December 2024 - Alsa-devel - mailman.alsa-project.org

[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 15 Jan '25

15 Jan '25

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

3 9

[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 13 Jan '25

13 Jan '25

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

2 1

[PATCH] ASoC: rt5682: Support the ALC5682I-VE codec
by derek.fang＠realtek.com 07 Jan '25

07 Jan '25

From: Derek Fang <derek.fang(a)realtek.com> The ALC5682I-VD and ALC5682I-VE use the same I2C codec driver with different calibration settings. This patch aims to handle their differences. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index ff9e14fad0cd..a8820435d1e0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -186,6 +186,12 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return -ENODEV; } + regmap_read(rt5682->regmap, RT5682_INT_DEVICE_ID, &val); + if (val == 0x6956) { + dev_dbg(&i2c->dev, "ALC5682I-VE device\n"); + rt5682->ve_ic = true; + } + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aa163ec40862..b4d72fc4a44d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -395,6 +395,7 @@ bool rt5682_volatile_register(struct device *dev, unsigned int reg) case RT5682_4BTN_IL_CMD_1: case RT5682_AJD1_CTRL: case RT5682_HP_CALIB_CTRL_1: + case RT5682_INT_DEVICE_ID: case RT5682_DEVICE_ID: case RT5682_I2C_MODE: case RT5682_HP_CALIB_CTRL_10: @@ -419,6 +420,7 @@ bool rt5682_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5682_RESET: + case RT5682_INT_DEVICE_ID: case RT5682_VERSION_ID: case RT5682_VENDOR_ID: case RT5682_DEVICE_ID: @@ -3139,7 +3141,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x7005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); @@ -3168,7 +3173,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x2005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b2d9e87af259..de43a5d99403 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -22,6 +22,7 @@ /* Info */ #define RT5682_RESET 0x0000 +#define RT5682_INT_DEVICE_ID 0x00f9 #define RT5682_VERSION_ID 0x00fd #define RT5682_VENDOR_ID 0x00fe #define RT5682_DEVICE_ID 0x00ff @@ -1446,6 +1447,7 @@ struct rt5682_priv { bool hw_init; bool first_hw_init; bool is_sdw; + bool ve_ic; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; -- 2.34.1

2 1

[PATCH v1] ASoC: tas2781: Fix occasional calibration failture
by Shenghao Ding 07 Jan '25

07 Jan '25

The root cause is that TAS2781 internal register is not unlocked before writing data into TAS2781_PRM_TEST_57_REG. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index fb8cd2284fe8..728bf78ae71f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), - X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true), X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), -- 2.34.1

2 1

[PATCH RESEND] ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK
by Marian Postevca 07 Jan '25

07 Jan '25

For 48Mhz MCLK systems the calculation of the HW rate is broken, and will not produce even one sane rate. Since es83xx supports the option to halve MCLK, calculate also rates with MCLK/2. Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/codecs/es8316.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 61729e5b50a8..f508df01145b 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -39,7 +39,9 @@ struct es8316_priv { struct snd_soc_jack *jack; int irq; unsigned int sysclk; - unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)]; + /* ES83xx supports halving the MCLK so it supports twice as many rates + */ + unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2]; struct snd_pcm_hw_constraint_list sysclk_constraints; bool jd_inverted; }; @@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq % ratio == 0) es8316->allowed_rates[count++] = freq / ratio; + + /* We also check if the halved MCLK produces a valid rate + * since the codec supports halving the MCLK. + */ + if ((freq / ratio) % 2 == 0) + es8316->allowed_rates[count++] = freq / ratio / 2; } if (count) { -- 2.45.2

2 1

[PATCH] ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise
by Zhang Yi 07 Jan '25

07 Jan '25

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..b06eead7e0f6 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B)) + cur_button = SND_JACK_BTN_0; + else + cur_button = SND_JACK_BTN_1; break; case 0x27: /* button volume down */ @@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

2 1

[PATCH 0/5] soundwire: amd: code improvements and new platform
by Vijendar Mukunda 02 Jan '25

02 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (5): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: add soundwire host wake interrupt enable/disable sequence soundwire: amd: clear soundwire pme wake status drivers/soundwire/amd_manager.c | 102 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 27 ++++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 131 insertions(+) -- 2.34.1

2 7

Add Roland QUAD-CAPTURE USB 2.0 support
by GitHub pull_request - opened 31 Dec '24

31 Dec '24

alsa-project/alsa-ucm-conf pull request #494 was opened from bjornmp: Add Roland QUAD-CAPTURE USB 2.0 support Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-lib-1.2.13 is breaking dssi-1.1.1 because of missing header
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-lib issue #431 was opened from kobboi: What was working with 1.2.12 no longer isn't: ``` /usr/include/alsa/ump_msg.h:1119:65: error: unknown type name ‘size_t’ 1119 | int snd_ump_msg_sysex_expand(const uint32_t *ump, uint8_t *buf, size_t maxlen, | ^~~~~~ /usr/include/alsa/ump_msg.h:13:1: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? 12 | #include <stdint.h> +++ |+#include <stddef.h> 13 | /usr/include/alsa/ump_msg.h:1120:30: error: unknown type name ‘size_t’ 1120 | size_t *filled); | ^~~~~~ /usr/include/alsa/ump_msg.h:1120:30: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? ``` Can this be addressed? Happy holidays! Issue URL : https://github.com/alsa-project/alsa-lib/issues/431 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Ask snd_seq_set_client_midi_version
by Correo Alternativo 30 Dec '24

30 Dec '24

Hi, I'm trying out MIDI 2.0 and found a problem, when using snd_seq_set_client_midi_version With SND_SEQ_CLIENT_LEGACY_MIDI I get the following sequence of events when connecting a midi device (using the method snd_seq_event_input or snd_seq_ump_event_input) (timidity command:timidity --interface=A --buffer-fragments=2,8 --output-mode=s1l --sampling-freq=44100): SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START But with SND_SEQ_CLIENT_UMP_MIDI_1_0 and SND_SEQ_CLIENT_UMP_MIDI_2_0 I don't get the SND_SEQ_EVENT_PORT_START events, I only get the following: SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM I don't know if I'm asking this in the right place, what mistake am I making? Just by modifying that it stops working.

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