From 88c4aff39452d7f0ab59f13730182a43bc7257c6 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 10 Sep 2026 13:10:46 +0100 Subject: [PATCH 01/30] ASoC: wm_adsp: Firmware search progress log should not look like an error In wm_adsp_request_firmware_file() only log the "Failed to request FILENAME" message when there is a real error (not when the file is missing). Add a new debug message to log the sequence of filenames tried during the file search. People have enabled debug messages, seen the "Failed to request" messages that are only logging the normal file search sequence, and reported them as errors. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260910121047.1592541-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 90c24c4b318e..b8f3035af3ba 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -775,9 +775,12 @@ static int wm_adsp_request_firmware_file(struct wm_adsp *dsp, s++; } + adsp_dbg(dsp, "Try '%s'\n", fw->filename); ret = wm_adsp_firmware_request(&fw->firmware, fw->filename, cs_dsp->dev); if (ret < 0) { - adsp_dbg(dsp, "Failed to request '%s': %d\n", fw->filename, ret); + if (ret != -ENOENT) + adsp_dbg(dsp, "Failed to request '%s': %d\n", fw->filename, ret); + kfree(fw->filename); fw->filename = NULL; if (ret != -ENOENT) From d56fe35c1bce1f547eb600695f60c139a508ff6e Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Wed, 9 Sep 2026 16:54:49 +0800 Subject: [PATCH 02/30] ASoC: rt712-sdca: reconfigure PLL2 to fix calibration time-out Add pll2 reconfiguration sequence in order to fix calibration time-out issue and to support 24.576MHz MCLK on specific platforms. Signed-off-by: Jack Yu Link: https://patch.msgid.link/20260909085449.862350-1-jack.yu@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca-sdw.c | 8 +++ sound/soc/codecs/rt712-sdca.c | 86 +++++++++++++++++++++++++++++-- sound/soc/codecs/rt712-sdca.h | 18 +++++++ 3 files changed, 109 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index c50e74e20a88..edba0367d9ce 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -18,12 +18,16 @@ static bool rt712_sdca_readable_register(struct device *dev, unsigned int reg) { switch (reg) { + case 0x004d: case 0x201a ... 0x201f: case 0x2029 ... 0x202a: case 0x202d ... 0x2034: case 0x2230 ... 0x2232: case 0x2f01 ... 0x2f0a: case 0x2f35 ... 0x2f36: + case 0x2f3a: + case 0x2f3d: + case 0x2f41: case 0x2f50: case 0x2f54: case 0x2f58 ... 0x2f5d: @@ -48,6 +52,7 @@ static bool rt712_sdca_readable_register(struct device *dev, unsigned int reg) static bool rt712_sdca_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { + case 0x004d: case 0x201b: case 0x201c: case 0x201d: @@ -56,6 +61,9 @@ static bool rt712_sdca_volatile_register(struct device *dev, unsigned int reg) case 0x2230: case 0x2f01: case 0x2f35: + case 0x2f3a: + case 0x2f3d: + case 0x2f41: case 0x320c: case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT712_SDCA_ENT_GE49, RT712_SDCA_CTL_DETECTED_MODE, 0): case SDW_SDCA_CTL(FUNC_NUM_HID, RT712_SDCA_ENT_HID01, RT712_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... diff --git a/sound/soc/codecs/rt712-sdca.c b/sound/soc/codecs/rt712-sdca.c index eda87eb9ab66..38052cb19790 100644 --- a/sound/soc/codecs/rt712-sdca.c +++ b/sound/soc/codecs/rt712-sdca.c @@ -73,14 +73,57 @@ static int rt712_sdca_index_update_bits(struct rt712_sdca_priv *rt712, return rt712_sdca_index_write(rt712, nid, reg, tmp); } +static void rt712_sdca_clk_patch(struct rt712_sdca_priv *rt712) +{ + rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0000); + regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x03); + usleep_range(1000, 1100); + regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x02); + usleep_range(1000, 1100); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x0080, 0x0000); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x001f, 0x0017); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF3, 0x0010, 0x0000); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF1, 0x0081, 0x0001); + regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x03); + usleep_range(1000, 1100); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF1, 0x0081, 0x0081); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x0080, 0x0080); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF2, 0x001f, 0x0000); + regmap_update_bits(rt712->regmap, RT712_PLL2_CONF3, 0x0010, 0x0010); + usleep_range(1000, 1100); + rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0081); +} + +static void rt712_sdca_clk_patch2(struct rt712_sdca_priv *rt712) +{ + rt712_sdca_index_update_bits(rt712, RT712_VENDOR_REG, 0x49, 0x0800, + 0x0000); + rt712_sdca_index_update_bits(rt712, RT712_VENDOR_REG, 0x49, 0xf000, + 0x0000); + rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0000); + rt712_sdca_index_update_bits(rt712, RT712_VENDOR_ANALOG_CTL, 0x0c, 0xc000, + 0xc000); + rt712_sdca_index_update_bits(rt712, RT712_VENDOR_ANALOG_CTL, 0x00, 0xc000, + 0xc000); + rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0081); + regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x02); + usleep_range(1000, 1100); + regmap_write(rt712->regmap, RT712_SDW_ROOT_CLK, 0x03); + usleep_range(1000, 1100); + rt712_sdca_index_write(rt712, RT712_VENDOR_REG, 0x65, 0x0000); +} + static int rt712_sdca_calibration(struct rt712_sdca_priv *rt712) { unsigned int val, loop_rc = 0, loop_dc = 0; struct device *dev; struct regmap *regmap = rt712->regmap; + unsigned int clk_base; int chk_cnt = 100; int ret = 0; + regmap_read(rt712->regmap, RT712_SDW_ROOT_CLK, &clk_base); + mutex_lock(&rt712->calibrate_mutex); dev = regmap_get_device(regmap); @@ -109,8 +152,35 @@ static int rt712_sdca_calibration(struct rt712_sdca_priv *rt712) if (ret < 0) goto _cali_fail_; } - if (loop_dc == chk_cnt) - dev_err(dev, "%s, calibration time-out!