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

[alsa-devel] [PATCH 29/30] sound/pci: Use kmemdup rather than duplicating its implementation
by Fuqian Huang 03 Jul '19

03 Jul '19

kmemdup is introduced to duplicate a region of memory in a neat way. Rather than kmalloc/kzalloc + memset, which the programmer needs to write the size twice (sometimes lead to mistakes), kmemdup improves readability, leads to smaller code and also reduce the chances of mistakes. Suggestion to use kmemdup rather than using kmalloc/kzalloc + memset. Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index b612a536a5a1..35bd3e7c8ce1 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2189,11 +2189,10 @@ static int snd_echo_resume(struct device *dev) u32 pipe_alloc_mask; int err; - commpage_bak = kmalloc(sizeof(*commpage), GFP_KERNEL); + commpage_bak = kmemdup(commpage, sizeof(*commpage), GFP_KERNEL); if (commpage_bak == NULL) return -ENOMEM; commpage = chip->comm_page; - memcpy(commpage_bak, commpage, sizeof(*commpage)); err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); if (err < 0) { -- 2.11.0

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[alsa-devel] [PATCH 0/3] pcm_file report EIO when fail to write to debug file
by Adam Miartus 03 Jul '19

03 Jul '19

Previously, I introduced a patch that may return EPIPE error when pcm_file plugin fails to write to output file for some reason. The purpose of reporting the error was to evade assert in code to allow API user error handling and recovery, or more clarity in debugging. Failing to write to a file usually means bad file descriptor, running out of free memory, insufficient file permissions or in case file is a pipe there may be other reasons preventing the write operation. All these are not easily recoverable by restaring playback as EPIPE error which is used to signal XRUN might suggest. Therefore, use EIO to signal a different kind of error. write_wav_header function is refactored a bit for more detailed error handling in case of failed write operation. Adam Miartus (3): pcm_file: use EIO instead of EPIPE when failing to write output file pcm_file: improve error checking in write_wav_header function pcm_file: in case of failed write clear file buffer variables src/pcm/pcm_file.c | 54 +++++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 41 insertions(+), 13 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: sunxi: sun50i-codec-analog: Add earpiece
by Luca Weiss 03 Jul '19

03 Jul '19

This adds the necessary registers and audio routes to play audio using the Earpiece, that's supported on the A64. Signed-off-by: Luca Weiss <luca(a)z3ntu.xyz> --- So, first of all: This is my first audio patch and I hope I didn't make too many mistakes :) , especially with the routes at the bottom of the patch. What I'm really unsure about, is how the enable & mute registers should be handled. Should I put both registers into a SOC_DOUBLE("Earpiece Playback Switch",...)? sound/soc/sunxi/sun50i-codec-analog.c | 51 +++++++++++++++++++++++++++ 1 file changed, 51 insertions(+) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index d105c90c3706..6c19fea992c5 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -49,6 +49,15 @@ #define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 #define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 +#define SUN50I_ADDA_EARPIECE_CTRL0 0x03 +#define SUN50I_ADDA_EARPIECE_CTRL0_EAR_RAMP_TIME 4 +#define SUN50I_ADDA_EARPIECE_CTRL0_ESPSR 0 + +#define SUN50I_ADDA_EARPIECE_CTRL1 0x04 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN 7 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE 6 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL 0 + #define SUN50I_ADDA_LINEOUT_CTRL0 0x05 #define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 #define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 @@ -172,6 +181,10 @@ static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), ); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_earpiece_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); /* volume / mute controls */ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { @@ -225,6 +238,19 @@ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { SUN50I_ADDA_LINEOUT_CTRL0_LEN, SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + SOC_SINGLE_TLV("Earpiece Playback Volume", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL, 0x1f, 0, + sun50i_codec_earpiece_vol_scale), + + SOC_SINGLE("Earpiece Playback Switch (enable)", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN, 1, 0), + + SOC_SINGLE("Earpiece Playback Switch", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0), + }; static const char * const sun50i_codec_hp_src_enum_text[] = { @@ -257,6 +283,20 @@ static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { sun50i_codec_lineout_src_enum), }; +static const char * const sun50i_codec_earpiece_src_enum_text[] = { + "DACR", "DACL", "Right Analog Mixer", "Left Analog Mixer", +}; + +static SOC_ENUM_SINGLE_DECL(sun50i_codec_earpiece_src_enum, + SUN50I_ADDA_EARPIECE_CTRL0, + SUN50I_ADDA_EARPIECE_CTRL0_ESPSR, + sun50i_codec_earpiece_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_earpiece_src[] = { + SOC_DAPM_ENUM("Earpiece Source Playback Route", + sun50i_codec_earpiece_src_enum), +}; + static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* DAC */ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, @@ -285,6 +325,10 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_MUX("Earpiece Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_earpiece_src), + SND_SOC_DAPM_OUTPUT("EARPIECE"), + /* Microphone inputs */ SND_SOC_DAPM_INPUT("MIC1"), @@ -388,6 +432,13 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, + + /* Earpiece Routes */ + { "Earpiece Source Playback Route", "DACL", "Left DAC" }, + { "Earpiece Source Playback Route", "DACR", "Right DAC" }, + { "Earpiece Source Playback Route", "Left Analog Mixer", "Left Mixer" }, + { "Earpiece Source Playback Route", "Right Analog Mixer", "Right Mixer" }, + { "EARPIECE", NULL, "Earpiece Source Playback Route" }, }; static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { -- 2.22.0

