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

[PATCH 0/4] ASoC: Intel: boards: Remove ignore_suspend flag from SSP0 dai link
by Cezary Rojewski 06 Apr '20

06 Apr '20

As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Link to first message in conversation: https://lkml.org/lkml/2020/3/18/54 Cezary Rojewski (4): ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link sound/soc/intel/boards/bdw-rt5650.c | 1 - sound/soc/intel/boards/bdw-rt5677.c | 1 - sound/soc/intel/boards/broadwell.c | 1 - sound/soc/intel/boards/haswell.c | 1 - 4 files changed, 4 deletions(-) -- 2.17.1

5 22

[PATCH] quirk for Pioneer DJ DJM-250MK2 usb sound card
by František Kučera 03 Apr '20

03 Apr '20

Hello, I have created a quirk that adds support for the Pioneer DJ DJM-250MK2 mixer (acts like a USB sound card, but Vendor Specific Class). Output (from computer to card) is working: 8 channels, 48 000 Hz, S24_3LE. Input is not working. I tried, but I was unable to make it working even under an officially supported OS with Pioneer drivers - Audacity or Mixxx see the channels, but there is no signal. It probably works only with Pioneer proprietary application. But I hope that even mere output support will be useful for someone. (it is usable for playback, just not for DVS) Franta Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/quirks-table.h | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d187aa6d50db..dcaf9eed9a41 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3592,5 +3592,47 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-250MK2 + * PCM is 8 channels out @ 48 fixed (endpoints 0x01). + * The output from computer to the mixer is usable. + * + * The input (phono or line to computer) is not working. + * It should be at endpoint 0x82 and probably also 8 channels, + * but it seems that it works only with Pioneer proprietary software. + * Even on officially supported OS, the Audacity was unable to record + * and Mixxx to recognize the control vinyls. + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC

4 6

[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 02 Apr '20

02 Apr '20

This commit adds alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..344b06df0e15 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3565,8 +3565,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->i2s_dev->name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3587,9 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

3 4

[PATCH] pcm: add chmap option to route plugin
by Jonas Holmberg 31 Mar '20

31 Mar '20

Add a config definition "chmap" to override the channel maps in the same way as in the hw and null plugins. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_route.c | 57 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 47 insertions(+), 10 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index bbcc6118..d3e5f3ff 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -104,6 +104,7 @@ typedef struct { int schannels; snd_pcm_route_params_t params; snd_pcm_chmap_t *chmap; + snd_pcm_chmap_query_t **chmap_override; } snd_pcm_route_t; #endif /* DOC_HIDDEN */ @@ -441,6 +442,7 @@ static int snd_pcm_route_close(snd_pcm_t *pcm) free(params->dsts); } free(route->chmap); + snd_pcm_free_chmaps(route->chmap_override); return snd_pcm_generic_close(pcm); } @@ -634,6 +636,9 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) snd_pcm_chmap_t *map, *slave_map; unsigned int src, dst, nsrcs; + if (route->chmap_override) + return _snd_pcm_choose_fixed_chmap(pcm, route->chmap_override); + slave_map = snd_pcm_generic_get_chmap(pcm); if (!slave_map) return NULL; @@ -660,8 +665,14 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) static snd_pcm_chmap_query_t **snd_pcm_route_query_chmaps(snd_pcm_t *pcm) { + snd_pcm_route_t *route = pcm->private_data; snd_pcm_chmap_query_t **maps; - snd_pcm_chmap_t *map = snd_pcm_route_get_chmap(pcm); + snd_pcm_chmap_t *map; + + if (route->chmap_override) + return _snd_pcm_copy_chmap_query(route->chmap_override); + + map = snd_pcm_route_get_chmap(pcm); if (!map) return NULL; maps = _snd_pcm_make_single_query_chmaps(map); @@ -818,10 +829,10 @@ err: return -EINVAL; } -static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, +static int find_matching_chmap(snd_pcm_chmap_query_t **chmaps, + snd_pcm_chmap_t *tt_chmap, snd_pcm_chmap_t **found_chmap, int *schannels) { - snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(spcm); int i; *found_chmap = NULL; @@ -854,7 +865,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, int size = sizeof(snd_pcm_chmap_t) + c->channels * sizeof(unsigned int); *found_chmap = malloc(size); if (!*found_chmap) { - snd_pcm_free_chmaps(chmaps); return -ENOMEM; } memcpy(*found_chmap, c, size); @@ -863,8 +873,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, } } - snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { SNDERR("Found no matching channel map"); return -EINVAL; @@ -1252,6 +1260,7 @@ pcm.name { SCHANNEL REAL # route value (0.0 - 1.0) } } + [chmap MAP] # Override channel maps; MAP is a string array } \endcode @@ -1292,6 +1301,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, snd_pcm_route_ttable_entry_t *ttable = NULL; unsigned int csize, ssize; unsigned int cused, sused; + snd_pcm_chmap_query_t **chmaps = NULL; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; @@ -1306,31 +1316,45 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (strcmp(id, "ttable") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } tt = n; continue; } + if (strcmp(id, "chmap") == 0) { + chmaps = _snd_pcm_parse_config_chmaps(n); + if (!chmaps) { + SNDERR("Invalid channel map for %s", id); + return -EINVAL; + } + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } if (!slave) { SNDERR("slave is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } if (!tt) { SNDERR("ttable is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } err = snd_pcm_slave_conf(root, slave, &sconf, 2, SND_PCM_HW_PARAM_FORMAT, 0, &sformat, SND_PCM_HW_PARAM_CHANNELS, 0, &schannels); - if (err < 0) + if (err < 0) { + snd_pcm_free_chmaps(chmaps); return err; + } if (sformat != SND_PCM_FORMAT_UNKNOWN && snd_pcm_format_linear(sformat) != 1) { snd_config_delete(sconf); SNDERR("slave format is not linear"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } @@ -1345,13 +1369,19 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(tt_chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); return err; } if (tt_chmap) { - err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); + if (!chmaps) + chmaps = snd_pcm_query_chmaps(spcm); + if (chmaps) + err = find_matching_chmap(chmaps, tt_chmap, &chmap, + &schannels); free(tt_chmap); - if (err < 0) { + if (chmaps && err < 0) { + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1360,12 +1390,14 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } ttable = malloc(csize * ssize * sizeof(snd_pcm_route_ttable_entry_t)); if (ttable == NULL) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return -ENOMEM; } @@ -1374,6 +1406,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1385,9 +1418,13 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, free(ttable); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); } else { - ((snd_pcm_route_t*) (*pcmp)->private_data)->chmap = chmap; + snd_pcm_route_t *route = (*pcmp)->private_data; + + route->chmap = chmap; + route->chmap_override = chmaps; } return err; -- 2.24.1