\n", __func__); + + if (loop_dc == chk_cnt) { + if (clk_base == RT712_CLK_FREQ_24_576MHZ) { + rt712_sdca_clk_patch(rt712); + rt712_sdca_clk_patch2(rt712); + } + rt712_sdca_index_write(rt712, RT712_VENDOR_REG, RT712_FSM_CTL, 0x4100); + rt712_sdca_index_write(rt712, RT712_VENDOR_CALI, + RT712_DAC_DC_CALI_CTL1, 0x7883); + rt712_sdca_index_write(rt712, RT712_VENDOR_CALI, + RT712_DAC_DC_CALI_CTL1, 0xf893); + rt712_sdca_index_read(rt712, RT712_VENDOR_CALI, + RT712_DAC_DC_CALI_CTL1, &val); + + for (loop_dc = 0; loop_dc < chk_cnt && + (val & RT712_DAC_DC_CALI_TRIGGER); loop_dc++) { + usleep_range(10000, 11000); + ret = rt712_sdca_index_read(rt712, RT712_VENDOR_CALI, + RT712_DAC_DC_CALI_CTL1, &val); + + if (ret < 0) + goto _cali_fail_; + } + + if (loop_dc == chk_cnt) + dev_err(dev, "%s, calibration time-out!\n", __func__); + else + dev_dbg(dev, "%s, calibration success!\n", __func__); + } if (loop_dc == chk_cnt || loop_rc == chk_cnt) ret = -ETIMEDOUT; @@ -1759,9 +1829,13 @@ static void rt712_sdca_va_io_init(struct rt712_sdca_priv *rt712) static void rt712_sdca_vb_io_init(struct rt712_sdca_priv *rt712) { - int ret = 0; unsigned int jack_func_status, mic_func_status, amp_func_status; struct device *dev = &rt712->slave->dev; + unsigned int clk_base; + int ret = 0; + + regmap_read(rt712->regmap, RT712_SDW_ROOT_CLK, &clk_base); + dev_dbg(dev, "%s clk_base=%x", __func__, clk_base); regmap_read(rt712->regmap, SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT712_SDCA_ENT0, RT712_SDCA_CTL_FUNC_STATUS, 0), &jack_func_status); @@ -1773,6 +1847,12 @@ static void rt712_sdca_vb_io_init(struct rt712_sdca_priv *rt712) __func__, jack_func_status, mic_func_status, amp_func_status); rt712_sdca_index_write(rt712, RT712_VENDOR_REG, RT712_JD_CTL3, 0x7778); + + if (clk_base == RT712_CLK_FREQ_24_576MHZ) { + rt712_sdca_clk_patch(rt712); + rt712_sdca_clk_patch2(rt712); + } + /* DMIC */ if ((mic_func_status & FUNCTION_NEEDS_INITIALIZATION) || (!rt712->first_hw_init)) { rt712_sdca_index_write(rt712, RT712_VENDOR_HDA_CTL, RT712_DMIC2_FU_IT_FLOAT_CTL, 0x1526); diff --git a/sound/soc/codecs/rt712-sdca.h b/sound/soc/codecs/rt712-sdca.h index 46740281a5c1..6229fe341bb5 100644 --- a/sound/soc/codecs/rt712-sdca.h +++ b/sound/soc/codecs/rt712-sdca.h @@ -162,6 +162,16 @@ struct rt712_dmic_kctrl_priv { #define RT712_EAPD_HIGH 0x2 #define RT712_EAPD_LOW 0x0 +/* SDW clock root frequency */ +#define RT712_SDW_ROOT_CLK 0x004d +#define RT712_SDW_SCALE_CLK0 0x0062 +#define RT712_SDW_SCALE_CLK1 0x0072 + +/* PLL2 config */ +#define RT712_PLL2_CONF1 0x2f3a +#define RT712_PLL2_CONF2 0x2f3d +#define RT712_PLL2_CONF3 0x2f41 + /* RC Calibration register */ #define RT712_RC_CAL 0x3201 @@ -254,6 +264,14 @@ enum rt712_sdca_version { RT712_VB, }; +enum { + RT712_CLK_FREQ_19_2_MHZ = 1, + RT712_CLK_FREQ_24MHZ = 2, + RT712_CLK_FREQ_24_576MHZ = 3, + RT712_CLK_FREQ_22_5792MHZ = 4, +}; + + int rt712_sdca_io_init(struct device *dev, struct sdw_slave *slave); int rt712_sdca_init(struct device *dev, struct regmap *regmap, struct regmap *mbq_regmap, struct sdw_slave *slave); From 4d855d747521505b54457c96bc73577bf74b2374 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 10 Sep 2026 12:44:56 +0100 Subject: [PATCH 03/30] ASoC: Rename snd_soc_dai_link_ch_map.ch_mask to cpu_ch_mask Rename the ch_mask member of snd_soc_dai_link_ch_map to cpu_ch_mask, as that is what it is used for. The CPU and codec channel masks are not necessarily the same, and are quite likely different. SoundWire and I2S/TDM both support assigning different sample slots to each codec, so for example channel 0 on each codec could map to different channels at the CPU. So it's quite normal that the channel mask at the CPU end is different for each codec, but the codec channel masks are the same for each codec. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260910114500.1586637-2-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/sdw_utils/soc_sdw_utils.c | 2 +- sound/soc/soc-pcm.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f46b2bc2a022..94c9b75e27e3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -699,7 +699,7 @@ struct snd_soc_dai_link_component { struct snd_soc_dai_link_ch_map { unsigned int cpu; unsigned int codec; - unsigned int ch_mask; + unsigned int cpu_ch_mask; }; struct snd_soc_dai_link { diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index a66dcc02fb59..f0cebcbf6288 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1596,7 +1596,7 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream, * ASoC will set the corresponding channel numbers for each cpu dai. */ for_each_link_ch_maps(rtd->dai_link, i, ch_maps) - ch_maps->ch_mask = ch_mask << (i * step); + ch_maps->cpu_ch_mask = ch_mask << (i * step); return 0; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0e49290a8c90..cb64ced21149 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1264,7 +1264,7 @@ static int __soc_pcm_hw_params(struct snd_pcm_substream *substream, */ for_each_rtd_ch_maps(rtd, j, ch_maps) if (ch_maps->cpu == i) - ch_mask |= ch_maps->ch_mask; + ch_mask |= ch_maps->cpu_ch_mask; /* fixup cpu channel number */ if (ch_mask) From 88b14c0d0bab5c0f3e7c641f274e3c70210c0e36 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 10 Sep 2026 12:44:57 +0100 Subject: [PATCH 04/30] ASoC: Add codec_ch_mask to snd_soc_dai_link_ch_map Add a codec_ch_mask member to snd_soc_dai_link_ch_map. The CPU and codec channel masks are not necessarily the same, and are quite likely different. SoundWire and I2S/TDM both support assigning different sample slots to each codec, so for example channel 0 on each codec could map to different channels at the CPU. It is also possible for one TX channel to map to multiple RX channels. So it isn't _always_ safe to assume that the total number of set bits in the CPU ch_mask is the same as the total number of enabled channels on the codec. For example consider this mapping on a capture stream: CPU0 CODEC0 cpu_ch_mask = 0x03 CPU1 CODEC0 cpu_ch_mask = 0x03 This could be either four TX channels on the codec split across two receiving CPUs, or two TX channels on the codec duplicated to two CPUs. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260910114500.1586637-3-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 94c9b75e27e3..5afc34b147b5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -700,6 +700,7 @@ struct snd_soc_dai_link_ch_map { unsigned int cpu; unsigned int codec; unsigned int cpu_ch_mask; + unsigned int codec_ch_mask; }; struct snd_soc_dai_link { From 6b382bdfe26a2232091bf743e454e6794295783e Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 10 Sep 2026 12:44:58 +0100 Subject: [PATCH 05/30] ASoC: soc-pcm: Apply snd_soc_dai_link_ch_map.codec_ch_mask to codec params In __soc_pcm_hw_params() if there is a snd_soc_dai_link_ch_map with non-zero codec_ch_mask, use that channel mask to restrict which channels are enabled on the codec. But only if there isn't a TDM mask. It is possible that a snd_soc_dai_link_ch_map could include the same codec multiple times on different CPUs so the for_each_rtd_ch_maps() loop accumulates the channel masks for all entries of that codec. If a TDM mask was also set, it takes priority and is used instead of any possible snd_soc_dai_link_ch_map entries. (They cannot be ANDed together because the bit positions are indicating different things: TDM is a bit for each TDM slot, codec_ch_mask is a bit for each codec channel.) This fixes a problem of incorrect TX channels enabled on the codec when multiple codecs are aggregated on a single capture link. For example: - Two CPUs with six 4-channel codecs. - The machine driver chooses to assign one channel from each codec to one channel on the CPU - But the codec hw_params() would be passed a channel count of 6, which (a) is more channels than the codec has and (b) allows enabling channels that should not be driving the audio bus. Fixes: ac950278b087 ("ASoC: add N cpus to M codecs dai link support") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260910114500.1586637-4-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cb64ced21149..3137c091bdb8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1206,7 +1206,9 @@ static int __soc_pcm_hw_params(struct snd_pcm_substream *substream, goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { - unsigned int tdm_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream); + unsigned int ch_mask = snd_soc_dai_tdm_mask_get(codec_dai, substream->stream); + struct snd_soc_dai_link_ch_map *ch_maps; + int j; /* * Skip CODECs which don't support the current stream type, @@ -1228,9 +1230,15 @@ static int __soc_pcm_hw_params(struct snd_pcm_substream *substream, /* copy params for each codec */ tmp_params = *params; - /* fixup params based on TDM slot masks */ - if (tdm_mask) - soc_pcm_codec_params_fixup(&tmp_params, tdm_mask); + /* fixup params based on TDM or ch_map masks */ + if (!ch_mask) { + for_each_rtd_ch_maps(rtd, j, ch_maps) + if (ch_maps->codec == i) + ch_mask |= ch_maps->codec_ch_mask; + } + + if (ch_mask) + soc_pcm_codec_params_fixup(&tmp_params, ch_mask); ret = snd_soc_dai_hw_params(codec_dai, substream, &tmp_params); From 290845e151cd4e307cc1f25319583aaceb4eeb30 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 10 Sep 2026 12:44:59 +0100 Subject: [PATCH 06/30] ASoC: sdw_utils: Set snd_soc_dai_link_ch_map.codec_ch_mask for capture In asoc_sdw_hw_params() set the codec_ch_mask member of struct snd_soc_dai_link_ch_map for capture streams. ASoC will then pass the correct number of channels to each codec hw_params(). This prevents trying to enable more channels on the codec DP than have been allocated bitslots in the SoundWire frame, which would cause bus clash errors. In theory codec_ch_mask could also be set for playback streams, but for those the CPU is the only sender so there is no risk of bus clash. For playback streams codec_ch_mask is set to 0 to preserve the existing behavior and avoid introducing bugs. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260910114500.1586637-5-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index f0cebcbf6288..293574a6ccca 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1565,7 +1565,7 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai_link_ch_map *ch_maps; int ch = params_channels(params); - unsigned int ch_mask; + unsigned int cpu_ch_mask, codec_ch_mask; int num_codecs; int step; int i; @@ -1575,8 +1575,9 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream, /* Identical data will be sent to all codecs in playback */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ch_mask = GENMASK(ch - 1, 0); + cpu_ch_mask = GENMASK(ch - 1, 0); step = 0; + codec_ch_mask = 0; } else { num_codecs = rtd->dai_link->num_codecs; @@ -1586,17 +1587,24 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - ch_mask = GENMASK(ch / num_codecs - 1, 0); - step = hweight_long(ch_mask); + cpu_ch_mask = GENMASK(ch / num_codecs - 1, 0); + step = hweight_long(cpu_ch_mask); + codec_ch_mask = cpu_ch_mask; } /* * The captured data will be combined from each cpu DAI if the dai * link has more than one codec DAIs. Set codec channel mask and * ASoC will set the corresponding channel numbers for each cpu dai. + * + * sdw_stream_add_slave() assigns different payload offsets to each + * codec in a capture stream, so that the same channels on each + * codec map to different channels on the CPU. */ - for_each_link_ch_maps(rtd->dai_link, i, ch_maps) - ch_maps->cpu_ch_mask = ch_mask << (i * step); + for_each_link_ch_maps(rtd->dai_link, i, ch_maps) { + ch_maps->cpu_ch_mask = cpu_ch_mask << (i * step); + ch_maps->codec_ch_mask = codec_ch_mask; + } return 0; } From b5b00a57868b1eabdf90a29a51f0eb732c609f3a Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 10 Sep 2026 12:45:00 +0100 Subject: [PATCH 07/30] ASoC: sdw_utils: cs_amp: Delete bogus and incorrect capture channel fixup Delete the asoc_sdw_cs_spk_feedback_rtd_init(). This is not needed now that the ASoC bug it was working around has been fixed. And it was broken anyway because it didn't match the way the core SoundWire code mapped codec channels to frame bitslots. This code was added to avoid a problem where multiple codec DP outputs were mapped to the same SoundWire frame bit slot. This would allow a user to break the SoundWire bus just by enabling mixer outputs using ALSA controls. As no production system has used the capture stream, this workaround was of little consequence and the problem of conflicting DP mappings was not investigated. The ASoC bug that enabled too many channels on each codec has now been fixed. So this workaround can be completely deleted. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260910114500.1586637-6-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 2 -- sound/soc/sdw_utils/soc_sdw_cs_amp.c | 46 ---------------------------- sound/soc/sdw_utils/soc_sdw_utils.c | 4 --- 3 files changed, 52 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 9b28e9aef4f1..9fbb69b9052d 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -250,8 +250,6 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct asoc_sdw_codec_info *info, bool playback); -int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd, - struct snd_soc_dai *dai); int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix); /* MAXIM codec support */ diff --git a/sound/soc/sdw_utils/soc_sdw_cs_amp.c b/sound/soc/sdw_utils/soc_sdw_cs_amp.c index 325ab7230481..6e21ef8f87e2 100644 --- a/sound/soc/sdw_utils/soc_sdw_cs_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_cs_amp.c @@ -14,7 +14,6 @@ #include #include -#define CS_AMP_CHANNELS_PER_AMP 4 #define CS35L56_SPK_VOLUME_0DB 400 /* 0dB Max */ int asoc_sdw_cs35l56_volume_limit(struct snd_soc_card *card, const char *name_prefix) @@ -64,51 +63,6 @@ int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai } EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_rtd_init, "SND_SOC_SDW_UTILS"); -int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) -{ - const struct snd_soc_dai_link *dai_link = rtd->dai_link; - const struct snd_soc_dai_link_ch_map *ch_map; - const struct snd_soc_dai_link_component *codec_dlc; - struct snd_soc_dai *codec_dai; - u8 ch_slot[8] = {}; - unsigned int amps_per_bus, ch_per_amp, mask; - int i, ret; - - WARN_ON(dai_link->num_cpus > ARRAY_SIZE(ch_slot)); - - /* - * CS35L56 has 4 TX channels. When the capture is aggregated the - * same bus slots will be allocated to all the amps on a bus. Only - * one amp on that bus can be transmitting in each slot so divide - * the available 4 slots between all the amps on a bus. - */ - amps_per_bus = dai_link->num_codecs / dai_link->num_cpus; - if ((amps_per_bus == 0) || (amps_per_bus > CS_AMP_CHANNELS_PER_AMP)) { - dev_err(rtd->card->dev, "Illegal num_codecs:%u / num_cpus:%u\n", - dai_link->num_codecs, dai_link->num_cpus); - return -EINVAL; - } - - ch_per_amp = CS_AMP_CHANNELS_PER_AMP / amps_per_bus; - - for_each_rtd_ch_maps(rtd, i, ch_map) { - codec_dlc = snd_soc_link_to_codec(rtd->dai_link, i); - codec_dai = snd_soc_find_dai(codec_dlc); - mask = GENMASK(ch_per_amp - 1, 0) << ch_slot[ch_map->cpu]; - - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0, mask, 4, 32); - if (ret < 0) { - dev_err(rtd->card->dev, "Failed to set TDM slot:%d\n", ret); - return ret; - } - - ch_slot[ch_map->cpu] += ch_per_amp; - } - - return 0; -} -EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_feedback_rtd_init, "SND_SOC_SDW_UTILS"); - int asoc_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct asoc_sdw_codec_info *info, diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 293574a6ccca..d2eeef4931c6 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -811,7 +811,6 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs35l56-sdw1c", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .rtd_init = asoc_sdw_cs_spk_feedback_rtd_init, }, }, .dai_num = 2, @@ -840,7 +839,6 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs35l56-sdw1c", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .rtd_init = asoc_sdw_cs_spk_feedback_rtd_init, }, }, .dai_num = 2, @@ -869,7 +867,6 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs35l56-sdw1c", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .rtd_init = asoc_sdw_cs_spk_feedback_rtd_init, }, }, .dai_num = 2, @@ -898,7 +895,6 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs35l56-sdw1c", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .rtd_init = asoc_sdw_cs_spk_feedback_rtd_init, }, }, .dai_num = 2, From 576725ded009f09a28da19852f7edf62dbc5f94c Mon Sep 17 00:00:00 2001 From: Ai Chao Date: Fri, 11 Sep 2026 16:19:32 +0800 Subject: [PATCH 08/30] ASoC: Intel: sof_es8336: Add a quirk for Huawei Matebook B3-420 Add DMI entry for Huawei Matebook B3-420 (BDZ-WXX9) with HEADPHONE_GPIO and HEADSET_MIC1 quirks. Similar to Huawei Matebook D (BOD-WXX9). On the same machine,audio routing between speakers and headphones works correctly when running Windows with the Huawei audio driver. However, after reinstalling Linux, both the speakers and headphones output sound simultaneously,indicating that the amplifier enable GPIOs are not being toggled correctly to separate the two outputs. Signed-off-by: Ai Chao Link: https://patch.msgid.link/20260911081932.2605407-1-aichao@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 