset also cancel packet processing in .pointer() callback. As a result: while (1): snd_pcm_hwsync() snd_pcm_mmap_begin() (queue/dequeue PCM frames here.) snd_pcm_mmap_commit() ->ioctl(SYNC_PTR) ->struct snd_pcm_ops.ack() = amdtp_stream_pcm_ack() ->fw_iso_context_flush_completions() Takashi Sakamoto (2): ALSA: firewire-lib: adopt SYNC_APPLPTR flag to flush packets after queueing/dequeueing PCM frames ALSA: firewire-lib: cancel processing packets in struct snd_pcm_ops.pointer() call sound/firewire/amdtp-stream.c | 34 ++-------------------------------- sound/firewire/amdtp-stream.h | 12 +++++++++++- 2 files changed, 13 insertions(+), 33 deletions(-) -- 2.17.1

2 4

[alsa-devel] What causes a STOP in the trigger?
by Rob 13 Sep '18

13 Sep '18

I'm seeing a weird issue where ALSA doesn't ever hit the pointer function, and we go straight from START to STOP. Debugging the DMA, we get IRQs, I call the period_elapsed. It looks like we hit STOP before we even get the first IRQ. I've checked my parentheses, and theres no overrun or underrun messages. I don't hit the close function, nor any other functions until i control-C. Once I do that, I get a large series of pointer() printks showing multiples of IRQ positions, as if it queued up the commands... I thought I was IRQing too fast maybe, but as mentioned above, the irqreturn function doesn't even get called before we STOP. What in aplay would instantly force a STOP? Audio plays a few frames (I can hear a chord), then stalls and repeats until i Ctrl-C. Thanks, Rob

2 1

[alsa-devel] [PATCH] ALSA: rawmidi: A lightweight function to discard pending bytes
by Takashi Iwai 13 Sep '18

13 Sep '18

For discarding the pending bytes on rawmidi, we process with a loop of snd_rawmidi_transmit() which is just a waste of CPU power. Implement a lightweight API function to discard the pending bytes and the proceed the ring buffer instantly, and use it instead of open codes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/rawmidi.h | 1 + sound/core/rawmidi.c | 22 ++++++++++++++++++++++ sound/core/seq/seq_virmidi.c | 4 +--- sound/usb/midi.c | 3 +-- 4 files changed, 25 insertions(+), 5 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 6665cb29e1a2..3b5a061132b6 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -171,6 +171,7 @@ int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream, unsigned char *buffer, int count); int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count); +int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream); /* main midi functions */ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 08d5662039e3..ee601d7f0926 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1236,6 +1236,28 @@ int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream, } EXPORT_SYMBOL(snd_rawmidi_transmit); +/** + * snd_rawmidi_proceed - Discard the all pending bytes and proceed + * @substream: rawmidi substream + * + * Return: the number of discarded bytes + */ +int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + int count = 0; + + spin_lock_irqsave(&runtime->lock, flags); + if (runtime->avail < runtime->buffer_size) { + count = runtime->buffer_size - runtime->avail; + __snd_rawmidi_transmit_ack(substream, count); + } + spin_unlock_irqrestore(&runtime->lock, flags); + return count; +} +EXPORT_SYMBOL(snd_rawmidi_proceed); + static long snd_rawmidi_kernel_write1(struct snd_rawmidi_substream *substream, const unsigned char __user *userbuf, const unsigned char *kernelbuf, diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index cb988efd1ed0..e5a40795914a 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -149,9 +149,7 @@ static void snd_vmidi_output_work(struct work_struct *work) /* discard the outputs in dispatch mode unless subscribed */ if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - char buf[32]; - while (snd_rawmidi_transmit(substream, buf, sizeof(buf)) > 0) - ; /* ignored */ + snd_rawmidi_proceed(substream); return; } diff --git a/sound/usb/midi.c b/sound/usb/midi.c index dcfc546d81b9..b737f0ec77d0 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1175,8 +1175,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, if (port->ep->umidi->disconnected) { /* gobble up remaining bytes to prevent wait in * snd_rawmidi_drain_output */ - while (!snd_rawmidi_transmit_empty(substream)) - snd_rawmidi_transmit_ack(substream, 1); + snd_rawmidi_proceed(substream); return; } tasklet_schedule(&port->ep->tasklet); -- 2.18.0

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable runtime PM on AMD/ATI GPU only for discrete GPU
by Takashi Iwai 13 Sep '18

