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[PATCH 0/6] ALSA: firewire-lib: code refactoring for processing rx packets
by Takashi Sakamoto 22 May '21

22 May '21

Hi, This patchset refactors mainly for processing rx packets. Except for devices handled by ALSA bebob and dice drivers, the devices doesn't take care of the value of syt field of CIP header for playback timing. To code it explicitly, the new flag is added in 1st patch. Some of devices based on DICE ASICs supports two pairs of packet streams. In the case, the sequence of rx packet in the streams should be different independently so that each of them deliver timing information from each peer. On the other hand, current implementation pools the sequence in AMDTP domain, then the rx packets are processed with the same pool. It is inconvenient and in 2nd patch the pool is prepared per streams. The other patches are for my future work. Regards Takashi Sakamoto (6): ALSA: firewire-lib: add flag to unaware of syt in CIP header ALSA: firewire-lib: pool sequence of packet in IT context independently ALSA: firewire-lib: code refactoring for generation of packet descriptors ALSA: firewire-lib: code refactoring for generation of syt sequence ALSA: firewire-lib: code refactoring for generation of data block sequence ALSA: firewire-lib: code refactoring for transfer delay sound/firewire/amdtp-stream.c | 267 +++++++++++++-------------- sound/firewire/amdtp-stream.h | 33 ++-- sound/firewire/motu/amdtp-motu.c | 4 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- 4 files changed, 146 insertions(+), 164 deletions(-) -- 2.27.0

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[PATCH] ALSA: hda/realtek: the bass speaker can't output sound on Yoga 9i
by Hui Wang 22 May '21

22 May '21

The Lenovo Yoga 9i has bass speaker, but the bass speaker can't work, that is because there is an i2s amplifier on that speaker, need to run ideapad_s740_coef() to initialize the amplifier. And also needs to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK to rename the speaker's mixer control name, otherwise the PA can't handle them. BugLink: http://bugs.launchpad.net/bugs/1926165 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4eab336dc739..c6b5db831ed0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6552,6 +6552,7 @@ enum { ALC295_FIXUP_ASUS_DACS, ALC295_FIXUP_HP_OMEN, ALC285_FIXUP_HP_SPECTRE_X360, + ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, }; static const struct hda_fixup alc269_fixups[] = { @@ -8118,6 +8119,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, }, + [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_ideapad_s740_coef, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8486,6 +8493,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), + SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -8701,6 +8709,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

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[PATCH] ALSA: hda/realtek: Headphone volume is controlled by Front mixer
by Hui Wang 22 May '21

22 May '21

On some ASUS and MSI machines, the audio codec is alc1220 and the Headphone is connected to audio mixer 0xf and DAC 0x5, in theory the Headphone volume is controlled by DAC 0x5 (Heapdhone Playback Volume), but somehow it is controlled by DAC 0x2 (Front Playback Volume), maybe this is a defect on the codec alc1220. Because of this issue, the PA couldn't switch the headphone and Lineout correctly, If we apply the quirk CLEVO_P950 to those machines, the Lineout and Headphone will share the audio mixer 0xc and DAC 0x2, and generate Headphone+LO mixer, then PA could handle them when switching between them. BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1206 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c6b5db831ed0..3d40d32ef3ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2603,6 +2603,28 @@ static const struct hda_model_fixup alc882_fixup_models[] = { {} }; +static const struct snd_hda_pin_quirk alc882_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec1220, 0x1043, "ASUS", ALC1220_FIXUP_CLEVO_P950, + {0x14, 0x01014010}, + {0x15, 0x01011012}, + {0x16, 0x01016011}, + {0x18, 0x01a19040}, + {0x19, 0x02a19050}, + {0x1a, 0x0181304f}, + {0x1b, 0x0221401f}, + {0x1e, 0x01456130}), + SND_HDA_PIN_QUIRK(0x10ec1220, 0x1462, "MS-7C35", ALC1220_FIXUP_CLEVO_P950, + {0x14, 0x01015010}, + {0x15, 0x01011012}, + {0x16, 0x01011011}, + {0x18, 0x01a11040}, + {0x19, 0x02a19050}, + {0x1a, 0x0181104f}, + {0x1b, 0x0221401f}, + {0x1e, 0x01451130}), + {} +}; + /* * BIOS auto configuration */ @@ -2644,6 +2666,7 @@ static int patch_alc882(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, alc882_fixups); + snd_hda_pick_pin_fixup(codec, alc882_pin_fixup_tbl, alc882_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); -- 2.25.1

