June 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: simple-card-utils.c: share asoc_graph_parse_dai()
by Kuninori Morimoto 14 Jun '23

14 Jun '23

Current Audio Graph Card/Card2 implements asoc_simple_parse_dai() on each driver, but these are same function. This patch share it as asoc_graph_parse_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/simple_card_utils.h | 3 + sound/soc/generic/audio-graph-card.c | 107 +------------------------- sound/soc/generic/audio-graph-card2.c | 107 +------------------------- sound/soc/generic/simple-card-utils.c | 105 +++++++++++++++++++++++++ 4 files changed, 110 insertions(+), 212 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 0e46f985eeda..9daef37fe9a8 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -195,6 +195,9 @@ int asoc_simple_remove(struct platform_device *pdev); int asoc_graph_card_probe(struct snd_soc_card *card); int asoc_graph_is_ports0(struct device_node *port); +int asoc_graph_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + int *is_single_link); #ifdef DEBUG static inline void asoc_simple_debug_dai(struct asoc_simple_priv *priv, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 4e85536a1b26..c6e0f9132193 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -55,60 +55,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_get_dai_id(struct device_node *ep) -{ - struct device_node *node; - struct device_node *endpoint; - struct of_endpoint info; - int i, id; - const u32 *reg; - int ret; - - /* use driver specified DAI ID if exist */ - ret = snd_soc_get_dai_id(ep); - if (ret != -ENOTSUPP) - return ret; - - /* use endpoint/port reg if exist */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret == 0) { - /* - * Because it will count port/endpoint if it doesn't have "reg". - * But, we can't judge whether it has "no reg", or "reg = <0>" - * only of_graph_parse_endpoint(). - * We need to check "reg" property - */ - if (of_property_present(ep, "reg")) - return info.id; - - node = of_get_parent(ep); - reg = of_get_property(node, "reg", NULL); - of_node_put(node); - if (reg) - return info.port; - } - node = of_graph_get_port_parent(ep); - - /* - * Non HDMI sound case, counting port/endpoint on its DT - * is enough. Let's count it. - */ - i = 0; - id = -1; - for_each_endpoint_of_node(node, endpoint) { - if (endpoint == ep) - id = i; - i++; - } - - of_node_put(node); - - if (id < 0) - return -ENODEV; - - return id; -} - static bool soc_component_is_pcm(struct snd_soc_dai_link_component *dlc) { struct snd_soc_dai *dai = snd_soc_find_dai_with_mutex(dlc); @@ -120,57 +66,6 @@ static bool soc_component_is_pcm(struct snd_soc_dai_link_component *dlc) return false; } -static int asoc_simple_parse_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - int *is_single_link) -{ - struct device_node *node; - struct of_phandle_args args; - int ret; - - if (!ep) - return 0; - - node = of_graph_get_port_parent(ep); - - /* Get dai->name */ - args.np = node; - args.args[0] = graph_get_dai_id(ep); - args.args_count = (of_graph_get_endpoint_count(node) > 1); - - /* - * FIXME - * - * Here, dlc->dai_name is pointer to CPU/Codec DAI name. - * If user unbinded CPU or Codec driver, but not for Sound Card, - * dlc->dai_name is keeping unbinded CPU or Codec - * driver's pointer. - * - * If user re-bind CPU or Codec driver again, ALSA SoC will try - * to rebind Card via snd_soc_try_rebind_card(), but because of - * above reason, it might can't bind Sound Card. - * Because Sound Card is pointing to released dai_name pointer. - * - * To avoid this rebind Card issue, - * 1) It needs to alloc memory to keep dai_name eventhough - * CPU or Codec driver was unbinded, or - * 2) user need to rebind Sound Card everytime - * if he unbinded CPU or Codec. - */ - ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) { - of_node_put(node); - return ret; - } - - dlc->of_node = node; - - if (is_single_link) - *is_single_link = of_graph_get_endpoint_count(node) == 1; - - return 0; -} - static void graph_parse_convert(struct device *dev, struct device_node *ep, struct asoc_simple_data *adata) @@ -231,7 +126,7 @@ static int graph_parse_node(struct asoc_simple_priv *priv, graph_parse_mclk_fs(top, ep, dai_props); - ret = asoc_simple_parse_dai(ep, dlc, cpu); + ret = asoc_graph_parse_dai(ep, dlc, cpu); if (ret < 0) return ret; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 25aa79dd55b3..542c4a114940 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -353,111 +353,6 @@ static const struct snd_soc_ops graph_ops = { .hw_params = asoc_simple_hw_params, }; -static int graph_get_dai_id(struct device_node *ep) -{ - struct device_node *node; - struct device_node *endpoint; - struct of_endpoint info; - int i, id; - const u32 *reg; - int ret; - - /* use