2 1

[PATCH v2 1/3] dt-bindings: sound: convert rockchip spdif bindings to yaml
by Johan Jonker 31 Mar '20

31 Mar '20

Current dts files with 'spdif' nodes are manually verified. In order to automate this process rockchip-spdif.txt has to be converted to yaml. Also rk3188.dtsi, rk3288.dtsi use an extra fallback string, so change this in the documentation. Changed: "rockchip,rk3188-spdif", "rockchip,rk3066-spdif" "rockchip,rk3288-spdif", "rockchip,rk3066-spdif" Signed-off-by: Johan Jonker <jbx6244(a)gmail.com> --- Changed V2: dmas and dma-names layout --- .../devicetree/bindings/sound/rockchip-spdif.txt | 45 ----------- .../devicetree/bindings/sound/rockchip-spdif.yaml | 94 ++++++++++++++++++++++ 2 files changed, 94 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt deleted file mode 100644 index ec20c1271..000000000 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Rockchip SPDIF transceiver - -The S/PDIF audio block is a stereo transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. - -Required properties: - -- compatible: should be one of the following: - - "rockchip,rk3066-spdif" - - "rockchip,rk3188-spdif" - - "rockchip,rk3228-spdif" - - "rockchip,rk3288-spdif" - - "rockchip,rk3328-spdif" - - "rockchip,rk3366-spdif" - - "rockchip,rk3368-spdif" - - "rockchip,rk3399-spdif" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the SPDIF interrupt. -- dmas: DMA specifiers for tx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should be "tx" -- clocks: a list of phandle + clock-specifier pairs, one for each entry - in clock-names. -- clock-names: should contain following: - - "hclk": clock for SPDIF controller - - "mclk" : clock for SPDIF bus - -Required properties on RK3288: - - rockchip,grf: the phandle of the syscon node for the general register - file (GRF) - -Example for the rk3188 SPDIF controller: - -spdif: spdif@1011e000 { - compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; - reg = <0x1011e000 0x2000>; - interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&dmac1_s 8>; - dma-names = "tx"; - clock-names = "hclk", "mclk"; - clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>; - #sound-dai-cells = <0>; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml new file mode 100644 index 000000000..d1c72c8a5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml @@ -0,0 +1,94 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip SPDIF transceiver + +description: + The S/PDIF audio block is a stereo transceiver that allows the + processor to receive and transmit digital audio via a coaxial or + fibre cable. + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + oneOf: + - const: rockchip,rk3066-spdif + - const: rockchip,rk3228-spdif + - const: rockchip,rk3328-spdif + - const: rockchip,rk3366-spdif + - const: rockchip,rk3368-spdif + - const: rockchip,rk3399-spdif + - items: + - enum: + - rockchip,rk3188-spdif + - rockchip,rk3288-spdif + - const: rockchip,rk3066-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: clock for SPDIF bus + - description: clock for SPDIF controller + + clock-names: + items: + - const: mclk + - const: hclk + + dmas: + maxItems: 1 + + dma-names: + const: tx + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + Required property on RK3288. + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +if: + properties: + compatible: + contains: + const: rockchip,rk3288-spdif + +then: + required: + - rockchip,grf + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/rk3188-cru-common.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + spdif: spdif@1011e000 { + compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif"; + reg = <0x1011e000 0x2000>; + interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>; + clock-names = "mclk", "hclk"; + dmas = <&dmac1_s 8>; + dma-names = "tx"; + }; -- 2.11.0