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[PATCH] ALSA: usb-audio: initialize the cur variable
by trix＠redhat.com 22 May '21

22 May '21

From: Tom Rix <trix(a)redhat.com> Static analysis reports this problem clock.c:338:10: warning: The right operand of '==' is a garbage value if (i == cur) ^ ~~~ In __uac_clock_find_source() the goto find_others skips over setting of the 'cur' variable. Since the pin loop variable starts with 1, it is safe to initialize cur to 0. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/usb/clock.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 772478c725c2e..edc458c50bf2c 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -258,9 +258,10 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, union uac23_clock_source_desc *source; union uac23_clock_selector_desc *selector; union uac23_clock_multiplier_desc *multiplier; - int ret, i, cur, err, pins, clock_id; + int ret, i, err, pins, clock_id; const u8 *sources; int proto = fmt->protocol; + int cur = 0; entity_id &= 0xff; -- 2.26.3

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[PATCH][next] ALSA: firewire-lib: Fix uninitialized variable err issue
by Colin King 22 May '21

22 May '21

From: Colin Ian King <colin.king(a)canonical.com> Currently in the case where the payload_length is less than the cip_header_size the error return variable err is not being set and function parse_ir_ctx_header can return an uninitialized error return value. Fix this by setting err to zero. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: c09010eeb373 ("ALSA: firewire-lib: handle the case that empty isochronous packet payload for CIP") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/firewire/amdtp-stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index af5c3629f1ac..242b1147d768 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -663,6 +663,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle, } else { // Handle the cycle so that empty packet arrives. cip_header = NULL; + err = 0; *data_blocks = 0; *syt = 0; } -- 2.31.1

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Motu M4 - where to report this issue?
by Georges Basile Stavracas Neto 21 May '21

21 May '21

Greetings ALSA community, I recently acquired a Motu M4 interface, and seems to be working well so far. Everyday usage is completely functional. However, I noticed an odd sound artifact that happens from time to time when using DAWs like Ardour and Waveform11. It's a windy, ghostly sound that happens whenever I open any DAWs (the simple act of opening them triggers this). Interestingly, audio apps that use PulseAudio do not suffer from this issue. It's hard to describe it, so I recorded a video reproducing this problem: https://youtu.be/bM8x-YuXLVI A few things I've discovered while trying to understand what's going on: * It only happens with apps that use JACK or ALSA APIs. Using PulseAudio (through PipeWire) does not trigger it. * Seems like this sound is not an artifact of the interface; it appears that this odd sound is sent to the interface through USB. (You can see this in the video.) * It's triggered per track; in the video, for example, the drums and keyboard tracks are affected independetly. I was oriented in #alsa at Freenode that the output of 'alsa-info' might be helpful, so here it is: http://alsa-project.org/db/?f=62f686710cc6c9344a8986707bf3d09248c514d1 (I'm running this on top of a 5.12.5 kernel, which should include this fix: https://bugzilla.kernel.org/show_bug.cgi?id=207023) I'd like to know where should I report this bug (kernel bugzilla?), and if there's any other information I can provide to help fixing it. With respect, Georges

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[PATCH 1/2] ASoC: Intel: common: Add entries for sdw codecs in ADL match table
by Kai Vehmanen 21 May '21

21 May '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> RT5682 and Max98373 are added with SDW0,SDW2 links respectively. Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- .../intel/common/soc-acpi-intel-adl-match.c | 45 +++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 39ac6d52106f..22c465f1d5d8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -223,6 +223,30 @@ static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link2_rt714_link0[] = { {} }; +static const struct snd_soc_acpi_adr_device mx8373_2_adr[] = { + { + .adr = 0x000223019F837300ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "Left" + }, + { + .adr = 0x000227019F837300ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "Right" + } +}; + +static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = { + { + .adr = 0x000021025D568200ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt5682" + } +}; + static const struct snd_soc_acpi_link_adr adl_rvp[] = { { .mask = BIT(0), @@ -232,6 +256,20 @@ static const struct snd_soc_acpi_link_adr adl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr adl_chromebook_base[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt5682_0_adr), + .adr_d = rt5682_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(mx8373_2_adr), + .adr_d = mx8373_2_adr, + }, + {} +}; + static const struct snd_soc_acpi_codecs adl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} @@ -295,6 +333,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt711.tplg", }, + { + .link_mask = 0x5, /* rt5682 on link0 & 2xmax98373 on link 2 */ + .links = adl_chromebook_base, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-sdw-max98373-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_sdw_machines); base-commit: aa736700f42fa0813e286ca2f9274ffaa25163b9 -- 2.31.0