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[alsa-devel] [PATCH] ALSA: usb-audio: fix Line6 Helix audio format rates
by Nicola Lunghi 03 Jul '19

03 Jul '19

Line6 Helix and HX stomp don't support retrieving the number of clock sample rate. Add a quirk to return the default value of 48Khz. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> --- sound/usb/format.c | 28 +++++++++++++++++++++++++--- 1 file changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index c02b51a82775..05442f6ada62 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -313,10 +313,32 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, tmp, sizeof(tmp)); if (ret < 0) { - dev_err(&dev->dev, - "%s(): unable to retrieve number of sample rates (clock %d)\n", + switch (chip->usb_id) { + /* LINE 6 HX pedals don't support getting the clock sample rate. + * Set the framerate to 48khz by default + */ + case USB_ID(0x0E41, 0x4244): /* HELIX */ + case USB_ID(0x0E41, 0x4246): /* HX STOMP */ + dev_warn(&dev->dev, + "%s(): line6 helix: unable to retrieve number of sample rates. Set it to default value (clock %d).\n", __func__, clock); - goto err; + fp->nr_rates = 1; + fp->rate_min = 48000; + fp->rate_max = 48000; + fp->rates = SNDRV_PCM_RATE_48000; + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + fp->rate_table[0] = 48000; + return 0; + default: + dev_err(&dev->dev, + "%s(): unable to retrieve number of sample rates (clock %d)\n", + __func__, clock); + goto err; + } } nr_triplets = (tmp[1] << 8) | tmp[0]; -- 2.19.1

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[alsa-devel] [PATCH 00/46] ASoC: consider CPU-Platform possibility
by Kuninori Morimoto 03 Jul '19