driver specified DAI ID if exist */ - ret = snd_soc_get_dai_id(ep); - if (ret != -ENOTSUPP) - return ret; - - /* use endpoint/port reg if exist */ - ret = of_graph_parse_endpoint(ep, &info); - if (ret == 0) { - /* - * Because it will count port/endpoint if it doesn't have "reg". - * But, we can't judge whether it has "no reg", or "reg = <0>" - * only of_graph_parse_endpoint(). - * We need to check "reg" property - */ - if (of_property_present(ep, "reg")) - return info.id; - - node = of_get_parent(ep); - reg = of_get_property(node, "reg", NULL); - of_node_put(node); - if (reg) - return info.port; - } - node = of_graph_get_port_parent(ep); - - /* - * Non HDMI sound case, counting port/endpoint on its DT - * is enough. Let's count it. - */ - i = 0; - id = -1; - for_each_endpoint_of_node(node, endpoint) { - if (endpoint == ep) - id = i; - i++; - } - - of_node_put(node); - - if (id < 0) - return -ENODEV; - - return id; -} - -static int asoc_simple_parse_dai(struct device_node *ep, - struct snd_soc_dai_link_component *dlc, - int *is_single_link) -{ - struct device_node *node; - struct of_phandle_args args; - int ret; - - if (!ep) - return 0; - - node = of_graph_get_port_parent(ep); - - /* Get dai->name */ - args.np = node; - args.args[0] = graph_get_dai_id(ep); - args.args_count = (of_graph_get_endpoint_count(node) > 1); - - /* - * FIXME - * - * Here, dlc->dai_name is pointer to CPU/Codec DAI name. - * If user unbinded CPU or Codec driver, but not for Sound Card, - * dlc->dai_name is keeping unbinded CPU or Codec - * driver's pointer. - * - * If user re-bind CPU or Codec driver again, ALSA SoC will try - * to rebind Card via snd_soc_try_rebind_card(), but because of - * above reason, it might can't bind Sound Card. - * Because Sound Card is pointing to released dai_name pointer. - * - * To avoid this rebind Card issue, - * 1) It needs to alloc memory to keep dai_name eventhough - * CPU or Codec driver was unbinded, or - * 2) user need to rebind Sound Card everytime - * if he unbinded CPU or Codec. - */ - ret = snd_soc_get_dai_name(&args, &dlc->dai_name); - if (ret < 0) { - of_node_put(node); - return ret; - } - - dlc->of_node = node; - - if (is_single_link) - *is_single_link = of_graph_get_endpoint_count(node) == 1; - - return 0; -} - static void graph_parse_convert(struct device_node *ep, struct simple_dai_props *props) { @@ -512,7 +407,7 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, graph_parse_mclk_fs(ep, dai_props); - ret = asoc_simple_parse_dai(ep, dlc, &is_single_links); + ret = asoc_graph_parse_dai(ep, dlc, &is_single_links); if (ret < 0) return ret; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index b5ac0f0d5e8e..6a3c9e4e1cfe 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1019,6 +1019,111 @@ int asoc_graph_is_ports0(struct device_node *np) } EXPORT_SYMBOL_GPL(asoc_graph_is_ports0); +static int graph_get_dai_id(struct device_node *ep) +{ + struct device_node *node; + struct device_node *endpoint; + struct of_endpoint info; + int i, id; + int ret; + + /* use driver specified DAI ID if exist */ + ret = snd_soc_get_dai_id(ep); + if (ret != -ENOTSUPP) + return ret; + + /* use endpoint/port reg if exist */ + ret = of_graph_parse_endpoint(ep, &info); + if (ret == 0) { + /* + * Because it will count port/endpoint if it doesn't have "reg". + * But, we can't judge whether it has "no reg", or "reg = <0>" + * only of_graph_parse_endpoint(). + * We need to check "reg" property + */ + if (of_property_present(ep, "reg")) + return info.id; + + node = of_get_parent(ep); + ret = of_property_present(node, "reg"); + of_node_put(node); + if (ret) + return info.port; + } + node = of_graph_get_port_parent(ep); + + /* + * Non HDMI sound case, counting port/endpoint on its DT + * is enough. Let's count it. + */ + i = 0; + id = -1; + for_each_endpoint_of_node(node, endpoint) { + if (endpoint == ep) + id = i; + i++; + } + + of_node_put(node); + + if (id < 0) + return -ENODEV; + + return id; +} + +int asoc_graph_parse_dai(struct device_node *ep, + struct snd_soc_dai_link_component *dlc, + int *is_single_link) +{ + struct device_node *node; + struct of_phandle_args args = {}; + int ret; + + if (!ep) + return 0; + + node = of_graph_get_port_parent(ep); + + /* Get dai->name */ + args.np = node; + args.args[0] = graph_get_dai_id(ep); + args.args_count = (of_graph_get_endpoint_count(node) > 1); + + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ + ret = snd_soc_get_dai_name(&args, &dlc->dai_name); + if (ret < 0) { + of_node_put(node); + return ret; + } + + dlc->of_node = node; + + if (is_single_link) + *is_single_link = of_graph_get_endpoint_count(node) == 1; + + return 0; +} +EXPORT_SYMBOL_GPL(asoc_graph_parse_dai); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); -- 2.25.1