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Motu M4 - where to report this issue?
by Georges Basile Stavracas Neto 21 May '21

21 May '21

Greetings ALSA community, I recently acquired a Motu M4 interface, and seems to be working well so far. Everyday usage is completely functional. However, I noticed an odd sound artifact that happens from time to time when using DAWs like Ardour and Waveform11. It's a windy, ghostly sound that happens whenever I open any DAWs (the simple act of opening them triggers this). Interestingly, audio apps that use PulseAudio do not suffer from this issue. It's hard to describe it, so I recorded a video reproducing this problem: https://youtu.be/bM8x-YuXLVI A few things I've discovered while trying to understand what's going on: * It only happens with apps that use JACK or ALSA APIs. Using PulseAudio (through PipeWire) does not trigger it. * Seems like this sound is not an artifact of the interface; it appears that this odd sound is sent to the interface through USB. (You can see this in the video.) * It's triggered per track; in the video, for example, the drums and keyboard tracks are affected independetly. I was oriented in #alsa at Freenode that the output of 'alsa-info' might be helpful, so here it is: http://alsa-project.org/db/?f=62f686710cc6c9344a8986707bf3d09248c514d1 (I'm running this on top of a 5.12.5 kernel, which should include this fix: https://bugzilla.kernel.org/show_bug.cgi?id=207023) I'd like to know where should I report this bug (kernel bugzilla?), and if there's any other information I can provide to help fixing it. With respect, Georges

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[PATCH v1] ASoC: dt-bindings: Convert imx-audmux binding to json schema
by Oleksij Rempel 21 May '21

21 May '21

Convert the imx-audmux binding to DT schema format using json-schema Signed-off-by: Oleksij Rempel <o.rempel(a)pengutronix.de> --- .../devicetree/bindings/sound/imx-audmux.txt | 28 ----- .../devicetree/bindings/sound/imx-audmux.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt deleted file mode 100644 index 2db4dcbee1b9..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ /dev/null @@ -1,28 +0,0 @@ -Freescale Digital Audio Mux (AUDMUX) device - -Required properties: - - - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used - on i.MX21, or "fsl,imx31-audmux" for the version - firstly used on i.MX31. - - - reg : Should contain AUDMUX registers location and length. - -An initial configuration can be setup using child nodes. - -Required properties of optional child nodes: - - - fsl,audmux-port : Integer of the audmux port that is configured by this - child node. - - - fsl,port-config : List of configuration options for the specific port. - For imx31-audmux and above, it is a list of tuples - <ptcr pdcr>. For imx21-audmux it is a list of pcr - values. - -Example: - -audmux@21d8000 { - compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; - reg = <0x021d8000 0x4000>; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000000..6078ce13bd46 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel(a)pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; -- 2.29.2

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[PATCH 0/7] ASoC: SOF: code cleanups for 5.14
by Kai Vehmanen 21 May '21

21 May '21

Set of code cleanups for issues found in code review. Jaska Uimonen (1): ASoC: SOF: topology: fix assignment to use le32_to_cpu Keyon Jie (1): ASoC: SOF: ops: print out the polling register Peter Ujfalusi (4): ASoC: SOF: Check desc->ops directly in acpi/pci/of probe functions ASoC: SOF: pci: No need to cast second time to save the desc ASoC: SOF: loader: Use snd_sof_dsp_block_read() instead sof_block_read() ASoC: SOF: Intel: hda: Remove conditions against CONFIG_PCI Pierre-Louis Bossart (1): ASoC: SOF: ops: don't return void value sound/soc/sof/intel/hda.c | 8 +++----- sound/soc/sof/loader.c | 2 +- sound/soc/sof/ops.h | 10 ++++++---- sound/soc/sof/sof-acpi-dev.c | 5 +---- sound/soc/sof/sof-of-dev.c | 5 +---- sound/soc/sof/sof-pci-dev.c | 7 ++----- sound/soc/sof/topology.c | 2 +- 7 files changed, 15 insertions(+), 24 deletions(-) base-commit: aa736700f42fa0813e286ca2f9274ffaa25163b9 -- 2.31.0

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