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

[alsa-devel] [PATCH] ASoC: xlnx: parse AES audio parameters
by Maruthi Srinivas Bayyavarapu 15 Jan '19

15 Jan '19

AES channel status carries various audio parameters. If channel status is detected, current patch extracts sample rate and bit depth parameters of the incoming stream during capture. Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu(a)xilinx.com> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 143 ++++++++++++++++++++++++++++++++++ 1 file changed, 143 insertions(+) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index d2194da..97177d3 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -13,6 +13,7 @@ #include <linux/of_irq.h> #include <linux/sizes.h> +#include <sound/asoundef.h> #include <sound/soc.h> #include <sound/pcm_params.h> @@ -63,6 +64,7 @@ #define PERIODS_MAX 6 #define PERIOD_BYTES_MIN 192 #define PERIOD_BYTES_MAX (50 * 1024) +#define XLNX_PARAM_UNKNOWN 0 enum bit_depth { BIT_DEPTH_8, @@ -117,6 +119,129 @@ static const struct snd_pcm_hardware xlnx_pcm_hardware = { .periods_max = PERIODS_MAX, }; +enum { + AES_TO_AES, + AES_TO_PCM, + PCM_TO_PCM, + PCM_TO_AES +}; + +static void xlnx_parse_aes_params(u32 chsts_reg1_val, u32 chsts_reg2_val, + struct device *dev) +{ + u32 padded, srate, bit_depth, status[2]; + + if (chsts_reg1_val & IEC958_AES0_PROFESSIONAL) { + status[0] = chsts_reg1_val & 0xff; + status[1] = (chsts_reg1_val >> 16) & 0xff; + + switch (status[0] & IEC958_AES0_PRO_FS) { + case IEC958_AES0_PRO_FS_44100: + srate = 44100; + break; + case IEC958_AES0_PRO_FS_48000: + srate = 48000; + break; + case IEC958_AES0_PRO_FS_32000: + srate = 32000; + break; + case IEC958_AES0_PRO_FS_NOTID: + default: + srate = XLNX_PARAM_UNKNOWN; + break; + } + + switch (status[1] & IEC958_AES2_PRO_SBITS) { + case IEC958_AES2_PRO_WORDLEN_NOTID: + case IEC958_AES2_PRO_SBITS_20: + padded = 0; + break; + case IEC958_AES2_PRO_SBITS_24: + padded = 4; + break; + default: + bit_depth = XLNX_PARAM_UNKNOWN; + goto log_params; + } + + switch (status[1] & IEC958_AES2_PRO_WORDLEN) { + case IEC958_AES2_PRO_WORDLEN_20_16: + bit_depth = 16 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_22_18: + bit_depth = 18 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_23_19: + bit_depth = 19 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_24_20: + bit_depth = 20 + padded; + break; + case IEC958_AES2_PRO_WORDLEN_NOTID: + default: + bit_depth = XLNX_PARAM_UNKNOWN; + break; + } + + } else { + status[0] = (chsts_reg1_val >> 24) & 0xff; + status[1] = chsts_reg2_val & 0xff; + + switch (status[0] & IEC958_AES3_CON_FS) { + case IEC958_AES3_CON_FS_44100: + srate = 44100; + break; + case IEC958_AES3_CON_FS_48000: + srate = 48000; + break; + case IEC958_AES3_CON_FS_32000: + srate = 32000; + break; + default: + srate = XLNX_PARAM_UNKNOWN; + break; + } + + if (status[1] & IEC958_AES4_CON_MAX_WORDLEN_24) + padded = 4; + else + padded = 0; + + switch (status[1] & IEC958_AES4_CON_WORDLEN) { + case IEC958_AES4_CON_WORDLEN_20_16: + bit_depth = 16 + padded; + break; + case IEC958_AES4_CON_WORDLEN_22_18: + bit_depth = 18 + padded; + break; + case IEC958_AES4_CON_WORDLEN_23_19: + bit_depth = 19 + padded; + break; + case IEC958_AES4_CON_WORDLEN_24_20: + bit_depth = 20 + padded; + break; + case IEC958_AES4_CON_WORDLEN_21_17: + bit_depth = 17 + padded; + break; + case IEC958_AES4_CON_WORDLEN_NOTID: + default: + bit_depth = XLNX_PARAM_UNKNOWN; + break; + } + } + +log_params: + if (srate != XLNX_PARAM_UNKNOWN) + dev_info(dev, "sample rate = %d\n", srate); + else + dev_info(dev, "sample rate = unknown\n"); + + if (bit_depth != XLNX_PARAM_UNKNOWN) + dev_info(dev, "bit_depth = %d\n", bit_depth); + else + dev_info(dev, "bit_depth = unknown\n"); +} + static int xlnx_formatter_pcm_reset(void __iomem *mmio_base) { u32 val, retries = 0; @@ -302,8 +427,12 @@ static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { u32 low, high, active_ch, val, bytes_per_ch, bits_per_sample; + u32 aes_reg1_val, aes_reg2_val; int status; u64 size; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, + DRV_NAME); struct snd_pcm_runtime *runtime = substream->runtime; struct xlnx_pcm_stream_param *stream_data = runtime->private_data; @@ -311,6 +440,20 @@ static int xlnx_formatter_pcm_hw_params(struct snd_pcm_substream *substream, if (active_ch > stream_data->ch_limit) return -EINVAL; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && + stream_data->xfer_mode == AES_TO_PCM) { + val = readl(stream_data->mmio + XLNX_AUD_STS); + if (val & AUD_STS_CH_STS_MASK) { + aes_reg1_val = readl(stream_data->mmio + + XLNX_AUD_CH_STS_START); + aes_reg2_val = readl(stream_data->mmio + + XLNX_AUD_CH_STS_START + 0x4); + + xlnx_parse_aes_params(aes_reg1_val, aes_reg2_val, + component->dev); + } + } + size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) -- 2.7.4

