July 2020 - Alsa-devel - mailman.alsa-project.org

Lenovo X270 with HDA ALC298
by Yasushi SHOJI 27 Jul '20

27 Jul '20

Hi all, I noticed that my 3 years old Lenovo X270 with Intel HDA & RealTek ALC298 doesn't work with 3.5 mm audio combo jack with 4 pole CTIA. A bit of googling showed me that `snd-hda-intel` module can take some options like `model` to run some fix ups. and I have found that `model=alc298-dell1` seems to at least work. But its gain is way too low. It's mostly just noise. It sounds like my mic is on the line-in. So, I opened `patch_realtek.c` to see what's going on. I found the following but not sure what the values for `.v.pins` are for. ``` [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ { } }, .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, ``` In `apply_fix(), these `.v.pins` values are passed to `snd_hda_apply_pincfgs()` but I can't find what values mean. Would someone please enlighten me? My questions are 1) How can I found out which pin goes where. There is x270 schematics on the net but not sure it's correct. 2) What does the value, ie, 0x01a1913d, represent? Thanks, -- yashi

2 1

/dev/sequencer descriptions
by Andrey Konovalov 27 Jul '20

27 Jul '20

Hi Takashi, We're working on syzkaller descriptions for /dev/sequencer [1], but have some troubles understanding its functionality. The main question is: is sound/synth/emux/ code reachable when tested in a VM (we're using QEMU with -soundhw all), or does it require some specific hardware to be present? In our setup (with Debian Stretch userspace image) we have /dev/sequencer with SNDRV_MINOR_OSS_SEQUENCER and /dev/sequencer2 with SNDRV_MINOR_OSS_MUSIC. For the former, there are no synth devices as reported by SNDCTL_SEQ_NRSYNTHS. For the latter, synth devices are there, but the load_patch() callback is not reachable as this check [2] fails. Is that code impossible to test in a VM at all? Or are there some specific kernel options/drivers that need to be enabled? Thanks! [1] https://github.com/google/syzkaller/pull/1952 [2] https://elixir.bootlin.com/linux/v5.7.8/source/sound/core/seq/oss/seq_oss_s…

2 2

[PATCH v2 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 27 Jul '20

27 Jul '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes for v2: - fix wrong *-gpios entries in the example, - mark sound-dai properties as required. --- .../bindings/sound/samsung,midas-audio.yaml | 108 +++++++++++++++++++++ 1 file changed, 108 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..fb685e51 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,108 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the I2S controller + required: + - sound-dai + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM1811 CODEC node + required: + - sound-dai + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + fm-sel-gpios = <&gpaa0 3 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

2 5

[PATCH] usb: endpoint : remove needless check before usb_free_coherent()
by Xu Wang 27 Jul '20

27 Jul '20

usb_free_coherent() is safe with NULL addr and this check is not required. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/usb/endpoint.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 88760268fb55..3a2b2a309a71 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -615,9 +615,8 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) for (i = 0; i < ep->nurbs; i++) release_urb_ctx(&ep->urb[i]); - if (ep->syncbuf) - usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, - ep->syncbuf, ep->sync_dma); + usb_free_coherent(ep->chip->dev, SYNC_URBS * 4, + ep->syncbuf, ep->sync_dma); ep->syncbuf = NULL; ep->nurbs = 0; -- 2.17.1

2 1

[PATCH 1/3] ASoC: samsung: Document DT bindings for Midas sound subsystem
by Sylwester Nawrocki 27 Jul '20