9b016136c639..f1e62c2e79fe 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -360,6 +360,15 @@ static const struct dmi_system_id sof_es8336_quirk_table[] = { .driver_data = (void *)(SOF_ES8336_HEADPHONE_GPIO | SOC_ES8336_HEADSET_MIC1) }, + { + .callback = sof_es8336_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HUAWEI"), + DMI_MATCH(DMI_PRODUCT_NAME, "NDZ-WXX9"), + }, + .driver_data = (void *)(SOF_ES8336_HEADPHONE_GPIO | + SOC_ES8336_HEADSET_MIC1) + }, {} }; From a5e22cba3549b3b9ca592a6bc62329c9b85ce285 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 16 Sep 2026 18:18:03 +0800 Subject: [PATCH 09/30] ASoC: rt721: Reset codec to fix abnormal sound The audio output may become abnormal after a warm reboot from Windows. Reset the codec once during hardware initialization to restore it to a known state and prevent the issue. Signed-off-by: Oder Chiou Link: https://patch.msgid.link/20260916101803.2301508-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt721-sdca-sdw.c | 3 +++ sound/soc/codecs/rt721-sdca.c | 17 +++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c index eae7d662efae..910583162d3e 100644 --- a/sound/soc/codecs/rt721-sdca-sdw.c +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -70,6 +70,7 @@ static bool rt721_sdca_mbq_readable_register(struct device *dev, unsigned int re case 0x0310100: case 0x2000000 ... 0x2000003: case 0x2000013: + case 0x2000026: case 0x200002c: case 0x200003c: case 0x2000046: @@ -142,6 +143,7 @@ static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int re case 0x200000d: case 0x2000019: case 0x2000020: + case 0x2000026: case 0x200002c: case 0x2000030: case 0x2000046: @@ -155,6 +157,7 @@ static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int re case 0x5810039: case 0x5b10018: case 0x5b10019: + case 0x6100006: return true; default: return false; diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index a9479d0e4941..738644018fb9 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -1497,6 +1497,15 @@ int rt721_sdca_init(struct device *dev, struct regmap *regmap, &soc_sdca_dev_rt721, rt721_sdca_dai, ARRAY_SIZE(rt721_sdca_dai)); } +static void rt721_sdca_reset(struct rt721_sdca_priv *rt721) +{ + rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_VENDOR_REG, + RT721_VD_HIDDEN_CTRL, RT721_HIDDEN_REG_SW_RESET, + RT721_HIDDEN_REG_SW_RESET); + rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_RESET_CTL, 0x1, 0x1); +} + int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave) { struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); @@ -1530,9 +1539,17 @@ int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave) } pm_runtime_get_noresume(&slave->dev); + + if (!rt721->first_hw_init) + rt721_sdca_reset(rt721); + rt721_sdca_dmic_preset(rt721); rt721_sdca_amp_preset(rt721); rt721_sdca_jack_preset(rt721); + + if (rt721->hs_jack && (!rt721->first_hw_init)) + rt721_sdca_jack_init(rt721); + if (rt721->first_hw_init) { regcache_cache_bypass(rt721->regmap, false); regcache_mark_dirty(rt721->regmap); From 11fc0048a6930f4fca44fe3bd16a0023e78846a2 Mon Sep 17 00:00:00 2001 From: Sasha Levin Date: Sun, 13 Sep 2026 13:31:32 -0400 Subject: [PATCH 10/30] ASoC: ux500: Parenthesize MSP_{RX,TX}_CLKPOL_BIT() arguments arm allmodconfig fails to build with gcc: In file included from sound/soc/ux500/ux500_msp_i2s.c:20: sound/soc/ux500/ux500_msp_i2s.h:151:38: error: suggest parentheses around arithmetic in operand of '^' [-Werror=parentheses] sound/soc/ux500/ux500_msp_i2s.c:204:21: note: in expansion of macro 'MSP_TX_CLKPOL_BIT' cc1: all warnings being treated as errors The macros never parenthesized their argument: #define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT) That went unnoticed while every caller passed a plain variable, but configure_protocol() now passes an XOR expression, which binds as "a ^ (b & MASK)" rather than "(a ^ b) & MASK", and gcc rightly complains. No functional change: tx_clk_pol and rx_clk_pol only ever hold MSP_FALLING_EDGE (0) or MSP_RISING_EDGE (1), and bclk_inverted is a bool, so masking before or after the XOR gives the same 0/1 result. Parenthesize the argument anyway - it fixes the build and stops the macros from silently mis-evaluating a future composite argument. Fixes: 9ccbacf5a012 ("ASoC: ux500: Validate MSP DAI configuration") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202609051547.G9SJp8UQ-lkp@intel.com/ Assisted-by: LLM Signed-off-by: Sasha Levin Reviewed-by: Linus Walleij Link: https://patch.msgid.link/20260913173132.1172003-1-sashal@kernel.org Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 2bf2699bdc49..c66ef455e138 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -147,8 +147,8 @@ enum msp_direction { #define RCKPOL_MASK BIT(0) #define TCKPOL_MASK BIT(0) #define SPICKM_MASK (BIT(1) | BIT(0)) -#define MSP_RX_CLKPOL_BIT(n) ((n & RCKPOL_MASK) << RCKPOL_SHIFT) -#define MSP_TX_CLKPOL_BIT(n) ((n & TCKPOL_MASK) << TCKPOL_SHIFT) +#define MSP_RX_CLKPOL_BIT(n) (((n) & RCKPOL_MASK) << RCKPOL_SHIFT) +#define MSP_TX_CLKPOL_BIT(n) (((n) & TCKPOL_MASK) << TCKPOL_SHIFT) #define P1ELEN_SHIFT 0 #define P1FLEN_SHIFT 3 From c17ae8c26eac16ad244daef44044d714f68a2ddc Mon Sep 17 00:00:00 2001 From: HyeongJun An Date: Tue, 15 Sep 2026 18:25:15 +0900 Subject: [PATCH 11/30] ASoC: hdmi-codec: Report a change when the channel status moves The put() callback of "IEC958 Playback Default" stores all 24 channel status bytes and then returns 0. The core notifies userspace only on a positive return, so a write that changes what the get() callback hands back is never announced, and a mixer holding the control open keeps showing the old value. Compare the stored bytes and return 1 when they move, the way