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[alsa-devel] Applied "ASoC: soc-core: fix init platform memory handling" to the asoc tree
by Mark Brown 15 Jan '19

15 Jan '19

The patch ASoC: soc-core: fix init platform memory handling 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 09ac6a817bd687e7f5dac00470262efdd72f9319 Mon Sep 17 00:00:00 2001 From: Curtis Malainey <cujomalainey(a)chromium.org> Date: Thu, 10 Jan 2019 16:21:04 -0800 Subject: [PATCH] ASoC: soc-core: fix init platform memory handling snd_soc_init_platform initializes pointers to snd_soc_dai_link which is statically allocated and it does this by devm_kzalloc. In the event of an EPROBE_DEFER the memory will be freed and the pointers are left dangling. snd_soc_init_platform sees the dangling pointers and assumes they are pointing to initialized memory and does not reallocate them on the second probe attempt which results in a use after free bug since devm has freed the memory from the first probe attempt. Since the intention for snd_soc_dai_link->platform is that it can be set statically by the machine driver we need to respect the pointer in the event we did not set it but still catch dangling pointers. The solution is to add a flag to track whether the pointer was dynamically allocated or not. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 11 ++++++----- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7..e665f111b0d2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -985,6 +985,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* + * This driver uses legacy platform naming. Set by the core, machine + * drivers should not modify this value. + */ + unsigned int legacy_platform:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0934b36645b3..cdcc417c94ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { + if (!platform || dai_link->legacy_platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); if (!platform) return -ENOMEM; - dai_link->platform = platform; - platform->name = dai_link->platform_name; - platform->of_node = dai_link->platform_of_node; - platform->dai_name = NULL; + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; } /* if there's no platform we match on the empty platform */ -- 2.20.1

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[alsa-devel] [PATCH] make huawei matebook X right speaker work
by Tomas Espeleta 14 Jan '19