13 Sep '18

The recent change of vga_switcheroo allowed the runtime PM for HD-audio on AMD GPU, but this also resulted in a regression. When the HD-audio controller driver gets runtime-suspended, HD-audio link is turned off, and the hotplug notification is ignored. This leads to the inconsistent audio state (the connection isn't notified and ELD is ignored). The best fix would be to implement the proper ELD notification via the audio component, but it's still not ready. As a quick workaround, this patch adds the check of runtime_idle and allows the runtime suspend only when the vga_switcheroo is bound with discrete GPU. That is, a system with a single GPU would be again without runtime PM (of the controller), as well as the APU that needs to keep the HD-audio link for the hotplug notification and ELD read out. For identifying the audio client connection type, a new API is added to vga_switcheroo. Other than that, the patch does a pretty simple job. Note that this patch won't influence on the HD-audio *codec* runtime PM. When a codec is unused and the system goes to S3/S4 and resumed, the codec won't be resumed immediately because it's unused. This likely keeps the hotplug notification broken, but this is basically the expected behavior, a cost needed for the power-saving. The hotplug state will be activated once when you access to the device (or triggering the power up via reading a codec#* proc file or such). Again, this issue will be solved by the proper audio component notification, but it's still not ready. Until then, user would need to disable the power-saving explicitly if the ELD hotplug notification is required. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200945 Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/vga/vga_switcheroo.c | 20 ++++++++++++++++++++ include/linux/vga_switcheroo.h | 1 + sound/pci/hda/hda_intel.c | 8 ++++++++ 3 files changed, 29 insertions(+) diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c index a96bf46bc483..89598adbf6be 100644 --- a/drivers/gpu/vga/vga_switcheroo.c +++ b/drivers/gpu/vga/vga_switcheroo.c @@ -485,6 +485,26 @@ enum vga_switcheroo_state vga_switcheroo_get_client_state(struct pci_dev *pdev) } EXPORT_SYMBOL(vga_switcheroo_get_client_state); +/** + * vga_switcheroo_get_client_id() - obtain client id of a given client + * @pdev: client pci device + */ +enum vga_switcheroo_client_id vga_switcheroo_get_client_id(struct pci_dev *pdev) +{ + struct vga_switcheroo_client *client; + enum vga_switcheroo_client_id ret; + + mutex_lock(&vgasr_mutex); + client = find_client_from_pci(&vgasr_priv.clients, pdev); + if (!client) + ret = VGA_SWITCHEROO_UNKNOWN_ID; + else + ret = client_id(client); + mutex_unlock(&vgasr_mutex); + return ret; +} +EXPORT_SYMBOL(vga_switcheroo_get_client_id); + /** * vga_switcheroo_unregister_client() - unregister client * @pdev: client pci device diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h index a34539b7f750..e52fcd7b68f3 100644 --- a/include/linux/vga_switcheroo.h +++ b/include/linux/vga_switcheroo.h @@ -167,6 +167,7 @@ int vga_switcheroo_process_delayed_switch(void); bool vga_switcheroo_client_probe_defer(struct pci_dev *pdev); enum vga_switcheroo_state vga_switcheroo_get_client_state(struct pci_dev *dev); +enum vga_switcheroo_client_id vga_switcheroo_get_client_id(struct pci_dev *dev); int vga_switcheroo_init_domain_pm_ops(struct device *dev, struct dev_pm_domain *domain); void vga_switcheroo_fini_domain_pm_ops(struct device *dev); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1b2ce304152a..48ab090648a7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1201,6 +1201,14 @@ static int azx_runtime_idle(struct device *dev) azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; +#ifdef SUPPORT_VGA_SWITCHEROO + /* ELD notification gets broken on AMD GPUs when HD-audio bus is off */ + if ((chip->pci->vendor == PCI_VENDOR_ID_ATI || + chip->pci->vendor == PCI_VENDOR_ID_AMD) && + vga_switcheroo_get_client_id(chip->pci) != VGA_SWITCHEROO_DIS) + return -EBUSY; +#endif + return 0; } -- 2.18.0

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[alsa-devel] [PATCH RFC 0/5] Add devres support to card resources
by Takashi Iwai 13 Sep '18

13 Sep '18

Hi, the devres or device-managed resource allocation is a popular method among many drivers but we didn't have such a mechanism for the core sound card objects. Here is an attempt to implement that. The first patch gives the devm-version of page allocator helper, and the second patch provides the devm-version of snd_card_new(). The rest patches are examples of the new API usages. As of now, this patch series is still an RFC. The usefulness of this is still unclear, but I believe this can be used in all legacy ALSA drivers pretty well. For ASoC, this would help very little, supposedly, though. The latest patches are found in topic/devres branch of my sound.git tree, too. thanks, Takashi === Takashi Iwai (5): ALSA: core: Add device-managed page allocator helper ALSA: core: Add managed card creation ALSA: intel8x0: Allocate resources with device-managed APIs ALSA: atiixp: Allocate resources with device-managed APIs ALSA: hda: Allocate resources with device-managed APIs include/sound/core.h | 5 ++ include/sound/memalloc.h | 4 + sound/core/init.c | 95 +++++++++++++++++++++- sound/core/memalloc.c | 88 +++++++++++++++----- sound/pci/atiixp.c | 104 +++++++----------------- sound/pci/atiixp_modem.c | 104 +++++++----------------- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 43 +++------- sound/pci/intel8x0.c | 143 +++++++++++---------------------- sound/pci/intel8x0m.c | 143 +++++++++++---------------------- 10 files changed, 332 insertions(+), 398 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH] ALSA: intel8x0m: Register irq handler after register initializations
by Takashi Iwai 13 Sep '18