2 1

[PATCH v2] sound: usb: quirks: add flag for HEM devices to enable native DSD playback
by Lukasz Tyl 14 Jun '23

14 Jun '23

This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for HEM devices which supports native DSD. Prior to this change Linux kernel was not enabling native DSD playback for HEM devices, and as a result, DSD audio was being converted to PCM "on the fly". HEM devices, when connected to the system, would only play audio in PCM format, even if the source material was in DSD format. With the addition of new VENDOR_FLG in the quircks.c file, the devices are now correctly recognized, and raw DSD data is transmitted to the device, allowing for native DSD playback. Signed-off-by: Lukasz Tyl <ltyl(a)hem-e.com> --- Changes in v2: - correct entry position (entry sorted by vendor id). --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3ecd1ba7fd4b..6cf55b7f7a04 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2191,6 +2191,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2ab6, /* T+A devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x3336, /* HEM devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3353, /* Khadas devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3842, /* EVGA */ -- 2.34.1

1 0

[PATCH] ASoC: SOF: amd: Add support for IPC with a reply_size set to zero
by Venkata Prasad Potturu 14 Jun '23

14 Jun '23

Add support for IPC tx_message with a reply_size set to zero, return zero when message reply_size is zero at acp_dsp_ipc_get_reply(). Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp-ipc.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index 749e856dc601..8a0fc635a997 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -130,6 +130,13 @@ static void acp_dsp_ipc_get_reply(struct snd_sof_dev *sdev) memcpy(msg->reply_data, &reply, sizeof(reply)); ret = reply.error; } else { + /* + * To support an IPC tx_message with a + * reply_size set to zero. + */ + if (!msg->reply_size) + goto out; + /* reply correct size ? */ if (reply.hdr.size != msg->reply_size && !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { -- 2.25.1

2 1

[PATCH] ASoC: dt-bindings: ti,tlv320aic3x: Add missing type for "gpio-reset"
by Rob Herring 14 Jun '23

14 Jun '23

"gpio-reset" may be deprecated, but it still needs a type. Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml index e8ca9f3369f8..206f6d61e362 100644 --- a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -61,6 +61,7 @@ properties: GPIO specification for the active low RESET input. gpio-reset: + $ref: /schemas/types.yaml#/definitions/uint32-matrix maxItems: 1 description: Deprecated, please use reset-gpios instead. -- 2.39.2