14 Jan '19

Hard coded coefficients to make Huawuei Matebook X right speaker work. The Matebook X has a ALC298, please refer to bug 197801 on how these numbers were reverse engineered from the Windows driver The reversed engineered sequence represents a repeating pattern of verbs, and the only values that are changing periodically are written on indexes 0x23 and 0x25: 0x500, 0x23 0x400, VALUE1 0x500, 0x25 0x400, VALUE2 * fixes bug 197801 * skipped reading sequences (0x500 - 0xc00 sequences are ignored) * static values from reverse engineering are used Signed-off-by: Tomas Espeleta <tomas.espeleta(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 70 +++++++++++++++++++++++++++++++++-- 1 file changed, 67 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0b3e7a18ca78..ec3972d9e30d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3628,6 +3628,64 @@ static void alc269_x101_hp_automute_hook(struct hda_codec *codec, vref); } +struct hda_alc298_mbxinit { + unsigned char value_0x23; + unsigned char value_0x25; +}; + +void alc298_stereo_seq(struct hda_codec *codec, + struct hda_alc298_mbxinit *initval, int first) +{ + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x0); + alc_write_coef_idx(codec, 0x26, 0xb000); + + if (first) + snd_hda_codec_write(codec, 0x21, 0, 0xf09, 0x0); + + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x23, initval->value_0x23); + + if (initval->value_0x23 != 0x1e) + alc_write_coef_idx(codec, 0x25, initval->value_0x25); + + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, 0x4b0, 0x10); +} + +void alc298_fixup_huawei_mbx_stereo(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + // Initialization magic + static struct hda_alc298_mbxinit dac_init[] = { + {0xc, 0x0}, {0xd, 0x0}, {0xe, 0x0}, {0xf, 0x0}, + {0x10, 0x0}, {0x1a, 0x40}, {0x1b, 0x82}, {0x1c, 0x0}, + {0x1d, 0x0}, {0x1e, 0x0}, {0x1f, 0x0}, + {0x20, 0xc2}, {0x21, 0xc8}, {0x22, 0x26}, {0x23, 0x24}, + {0x27, 0xff}, {0x28, 0xff}, {0x29, 0xff}, {0x2a, 0x8f}, + {0x2b, 0x2}, {0x2c, 0x48}, {0x2d, 0x34}, {0x2e, 0x0}, + {0x2f, 0x0}, + {0x30, 0x0}, {0x31, 0x0}, {0x32, 0x0}, {0x33, 0x0}, + {0x34, 0x0}, {0x35, 0x1}, {0x36, 0x93}, {0x37, 0xc}, + {0x38, 0x0}, {0x39, 0x0}, {0x3a, 0xf8}, {0x38, 0x80}, + {} + }; + struct hda_alc298_mbxinit *seq = dac_init; + + // Start + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x0); + snd_hda_codec_write(codec, 0x6, 0, 0x73e, 0x80); + alc_write_coef_idx(codec, 0x26, 0xf000); + alc_write_coef_idx(codec, 0x22, 0x31); + alc_write_coef_idx(codec, 0x23, 0xb); + alc_write_coef_idx(codec, 0x25, 0x0); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x26); + snd_hda_codec_write(codec, 0x20, 0, 0x4b0, 0x10); + + for (; seq->value_0x23; seq++) + alc298_stereo_seq(codec, seq, seq == dac_init); +} + static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3638,7 +3696,6 @@ static void alc269_fixup_x101_headset_mic(struct hda_codec *codec, } } - /* update mute-LED according to the speaker mute state via mic VREF pin */ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) { @@ -5582,6 +5639,7 @@ enum { ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC298_FIXUP_HUAWEI_MBX_STEREO, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -5880,6 +5938,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_MIC_MUTE_LED }, + [ALC298_FIXUP_HUAWEI_MBX_STEREO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_huawei_mbx_stereo, + .chained = true, + .chain_id = ALC255_FIXUP_MIC_MUTE_LED + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -6778,8 +6842,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), - SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), - SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), + SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC298_FIXUP_HUAWEI_MBX_STEREO), + SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC298_FIXUP_HUAWEI_MBX_STEREO), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- 2.20.1

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[alsa-devel] Fwd: [Alsa-user] AlsaMixer v1.1.8 Intel HDA -8 master volume
by makepost＠firemail.cc 14 Jan '19

14 Jan '19

Hello alsa-devel, forwarding from the silent alsa-user, with clarification. Found out this is specific for alsa-utils' alsamixer/volume_mapping.c, in which Clang 7.0.1 warns about exp10 implicit declaration. Broke after removal of _GNU_SOURCE in v1.1.7 apparently.[1] Uclibc-ng has exp10 since 1.0.12 according to its release notes, and exp10 is likely behind _GNU_SOURCE like other non-standard features. Should the exp10 fallback macro lower in the file have been removed instead?[2] [1] http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=6e90e4fe8e0052… [2] http://git.alsa-project.org/?p=alsa-utils.git;a=blob;f=alsamixer/volume_map… -------- Original Message -------- Subject: [Alsa-user] AlsaMixer v1.1.8 Intel HDA -8 master volume Date: 2019-01-11 21:48 From: makepost(a)firemail.cc To: alsa-user(a)lists.sourceforge.net Hi, after upgrading alsa-lib and alsa-utils to 1.1.8, Master, Headphones and Speaker in AlsaMixer are stuck at -8 percentage. Adjusting attempts immediately set max volume, still showing -8. Muting and unmuting work fine. PCM control is not affected. Realtek ALC283, vanilla 4.20.0 kernel. Attaching codec info and amixer output. What other info should I post for debug? _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user