snd_hda_spdif_default_put() does. The same shape is in img-spdif-out and uniperif_player. No board with this codec was to hand. The change is a comparison of driver state with no hardware behaviour in it, and mixer-test counts the missing notification as event_missing. Fixes: 7a8e1d44211e ("ASoC: hdmi-codec: Add iec958 controls") Signed-off-by: HyeongJun An Assisted-by: Claude:claude-opus-5 Link: https://patch.msgid.link/20260915092515.2638542-1-sammiee5311@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index bc2c22436ba6..7aa50c5bd3df 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -426,10 +426,14 @@ static int hdmi_codec_iec958_default_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + if (!memcmp(hcp->iec_status, ucontrol->value.iec958.status, + sizeof(hcp->iec_status))) + return 0; + memcpy(hcp->iec_status, ucontrol->value.iec958.status, sizeof(hcp->iec_status)); - return 0; + return 1; } static int hdmi_codec_iec958_mask_get(struct snd_kcontrol *kcontrol, From 03a5699a0a04309c597683967aaaf25d1e555ea2 Mon Sep 17 00:00:00 2001 From: Jiangshan Yi Date: Mon, 14 Sep 2026 18:47:12 +0800 Subject: [PATCH 12/30] ASoC: codecs: rt712-sdca-dmic: fix uninitialized stream_config->type stream_config is not initialized before being passed to sdw_stream_add_slave(). The type field may contain garbage and is later copied to stream->type by sdw_config_stream(). Zero-initialize stream_config so type defaults to SDW_STREAM_PCM. While at it, use snd_sdw_params_to_config() helper instead of open-coding the same logic. Fixes: 63a511284c9e ("ASoC: rt712-sdca: Add RT712 SDCA driver for Mic topology") Cc: stable@vger.kernel.org Signed-off-by: Jiangshan Yi Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20260914104712.379574-1-yijiangshan@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/codecs/rt712-sdca-dmic.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt712-sdca-dmic.c b/sound/soc/codecs/rt712-sdca-dmic.c index 8860d81134e7..a9f3aa4e143a 100644 --- a/sound/soc/codecs/rt712-sdca-dmic.c +++ b/sound/soc/codecs/rt712-sdca-dmic.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include "rt712-sdca.h" #include "rt712-sdca-dmic.h" @@ -632,10 +633,10 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct rt712_sdca_dmic_priv *rt712 = snd_soc_component_get_drvdata(component); - struct sdw_stream_config stream_config; + struct sdw_stream_config stream_config = {0}; struct sdw_port_config port_config; struct sdw_stream_runtime *sdw_stream; - int retval, num_channels; + int retval; unsigned int sampling_rate; dev_dbg(dai->dev, "%s %s", __func__, dai->name); @@ -647,13 +648,8 @@ static int rt712_sdca_dmic_hw_params(struct snd_pcm_substream *substream, if (!rt712->slave) return -EINVAL; - stream_config.frame_rate = params_rate(params); - stream_config.ch_count = params_channels(params); - stream_config.bps = snd_pcm_format_width(params_format(params)); - stream_config.direction = SDW_DATA_DIR_TX; - - num_channels = params_channels(params); - port_config.ch_mask = GENMASK(num_channels - 1, 0); + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, &stream_config, &port_config); port_config.num = 2; retval = sdw_stream_add_slave(rt712->slave, &stream_config, From 3482062c786ce4233f8ed3224d824184f53ec154 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 14 Sep 2026 13:26:11 +0100 Subject: [PATCH 13/30] ASoC: cs-amp-lib: Prevent NULL pointer if efi variable is zero length In cs_amp_alloc_get_efi_variable() the first call to cs_amp_get_efi_variable() might return EFI_SUCCESS if the variable exists with zero length. Trap this and return -ENOENT to prevent returning an unexpected NULL pointer. The first cs_amp_get_efi_variable() call was assumed to return EFI_BUFFER_TOO_SMALL if the variable existed, but if instead it returned EFI_SUCCESS this would be converted to 0 by cs_amp_convert_efi_status() and then be returned as a NULL pointer. Fixes: 00fd40bc7acec ("ASoC: cs-amp-lib: Support Dell SSIDExV2 UEFI variable") Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20260914122611.2783563-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/cs-amp-lib.c b/sound/soc/codecs/cs-amp-lib.c index 41a9a5b005c6..9bc19d2e1639 100644 --- a/sound/soc/codecs/cs-amp-lib.c +++ b/sound/soc/codecs/cs-amp-lib.c @@ -317,6 +317,8 @@ static void *cs_amp_alloc_get_efi_variable(efi_char16_t *name, unsigned long size = 0; status = cs_amp_get_efi_variable(name, guid, NULL, &size, NULL); + if (status == EFI_SUCCESS) + return ERR_PTR(-ENOENT); if (status != EFI_BUFFER_TOO_SMALL) return ERR_PTR(cs_amp_convert_efi_status(status)); From 29218a4d11a31a8157389bc2b9e62dd768d7ea42 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 10 Sep 2026 21:46:46 +0530 Subject: [PATCH 14/30] ASoC: amd: acp: bounds-check SoundWire link ID in machine drivers Add a bounds check in create_sdw_dailink() to validate that the SoundWire link ID derived from link_mask does not exceed the maximum supported by the platform. If the link ID is out of range or link_mask is zero, log an error and return -EINVAL to prevent accessing invalid CPU pin ID tables. Applied to both acp-sdw-sof-mach.c and acp-sdw-legacy-mach.c. Fixes: 6d8348ddc56e ("ASoC: amd: acp: refactor SoundWire machine driver code") Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20260910161728.1452808-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 10 ++++++++++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 9 +++++++++ 2 files changed, 19 insertions(+) diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c index 6eac42bac855..2ea226a195c3 100644 --- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -205,6 +205,16 @@ static int create_sdw_dailink(struct snd_soc_card *card, return -EINVAL; } + if (!soc_end->link_mask) { + dev_err(dev, "invalid