3 2

[PATCH v2] ALSA: compress: allow setting codec params after next track
by Srinivas Kandagatla 14 Jun '23

14 Jun '23

For gapless playback it is possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Or if DSP's like QDSP have abililty to switch decoders on single stream for each track, then this call could be used to set new codec parameters. Existing code does not allow to change this profile while doing gapless playback. Reuse existing SNDRV_COMPRESS_SET_PARAMS to set this new track params along some additional checks to enforce proper state machine. With this new changes now the user can call SNDRV_COMPRESS_SET_PARAMS anytime after setting next track and additional check in write should also ensure that params are set before writing new data. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1/RFC: - removed introduction of new IOCTL, as suggested. - update the state-machine doc. .../sound/designs/compress-offload.rst | 52 +++++++++---------- sound/core/compress_offload.c | 10 ++-- 2 files changed, 33 insertions(+), 29 deletions(-) diff --git a/Documentation/sound/designs/compress-offload.rst b/Documentation/sound/designs/compress-offload.rst index 935f325dbc77..205cadcabe70 100644 --- a/Documentation/sound/designs/compress-offload.rst +++ b/Documentation/sound/designs/compress-offload.rst @@ -256,32 +256,32 @@ Gapless Playback SM For Gapless, we move from running state to partial drain and back, along with setting of meta_data and signalling for next track :: - - +----------+ - compr_drain_notify() | | - +------------------------>| RUNNING | - | | | - | +----------+ - | | - | | - | | compr_next_track() - | | - | V - | +----------+ - | | | - | |NEXT_TRACK| - | | | - | +----------+ - | | - | | - | | compr_partial_drain() - | | - | V - | +----------+ - | | | - +------------------------ | PARTIAL_ | - | DRAIN | - +----------+ + +----------+ + compr_drain_notify() | | compr_set_params() iff next-track set. + +------------------------>| RUNNING |----------------------+ + | | | | + | +----------+ | + | | | + | | | + | | compr_next_track() | + | | V + | V | + | +----------+ | + | | | | + | |NEXT_TRACK| | + | | | | + | +----------+ | + | | | + | +----------------------------+ + | | + | | compr_partial_drain() + | | + | V + | +----------+ + | | | + +------------------------ | PARTIAL_ | + | DRAIN | + +----------+ Not supported ============= diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 243acad89fd3..fe67228e74b3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -294,6 +294,9 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: case SNDRV_PCM_STATE_RUNNING: + /* Make sure next track params are set before writing new data */ + if (stream->next_track) + return -EPERM; break; default: mutex_unlock(&stream->device->lock); @@ -589,7 +592,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) struct snd_compr_params *params; int retval; - if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN || + (stream->runtime->state == SNDRV_PCM_STATE_RUNNING && stream->next_track)) { /* * we should allow parameter change only when stream has been * opened not in other cases @@ -615,7 +619,8 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) stream->metadata_set = false; stream->next_track = false; - stream->runtime->state = SNDRV_PCM_STATE_SETUP; + if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) + stream->runtime->state = SNDRV_PCM_STATE_SETUP; } else { return -EPERM; } @@ -956,7 +961,6 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) return retval; } - stream->next_track = false; return snd_compress_wait_for_drain(stream); } -- 2.21.0