4 9

[PATCH] ASoC: qcom: q6asm-dai: Add SNDRV_PCM_INFO_BATCH flag
by Stephan Gerhold 31 Mar '20

31 Mar '20

At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- I'm still quite confused about the true meaning of SNDRV_PCM_INFO_BATCH, so please correct me if I'm wrong :) The tsched=0 workaround can be found in Linaro distributions for QCOM devices for example: - https://git.linaro.org/ci/fai.git/commit/?id=63494268b654d80df033f4cdeccf8f… - https://github.com/ndechesne/meta-qcom/commit/7035dfeadd1c434fc7613730ac380… This patch allows removing that workaround since audio then works without any configuration changes. --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index f6c7cddf08e8..125af00bba53 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -78,7 +78,7 @@ struct q6asm_dai_data { }; static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -100,7 +100,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = { }; static struct snd_pcm_hardware q6asm_dai_hardware_playback = { - .info = (SNDRV_PCM_INFO_MMAP | + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | -- 2.26.0

5 4

[PATCH v2 1/3] ASoC: dapm: connect virtual mux with default value
by 이경택 31 Mar '20

31 Mar '20

Since a virtual mixer has no backing registers to decide which path to connect, it will try to match with initial state. This is to ensure that the default mixer choice will be correctly powered up during initialization. Invert flag is used to select initial state of the virtual switch. Since actual hardware can't be disconnected by virtual switch, connected is better choice as initial state in many cases. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dapm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9fb54e6fe254..17962564866d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -802,7 +802,13 @@ static void dapm_set_mixer_path_status(struct snd_soc_dapm_path *p, int i, val = max - val; p->connect = !!val; } else { - p->connect = 0; + /* since a virtual mixer has no backing registers to + * decide which path to connect, it will try to match + * with initial state. This is to ensure + * that the default mixer choice will be + * correctly powered up during initialization. + */ + p->connect = invert; } } -- 2.21.0

2 1

[PATCH 2/3] ASoC: topology: use name_prefix for new kcontrol
by 이경택 31 Mar '20

31 Mar '20

Current topology doesn't add prefix of the component to newly created kcontrol. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..a152409e8746 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -362,7 +362,7 @@ static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, struct snd_soc_component *comp = tplg->comp; return soc_tplg_add_dcontrol(comp->card->snd_card, - comp->dev, k, NULL, comp, kcontrol); + comp->dev, k, comp->name_prefix, comp, kcontrol); } /* remove a mixer kcontrol */ -- 2.21.0

