June 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/7] ASoC: simple-card: add asoc_simple_card_convert_fixup/parse_routing
by Kuninori Morimoto 15 Jun '17

15 Jun '17

Hi Mark Simple/OF-Audio card are using same DT properties, but not using same functions for it. These patches adds such function on simple-card-utils and uses it from simple/simple-scu/audio/audio-scu cards. These are for convert_rate/convert_channels/routing Kuninori Morimoto (7): ASoC: simple-card-utils: add asoc_simple_card_convert_fixup() ASoC: simple-scu-card: use asoc_simple_card_convert_fixup() ASoC: audio-graph-scu-card: use asoc_simple_card_convert_fixup() ASoC: simple-card-utils: add asoc_simple_card_of_parse_routing() ASoC: simple-card: use asoc_simple_card_of_parse_routing() ASoC: simple-scu-card: use asoc_simple_card_of_parse_routing() ASoC: audio-graph-scu-card: use asoc_simple_card_of_parse_routing() include/sound/simple_card_utils.h | 14 ++++++++ sound/soc/generic/audio-graph-scu-card.c | 28 +++------------ sound/soc/generic/simple-card-utils.c | 62 ++++++++++++++++++++++++++++++++ sound/soc/generic/simple-card.c | 10 ++---- sound/soc/generic/simple-scu-card.c | 26 +++----------- 5 files changed, 88 insertions(+), 52 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/4] ASoC: rsnd: add PCM rule
by Kuninori Morimoto 15 Jun '17

15 Jun '17

Hi Mark These patches adds hw_constraints rule to Renesas sound driver. It supports TDM and Multi SSI features, possible Channel / Sampling Rate are depends on its situation. Kuninori Morimoto (4): ASoC: rsnd: rename "slots" related variable ASoC: rsnd: add rsnd_adg_clk_query() ASoC: rsnd: add rsnd_ssi_clk_query() ASoC: rsnd: add rsnd_soc_hw_rule/constraint() sound/soc/sh/rcar/adg.c | 53 ++++++------ sound/soc/sh/rcar/core.c | 208 ++++++++++++++++++++++++++++++++++++++++------- sound/soc/sh/rcar/dvc.c | 7 +- sound/soc/sh/rcar/rsnd.h | 27 ++++-- sound/soc/sh/rcar/ssi.c | 98 +++++++++++++--------- 5 files changed, 289 insertions(+), 104 deletions(-) -- 1.9.1

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[alsa-devel] Multiplexable codecs?
by Ricard Wanderlof 15 Jun '17

15 Jun '17

I have a usecase where there is a requirement to have two codecs which are multiplexed on the same I2S channel. (In fact one of the 'codecs' is actually a digital microphone, but it generates an I2S stream just the same). In the ALSA SoC framework I know there is support for having separate codecs on the right and left channels, and also for having different codecs for playback and capture. But is there any way in the framework that two codecs can be toggled between at runtime? Normally a machine driver would configure a single codec which essentially is statically allocated. Of course the whole driver could be removed and another one loaded, but the thing here is that it is only for capture that the codecs are multiplexed, for playback it's always the same codec, and I don't want to disrupt playback just because a change is required on the capture side. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [GIT PULL] KBL firmware update
by Vinod Koul 15 Jun '17

15 Jun '17

Hi, Please consider this pull request to get updated audio dsp fw for KBl platform. This version updates to 9.21.00.2042. The following changes since commit 37857004a430e96dc837db7f967b6d0279053de8: linux-firmware: add firmware image for Redpine 9113 chipset (2017-06-01 10:22:17 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git kbl for you to fetch changes up to cb8cdaaff39c1a453350bcd8c9fa4635661cc135: linux-firmware: intel: Update Kabylake audio firmware (2017-06-13 14:54:03 +0530) ---------------------------------------------------------------- Subhransu S. Prusty (1): linux-firmware: intel: Update Kabylake audio firmware WHENCE | 4 +++- intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v2042.bin | Bin 0 -> 238920 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_kbl_v2042.bin Thanks -- ~Vinod

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[alsa-devel] [PATCH] ASoC: rockchip: Fix an error handling in 'rockchip_i2s_probe'
by Christophe JAILLET 15 Jun '17