27 Jul '20

This patch adds documentation of DT biding for the Midas sound complex. Partially based on the *txt version by Simon Shields <simon(a)lineageos.org>. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,midas-audio.yaml | 105 +++++++++++++++++++++ 1 file changed, 105 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml new file mode 100644 index 0000000..4a82829 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,midas-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Midas audio complex with WM1811 codec + +maintainers: + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,midas-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the I2S controller + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM1811 CODEC node + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components; each entry is + a pair of strings, the first being the connection's sink, the second + being the connection's source; valid names for sources and sinks are + the WM1811's pins (as documented in its binding), and the jacks + on the board: HP, SPK, Main Mic, Sub Mic, Headset Mic. + + mic-bias-supply: + description: Supply for the micbias on the Main microphone + + submic-bias-supply: + description: Supply for the micbias on the Sub microphone + + fm-sel-gpios: + description: GPIO pin for FM selection + + lineout-sel-gpios: + description: GPIO pin for line out selection + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - mic-bias-supply + - submic-bias-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,midas-audio"; + model = "Midas"; + + mic-bias-gpios = <&gpf1 7 GPIO_ACTIVE_HIGH>; + submic-bias-gpios = <&gpf2 0 GPIO_ACTIVE_HIGH>; + + mic-bias-supply = <&mic_bias_reg>; + submic-bias-supply = <&submic_bias_reg>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + "SPK", "SPKOUTRN", + "SPK", "SPKOUTRP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + "IN1RP", "Sub Mic", + "IN1LP", "Sub Mic"; + + cpu { + sound-dai = <&i2s0>; + }; + + codec { + sound-dai = <&wm1811>; + }; + + }; -- 2.7.4

3 6

[PATCH 0/2] ASoC: intel: use asoc_substream_to_rtd()
by Kuninori Morimoto 27 Jul '20

27 Jul '20

Hi Mark I have posted "ASoC: add asoc_substream_to_rtd() macro" patch-set to ALSA SoC ML (= see Link), and then Pierre-Louis wanted that Intel patch was separated for boards. The patches which are not for Intel were already accepted. This is for Intel, and Intel boards. Link: https://lore.kernel.org/r/87y2nf0yw2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/877duz0ysw.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (2): ASoC: intel/boards: use asoc_substream_to_rtd() ASoC: intel: use asoc_substream_to_rtd() sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 16 ++++++++-------- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 ++-- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 4 ++-- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 ++-- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 4 ++-- sound/soc/intel/boards/sof_pcm512x.c | 4 ++-- sound/soc/intel/boards/sof_rt5682.c | 4 ++-- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 12 ++++++------ sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- 34 files changed, 62 insertions(+), 62 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: dt-bindings: q6asm: Add Q6ASM_DAI_{TX_RX, TX, RX} defines
by Stephan Gerhold 27 Jul '20

27 Jul '20

Right now the direction of a DAI has to be specified as a literal number in the device tree, e.g.: dai@0 { reg = <0>; direction = <2>; }; but this does not make it immediately clear that this is a playback/RX-only DAI. Actually, q6asm-dai.c has useful defines for this. Move them to the dt-bindings header to allow using them in the dts(i) files. The example above then becomes: dai@0 { reg = <0>; direction = <Q6ASM_DAI_RX>; }; which is immediately recognizable as playback/RX-only DAI. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Documentation/devicetree/bindings/sound/qcom,q6asm.txt | 9 +++++---- include/dt-bindings/sound/qcom,q6asm.h | 4 ++++ sound/soc/qcom/qdsp6/q6asm-dai.c | 3 --- 3 files changed, 9 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt index 6b9a88d0ea3f..8c4883becae9 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -39,9 +39,9 @@ configuration of each dai. Must contain the following properties. Usage: Required for Compress offload dais Value type: <u32> Definition: Specifies the direction of the dai stream - 0 for both tx and rx - 1 for only tx (Capture/Encode) - 2 for only rx (Playback/Decode) + Q6ASM_DAI_TX_RX (0) for both tx and rx + Q6ASM_DAI_TX (1) for only tx (Capture/Encode) + Q6ASM_DAI_RX (2) for only rx (Playback/Decode) - is-compress-dai: Usage: Required for Compress offload dais @@ -50,6 +50,7 @@ configuration of each dai. Must contain the following properties. = EXAMPLE +#include <dt-bindings/sound/qcom,q6asm.h> apr-service@7 { compatible = "qcom,q6asm"; @@ -62,7 +63,7 @@ apr-service@7 { dai@0 { reg = <0>; - direction = <2>; + direction = <Q6ASM_DAI_RX>; is-compress-dai; }; }; diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h index 1eb77d87c2e8..f59d74f14395 100644 --- a/include/dt-bindings/sound/qcom,q6asm.h +++ b/include/dt-bindings/sound/qcom,q6asm.h @@ -19,4 +19,8 @@ #define MSM_FRONTEND_DAI_MULTIMEDIA15 14 #define MSM_FRONTEND_DAI_MULTIMEDIA16 15 +#define Q6ASM_DAI_TX_RX 0 +#define Q6ASM_DAI_TX 1 +#define Q6ASM_DAI_RX 2 + #endif /* __DT_BINDINGS_Q6_ASM_H__ */ diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a2acb7564eb8..9b7b218f2a20 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -37,9 +37,6 @@ #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024) #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4) #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4) -#define Q6ASM_DAI_TX_RX 0 -#define Q6ASM_DAI_TX 1 -#define Q6ASM_DAI_RX 2 #define ALAC_CH_LAYOUT_MONO ((101 << 16) | 1) #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2) -- 2.27.0