3 3

[PATCH 0/7] ASoC: mediatek: Add support for MT79xx SoC
by Maso Hunag 14 Jun '23

14 Jun '23

From: Maso Huang <maso.huang(a)mediatek.com> This series of patches adds support for MediaTek AFE of MT79xx SoC. Patches are based on broonie tree "for-next" branch. Maso Huang (7): ASoC: mediatek: mt79xx: add common header ASoC: mediatek: mt79xx: support audio clock control ASoC: mediatek: mt79xx: add platform driver ASoC: mediatek: mt79xx: support etdm in platform driver ASoC: mediatek: mt79xx: add machine driver with wm8960 ASoC: dt-bindings: mediatek,mt79xx-wm8960: add mt79xx-wm8960 document ASoC: dt-bindings: mediatek,mt79xx-afe: add audio afe document .../bindings/sound/mediatek,mt79xx-afe.yaml | 102 +++ .../sound/mediatek,mt79xx-wm8960.yaml | 53 ++ sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt79xx/Makefile | 10 + sound/soc/mediatek/mt79xx/mt79xx-afe-clk.c | 123 ++++ sound/soc/mediatek/mt79xx/mt79xx-afe-clk.h | 18 + sound/soc/mediatek/mt79xx/mt79xx-afe-common.h | 49 ++ sound/soc/mediatek/mt79xx/mt79xx-afe-pcm.c | 608 ++++++++++++++++++ sound/soc/mediatek/mt79xx/mt79xx-dai-etdm.c | 421 ++++++++++++ sound/soc/mediatek/mt79xx/mt79xx-reg.h | 206 ++++++ sound/soc/mediatek/mt79xx/mt79xx-wm8960.c | 185 ++++++ 12 files changed, 1796 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt79xx-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt79xx-wm8960.yaml create mode 100644 sound/soc/mediatek/mt79xx/Makefile create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-clk.c create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-clk.h create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-common.h create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-afe-pcm.c create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-dai-etdm.c create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-reg.h create mode 100644 sound/soc/mediatek/mt79xx/mt79xx-wm8960.c -- 2.18.0

5 13

[PATCH v3 0/3] Add Loongson I2S controller support
by YingKun Meng 14 Jun '23

14 Jun '23

Hi all, This patchset adds support for Loongson I2S controller, and introduce a ASoC machine driver for loongson platform. The Loongson I2S controller is available on Loongson 7a2000/2k2000 chips, works as a PCI device. It has two private DMA controllers used to playback and capture. Each DMA controller has one channel. The ASoC machine driver adds support for audio device which using loongson I2S controller to tranfser the audio data. The audio device uses "PRP0001" as its ACPI device ID, which provides a means to use the existing DT-compatible device identification in ACPI. Thanks. --- Changes v2 -> v3 - Patch 1 Split the driver into PCI driver and component driver. Move interrupt requesting from pcm_open to pcm_create. Use pcim_xxx() API to get iomap of I2S register. Add condition checking for the number of periods. Add set_fmt() callback for DAI ops. Add volatile registers for regmap of I2S. Remove redundant condition checking. Update comment style. - Patch 2 Remove redundant SoC DAI link. Update comment style. - Patch 3 No change. Changes v1 -> v2 - Patch 1 Replace pci_xxx() APIs with pcim_xxx() APIs. Fix build errors from lkp(a)intel.com. Minor changes in log printing. - Patch 2 Use ACPI bindings to reference I2S and codec nodes. - Patch 3 Add chip model restriction for title and description. Add 'required' option for sound-dai property. Yingkun Meng (3): ASoC: Add support for Loongson I2S controller ASoC: loongson: Add Loongson ASoC Sound Card Support ASoC: dt-bindings: Add support for Loongson audio card .../sound/loongson,ls-audio-card.yaml | 70 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/loongson/Kconfig | 26 ++ sound/soc/loongson/Makefile | 8 + sound/soc/loongson/loongson_card.c | 230 ++++++++++++ sound/soc/loongson/loongson_dma.c | 350 ++++++++++++++++++ sound/soc/loongson/loongson_dma.h | 16 + sound/soc/loongson/loongson_i2s.c | 269 ++++++++++++++ sound/soc/loongson/loongson_i2s.h | 71 ++++ sound/soc/loongson/loongson_i2s_pci.c | 171 +++++++++ 11 files changed, 1213 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml create mode 100644 sound/soc/loongson/Kconfig create mode 100644 sound/soc/loongson/Makefile create mode 100644 sound/soc/loongson/loongson_card.c create mode 100644 sound/soc/loongson/loongson_dma.c create mode 100644 sound/soc/loongson/loongson_dma.h create mode 100644 sound/soc/loongson/loongson_i2s.c create mode 100644 sound/soc/loongson/loongson_i2s.h create mode 100644 sound/soc/loongson/loongson_i2s_pci.c base-commit: a11e6515b019da62266b731ff20bc6863f00df4d -- 2.33.0