15 Jun '17

If this memory allocation fail, we must disable what has been enabled. Do not return immediately but go thrue the error handling path instead. Also use 'devm_kmemdup' instead of 'devm_kzalloc+memcpy' to simplify code. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/rockchip/rockchip_i2s.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af5e0da144f7..f3aa3dfe914b 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -643,12 +643,13 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_pm_disable; } - soc_dai = devm_kzalloc(&pdev->dev, + soc_dai = devm_kmemdup(&pdev->dev, &rockchip_i2s_dai sizeof(*soc_dai), GFP_KERNEL); - if (!soc_dai) - return -ENOMEM; + if (!soc_dai) { + err = -ENOMEM; + goto err_pm_disable; + } - memcpy(soc_dai, &rockchip_i2s_dai, sizeof(*soc_dai)); if (!of_property_read_u32(node, "rockchip,playback-channels", &val)) { if (val >= 2 && val <= 8) soc_dai->playback.channels_max = val; -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.
by Liam Girdwood 15 Jun '17

15 Jun '17

Add some DAPM widget types to better support the construction of DAPM graphs within DSPs Changes v1. o Added some documentation. o Split codec widget into encoder and decoder to avoid confusion with HW CODECS. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Documentation/sound/soc/dapm.rst | 18 ++++++++++++++++++ include/sound/soc-dapm.h | 7 +++++++ include/uapi/sound/asoc.h | 10 +++++++++- sound/soc/soc-topology.c | 8 ++++++++ 4 files changed, 42 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index a27f42befa4d..78d63f6a35fa 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -105,6 +105,24 @@ Pre Special PRE widget (exec before all others) Post Special POST widget (exec after all others) +Buffer + Inter widget audio data buffer within a DSP. +Pipeline + Collection of widgets within a DSP, can represent a sequence of DSP + widgets and their collective configuration. +Effect + Widget that performs an audio processing effect. +SRC + Sample Rate Converter within DSP or CODEC +ASRC + Asynchronous Sample Rate Converter within DSP or CODEC +Encoder + Widget that encodes audio data from one format (usually PCM) to another + usually more compressed format. +Decoder + Widget that decodes audio data from a compressed format to an + uncompressed format like PCM. + (Widgets are defined in include/sound/soc-dapm.h) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a466f4bdc835..68fea6d97199 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -510,6 +510,13 @@ enum snd_soc_dapm_type { snd_soc_dapm_dai_out, snd_soc_dapm_dai_link, /* link between two DAI structures */ snd_soc_dapm_kcontrol, /* Auto-disabled kcontrol */ + snd_soc_dapm_buffer, /* DSP/CODEC internal buffer */ + snd_soc_dapm_pipeline, /* DSP/CODEC internal pipeline */ + snd_soc_dapm_effect, /* DSP/CODEC effect component */ + snd_soc_dapm_src, /* DSP/CODEC SRC component */ + snd_soc_dapm_asrc, /* DSP/CODEC ASRC component */ + snd_soc_dapm_encoder, /* FW/SW audio encoder component */ + snd_soc_dapm_decoder, /* FW/SW audio decoder component */ }; enum snd_soc_dapm_subclass { diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 6702533c8bd8..8a4fdf3e7e2e 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -73,7 +73,15 @@ #define SND_SOC_TPLG_DAPM_DAI_IN 13 #define SND_SOC_TPLG_DAPM_DAI_OUT 14 #define SND_SOC_TPLG_DAPM_DAI_LINK 15 -#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DAI_LINK +#define SND_SOC_TPLG_DAPM_BUFFER 16 +#define SND_SOC_TPLG_DAPM_PIPELINE 17 +#define SND_SOC_TPLG_DAPM_EFFECT 18 +#define SND_SOC_TPLG_DAPM_SIGGEN 19 +#define SND_SOC_TPLG_DAPM_SRC 20 +#define SND_SOC_TPLG_DAPM_ASRC 21 +#define SND_SOC_TPLG_DAPM_ENCODER 22 +#define SND_SOC_TPLG_DAPM_DECODER 23 +#define SND_SOC_TPLG_DAPM_LAST SND_SOC_TPLG_DAPM_DECODER /* Header magic number and string sizes */ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 12e189701924..b82d2755c2ac 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -242,6 +242,14 @@ static const struct soc_tplg_map dapm_map[] = { {SND_SOC_TPLG_DAPM_DAI_IN, snd_soc_dapm_dai_in}, {SND_SOC_TPLG_DAPM_DAI_OUT, snd_soc_dapm_dai_out}, {SND_SOC_TPLG_DAPM_DAI_LINK, snd_soc_dapm_dai_link}, + {SND_SOC_TPLG_DAPM_BUFFER, snd_soc_dapm_buffer}, + {SND_SOC_TPLG_DAPM_PIPELINE, snd_soc_dapm_pipeline}, + {SND_SOC_TPLG_DAPM_EFFECT, snd_soc_dapm_effect}, + {SND_SOC_TPLG_DAPM_SIGGEN, snd_soc_dapm_siggen}, + {SND_SOC_TPLG_DAPM_SRC, snd_soc_dapm_src}, + {SND_SOC_TPLG_DAPM_ASRC, snd_soc_dapm_asrc}, + {SND_SOC_TPLG_DAPM_ENCODER, snd_soc_dapm_encoder}, + {SND_SOC_TPLG_DAPM_DECODER, snd_soc_dapm_decoder}, }; static int tplc_chan_get_reg(struct soc_tplg *tplg, -- 2.11.0