13 Sep '18

The interrupt handler has to be acquired after the other resource initialization when allocated with IRQF_SHARED. Otherwise it's triggered before the resource gets ready, and may lead to unpleasant behavior. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/intel8x0m.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 943a726b1c1b..c84629190cba 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1171,16 +1171,6 @@ static int snd_intel8x0m_create(struct snd_card *card, } port_inited: - if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { - dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); - snd_intel8x0m_free(chip); - return -EBUSY; - } - chip->irq = pci->irq; - pci_set_master(pci); - synchronize_irq(chip->irq); - /* initialize offsets */ chip->bdbars_count = 2; tbl = intel_regs; @@ -1224,11 +1214,21 @@ static int snd_intel8x0m_create(struct snd_card *card, chip->int_sta_reg = ICH_REG_GLOB_STA; chip->int_sta_mask = int_sta_masks; + pci_set_master(pci); + if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) { snd_intel8x0m_free(chip); return err; } + if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, + KBUILD_MODNAME, chip)) { + dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); + snd_intel8x0m_free(chip); + return -EBUSY; + } + chip->irq = pci->irq; + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_intel8x0m_free(chip); return err; -- 2.18.0

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[alsa-devel] [RFC - AAF PCM plugin 0/5] Introduce AVTP Audio Format (AAF) plugin
by Andre Guedes 13 Sep '18

13 Sep '18

Hi all, This patchset introduces the AAF plugin to the alsa-plugins project which enables TSN audio applications on Linux. For those not familiarized with TSN, it stands for Time-Sensitive Networking, and it's a set of IEEE technologies (enhancements to IEEE 802.1Q and Ethernet protocols) that provide precise time synchronization, bounded latency and application interoperability to bridged networks. Those technologies enable time-sensitive applications such as audio, video and control loops to run on top of bridged networks, co-existing with regular applications. TSN technologies are a super set of Audio Video Bridging (AVB) technologies also developed by IEEE. AVTP is the protocol defined to transport data in a TSN system, and AAF is one of the formats defined by AVTP to encapsulate audio data. AAF is specified in Chapter 7 from the AVTP spec [1]. This work is part of the effort to enable TSN technologies on upstream Linux ecosystem. All building-blocks required to enable TSN audio applications have been already developed and pushed to upstream projects. Time synchronization features are provided by linuxptp project [2], bounded latency features are provided by Linux Traffic Control subsystem since kernel version 4.15 [3], and AVTP packetization is provided by libavtp [4]. What is currently missing in the ecosystem to enable TSN audio applications is a piece of software that plumbs it all together and interfaces with Linux Audio system properly. That's the point of the AAF plugin introduced here. The AAF plugin is a PCM plugin that uses AVTP protocol to transmit/receive audio data through a TSN capable network. When operating in playback mode, the plugin reads PCM samples from the audio buffer, encapsulates into AVTP packets and transmits to the network, mimicking a typical AVTP Talker. When operating in capture mode, the plugin receives AVTP packets from the network, retrieves the PCM samples, and present them to alsa-lib layer at the presentation time, mimicking a typical AVTP Listener. Details about the plugin implementation are provided in the patches description, and information on how to configure and run the AAF plugin is provided in doc/aaf.txt file. Best regards, Andre [1] 1722-2016 - IEEE Standard for a Transport Protocol for Time-Sensitive Applications in Bridged Local Area Networks [2] http://linuxptp.sourceforge.net [3] https://patchwork.ozlabs.org/cover/826678/ [4] https://github.com/AVnu/libavtp Andre Guedes (5): aaf: Introduce plugin skeleton aaf: Load configuration parameters aaf: Implement Playback mode support aaf: Prepare for Capture mode support aaf: Implement Capture mode support Makefile.am | 3 + aaf/Makefile.am | 9 + aaf/pcm_aaf.c | 1172 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.ac | 9 + doc/aaf.txt | 110 ++++++ 5 files changed, 1303 insertions(+) create mode 100644 aaf/Makefile.am create mode 100644 aaf/pcm_aaf.c create mode 100644 doc/aaf.txt -- 2.14.4

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Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Connor McAdams 12 Sep '18