1 0

[PATCH] sound: usb: quirks: add flag for HEM devices to enable native DSD playback
by Lukasz Tyl 14 Jun '23

14 Jun '23

This commit adds new DEVICE_FLG with QUIRK_FLAG_DSD_RAW and Vendor Id for HEM devices which supports native DSD. Prior to this change Linux kernel was not enabling native DSD playback for HEM devices, and as a result, DSD audio was being converted to PCM "on the fly". HEM devices, when connected to the system, would only play audio in PCM format, even if the source material was in DSD format. With the addition of new VENDOR_FLG in the quircks.c file, the devices are now correctly recognized, and raw DSD data is transmitted to the device, allowing for native DSD playback. Signed-off-by: Lukasz Tyl <ltyl(a)hem-e.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3ecd1ba7fd4b..1414bf15ac21 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2197,6 +2197,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0xc502, /* HiBy devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x3336, /* HEM devices */ + QUIRK_FLAG_DSD_RAW), {} /* terminator */ }; -- 2.34.1

1 0

[ PATCH RESEND v2 3/3] ASoC: dt-bindings: Add support for Loongson audio card
by YingKun Meng 14 Jun '23

14 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The audio card uses loongson I2S controller present in 7axxx/2kxxx chips to transfer audio data. On loongson platform, the chip has only one I2S controller. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- v1 -> v2: * Add chip model restriction for title and description. * Add 'required' restriction for sound-dai property. .../sound/loongson,ls-audio-card.yaml | 70 +++++++++++++++++++ 1 file changed, 70 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml diff --git a/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml new file mode 100644 index 000000000000..61e8babed402 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/loongson,ls-audio-card.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/loongson,ls-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Loongson 7axxx/2kxxx ASoC audio sound card driver + +maintainers: + - Yingkun Meng <mengyingkun(a)loongson.cn> + +description: + The binding describes the sound card present in loongson + 7axxx/2kxxx platform. The sound card is an ASoC component + which uses Loongson I2S controller to transfer the audio data. + +properties: + compatible: + const: loongson,ls-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + mclk-fs: + $ref: simple-card.yaml#/definitions/mclk-fs + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + properties: + sound-dai: + maxItems: 1 + required: + - sound-dai + +required: + - compatible + - model + - mclk-fs + - cpu + - codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "loongson,ls-audio-card"; + model = "loongson-audio"; + mclk-fs = <512>; + + cpu { + sound-dai = <&i2s>; + }; + codec { + sound-dai = <&es8323>; + }; + }; -- 2.33.0

1 0

Re: [PATCH 7/7] ASoC: dt-bindings: mediatek,mt79xx-afe: add audio afe document
by Krzysztof Kozlowski 14 Jun '23

14 Jun '23

On 14/06/2023 11:19, Maso Huang (黃加竹) wrote: > On Wed, 2023-06-14 at 10:21 +0200, Krzysztof Kozlowski wrote: >> >> External email : Please do not click links or open attachments until >> you have verified the sender or the content. >> On 14/06/2023 09:37, Maso Huang (黃加竹) wrote: >>>>>> I already saw AFE, why it cannot be part of existing bindings? >>>> >>>> Can you answer this? >>>> >>> >>> Did you mean mtk-afe-pcm.txt? >>> If yes, I'll modify mtk-afe-pcm.txt to yaml format, and add mt7986 >> to >>> its compatible list. >>> >> >> No, I meant mediatek,mt8188-afe.yaml. >> >> Aren't you working on some old tree? If so, please don't... >> >> Best regards, >> Krzysztof >> > > Hi Krzysztof, > AFE is common name for our audio hardware, and the design might be > different for soc, like clock. > > And the design is the same for mt7981/mt7986/mt7988. > Is it better to create a new dtbinding file mediatk,mt7986-afe.yaml? Is it different? That was my question whether it can be part of existing bindings. Best regards, Krzysztof

1 0