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[alsa-devel] [PATCH v3 0/5] Stripe control functionality
by Sameer Pujar 14 Jan '19

14 Jan '19

Background ========== Azalia Controller fetches command and audio data via DMA and send them to codec through SDO(Serial Data Out) lines. SDO is for outboung and it broadcasts to all codecs. There is atleast one SDO line present, but there can be multiple SDO lines supported for extended bandwidth. This is essential when a platform supports multiple hdmi/dp sinks and there is a requirement for higher resolution audio playback. In such cases simultaneous audio playback data can be striped across multiple SDOs. Global Capabilities(GCAP) Register indicates the capabilities of the controller. Bits 2:1 of GCAP can be read to know the number of supported SDO lines (below is from HD audio spec) 00: 1 SDO 01: 2 SDO 10: 4 SDO 11: Reserved Stripe control verb reports and controls the stripe capability of an Audio Output Converter. This verb needs to be implemented only for an audio output converter and only if the stripe bit of the Audio Widget Capabilities parameter is 1. Stripe control: get code(0xf24) and set code(0x724) Change log ========== v1-->v2: -------- Patch 1: "ALSA: hda: add verbs for stripe control" * no change Patch 2: "ALSA: hda: Add api to program stripe control bits" * no change Patch 3: "ALSA: hda: add register offset for stripe control" * added mask for stripe control programming Patch 4: "ALSA: hda: program stripe bits for controller" * used stripe control mask instead of hard coded value Patch 5: "ALSA: hda: program stripe control for codec" * added conditional check to know if striping is supported. If supported, then only stripe verb is implemented. v2-->v3: -------- Patch 1: "ALSA: hda: add verbs for stripe control" * no change Patch 2: "ALSA: hda: Add api to program stripe control bits" * added braces around stripe value calucation as per spec Patch 3: "ALSA: hda: add register offset for stripe control" * no change Patch 4: "ALSA: hda: program stripe bits for controller" * no change Patch 5: "ALSA: hda: program stripe control for codec" * no change =========== Sameer Pujar (5): ALSA: hda: add verbs for stripe control ALSA: hda: Add api to program stripe control bits ALSA: hda: add register offset for stripe control ALSA: hda: program stripe bits for controller ALSA: hda: program stripe control for codec include/sound/hda_register.h | 2 ++ include/sound/hda_verbs.h | 2 ++ include/sound/hdaudio.h | 3 +++ sound/hda/hdac_stream.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 10 +++++++++- 5 files changed, 56 insertions(+), 1 deletion(-) -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[alsa-devel] [PATCH 1/3] drm/i915/audio: Track temporary rpm wakerefs
by Chris Wilson 14 Jan '19

14 Jan '19

Track the temporary rpm wakerefs used within audio so that they may be marked as complete and the tracking cancelled upon release. Signed-off-by: Chris Wilson <chris(a)chris-wilson.co.uk> --- drivers/gpu/drm/i915/intel_audio.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index de26cd0a5497..92e27359c2e3 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -756,12 +756,13 @@ static void i915_audio_component_codec_wake_override(struct device *kdev, bool enable) { struct drm_i915_private *dev_priv = kdev_to_i915(kdev); + intel_wakeref_t wakeref; u32 tmp; if (!IS_GEN(dev_priv, 9)) return; - i915_audio_component_get_power(kdev); + wakeref = intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); /* * Enable/disable generating the codec wake signal, overriding the @@ -779,7 +780,7 @@ static void i915_audio_component_codec_wake_override(struct device *kdev, usleep_range(1000, 1500); } - i915_audio_component_put_power(kdev); + intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO, wakeref); } /* Get CDCLK in kHz */ @@ -850,12 +851,13 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, int port, struct i915_audio_component *acomp = dev_priv->audio_component; struct intel_encoder *encoder; struct intel_crtc *crtc; + intel_wakeref_t wakeref; int err = 0; if (!HAS_DDI(dev_priv)) return 0; - i915_audio_component_get_power(kdev); + wakeref = intel_display_power_get(dev_priv, POWER_DOMAIN_AUDIO); mutex_lock(&dev_priv->av_mutex); /* 1. get the pipe */ @@ -875,7 +877,7 @@ static int i915_audio_component_sync_audio_rate(struct device *kdev, int port, unlock: mutex_unlock(&dev_priv->av_mutex); - i915_audio_component_put_power(kdev); + intel_display_power_put(dev_priv, POWER_DOMAIN_AUDIO, wakeref); return err; } -- 2.20.1