12 Sep '18

You get that terminal output while running the program? Those vfio_region_writes shouldn't be visible from the terminal... they should be fed into QemuHDADump, and then it outputs it's own stuff. If that is the output you're seeing, you might not have set up logging to go to stderr. On Wed, Sep 12, 2018 at 1:05 PM, Håvard <hovardslill(a)gmail.com> wrote: > I think this is right! > > I first booted up, then tested speakers (That I could not hear) and then > tested microphone (Which worked perfectly) in windows settings. > > -Håvard > > Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams > <conmanx360(a)gmail.com>: >> >> Also, sorry to keep sending more replies, make sure the only trace you >> have on is vfio_region_write, and not the read one. The read spams up >> the console. >> >> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >> wrote: >> > Reason being, it may not have identified the CORB buffer properly or >> > something. That's very odd to have no data. Never seen that before. >> > >> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams <conmanx360(a)gmail.com> >> > wrote: >> >> Could you try again and make a copy of your terminal output? >> >> >> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> wrote: >> >>> Hi again. >> >>> Sorry for not noticing all files were empty, did it not run for long >> >>> enough? >> >>> >> >>> -Håvard >> >>> >> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard <hovardslill(a)gmail.com>: >> >>>> >> >>>> Sorry my bad again it was working perfectly, just missed a step. >> >>>> >> >>>> I'll add the dumps as an attatchment. I had both external headset and >> >>>> microphone plugged in and let the VM run in the background for ~30min >> >>>> while >> >>>> doing other stuff. I did not play music or test my microphone in the >> >>>> VM >> >>>> either, If i did anything wrong, please tell me! >> >>>> >> >>>> It stopped at 0x104f4 >> >>>> >> >>>> -Håvard >> >>>> >> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >> >>>> <conmanx360(a)gmail.com>: >> >>>>> >> >>>>> Hm.... that's odd. They should show up in the folder you ran the >> >>>>> command from. Does your console show any "DumpMem entered..." or >> >>>>> something like that? You may have a permissions error. I've had some >> >>>>> people report that as an issue before. >> >>>>> >> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >> >>>>> wrote: >> >>>>> > Hi! >> >>>>> > >> >>>>> > Took a while, but I think I got it working. Did not see any >> >>>>> > "frame[xx]" >> >>>>> > files though. What dumps do you want? >> >>>>> > >> >>>>> > -Håvard >> >>>>> > >> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard >> >>>>> > <hovardslill(a)gmail.com>: >> >>>>> >> >> >>>>> >> Sorry about that, I forgot to enable a kernel config... >> >>>>> >> >> >>>>> >> -Håvard >> >>>>> >> >> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >> >>>>> >> <conmanx360(a)gmail.com>: >> >>>>> >>> >> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >> >>>>> >>> controls. >> >>>>> >>> You can check what module is loaded by doing lspci -v . It >> >>>>> >>> should say >> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the vfio-bind >> >>>>> >>> script >> >>>>> >>> before trying it? >> >>>>> >>> >> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard <hovardslill(a)gmail.com> >> >>>>> >>> wrote: >> >>>>> >>> > Thank you for the offer! >> >>>>> >>> > >> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >> >>>>> >>> > seem to >> >>>>> >>> > bind my >> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >> >>>>> >>> > to >> >>>>> >>> > function as >> >>>>> >>> > normal when I (think I) have bound it to a stud. >> >>>>> >>> > >> >>>>> >>> > -Håvard >> >>>>> >>> > >> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >> >>>>> >>> > <conmanx360(a)gmail.com>: >> >>>>> >>> >> >> >>>>> >>> >> One thing you could try, is using the program I used to >> >>>>> >>> >> reverse >> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. If >> >>>>> >>> >> your >> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >> >>>>> >>> >> you >> >>>>> >>> >> could >> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >> >>>>> >>> >> capture >> >>>>> >>> >> the commands. I'd be willing to look through the dumps to see >> >>>>> >>> >> if >> >>>>> >>> >> there >> >>>>> >>> >> are any special verbs or anything. >> >>>>> >>> >> >> >>>>> >>> >> You can find the program here: >> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >> >>>>> >>> >> >> >>>>> >>> >> Let me know. >> >>>>> >>> >> >> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard >> >>>>> >>> >> <hovardslill(a)gmail.com> >> >>>>> >>> >> wrote: >> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response before >> >>>>> >>> >> > I >> >>>>> >>> >> > read >> >>>>> >>> >> > your >> >>>>> >>> >> > last one. >> >>>>> >>> >> > >> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >> >>>>> >>> >> > headset >> >>>>> >>> >> > multijacks. >> >>>>> >>> >> > >> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the tip! >> >>>>> >>> >> > >> >>>>> >>> >> > -Håvard >> >>>>> >>> >> > >> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >> >>>>> >>> >> > <hovardslill(a)gmail.com>: >> >>>>> >>> >> > >> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >> >>>>> >>> >> >> >> >>>>> >>> >> >> Do you know where I can ask around for more help on the >> >>>>> >>> >> >> issue? >> >>>>> >>> >> >> I >> >>>>> >>> >> >> don't >> >>>>> >>> >> >> want to give up yet. >> >>>>> >>> >> >> >> >>>>> >>> >> >> -Håvard >> >>>>> >>> >> >> >> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >> >>>>> >>> >> >> <tiwai(a)suse.de>: >> >>>>> >>> >> >> >> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >> >>>>> >>> >> >>> Håvard wrote: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >> >>>>> >>> >> >>> > automatically >> >>>>> >>> >> >>> > changes >> >>>>> >>> >> >>> > to >> >>>>> >>> >> >>> it >> >>>>> >>> >> >>> > in pavucontrol. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Then it likely requires some additional initialization >> >>>>> >>> >> >>> outside >> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >> >>>>> >>> >> >>> vendor-specific. >> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >> >>>>> >>> >> >>> Windows >> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Takashi >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >> >>>>> >>> >> >>> > > Håvard wrote: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Thank you for replying! >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic Playback >> >>>>> >>> >> >>> > > > Volume" >> >>>>> >>> >> >>> > > > in >> >>>>> >>> >> >>> > > hdaanalyzer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >> >>>>> >>> >> >>> > > > made white noise come from the headset. However it >> >>>>> >>> >> >>> > > > did >> >>>>> >>> >> >>> > > > not >> >>>>> >>> >> >>> > > > change or >> >>>>> >>> >> >>> > > react >> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >> >>>>> >>> >> >>> > > > physically. >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >> >>>>> >>> >> >>> > > > either >> >>>>> >>> >> >>> > > > alsamixer >> >>>>> >>> >> >>> or >> >>>>> >>> >> >>> > > > hdaanalyzer. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > It's a mixer mute switch. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > The microphone works perfectly fine under Windows, >> >>>>> >>> >> >>> > > > so I >> >>>>> >>> >> >>> > > > don't >> >>>>> >>> >> >>> > > > think >> >>>>> >>> >> >>> it is >> >>>>> >>> >> >>> > > > the mic pin. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >> >>>>> >>> >> >>> > > wrong >> >>>>> >>> >> >>> > > pin, >> >>>>> >>> >> >>> > > too. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Takashi >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > -Håvard >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi Iwai >> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >> >>>>> >>> >> >>> >: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >> >>>>> >>> >> >>> > > > > Håvard wrote: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Additional relevant info: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three years >> >>>>> >>> >> >>> > > > > > ago >> >>>>> >>> >> >>> > > > > > on >> >>>>> >>> >> >>> > > > > > Sun >> >>>>> >>> >> >>> > > > > > Jun >> >>>>> >>> >> >>> > > 17:15:54 >> >>>>> >>> >> >>> > > > > CEST >> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >> >>>>> >>> >> >>> > > > > > but did >> >>>>> >>> >> >>> > > > > > not >> >>>>> >>> >> >>> > > > > > touch >> >>>>> >>> >> >>> on the >> >>>>> >>> >> >>> > > > > > external microphone problem: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > alsa-info.sh: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help the >> >>>>> >>> >> >>> > > > > > issue. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > There are several of similar bug reports around >> >>>>> >>> >> >>> > > > > > the >> >>>>> >>> >> >>> > > > > > web >> >>>>> >>> >> >>> experiencing >> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >> >>>>> >>> >> >>> > > > > > seems to >> >>>>> >>> >> >>> > > > > > happen >> >>>>> >>> >> >>> > > > > > with >> >>>>> >>> >> >>> all >> >>>>> >>> >> >>> > > > > laptops >> >>>>> >>> >> >>> > > > > > under the G751 name. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >> >>>>> >>> >> >>> > > > > and >> >>>>> >>> >> >>> > > > > unmute/adjust >> >>>>> >>> >> >>> "Mic >> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >> >>>>> >>> >> >>> > > > > you >> >>>>> >>> >> >>> > > > > hear >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from NID >> >>>>> >>> >> >>> > > > > 0x18 >> >>>>> >>> >> >>> > > > > to >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> mixer >> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >> >>>>> >>> >> >>> > > > > outputs. >> >>>>> >>> >> >>> > > > > So >> >>>>> >>> >> >>> > > > > this >> >>>>> >>> >> >>> > > > > can >> >>>>> >>> >> >>> be >> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from >> >>>>> >>> >> >>> the ext >> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >> >>>>> >>> >> >>> > > > > something >> >>>>> >>> >> >>> > > > > outside >> >>>>> >>> >> >>> > > > > HD-audio codec. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > Takashi >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> >> >>>>> >>> >> >> >> >>>>> >>> >> > _______________________________________________ >> >>>>> >>> >> > Alsa-devel mailing list >> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >> >>>>> >>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Håvard 12 Sep '18