3 2

[PATCH] ASoC: AMD: Restore PME_EN state at Power On
by Akshu Agrawal 27 Jul '20

27 Jul '20

PME_EN state needs to restored to the value set by fmw. For the devices which are not using I2S wake event which gets enabled by PME_EN bit, keeping PME_EN enabled burns considerable amount of power as it blocks low power state. For the devices using I2S wake event, PME_EN gets enabled in fmw and the state should be maintained after ACP Power On. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index ebf4388b6262..31b797c8bfe6 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -19,10 +19,12 @@ struct acp3x_dev_data { bool acp3x_audio_mode; struct resource *res; struct platform_device *pdev[ACP3x_DEVS]; + u32 pme_en; }; -static int acp3x_power_on(void __iomem *acp3x_base) +static int acp3x_power_on(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; u32 val; int timeout; @@ -39,10 +41,10 @@ static int acp3x_power_on(void __iomem *acp3x_base) while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); if (!val) { - /* Set PME_EN as after ACP power On, - * PME_EN gets cleared + /* ACP power On clears PME_EN. + * Restore the value to its prior state */ - rv_writel(0x1, acp3x_base + mmACP_PME_EN); + rv_writel(adata->pme_en, acp3x_base + mmACP_PME_EN); return 0; } udelay(1); @@ -74,12 +76,13 @@ static int acp3x_reset(void __iomem *acp3x_base) return -ETIMEDOUT; } -static int acp3x_init(void __iomem *acp3x_base) +static int acp3x_init(struct acp3x_dev_data *adata) { + void __iomem *acp3x_base = adata->acp3x_base; int ret; /* power on */ - ret = acp3x_power_on(acp3x_base); + ret = acp3x_power_on(adata); if (ret) { pr_err("ACP3x power on failed\n"); return ret; @@ -151,7 +154,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - ret = acp3x_init(adata->acp3x_base); + /* Save ACP_PME_EN state */ + adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN); + ret = acp3x_init(adata); if (ret) goto disable_msi; @@ -274,7 +279,7 @@ static int snd_acp3x_resume(struct device *dev) struct acp3x_dev_data *adata; adata = dev_get_drvdata(dev); - ret = acp3x_init(adata->acp3x_base); + ret = acp3x_init(adata); if (ret) { dev_err(dev, "ACP init failed\n"); return ret; -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Fix add a "ultra_low_power" function for intel reference board (alc256)
by Takashi Iwai 27 Jul '20

27 Jul '20

From: PeiSen Hou <pshou(a)realtek.com> Intel requires to enable power saving mode for intel reference board (alc256) Signed-off-by: PeiSen Hou <pshou(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3f2512942daf..0f640d99a396 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7590,7 +7590,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), - SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC225_FIXUP_HEADSET_JACK), + SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), -- 2.16.4

1 0

[PATCH] alsaucm: Fix ending with quotes commands
by Cheng Yueh 27 Jul '20

27 Jul '20

If we use 'alsaucm -n -b -' with 'get "CapturePCM/Internal Mic"' then the alsaucm will report error and stop immediately. The reason is that the parse_line in usecase.c appends an empty argument if a command ends with quotes. This change adds a patch to fix the parse_line function in usecase.c. Signed-off-by: Cheng Yueh <cyueh(a)chromium.org> --- alsaucm/usecase.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsaucm/usecase.c b/alsaucm/usecase.c index 1ed91ea..d39a159 100644 --- a/alsaucm/usecase.c +++ b/alsaucm/usecase.c @@ -130,6 +130,8 @@ static int parse_line(struct context *context, char *line) *line == '\n')) line++; c = *line; + if (c == '\0') + return 0; if (c == '\"' || c == '\'') { start = ++line; while (*line && *line != c) -- 2.26.2

2 1