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[alsa-devel] [PATCH v2 0/5] Stripe control functionality
by Sameer Pujar 14 Jan '19

14 Jan '19

Background ========== Azalia Controller fetches command and audio data via DMA and send them to codec through SDO(Serial Data Out) lines. SDO is for outboung and it broadcasts to all codecs. There is atleast one SDO line present, but there can be multiple SDO lines supported for extended bandwidth. This is essential when a platform supports multiple hdmi/dp sinks and there is a requirement for higher resolution audio playback. In such cases simultaneous audio playback data can be striped across multiple SDOs. Global Capabilities(GCAP) Register indicates the capabilities of the controller. Bits 2:1 of GCAP can be read to know the number of supported SDO lines (below is from HD audio spec) 00: 1 SDO 01: 2 SDO 10: 4 SDO 11: Reserved Stripe control verb reports and controls the stripe capability of an Audio Output Converter. This verb needs to be implemented only for an audio output converter and only if the stripe bit of the Audio Widget Capabilities parameter is 1. Stripe control: get code(0xf24) and set code(0x724) Change log ========== v1-->v2: -------- Patch 1: "ALSA: hda: add verbs for stripe control" * no change Patch 2: "ALSA: hda: Add api to program stripe control bits" * no change Patch 3: "ALSA: hda: add register offset for stripe control" * added mask for stripe control programming Patch 4: "ALSA: hda: program stripe bits for controller" * used stripe control mask instead of hard coded value Patch 5: "ALSA: hda: program stripe control for codec" * added conditional check to know if striping is supported. If supported, then only stripe verb is implemented. =========== Sameer Pujar (5): ALSA: hda: add verbs for stripe control ALSA: hda: Add api to program stripe control bits ALSA: hda: add register offset for stripe control ALSA: hda: program stripe bits for controller ALSA: hda: program stripe control for codec include/sound/hda_register.h | 2 ++ include/sound/hda_verbs.h | 2 ++ include/sound/hdaudio.h | 3 +++ sound/hda/hdac_stream.c | 40 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 10 +++++++++- 5 files changed, 56 insertions(+), 1 deletion(-) -- 2.7.4 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

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[alsa-devel] [PATCH 0/4] slimbus: minor fixes.
by Srinivas Kandagatla 14 Jan '19

14 Jan '19

Here are some minor fixes to slimbus core for issues found while testing wcd9335. First one is make the slimbus core match full device id, second one is to avoid using big lock in status callback function, third one is to use status function rather than directly invoking the status ops. finally last one is to fix missing spinlock init. Thanks, srini Srinivas Kandagatla (4): slimbus: core: match full device id slimbus: core: do not call device_status under a lock slimbus: core: use slim_device_update_status() to update status slimbus: core: add missing spin_lock_init on txn_lock drivers/slimbus/core.c | 45 +++++++++++++++++++++++------------------- 1 file changed, 25 insertions(+), 20 deletions(-) -- 2.20.1

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[alsa-devel] next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on bcm2837-rpi-3-b-32
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on bcm2837-rpi-3-b-32 Summary: Start: 727b373740d1 Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3c42c459b5145f216edd41 Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: bcm2837-rpi-3-b-32 CPU arch: arm Lab: lab-baylibre Config: bcm2835_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [727b373740d1d3c6d3273cfc6e81d5e9a229b12d] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 727b373740d1d3c6d3273cfc6e81d5e9a229b12d # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [52bb35a9520ac1746fb34687c5582695813aa00a] Merge remote-tracking branch 'char-misc.current/char-misc-linus' git bisect bad 52bb35a9520ac1746fb34687c5582695813aa00a # bad: [4baf5101d553aae944fa7dc7bd13772eb559d78b] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad 4baf5101d553aae944fa7dc7bd13772eb559d78b # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