12 Sep '18

I think this is right! I first booted up, then tested speakers (That I could not hear) and then tested microphone (Which worked perfectly) in windows settings. (My first email was dissaproved as it was too large) Den ons. 12. sep. 2018 kl. 19:05 skrev Håvard <hovardslill(a)gmail.com>: > I think this is right! > > I first booted up, then tested speakers (That I could not hear) and then > tested microphone (Which worked perfectly) in windows settings. > > -Håvard > > Den ons. 12. sep. 2018 kl. 18:49 skrev Connor McAdams < > conmanx360(a)gmail.com>: > >> Also, sorry to keep sending more replies, make sure the only trace you >> have on is vfio_region_write, and not the read one. The read spams up >> the console. >> >> On Wed, Sep 12, 2018 at 12:43 PM, Connor McAdams <conmanx360(a)gmail.com> >> wrote: >> > Reason being, it may not have identified the CORB buffer properly or >> > something. That's very odd to have no data. Never seen that before. >> > >> > On Wed, Sep 12, 2018 at 12:42 PM, Connor McAdams <conmanx360(a)gmail.com> >> wrote: >> >> Could you try again and make a copy of your terminal output? >> >> >> >> On Wed, Sep 12, 2018 at 12:41 PM, Håvard <hovardslill(a)gmail.com> >> wrote: >> >>> Hi again. >> >>> Sorry for not noticing all files were empty, did it not run for long >> enough? >> >>> >> >>> -Håvard >> >>> >> >>> Den ons. 12. sep. 2018 kl. 18:39 skrev Håvard <hovardslill(a)gmail.com >> >: >> >>>> >> >>>> Sorry my bad again it was working perfectly, just missed a step. >> >>>> >> >>>> I'll add the dumps as an attatchment. I had both external headset and >> >>>> microphone plugged in and let the VM run in the background for >> ~30min while >> >>>> doing other stuff. I did not play music or test my microphone in the >> VM >> >>>> either, If i did anything wrong, please tell me! >> >>>> >> >>>> It stopped at 0x104f4 >> >>>> >> >>>> -Håvard >> >>>> >> >>>> Den ons. 12. sep. 2018 kl. 00:31 skrev Connor McAdams >> >>>> <conmanx360(a)gmail.com>: >> >>>>> >> >>>>> Hm.... that's odd. They should show up in the folder you ran the >> >>>>> command from. Does your console show any "DumpMem entered..." or >> >>>>> something like that? You may have a permissions error. I've had some >> >>>>> people report that as an issue before. >> >>>>> >> >>>>> On Tue, Sep 11, 2018 at 5:08 PM, Håvard <hovardslill(a)gmail.com> >> wrote: >> >>>>> > Hi! >> >>>>> > >> >>>>> > Took a while, but I think I got it working. Did not see any >> "frame[xx]" >> >>>>> > files though. What dumps do you want? >> >>>>> > >> >>>>> > -Håvard >> >>>>> > >> >>>>> > Den tir. 11. sep. 2018 kl. 21:27 skrev Håvard < >> hovardslill(a)gmail.com>: >> >>>>> >> >> >>>>> >> Sorry about that, I forgot to enable a kernel config... >> >>>>> >> >> >>>>> >> -Håvard >> >>>>> >> >> >>>>> >> Den tir. 11. sep. 2018 kl. 21:20 skrev Connor McAdams >> >>>>> >> <conmanx360(a)gmail.com>: >> >>>>> >>> >> >>>>> >>> When it's bound as a stub, it shouldn't show up in alsamixer >> >>>>> >>> controls. >> >>>>> >>> You can check what module is loaded by doing lspci -v . It >> should say >> >>>>> >>> kernel driver in use: pci-stub. Also, did you run the vfio-bind >> >>>>> >>> script >> >>>>> >>> before trying it? >> >>>>> >>> >> >>>>> >>> On Tue, Sep 11, 2018 at 3:18 PM, Håvard <hovardslill(a)gmail.com> >> >>>>> >>> wrote: >> >>>>> >>> > Thank you for the offer! >> >>>>> >>> > >> >>>>> >>> > I run gentoo, but your github guide is very useful. I can't >> seem to >> >>>>> >>> > bind my >> >>>>> >>> > audio driver to a pci-stud. Is the Sound card still supposed >> to >> >>>>> >>> > function as >> >>>>> >>> > normal when I (think I) have bound it to a stud. >> >>>>> >>> > >> >>>>> >>> > -Håvard >> >>>>> >>> > >> >>>>> >>> > Den tir. 11. sep. 2018 kl. 20:26 skrev Connor McAdams >> >>>>> >>> > <conmanx360(a)gmail.com>: >> >>>>> >>> >> >> >>>>> >>> >> One thing you could try, is using the program I used to >> reverse >> >>>>> >>> >> engineer the Sound Blaster Z series of cards, QemuHDADump. >> If your >> >>>>> >>> >> processor supports pci-passthrough with a virtual machine, >> you >> >>>>> >>> >> could >> >>>>> >>> >> run a Windows virtual machine with the sound card in it and >> >>>>> >>> >> capture >> >>>>> >>> >> the commands. I'd be willing to look through the dumps to >> see if >> >>>>> >>> >> there >> >>>>> >>> >> are any special verbs or anything. >> >>>>> >>> >> >> >>>>> >>> >> You can find