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

[alsa-devel] [PATCH v6 0/3] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 27 Aug '19

27 Aug '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver which is enough to get HDMI audio working on the A83T, A64, H3 and H5 platforms. I've dropped a lot of the functionality that was presented earlier in favour of getting initial HDMI audio delivered. H6 and multi-channel HDMI will follow shortly. My test branch for this can be found at https://github.com/codekipper/linux-sunxi/commits/upstream-i2s , I've been using a Pine64 to test with; validating the new SoC block with HDMI audio and ensuring that I've not broken the old block by making sure that the audio codec still works. BR, CK --- v6 changes compared to v5 are: - removed patches for multi-channel and H6 HDMI audio. - removed patch for 20, 24 and 32 bit (will push support for just 20 and 24bit) - ditched tdm patches as support has already been added. - added fix for A83T reg map. v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Marcus Cooper (3): ASoC: sun4i-i2s: incorrect regmap for A83T ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Adjust LRCLK width sound/soc/sunxi/sun4i-i2s.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) -- 2.23.0

4 12

[alsa-devel] [PATCH 00/21] ASoC: sun4i-i2s: Number of fixes and TDM Support
by Maxime Ripard 27 Aug '19

27 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> Hi, This series aims at fixing a number of issues in the current i2s driver, mostly related to the i2s master support and the A83t support. It also uses that occasion to cleanup a few things and simplify the driver. Finally, it builds on those fixes and cleanups to introduce TDM and DSP formats support. Let me know what you think, Maxime Marcus Cooper (1): ASoC: sun4i-i2s: Fix the MCLK and BCLK dividers on newer SoCs Maxime Ripard (20): ASoC: sun4i-i2s: Register regmap and PCM before our component ASoC: sun4i-i2s: Switch to devm for PCM register ASoC: sun4i-i2s: Replace call to params_channels by local variable ASoC: sun4i-i2s: Move the channel configuration to a callback ASoC: sun4i-i2s: Move the format configuration to a callback ASoC: sun4i-i2s: Rework MCLK divider calculation ASoC: sun4i-i2s: Don't use the oversample to calculate BCLK ASoC: sun4i-i2s: Use module clock as BCLK parent on newer SoCs ASoC: sun4i-i2s: RX and TX counter registers are swapped ASoC: sun4i-i2s: Use the actual format width instead of an hardcoded one ASoC: sun4i-i2s: Fix LRCK and BCLK polarity offsets on newer SoCs ASoC: sun4i-i2s: Fix the LRCK polarity ASoC: sun4i-i2s: Fix WSS and SR fields for the A83t ASoC: sun4i-i2s: Fix MCLK Enable bit offset on A83t ASoC: sun4i-i2s: Fix the LRCK period on A83t ASoC: sun4i-i2s: Remove duplicated quirks structure ASoC: sun4i-i2s: Pass the channels number as an argument ASoC: sun4i-i2s: Support more channels ASoC: sun4i-i2s: Add support for TDM slots ASoC: sun4i-i2s: Add support for DSP formats sound/soc/sunxi/sun4i-i2s.c | 660 ++++++++++++++++++++----------------- 1 file changed, 372 insertions(+), 288 deletions(-) base-commit: d45331b00ddb179e291766617259261c112db872 -- git-series 0.9.1

4 50

[alsa-devel] [PATCH 1/2] Revert "ASoC: sun4i-i2s: Remove duplicated quirks structure"
by Maxime Ripard 27 Aug '19