zero link_mask\n"); + return -EINVAL; + } + if ((ffs(soc_end->link_mask) - 1) >= amd_ctx->max_sdw_links) { + dev_err(dev, "link_id %d exceeds max_sdw_links %d\n", + ffs(soc_end->link_mask) - 1, amd_ctx->max_sdw_links); + return -EINVAL; + } + switch (amd_ctx->acp_rev) { case ACP63_PCI_REV: ret = get_acp63_cpu_pin_id(ffs(soc_end->link_mask - 1), diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index a9cd1f335167..6c74e67b134f 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -121,6 +121,15 @@ static int create_sdw_dailink(struct snd_soc_card *card, return -EINVAL; } + if (!sof_end->link_mask) { + dev_err(dev, "invalid zero link_mask\n"); + return -EINVAL; + } + if ((ffs(sof_end->link_mask) - 1) >= amd_ctx->max_sdw_links) { + dev_err(dev, "link_id %d exceeds max_sdw_links %d\n", + ffs(sof_end->link_mask) - 1, amd_ctx->max_sdw_links); + return -EINVAL; + } switch (amd_ctx->acp_rev) { case ACP63_PCI_REV: ret = get_acp63_cpu_pin_id(ffs(sof_end->link_mask - 1), From 0b7d55d3a91200f2b1ed710f525a944b0a7d6369 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 10 Sep 2026 21:46:47 +0530 Subject: [PATCH 15/30] ASoC: amd: acp: refactor codec config count in SOF SoundWire machine driver num_devs was used both as the endpoint count and as the output for asoc_sdw_parse_sdw_endpoints(), which overwrites it with the codec configuration count. Introduce a separate num_confs variable to hold the codec conf count so the two values remain distinct across codec_conf allocation and card->num_configs assignment. Fixes: 6d8348ddc56e ("ASoC: amd: acp: refactor SoundWire machine driver code") Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20260910161728.1452808-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 6c74e67b134f..b7926967593f 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -286,6 +286,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) int num_devs = 0; int num_ends = 0; int num_aux = 0; + int num_confs; int num_links; int be_id = 0; int ret; @@ -296,6 +297,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) return ret; } + num_confs = num_ends; /* One per DAI link, worst case is a DAI link for every endpoint */ struct asoc_sdw_dailink *sof_dais __free(kfree) = kzalloc_objs(*sof_dais, num_ends); @@ -312,7 +314,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) if (!sof_aux) return -ENOMEM; - ret = asoc_sdw_parse_sdw_endpoints(dev, ctx, sof_aux, sof_dais, sof_ends, &num_devs); + ret = asoc_sdw_parse_sdw_endpoints(dev, ctx, sof_aux, sof_dais, sof_ends, &num_confs); if (ret < 0) return ret; @@ -324,7 +326,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) dev_dbg(dev, "sdw %d, dmic %d", sdw_be_num, dmic_num); - codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); + codec_conf = devm_kcalloc(dev, num_confs, sizeof(*codec_conf), GFP_KERNEL); if (!codec_conf) return -ENOMEM; @@ -335,7 +337,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) return -ENOMEM; card->codec_conf = codec_conf; - card->num_configs = num_devs; + card->num_configs = num_confs; card->dai_link = dai_links; card->num_links = num_links; card->aux_dev = sof_aux; From 27098aaf28b96ab4e6891709062c343566d4882b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 10 Sep 2026 21:46:48 +0530 Subject: [PATCH 16/30] ASoC: amd: acp: fix ffs() operator precedence for SoundWire link ID ffs(link_mask - 1) computes ffs on (link_mask - 1) instead of subtracting 1 from the result of ffs(link_mask). For a typical power-of-2 link_mask this returns the wrong link ID, causing cpu_pin_id lookup to select the incorrect SoundWire manager. Fix the operator precedence to ffs(link_mask) - 1 in both acp-sdw-sof-mach.c and acp-sdw-legacy-mach.c. Fixes: 6d8348ddc56e ("ASoC: amd: acp: refactor SoundWire machine driver code") Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20260910161728.1452808-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 4 ++-- sound/soc/amd/acp/acp-sdw-sof-mach.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-legacy-mach.c b/sound/soc/amd/acp/acp-sdw-legacy-mach.c index 2ea226a195c3..1a05d4288a46 100644 --- a/sound/soc/amd/acp/acp-sdw-legacy-mach.c +++ b/sound/soc/amd/acp/acp-sdw-legacy-mach.c @@ -217,7 +217,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, switch (amd_ctx->acp_rev) { case ACP63_PCI_REV: - ret = get_acp63_cpu_pin_id(ffs(soc_end->link_mask - 1), + ret = get_acp63_cpu_pin_id(ffs(soc_end->link_mask) - 1, *be_id, &cpu_pin_id, dev); if (ret) return ret; @@ -225,7 +225,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, case ACP70_PCI_REV: case ACP71_PCI_REV: case ACP72_PCI_REV: - ret = get_acp70_cpu_pin_id(ffs(soc_end->link_mask - 1), + ret = get_acp70_cpu_pin_id(ffs(soc_end->link_mask) - 1, *be_id, &cpu_pin_id, dev); if (ret) return ret; diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index b7926967593f..e6d545fd665e 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -132,7 +132,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, } switch (amd_ctx->acp_rev) { case ACP63_PCI_REV: - ret = get_acp63_cpu_pin_id(ffs(sof_end->link_mask - 1), + ret = get_acp63_cpu_pin_id(ffs(sof_end->link_mask) - 1, *be_id, &cpu_pin_id, dev); if (ret) return ret; @@ -140,7 +140,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, case ACP70_PCI_REV: case ACP71_PCI_REV: case ACP72_PCI_REV: - ret = get_acp70_cpu_pin_id(ffs(sof_end->link_mask - 1), + ret = get_acp70_cpu_pin_id(ffs(sof_end->link_mask) - 1, *be_id, &cpu_pin_id, dev); if (ret) return ret; From d57616f8be5601d210bbb0f677b9cb88a5186c3c Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 10 Sep 2026 21:46:49 +0530 Subject: [PATCH 17/30] ASoC: amd: acp: fix card name length warning in SOF SoundWire machine driver The ALSA snd_card driver[] field is 16 bytes (including the NUL