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[alsa-devel] [PATCH v4 0/3] ASoC: Intel: Add Kabylake machine driver for RT5663+RT5514+MAX98927
by Harsha Priya 14 Jun '17

14 Jun '17

This patch set adds a machine driver for Kabylake platform which has RT5663, RT5514 and MAX9827 codecs. It also adds the quirk data for the platform driver to load the correct machine driver. For this particular machine, * the RT5663 is a headset codec connected on SSP0 * the RT5514 is a dmic codec connected on SSP1 * the MAX98987 is a speaker codec connected on SSP1 Harsha Priya (3): ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927 ASoC: Intel: Convert all sst_codecs data definition to c99 style ASoC: Intel: Add Kabylake RT5663+RT5514+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 640 +++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 37 +- 4 files changed, 690 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c -- 1.9.1

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[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Add Kabylake machine driver for RT5663+RT5514+MAX98927
by Harsha Priya 14 Jun '17

14 Jun '17

This patch set adds a machine driver for Kabylake platform which has RT5663, RT5514 and MAX9827 codecs. It also adds the quirk data for the platform driver to load the correct machine driver. For this particular machine, * the RT5663 is a headset codec connected on SSP0 * the RT5514 is a dmic codec connected on SSP1 * the MAX98987 is a speaker codec connected on SSP1 Harsha Priya (3): ASoC: Intel: Add Kabylake machine driver for RT5514, RT5663 and MAX98927 ASoC: Intel: Convert all sst_codecs data defintion to c99 style ASoC: Intel: Add Kabylake RT5663+RT5514+MAX98927 machine driver entry sound/soc/intel/Kconfig | 15 + sound/soc/intel/boards/Makefile | 2 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 640 +++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 37 +- 4 files changed, 690 insertions(+), 4 deletions(-) create mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c -- 1.9.1

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[alsa-devel] [PATCH] ALSA: pcm: Don't treat NULL chmap as a fatal error
by Takashi Iwai 14 Jun '17

14 Jun '17

The standard PCM chmap helper callbacks treat the NULL info->chmap as a fatal error and spews the kernel warning with stack trace when CONFIG_SND_DEBUG is on. This was OK, originally it was supposed to be always static and non-NULL. But, as the recent addition of Intel LPE audio driver shows, the chmap content may vary dynamically, and it can be even NULL when disconnected. The user still sees the kernel warning unnecessarily. For clearing such a confusion, this patch simply removes the snd_BUG_ON() in each place, just returns an error without warning. Cc: <stable(a)vger.kernel.org> # v4.11+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5088d4b8db22..009e6c98754e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2492,7 +2492,7 @@ static int pcm_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_substream *substream; const struct snd_pcm_chmap_elem *map; - if (snd_BUG_ON(!info->chmap)) + if (!info->chmap) return -EINVAL; substream = snd_pcm_chmap_substream(info, idx); if (!substream) @@ -2524,7 +2524,7 @@ static int pcm_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int __user *dst; int c, count = 0; - if (snd_BUG_ON(!info->chmap)) + if (!info->chmap) return -EINVAL; if (size < 8) return -ENOMEM; -- 2.13.1