the program here: >> >>>>> >>> >> https://github.com/Conmanx360/QemuHDADump >> >>>>> >>> >> >> >>>>> >>> >> Let me know. >> >>>>> >>> >> >> >>>>> >>> >> On Tue, Sep 11, 2018 at 2:15 PM, Håvard < >> hovardslill(a)gmail.com> >> >>>>> >>> >> wrote: >> >>>>> >>> >> > Sorry, my gmail didn't update so I wrote my response >> before I >> >>>>> >>> >> > read >> >>>>> >>> >> > your >> >>>>> >>> >> > last one. >> >>>>> >>> >> > >> >>>>> >>> >> > It is a separate mic port as the G751JT doesn't have any >> headset >> >>>>> >>> >> > multijacks. >> >>>>> >>> >> > >> >>>>> >>> >> > I'll try and dig around in the kernel! Thanks for the tip! >> >>>>> >>> >> > >> >>>>> >>> >> > -Håvard >> >>>>> >>> >> > >> >>>>> >>> >> > Den tir. 11. sep. 2018 kl. 20:09 skrev Håvard >> >>>>> >>> >> > <hovardslill(a)gmail.com>: >> >>>>> >>> >> > >> >>>>> >>> >> >> Thank you for taking your time and trying to help! :) >> >>>>> >>> >> >> >> >>>>> >>> >> >> Do you know where I can ask around for more help on the >> issue? >> >>>>> >>> >> >> I >> >>>>> >>> >> >> don't >> >>>>> >>> >> >> want to give up yet. >> >>>>> >>> >> >> >> >>>>> >>> >> >> -Håvard >> >>>>> >>> >> >> >> >>>>> >>> >> >> Den tir. 11. sep. 2018 kl. 20:02 skrev Takashi Iwai >> >>>>> >>> >> >> <tiwai(a)suse.de>: >> >>>>> >>> >> >> >> >>>>> >>> >> >>> On Tue, 11 Sep 2018 19:14:37 +0200, >> >>>>> >>> >> >>> Håvard wrote: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Yes, microphone gets detected instantly and it >> automatically >> >>>>> >>> >> >>> > changes >> >>>>> >>> >> >>> > to >> >>>>> >>> >> >>> it >> >>>>> >>> >> >>> > in pavucontrol. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Then it likely requires some additional initialization >> outside >> >>>>> >>> >> >>> HD-audio. It's hard to know, as it's pretty much >> >>>>> >>> >> >>> vendor-specific. >> >>>>> >>> >> >>> You can dig down the Windows, but I have no idea about >> Windows >> >>>>> >>> >> >>> implementation, so can't give any hints, unfortunately. >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> Takashi >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > Den tir. 11. sep. 2018 kl. 18:52 skrev Takashi Iwai >> >>>>> >>> >> >>> > <tiwai(a)suse.de>: >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> > > On Tue, 11 Sep 2018 18:40:23 +0200, >> >>>>> >>> >> >>> > > Håvard wrote: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Thank you for replying! >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Enabling loopback in alsamixer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/lNo6e7T.png >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > And unmuting more and more things in "Mic Playback >> >>>>> >>> >> >>> > > > Volume" >> >>>>> >>> >> >>> > > > in >> >>>>> >>> >> >>> > > hdaanalyzer: >> >>>>> >>> >> >>> > > > http://i.imgur.com/H0HiOhy.png >> >>>>> >>> >> >>> > > > made white noise come from the headset. However it >> did >> >>>>> >>> >> >>> > > > not >> >>>>> >>> >> >>> > > > change or >> >>>>> >>> >> >>> > > react >> >>>>> >>> >> >>> > > > at all when I talked or even muted the microphone >> >>>>> >>> >> >>> > > > physically. >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > I couldn't find "Mic Playback Switch" anywhere in >> either >> >>>>> >>> >> >>> > > > alsamixer >> >>>>> >>> >> >>> or >> >>>>> >>> >> >>> > > > hdaanalyzer. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > It's a mixer mute switch. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > The microphone works perfectly fine under Windows, >> so I >> >>>>> >>> >> >>> > > > don't >> >>>>> >>> >> >>> > > > think >> >>>>> >>> >> >>> it is >> >>>>> >>> >> >>> > > > the mic pin. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > But the fact above indicates the possibility of the >> wrong >> >>>>> >>> >> >>> > > pin, >> >>>>> >>> >> >>> > > too. >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Does the jack detection of the ext mic pin work? >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > Takashi >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > -Håvard >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > Den man. 10. sep. 2018 kl. 22:39 skrev Takashi Iwai >> >>>>> >>> >> >>> > > > <tiwai(a)suse.de >> >>>>> >>> >> >>> >: >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > > > On Thu, 06 Sep 2018 20:44:30 +0200, >> >>>>> >>> >> >>> > > > > Håvard wrote: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Additional