27 Aug '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> This reverts commit 3e9acd7ac6933cdc20c441bbf9a38ed9e42e1490. It turns out that while one I2S controller is described in the A83t datasheet, the driver supports another, undocumented, one that has been inherited from the older SoCs, while the documented one uses the new design. Fixes: 3e9acd7ac693 ("ASoC: sun4i-i2s: Remove duplicated quirks structure") Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- sound/soc/sunxi/sun4i-i2s.c | 25 ++++++++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 57bf2a33753e..a6a3f772fdf0 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1097,6 +1097,11 @@ static const struct sun4i_i2s_quirks sun6i_a31_i2s_quirks = { .set_fmt = sun4i_i2s_set_soc_fmt, }; +/* + * This doesn't describe the TDM controller documented in the A83t + * datasheet, but the three undocumented I2S controller that use the + * older design. + */ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1115,6 +1120,24 @@ static const struct sun4i_i2s_quirks sun8i_a83t_i2s_quirks = { .set_fmt = sun8i_i2s_set_soc_fmt, }; +static const struct sun4i_i2s_quirks sun8i_h3_i2s_quirks = { + .has_reset = true, + .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, + .sun4i_i2s_regmap = &sun8i_i2s_regmap_config, + .field_clkdiv_mclk_en = REG_FIELD(SUN4I_I2S_CLK_DIV_REG, 8, 8), + .field_fmt_wss = REG_FIELD(SUN4I_I2S_FMT0_REG, 0, 2), + .field_fmt_sr = REG_FIELD(SUN4I_I2S_FMT0_REG, 4, 6), + .bclk_dividers = sun8i_i2s_clk_div, + .num_bclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .mclk_dividers = sun8i_i2s_clk_div, + .num_mclk_dividers = ARRAY_SIZE(sun8i_i2s_clk_div), + .get_bclk_parent_rate = sun8i_i2s_get_bclk_parent_rate, + .get_sr = sun8i_i2s_get_sr_wss, + .get_wss = sun8i_i2s_get_sr_wss, + .set_chan_cfg = sun8i_i2s_set_chan_cfg, + .set_fmt = sun8i_i2s_set_soc_fmt, +}; + static const struct sun4i_i2s_quirks sun50i_a64_codec_i2s_quirks = { .has_reset = true, .reg_offset_txdata = SUN8I_I2S_FIFO_TX_REG, @@ -1296,7 +1319,7 @@ static const struct of_device_id sun4i_i2s_match[] = { }, { .compatible = "allwinner,sun8i-h3-i2s", - .data = &sun8i_a83t_i2s_quirks, + .data = &sun8i_h3_i2s_quirks, }, { .compatible = "allwinner,sun50i-a64-codec-i2s", -- 2.21.0

3 4

[alsa-devel] [PATCH 5.2 010/162] ASoC: samsung: odroid: fix a double-free issue for cpu_dai
by Greg Kroah-Hartman 27 Aug '19

27 Aug '19

[ Upstream commit 2abee12c0ab1924a69993d2c063a39a952e7d836 ] The cpu_dai variable is still being used after the of_node_put() call, which may result in double-free: of_node_put(cpu_dai); ---> released here ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { ... goto err_put_clk_i2s; --> jump to err_put_clk_i2s ... err_put_clk_i2s: clk_put(priv->clk_i2s_bus); err_put_sclk: clk_put(priv->sclk_i2s); err_put_cpu_dai: of_node_put(cpu_dai); --> double-free here Fixes: d832d2b246c5 ("ASoC: samsung: odroid: Fix of_node refcount unbalance") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-3-git-send-email-wen.yang99@zte.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/samsung/odroid.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 95c35e3ff3303..d606e48fe551a 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -299,7 +299,6 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = PTR_ERR(priv->clk_i2s_bus); goto err_put_sclk; } - of_node_put(cpu_dai); ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { @@ -307,6 +306,7 @@ static int odroid_audio_probe(struct platform_device *pdev) goto err_put_clk_i2s; } + of_node_put(cpu_dai); of_node_put(codec); return 0; -- 2.20.1

1 0

[alsa-devel] [PATCH 5.2 009/162] ASoC: samsung: odroid: fix an use-after-free issue for codec
by Greg Kroah-Hartman 27 Aug '19

27 Aug '19

[ Upstream commit 9b6d104a6b150bd4d3e5b039340e1f6b20c2e3c1 ] The codec variable is still being used after the of_node_put() call, which may result in use-after-free. Fixes: bc3cf17b575a ("ASoC: samsung: odroid: Add support for secondary CPU DAI") Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Krzysztof Kozlowski <krzk(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Link: https://lore.kernel.org/r/1562989575-33785-2-git-send-email-wen.yang99@zte.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/samsung/odroid.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index e688169ff12ab..95c35e3ff3303 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -275,9 +275,8 @@ static int odroid_audio_probe(struct platform_device *pdev) } of_node_put(cpu); - of_node_put(codec); if (ret < 0) - return ret; + goto err_put_node; ret = snd_soc_of_get_dai_link_codecs(dev, codec, codec_link); if (ret < 0) @@ -308,6 +307,7 @@ static int odroid_audio_probe(struct platform_device *pdev) goto err_put_clk_i2s; } + of_node_put(codec); return 0; err_put_clk_i2s: @@ -317,6 +317,8 @@ err_put_sclk: err_put_cpu_dai: of_node_put(cpu_dai); snd_soc_of_put_dai_link_codecs(codec_link); +err_put_node: + of_node_put(codec); return ret; } -- 2.20.1