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[alsa-devel] [PATCH] topology: Add support for missing fields parser.
by Liam Girdwood 14 Jun '17

14 Jun '17

The alsa-lib topology parser is missing some fields for certain objects that are part of the ABI. This patch adds the missing fields to the parser. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/pcm.c | 183 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 183 insertions(+) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index daef20e..0f4deb4 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -383,6 +383,49 @@ int tplg_parse_stream_caps(snd_tplg_t *tplg, tplg_dbg("\t\t%s: %d\n", id, sc->channels_max); continue; } + + if (strcmp(id, "periods_min") == 0) { + sc->periods_min = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->periods_min); + continue; + } + + if (strcmp(id, "periods_max") == 0) { + sc->periods_max = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->periods_max); + continue; + } + + if (strcmp(id, "period_size_min") == 0) { + sc->period_size_min = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->period_size_min); + continue; + } + + if (strcmp(id, "period_size_max") == 0) { + sc->period_size_max = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->period_size_max); + continue; + } + + if (strcmp(id, "buffer_size_min") == 0) { + sc->buffer_size_min = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->buffer_size_min); + continue; + } + + if (strcmp(id, "buffer_size_max") == 0) { + sc->buffer_size_max = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->buffer_size_max); + continue; + } + + if (strcmp(id, "sig_bits") == 0) { + sc->sig_bits = atoi(val); + tplg_dbg("\t\t%s: %d\n", id, sc->sig_bits); + continue; + } + } return 0; @@ -572,6 +615,17 @@ int tplg_parse_pcm(snd_tplg_t *tplg, continue; } + if (strcmp(id, "compress") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (strcmp(val, "true") == 0) + pcm->compress = 1; + + tplg_dbg("\t%s: %s\n", id, val); + continue; + } + if (strcmp(id, "dai") == 0) { err = tplg_parse_compound(tplg, n, tplg_parse_fe_dai, elem); @@ -655,6 +709,26 @@ int tplg_parse_dai(snd_tplg_t *tplg, continue; } + if (strcmp(id, "playback") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + dai->playback = atoi(val); + tplg_dbg("\t%s: %d\n", id, dai->playback); + continue; + } + + + if (strcmp(id, "capture") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + dai->capture = atoi(val); + tplg_dbg("\t%s: %d\n", id, dai->capture); + continue; + } + + /* stream capabilities */ if (strcmp(id, "pcm") == 0) { err = tplg_parse_compound(tplg, n, @@ -997,6 +1071,23 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, continue; } + if (strcmp(id, "bclk_freq") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->bclk_rate = atoi(val); + continue; + } + + if (strcmp(id, "bclk_invert") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "true")) + hw_cfg->invert_bclk = true; + continue; + } + if (strcmp(id, "fsync") == 0) { if (snd_config_get_string(n, &val) < 0) return -EINVAL; @@ -1005,6 +1096,98 @@ int tplg_parse_hw_config(snd_tplg_t *tplg, snd_config_t *cfg, hw_cfg->fsync_master = true; continue; } + + if (strcmp(id, "fsync_invert") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "true")) + hw_cfg->invert_fsync = true; + continue; + } + + if (strcmp(id, "fsync_freq") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->fsync_rate = atoi(val); + continue; + } + + if (strcmp(id, "mclk_freq") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->mclk_rate = atoi(val); + continue; + } + + if (strcmp(id, "mclk") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "master")) + hw_cfg->mclk_direction = true; + continue; + } + + if (strcmp(id, "pm_gate_clocks") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + if (!strcmp(val, "true")) + hw_cfg->clock_gated = true; + continue; + } + + if (strcmp(id, "tdm_slots") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tdm_slots = atoi(val); + continue; + } + + if (strcmp(id, "tdm_slot_width") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tdm_slot_width = atoi(val); + continue; + } + + if (strcmp(id, "tx_slots") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tx_slots = atoi(val); + continue; + } + + if (strcmp(id, "rx_slots") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->rx_slots = atoi(val); + continue; + } + + if (strcmp(id, "tx_channels") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->tx_channels = atoi(val); + continue; + } + + if (strcmp(id, "rx_channels") == 0) { + if (snd_config_get_string(n, &val) < 0) + return -EINVAL; + + hw_cfg->rx_channels = atoi(val); + continue; + } + } return 0; -- 1.9.1

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