03 Jul '19

Hi Mark My posted patch (= "ASoC: xxx: don't select unnecessary Platform") removed Platform component settings from Card driver if it selects CPU as Platform. ALSA SoC will ignore duplicate component. I thought these Card driver don't need Platform, but it was wrong. Some CPU component is using generic DMAEngine component. In such case, CPU and Platform have same name but different component, and some Card needs both. My posted above patch breaks (not all but) few Cards. We can revert one-by-one if we get report, but better to revert all from stable kernel point of view. These patches revert all patches. I'm so sorry to bother you and users. I hope these can fixup issue. Kuninori Morimoto (46): ASoC: vc4: vc4_htmi: consider CPU-Platform possibility ASoC: atmel: atmel-classd: consider CPU-Platform possibility ASoC: atmel: atmel-pdmic: consider CPU-Platform possibility ASoC: atmel: atmel_wm8904: consider CPU-Platform possibility unnecessary Platform" ASoC: amtel: mikroe-proto: consider CPU-Platform possibility ASoC: atmel: sam9g20_wm8731: consider CPU-Platform possibility ASoC: atmel: sam9x5_wm8731: consider CPU-Platform possibility ASoC: atmel: tse850-pcm5142: consider CPU-Platform possibility ASoC: fsl: eukrea-tlv320: consider CPU-Platform possibility ASoC: fsl: fsl-asoc-card: consider CPU-Platform possibility ASoC: fsl: imx-es8328: consider CPU-Platform possibility ASoC: fsl: imx-sgtl5000: consider CPU-Platform possibility ASoC: fsl: imx-spdif: consider CPU-Platform possibility ASoC: fsl: imx-audmix: consider CPU-Platform possibility ASoC: kirkwood: armada-370-db: consider CPU-Platform possibility ASoC: mxs: mxs-sgtl5000: consider CPU-Platform possibility ASoC: qcom: apq8016_sbc: consider CPU-Platform possibility ASoC: qcom: storm: consider CPU-Platform possibility ASoC: rockchip: rk3288_hdmi_analog: consider CPU-Platform possibility ASoC: rockchip: rockchip_max98090: consider CPU-Platform possibility ASoC: rockchip: rockchip_rt5645: consider CPU-Platform possibility ASoC: samsung: arndale_rt5631: consider CPU-Platform possibility ASoC: samsung: smdk_wm8994: consider CPU-Platform possibility ASoC: samsung: snow: consider CPU-Platform possibility ASoC: samsung: tm2_wm5110: consider CPU-Platform possibility ASoC: sirf: sirf-audio: consider CPU-Platform possibility ASoC: sunxi: sun4i-codec: consider CPU-Platform possibility ASoC: tegra: tegra_alc5632: consider CPU-Platform possibility ASoC: tegra: tegra_max98090: consider CPU-Platform possibility ASoC: tegra: tegra_rt5640: consider CPU-Platform possibility ASoC: tegra: tegra_rt5677: consider CPU-Platform possibility ASoC: tegra: tegra_sgtl5000: consider CPU-Platform possibility ASoC: tegra: tegra_wm8753: consider CPU-Platform possibility ASoC: tegra: tegra_wm8903: consider CPU-Platform possibility ASoC: tegra: tegra_wm9712: consider CPU-Platform possibility ASoC: tegra: trimslice: consider CPU-Platform possibility ASoC: ti: davinci-evm: consider CPU-Platform possibility ASoC: ti: omap-abe-twl6040: consider CPU-Platform possibility ASoC: ti: omap-hdmi: consider CPU-Platform possibility ASoC: ti: omap-twl4030: consider CPU-Platform possibility ASoC: ti: rx51: consider CPU-Platform possibility ASoC: ux500: mop500: consider CPU-Platform possibility ASoC: simple-card-utils: consider CPU-Platform possibility ASoC: qcom: consider CPU-Platform possibility ASoC: rockchip: rk3399_gru_sound: consider CPU-Platform possibility ASoC: soc-utils: respawn dummy Platform drivers/gpu/drm/vc4/vc4_hdmi.c | 4 ++++ sound/soc/atmel/atmel-classd.c | 5 ++++- sound/soc/atmel/atmel-pdmic.c | 5 ++++- sound/soc/atmel/atmel_wm8904.c | 4 +++- sound/soc/atmel/mikroe-proto.c | 7 +++++-- sound/soc/atmel/sam9g20_wm8731.c | 5 ++++- sound/soc/atmel/sam9x5_wm8731.c | 5 ++++- sound/soc/atmel/tse850-pcm5142.c | 4 +++- sound/soc/fsl/eukrea-tlv320.c | 5 ++++- sound/soc/fsl/fsl-asoc-card.c | 6 +++++- sound/soc/fsl/imx-audmix.c | 14 ++++++++++---- sound/soc/fsl/imx-es8328.c | 5 ++++- sound/soc/fsl/imx-sgtl5000.c | 5 ++++- sound/soc/fsl/imx-spdif.c | 5 ++++- sound/soc/generic/simple-card-utils.c | 15 +++------------ sound/soc/kirkwood/armada-370-db.c | 12 +++++++++--- sound/soc/mxs/mxs-sgtl5000.c | 8 ++++++-- sound/soc/qcom/apq8016_sbc.c | 10 +++++++--- sound/soc/qcom/common.c | 4 +--- sound/soc/qcom/storm.c | 4 +++- sound/soc/rockchip/rk3288_hdmi_analog.c | 5 ++++- sound/soc/rockchip/rk3399_gru_sound.c | 19 +++++++++++++------ sound/soc/rockchip/rockchip_max98090.c | 5 ++++- sound/soc/rockchip/rockchip_rt5645.c | 5 ++++- sound/soc/samsung/arndale_rt5631.c | 6 +++++- sound/soc/samsung/smdk_wm8994.c | 9 +++++++-- sound/soc/samsung/snow.c | 10 +++++++--- sound/soc/samsung/tm2_wm5110.c | 12 +++++++++--- sound/soc/sirf/sirf-audio.c | 5 ++++- sound/soc/soc-utils.c | 25 +++++++++++++++++++++++++ sound/soc/sunxi/sun4i-codec.c | 5 ++++- sound/soc/tegra/tegra_alc5632.c | 7 ++++++- sound/soc/tegra/tegra_max98090.c | 5 ++++- sound/soc/tegra/tegra_rt5640.c | 5 ++++- sound/soc/tegra/tegra_rt5677.c | 6 +++++- sound/soc/tegra/tegra_sgtl5000.c | 7 ++++++- sound/soc/tegra/tegra_wm8753.c | 5 ++++- sound/soc/tegra/tegra_wm8903.c | 5 ++++- sound/soc/tegra/tegra_wm9712.c | 5 ++++- sound/soc/tegra/trimslice.c | 6 +++++- sound/soc/ti/davinci-evm.c | 5 ++++- sound/soc/ti/omap-abe-twl6040.c | 22 ++++++++++++++-------- sound/soc/ti/omap-hdmi.c | 5 ++++- sound/soc/ti/omap-twl4030.c | 12 ++++++++++-- sound/soc/ti/rx51.c | 5 ++++- sound/soc/ux500/mop500.c | 8 ++++++-- 46 files changed, 261 insertions(+), 85 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 1/2] ASoC: SOF: ipc: update sof_ipc_stream_params
by Keyon Jie 03 Jul '19