1 0

[alsa-devel] [PATCH 1/1] ASoC: topology: Fix variable overflow in split_format
by Marcin Pietraszko 27 Aug '19

27 Aug '19

The 1 which was 32-bit long, when left-shifted by more than 31 caused an overflow, which resulted in some of the formats being not supported. Signed-off-by: Marcin Pietraszko <marcin.pietraszko(a)linux.intel.com> --- src/topology/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 5f586dc..00a35b9 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -321,7 +321,7 @@ static int split_format(struct snd_soc_tplg_stream_caps *caps, char *str) return -EINVAL; } - caps->formats |= 1 << format; + caps->formats |= 1ull << format; s = strtok(NULL, ", "); i++; } -- 2.7.4

3 2

[alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby R3 to quirks for native DSD support
by ilya.pshonkin＠netforce.ua 26 Aug '19

26 Aug '19

From: Sudokamikaze <sudokamikaze(a)protonmail.com> This patch adds quirk VID/PID IDs for Hiby R3 portable DSD player DSD support Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> --- sound/usb/quirks.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..f90418149e4e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1588,6 +1588,12 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + + /* ESS Sabre based USB DACs */ + case USB_ID(0xc502, 0x0051): /* Hiby R3 */ + if (fp->altsetting == 4) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ -- 2.23.0

2 1

Re: [alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby R3 to quirks for native DSD support
by Takashi Iwai 26 Aug '19

26 Aug '19

On Mon, 26 Aug 2019 18:05:05 +0200, Sudo Kamikaze wrote: > > From: Sudokamikaze <sudokamikaze(a)protonmail.com> > > This patch adds quirk VID/PID IDs for Hiby R3 portable DSD player DSD support > > Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> > --- > sound/usb/quirks.c | 6 ++++++ > 1 file changed, 6 insertions(+) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 78858918cbc1..f90418149e4e 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -1588,6 +1588,12 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, > if (fp->altsetting == 2) > return SNDRV_PCM_FMTBIT_DSD_U32_BE; > break; > + > + /* ESS Sabre based USB DACs */ > + case USB_ID(0xc502, 0x0051): /* Hiby R3 */ > + if (fp->altsetting == 4) > + return SNDRV_PCM_FMTBIT_DSD_U32_BE; > + break; Is fp->dsd_raw set for your device? IOW, instead of adding an individual entry here, adding the vendor 0xc502 with fp->dsd_raw check like below would work? thanks, Takashi --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* ESS Sabre */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break;

1 0

[alsa-devel] [PATCH] ASoC: es8316: limit headphone mixer volume
by Katsuhiro Suzuki 26 Aug '19

26 Aug '19

This patch limits Headphone mixer volume to 4 from 7. Because output sound suddenly becomes very loudly with many noise if set volume over 4. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 8dfb5dbeebbf..bc4141e1eb7f 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -91,7 +91,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = { SOC_DOUBLE_TLV("Headphone Playback Volume", ES8316_CPHP_ICAL_VOL, 4, 0, 3, 1, hpout_vol_tlv), SOC_DOUBLE_TLV("Headphone Mixer Volume", ES8316_HPMIX_VOL, - 0, 4, 7, 0, hpmixer_gain_tlv), + 0, 4, 4, 0, hpmixer_gain_tlv), SOC_ENUM("Playback Polarity", dacpol), SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL, -- 2.23.0.rc1

3 6

[alsa-devel] [PATCH] ALSA: oxfw: fix to handle correct stream for PCM playback
by Takashi Sakamoto 26 Aug '19

26 Aug '19

When userspace application calls ioctl(2) to configure hardware for PCM playback substream, ALSA OXFW driver handles incoming AMDTP stream. In this case, outgoing AMDTP stream should be handled. This commit fixes the bug for v5.3-rc kernel. Fixes: 4f380d007052 ("ALSA: oxfw: configure packet format in pcm.hw_params callback") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9ea39348cdf5..7c6d1c277d4d 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -248,7 +248,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(hw_params); mutex_lock(&oxfw->mutex); - err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->tx_stream, + err = snd_oxfw_stream_reserve_duplex(oxfw, &oxfw->rx_stream, rate, channels); if (err >= 0) ++oxfw->substreams_count; -- 2.20.1

2 1