2 1

[PATCH] pcm: add chmap option to route plugin
by jonashg＠axis.com 31 Mar '20

31 Mar '20

From: Jonas Holmberg <jonashg(a)axis.com> Add a config definition "chmap" to override the channel maps in the same way as in the hw and null plugins. Signed-off-by: Jonas Holmberg <jonashg(a)axis.com> --- src/pcm/pcm_route.c | 57 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 47 insertions(+), 10 deletions(-) diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index bbcc6118..a7b6552c 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -104,6 +104,7 @@ typedef struct { int schannels; snd_pcm_route_params_t params; snd_pcm_chmap_t *chmap; + snd_pcm_chmap_query_t **chmap_override; } snd_pcm_route_t; #endif /* DOC_HIDDEN */ @@ -441,6 +442,7 @@ static int snd_pcm_route_close(snd_pcm_t *pcm) free(params->dsts); } free(route->chmap); + snd_pcm_free_chmaps(route->chmap_override); return snd_pcm_generic_close(pcm); } @@ -634,6 +636,9 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) snd_pcm_chmap_t *map, *slave_map; unsigned int src, dst, nsrcs; + if (route->chmap_override) + return _snd_pcm_choose_fixed_chmap(pcm, route->chmap_override); + slave_map = snd_pcm_generic_get_chmap(pcm); if (!slave_map) return NULL; @@ -660,8 +665,14 @@ static snd_pcm_chmap_t *snd_pcm_route_get_chmap(snd_pcm_t *pcm) static snd_pcm_chmap_query_t **snd_pcm_route_query_chmaps(snd_pcm_t *pcm) { + snd_pcm_route_t *route = pcm->private_data; snd_pcm_chmap_query_t **maps; - snd_pcm_chmap_t *map = snd_pcm_route_get_chmap(pcm); + snd_pcm_chmap_t *map; + + if (route->chmap_override) + return _snd_pcm_copy_chmap_query(route->chmap_override); + + map = snd_pcm_route_get_chmap(pcm); if (!map) return NULL; maps = _snd_pcm_make_single_query_chmaps(map); @@ -818,10 +829,10 @@ err: return -EINVAL; } -static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, +static int find_matching_chmap(snd_pcm_chmap_query_t **chmaps, + snd_pcm_chmap_t *tt_chmap, snd_pcm_chmap_t **found_chmap, int *schannels) { - snd_pcm_chmap_query_t** chmaps = snd_pcm_query_chmaps(spcm); int i; *found_chmap = NULL; @@ -854,7 +865,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, int size = sizeof(snd_pcm_chmap_t) + c->channels * sizeof(unsigned int); *found_chmap = malloc(size); if (!*found_chmap) { - snd_pcm_free_chmaps(chmaps); return -ENOMEM; } memcpy(*found_chmap, c, size); @@ -863,8 +873,6 @@ static int find_matching_chmap(snd_pcm_t *spcm, snd_pcm_chmap_t *tt_chmap, } } - snd_pcm_free_chmaps(chmaps); - if (*found_chmap == NULL) { SNDERR("Found no matching channel map"); return -EINVAL; @@ -1252,6 +1260,7 @@ pcm.name { SCHANNEL REAL # route value (0.0 - 1.0) } } + [chmap MAP] # Override channel maps; MAP is a string array } \endcode @@ -1292,6 +1301,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, snd_pcm_route_ttable_entry_t *ttable = NULL; unsigned int csize, ssize; unsigned int cused, sused; + snd_pcm_chmap_query_t **chmaps = NULL; snd_config_for_each(i, next, conf) { snd_config_t *n = snd_config_iterator_entry(i); const char *id; @@ -1306,31 +1316,45 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (strcmp(id, "ttable") == 0) { if (snd_config_get_type(n) != SND_CONFIG_TYPE_COMPOUND) { SNDERR("Invalid type for %s", id); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } tt = n; continue; } + if (strcmp(id, "chmap") == 0) { + chmaps = _snd_pcm_parse_config_chmaps(n); + if (!chmaps) { + SNDERR("Invalid channel map for %s", id); + return -EINVAL; + } + continue; + } SNDERR("Unknown field %s", id); return -EINVAL; } if (!slave) { SNDERR("slave is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } if (!tt) { SNDERR("ttable is not defined"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } err = snd_pcm_slave_conf(root, slave, &sconf, 2, SND_PCM_HW_PARAM_FORMAT, 0, &sformat, SND_PCM_HW_PARAM_CHANNELS, 0, &schannels); - if (err < 0) + if (err < 0) { + snd_pcm_free_chmaps(chmaps); return err; + } if (sformat != SND_PCM_FORMAT_UNKNOWN && snd_pcm_format_linear(sformat) != 1) { snd_config_delete(sconf); SNDERR("slave format is not linear"); + snd_pcm_free_chmaps(chmaps); return -EINVAL; } @@ -1345,13 +1369,19 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(tt_chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); return err; } if (tt_chmap) { - err = find_matching_chmap(spcm, tt_chmap, &chmap, &schannels); + if (!chmaps) + chmaps = snd_pcm_query_chmaps(spcm); + if (chmaps) + err = find_matching_chmap(chmaps, tt_chmap, &chmap, + &schannels); free(tt_chmap); - if (err < 0) { + if (chmaps && err < 0) { + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1360,12 +1390,14 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, err = _snd_pcm_route_determine_ttable(tt, &csize, &ssize, chmap); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } ttable = malloc(csize * ssize * sizeof(snd_pcm_route_ttable_entry_t)); if (ttable == NULL) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return -ENOMEM; } @@ -1374,6 +1406,7 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, if (err < 0) { free(chmap); free(ttable); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); return err; } @@ -1385,9 +1418,13 @@ int _snd_pcm_route_open(snd_pcm_t **pcmp, const char *name, free(ttable); if (err < 0) { free(chmap); + snd_pcm_free_chmaps(chmaps); snd_pcm_close(spcm); } else { - ((snd_pcm_route_t*) (*pcmp)->private_data)->chmap = chmap; + snd_pcm_route_t *route = (*pcmp)->private_data; + + route->chmap = chmap; + route->chmap_override = chmaps; } return err; -- 2.24.1

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[PATCH] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 31 Mar '20

31 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 78def224f59c (ALSA: hda/realtek - Add Headset Mic supported) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

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