03 Jul '19

From: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> The host period bytes needs to be passed to firmware. Currently this field is used as notification for host about period copy completion. Therefore we need to split these two informations. Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> --- include/sound/sof/stream.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h index 643f175cb479..44acfa62fa69 100644 --- a/include/sound/sof/stream.h +++ b/include/sound/sof/stream.h @@ -83,10 +83,10 @@ struct sof_ipc_stream_params { uint16_t sample_valid_bytes; uint16_t sample_container_bytes; - /* for notifying host period has completed - 0 means no period IRQ */ uint32_t host_period_bytes; + uint16_t no_period_irq; /* 1 means period IRQ mode OFF */ - uint32_t reserved[2]; + uint16_t reserved[3]; uint16_t chmap[SOF_IPC_MAX_CHANNELS]; /**< channel map - SOF_CHMAP_ */ } __packed; -- 2.20.1

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[alsa-devel] [PATCH v2 00/23] ASoC: use devm_xxx
by Kuninori Morimoto 03 Jul '19

03 Jul '19

Hi Mark These are v2 patch of converting for devm_xxx(). v2 is caring about related empty functions. Kuninori Morimoto (23): ASoC: atmel: atmel-pcm-dma.c: use devm_snd_dmaengine_pcm_register() ASoC: sunxi: sun4i-i2s.c: use devm_snd_dmaengine_pcm_register() ASoC: tegra: tegra_pcm.c: use devm_snd_dmaengine_pcm_register() ASoC: ux500: ux500_pcm.c: use devm_snd_dmaengine_pcm_register() ASoC: au1x: i2sc.c: use devm_snd_soc_register_component() ASoC: au1x: ac97c.c: use devm_snd_soc_register_component() ASoC: au1x: psc-ac97.c: use devm_snd_soc_register_component() ASoC: au1x: psc-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-i2s.c: use devm_snd_soc_register_component() ASoC: cirrus: ep93xx-ac97.c: use devm_snd_soc_register_component() ASoC: rt5682: use devm_snd_soc_register_component() ASoC: ak4118: use devm_snd_soc_register_component() ASoC: cros_ec_codec: use devm_snd_soc_register_component() ASoC: fsl: imx-ssi.c: use devm_snd_soc_register_component() ASoC: fsl: mpc5200_psc_i2s.c: use devm_snd_soc_register_component() ASoC: fsl: mpc5200_psc_ac97.c: use devm_snd_soc_register_component() ASoC: pxa: pxa2xx-ac97.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_i2s.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_spdif.c: use devm_snd_soc_register_component() ASoC: tegra: tegra20_ac97.c: use devm_snd_soc_register_component() ASoC: tegra: tegra30_i2s.c: use devm_snd_soc_register_component() ASoC: ti: davinci-i2s.c: use devm_snd_soc_register_component() ASoC: ux500: ux500_msp_dai.c: use devm_snd_soc_register_component() sound/soc/atmel/atmel-pcm-dma.c | 9 ++------- sound/soc/atmel/atmel-pcm-pdc.c | 5 ----- sound/soc/atmel/atmel-pcm.h | 8 -------- sound/soc/atmel/atmel_ssc_dai.c | 11 ----------- sound/soc/au1x/ac97c.c | 4 +--- sound/soc/au1x/i2sc.c | 4 +--- sound/soc/au1x/psc-ac97.c | 6 ++---- sound/soc/au1x/psc-i2s.c | 6 ++---- sound/soc/cirrus/ep93xx-ac97.c | 10 +++------- sound/soc/cirrus/ep93xx-i2s.c | 7 ++----- sound/soc/codecs/ak4118.c | 11 ++--------- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/rt5682.c | 13 +++---------- sound/soc/fsl/imx-ssi.c | 8 ++------ sound/soc/fsl/mpc5200_psc_ac97.c | 3 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +-- sound/soc/pxa/pxa2xx-ac97.c | 3 +-- sound/soc/sunxi/sun4i-i2s.c | 4 +--- sound/soc/tegra/tegra20_ac97.c | 11 +++-------- sound/soc/tegra/tegra20_i2s.c | 11 +++-------- sound/soc/tegra/tegra20_spdif.c | 12 ++++-------- sound/soc/tegra/tegra30_i2s.c | 11 +++-------- sound/soc/tegra/tegra_pcm.c | 11 +++-------- sound/soc/tegra/tegra_pcm.h | 1 - sound/soc/ti/davinci-i2s.c | 8 ++------ sound/soc/ux500/ux500_msp_dai.c | 12 ++---------- sound/soc/ux500/ux500_pcm.c | 9 +-------- sound/soc/ux500/ux500_pcm.h | 1 - 28 files changed, 46 insertions(+), 158 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: line6: Fix write on zero-sized buffer
by Takashi Iwai 02 Jul '19

02 Jul '19

LINE6 drivers allocate the buffers based on the value returned from usb_maxpacket() calls. The manipulated device may return zero for this, and this results in the kmalloc() with zero size (and it may succeed) while the other part of the driver code writes the packet data with the fixed size -- which eventually overwrites. This patch adds a simple sanity check for the invalid buffer size for avoiding that problem. Reported-by: syzbot+219f00fb49874dcaea17(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/pcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 72c6f8e82a7e..78c2d6cab3b5 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -560,6 +560,11 @@ int line6_init_pcm(struct usb_line6 *line6, line6pcm->max_packet_size_out = usb_maxpacket(line6->usbdev, usb_sndisocpipe(line6->usbdev, ep_write), 1); + if (!line6pcm->max_packet_size_in || !line6pcm->max_packet_size_out) { + dev_err(line6pcm->line6->ifcdev, + "cannot get proper max packet size\n"); + return -EINVAL; + } spin_lock_init(&line6pcm->out.lock); spin_lock_init(&line6pcm->in.lock); -- 2.16.4

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[alsa-devel] [PATCH v6] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 02 Jul '19

02 Jul '19

The supported formats are S16_LE and S24_LE now. However, S24_LE is not supported when TDM is 0 and it is not in the right justified mode. We should remove 24-bit format in that situation to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- Changed the order of the conditional. Remove the snd_pcm_hw_constraint_msbits function. Use removing 24 bits format instead of fixing at 16 bits format. sound/soc/codecs/max98090.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..9fbb4c31bcf1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,24 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* + * When TDM = 0, remove 24-bit format support if it is not in right + * justified mode. + */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J && + !max98090->tdm_slots) + substream->runtime->hw.formats &= ~SNDRV_PCM_FMTBIT_S24_LE; + + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2334,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.410.gd8fdbe21b5-goog

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[alsa-devel] [PATCH -next] ASoC: madera: Remove duplicated include from cs47l35.c
by YueHaibing 02 Jul '19

02 Jul '19

Remove duplicated include. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs47l35.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 511d0d6fa962..e3585c1dab3d 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -19,7 +19,6 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/tlv.h> -#include <sound/tlv.h> #include <linux/irqchip/irq-madera.h> #include <linux/mfd/madera/core.h>

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