relevant info: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > A similar issue was also discussed three years >> ago >> >>>>> >>> >> >>> > > > > > on >> >>>>> >>> >> >>> > > > > > Sun >> >>>>> >>> >> >>> > > > > > Jun >> >>>>> >>> >> >>> > > 17:15:54 >> >>>>> >>> >> >>> > > > > CEST >> >>>>> >>> >> >>> > > > > > 2015 and was about his surround sound setup, >> but did >> >>>>> >>> >> >>> > > > > > not >> >>>>> >>> >> >>> > > > > > touch >> >>>>> >>> >> >>> on the >> >>>>> >>> >> >>> > > > > > external microphone problem: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > alsa-info.sh: >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> >>>>> >>> >> >>> >> http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Graph of ALC668 chipset from hdaanalyzer: >> >>>>> >>> >> >>> > > > > > http://i.imgur.com/c08DNJW.png >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > setting alsa-mode[1-8] does nothing to help the >> >>>>> >>> >> >>> > > > > > issue. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > There are several of similar bug reports >> around the >> >>>>> >>> >> >>> > > > > > web >> >>>>> >>> >> >>> experiencing >> >>>>> >>> >> >>> > > > > > similar issues, and on different distros. >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > Microphone works perfectly in windows >> >>>>> >>> >> >>> > > > > > >> >>>>> >>> >> >>> > > > > > I have a ASUS ROG G751JT, but this problem >> seems to >> >>>>> >>> >> >>> > > > > > happen >> >>>>> >>> >> >>> > > > > > with >> >>>>> >>> >> >>> all >> >>>>> >>> >> >>> > > > > laptops >> >>>>> >>> >> >>> > > > > > under the G751 name. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > When you enable the loopback volume and switch, >> and >> >>>>> >>> >> >>> > > > > unmute/adjust >> >>>>> >>> >> >>> "Mic >> >>>>> >>> >> >>> > > > > Playback Volume", and "Mic Playback Switch", do >> you >> >>>>> >>> >> >>> > > > > hear >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from the ext mic? It's a route directly from >> NID 0x18 >> >>>>> >>> >> >>> > > > > to >> >>>>> >>> >> >>> > > > > the >> >>>>> >>> >> >>> mixer >> >>>>> >>> >> >>> > > > > NID 0x0b, then output mixer NID 0x0c, then >> outputs. >> >>>>> >>> >> >>> > > > > So >> >>>>> >>> >> >>> > > > > this >> >>>>> >>> >> >>> > > > > can >> >>>>> >>> >> >>> be >> >>>>> >>> >> >>> > > > > used to verify the hardware routing. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > If you don't hear via this route, it means that >> the >> >>>>> >>> >> >>> > > > > input >> >>>>> >>> >> >>> > > > > from >> >>>>> >>> >> >>> the ext >> >>>>> >>> >> >>> > > > > mic pin itself is broken, and it implies that >> >>>>> >>> >> >>> > > > > something >> >>>>> >>> >> >>> > > > > outside >> >>>>> >>> >> >>> > > > > HD-audio codec. >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > > Takashi >> >>>>> >>> >> >>> > > > > >> >>>>> >>> >> >>> > > > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > > > >> >>>>> >>> >> >>> > > >> >>>>> >>> >> >>> > [2 <text/html; UTF-8 (quoted-printable)>] >> >>>>> >>> >> >>> > >> >>>>> >>> >> >>> >> >>>>> >>> >> >> >> >>>>> >>> >> > _______________________________________________ >> >>>>> >>> >> > Alsa-devel mailing list >> >>>>> >>> >> > Alsa-devel(a)alsa-project.org >> >>>>> >>> >> > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> >

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Re: [alsa-devel] Microphone detected, but no output for all ASUS G751xx with ALC668 chipset
by Håvard 12 Sep '18

12 Sep '18

Additional relevant info: A similar issue was also discussed three years ago on Sun Jun 17:15:54 CEST 2015 and was about his surround sound setup, but did not touch on the external microphone problem: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093317.html alsa-info.sh: http://www.alsa-project.org/db/?f=1d8616ba5977308e03db6c3a86e36e9e9b38d6f0 Graph of ALC668 chipset from hdaanalyzer: http://i.imgur.com/c08DNJW.png setting alsa-mode[1-8] does nothing to help the issue. There are several of similar bug reports around the web experiencing similar issues, and on different distros. Microphone works perfectly in windows I have a ASUS ROG G751JT, but this problem seems to happen with all laptops under the G751 name. These are my kernel configs: https://paste.pound-python.org/show/GRk4T6xNvmdp8W7K9slJ/ Thank you! -Håvard

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