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[alsa-devel] next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on sun8i-h3-libretech-all-h3-cc
by kernelci.org bot 14 Jan '19

14 Jan '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * next/pending-fixes boot bisection: v5.0-rc1-535-g727b373740d1 on sun8i-h3-libretech-all-h3-cc Summary: Start: 727b373740d1 Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' Details: https://kernelci.org/boot/id/5c3c423359b5145dde6edd37 Plain log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… HTML log: https://storage.kernelci.org//next/pending-fixes/v5.0-rc1-535-g727b373740d1… Result: 8780cf1142a5 ASoC: soc-core: defer card probe until all component is added to list Checks: revert: PASS verify: PASS Parameters: Tree: next URL: http://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git Branch: pending-fixes Target: sun8i-h3-libretech-all-h3-cc CPU arch: arm Lab: lab-baylibre Config: sunxi_defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit 8780cf1142a59568a3aa77959cbd76b2edb6fd81 Author: Ajit Pandey <ajitp(a)codeaurora.org> Date: Wed Jan 9 14:17:07 2019 +0530 ASoC: soc-core: defer card probe until all component is added to list DAI component probe is not called if it is not present in component list during sound card registration. Check if component is available in component list for platform and cpu dai before soundcard registration. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0462b3ec977a..eec92f17dd15 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,7 +1027,6 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; - /* * FIXME * @@ -1129,6 +1128,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if platform dai component is not added to + * component list. + */ + if (!soc_find_component(link->platform->of_node, link->platform->name)) + return -EPROBE_DEFER; + /* * CPU device may be specified by either name or OF node, but * can be left unspecified, and will be matched based on DAI @@ -1140,6 +1147,14 @@ static int soc_init_dai_link(struct snd_soc_card *card, link->name); return -EINVAL; } + + /* + * Defer card registartion if cpu dai component is not added to + * component list. + */ + if (!soc_find_component(link->cpu_of_node, link->cpu_name)) + return -EPROBE_DEFER; + /* * At least one of CPU DAI name or CPU device name/node must be * specified ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [6b529fb0a3eabf9c4cc3e94c11477250379ce6d8] Merge tag 'for-5.0-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/kdave/linux git bisect good 6b529fb0a3eabf9c4cc3e94c11477250379ce6d8 # bad: [727b373740d1d3c6d3273cfc6e81d5e9a229b12d] Merge remote-tracking branch 'drm-misc-fixes/for-linux-next-fixes' git bisect bad 727b373740d1d3c6d3273cfc6e81d5e9a229b12d # good: [34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d] Merge remote-tracking branch 'net/master' git bisect good 34aea5d2c46b2581ddf0b78a6119f85c1b3a8d1d # bad: [52bb35a9520ac1746fb34687c5582695813aa00a] Merge remote-tracking branch 'char-misc.current/char-misc-linus' git bisect bad 52bb35a9520ac1746fb34687c5582695813aa00a # bad: [4baf5101d553aae944fa7dc7bd13772eb559d78b] Merge remote-tracking branch 'sound-asoc-fixes/for-linus' git bisect bad 4baf5101d553aae944fa7dc7bd13772eb559d78b # good: [49ff71f99a8826eda5cee76b69f15600d1cbf4b2] Merge remote-tracking branch 'rdma-fixes/for-rc' git bisect good 49ff71f99a8826eda5cee76b69f15600d1cbf4b2 # good: [678e2b44c8e3fec3afc7202f1996a4500a50be93] ALSA: compress: prevent potential divide by zero bugs git bisect good 678e2b44c8e3fec3afc7202f1996a4500a50be93 # good: [667e9334fa64da2273e36ce131b05ac9e47c5769] ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode git bisect good 667e9334fa64da2273e36ce131b05ac9e47c5769 # bad: [239b8b34a856777e562373ae0de605536a7ccade] ASoC: Intel: Boards: move the codec PLL configuration to _init git bisect bad 239b8b34a856777e562373ae0de605536a7ccade # bad: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list git bisect bad 8780cf1142a59568a3aa77959cbd76b2edb6fd81 # good: [44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661] ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages git bisect good 44fabd8cdaaa3acb80ad2bb3b5c61ae2136af661 # first bad commit: [8780cf1142a59568a3aa77959cbd76b2edb6fd81] ASoC: soc-core: defer card probe until all component is added to list -------------------------------------------------------------------------------

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