terminator), leaving 15 usable characters. The SOF framework prepends a "sof-" prefix when registering the card, so card->name = "amd-soundwire" becomes driver name "sof-amd-soundwire" which is 17 characters and overflows the driver[16] buffer, triggering a kernel warning. Fix by shortening the card name to "amd-sdw"; the resulting driver name "sof-amd-sdw" fits within the 15-character limit. Signed-off-by: Vijendar Mukunda Reviewed-by: Mario Limonciello (AMD) Link: https://patch.msgid.link/20260910161728.1452808-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index e6d545fd665e..ec3e1f5f1052 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -390,7 +390,7 @@ static int mc_probe(struct platform_device *pdev) ctx->private = amd_ctx; card = &ctx->card; card->dev = &pdev->dev; - card->name = "amd-soundwire"; + card->name = "amd-sdw"; card->owner = THIS_MODULE; card->late_probe = asoc_sdw_card_late_probe; From 0030f62683d5061d43b80577b7ab27196f1adb4c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= Date: Mon, 14 Sep 2026 12:12:35 +0200 Subject: [PATCH 18/30] ASoC: adau1977: make the Kconfig symbols user selectable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit SND_SOC_ADAU1977_{SPI,I2C} are missing Kconfig text, so they don't show up in menuconfig and can't be selected by a user - only by another symbol such as a machine driver. Add the text to make these symbols selectable and usable with generic machine drivers like the simple audio card. Signed-off-by: Alvin Šipraga Reviewed-by: Nuno Sá Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-1-aa2f0cabd728@analog.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f9a47e262a77..d88593c2bac8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -524,13 +524,13 @@ config SND_SOC_ADAU1977 tristate config SND_SOC_ADAU1977_SPI - tristate + tristate "Analog Devices ADAU1977/ADAU1978/ADAU1979 CODEC - SPI" depends on SPI_MASTER select SND_SOC_ADAU1977 select REGMAP_SPI config SND_SOC_ADAU1977_I2C - tristate + tristate "Analog Devices ADAU1977/ADAU1978/ADAU1979 CODEC - I2C" depends on I2C select SND_SOC_ADAU1977 select REGMAP_I2C From 528a0da3e55b24d1113b3658e94cf432e0020913 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= Date: Mon, 14 Sep 2026 12:12:36 +0200 Subject: [PATCH 19/30] ASoC: adau1977-spi: drop __maybe_unused and of_match_ptr() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since commit 5ab23c7923a1 ("modpost: Create modalias for builtin modules") MODULE_DEVICE_TABLE() is enough to reference a match table and the data isn't discarded by the linker even when the driver is built-in and CONFIG_OF is disabled. Drop the of_match_ptr() wrapping so that OF matching keeps working regardless of CONFIG_OF. This also means we can drop __maybe_unused since it's always used. The entries in adau1977_spi_of_match were also erroneously indented with spaces - replace the indentation with tabs to conform with coding style. Signed-off-by: Alvin Šipraga Reviewed-by: Nuno Sá Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-2-aa2f0cabd728@analog.com Signed-off-by: Mark Brown --- sound/soc/codecs/adau1977-spi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c index 878cde9d1014..c98da5ba9e9e 100644 --- a/sound/soc/codecs/adau1977-spi.c +++ b/sound/soc/codecs/adau1977-spi.c @@ -53,18 +53,18 @@ static const struct spi_device_id adau1977_spi_ids[] = { }; MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); -static const struct of_device_id adau1977_spi_of_match[] __maybe_unused = { - { .compatible = "adi,adau1977" }, - { .compatible = "adi,adau1978" }, - { .compatible = "adi,adau1979" }, - { }, +static const struct of_device_id adau1977_spi_of_match[] = { + { .compatible = "adi,adau1977" }, + { .compatible = "adi,adau1978" }, + { .compatible = "adi,adau1979" }, + { }, }; MODULE_DEVICE_TABLE(of, adau1977_spi_of_match); static struct spi_driver adau1977_spi_driver = { .driver = { .name = "adau1977", - .of_match_table = of_match_ptr(adau1977_spi_of_match), + .of_match_table = adau1977_spi_of_match, }, .probe = adau1977_spi_probe, .id_table = adau1977_spi_ids, From 76a8fe25b97881223976363044924d5cf0511749 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Alvin=20=C5=A0ipraga?= Date: Mon, 14 Sep 2026 12:12:37 +0200 Subject: [PATCH 20/30] ASoC: adau1977-i2c: add OF match table for I2C MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Like for SPI, the I2C driver needs an OF match table for the kernel to be able to automatically load the driver when built as a module. Add one. Signed-off-by: Alvin Šipraga Reviewed-by: Nuno Sá Link: https://patch.msgid.link/20260914-asoc-adau1977-fixes-v1-3-aa2f0cabd728@analog.com Signed-off-by: Mark Brown --- sound/soc/codecs/adau1977-i2c.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c index d1c6c4ddf506..5a11cafdff36 100644 --- a/sound/soc/codecs/adau1977-i2c.c +++ b/sound/soc/codecs/adau1977-i2c.c @@ -34,9 +34,18 @@ static const struct i2c_device_id adau1977_i2c_ids[] = { }; MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); +static const struct of_device_id adau1977_i2c_of_match[] = { + { .compatible = "adi,adau1977" }, + { .compatible = "adi,adau1978" }, + { .compatible = "adi,adau1979" }, + { }, +}; +MODULE_DEVICE_TABLE(of, adau1977_i2c_of_match); + static struct i2c_driver adau1977_i2c_driver = { .driver = { .name = "adau1977", + .of_match_table = adau1977_i2c_of_match, }, .probe = adau1977_i2c_probe, .id_table = adau1977_i2c_ids, From 4cae4ca34992d210d115e8e891f1e88c9df5f24c Mon Sep 17 00:00:00 2001 From: Seiji Adachi Date: Thu, 17 Sep 2026 22:18:51 +0900 Subject: [PATCH 21/30] ASoC: amd: yc: Add Lenovo ThinkPad E16 Gen 2 (21JN) to quirks Signed-off-by: Seiji Adachi Link: https://patch.msgid.link/20260917131851.2122144-1-seiji_adachi@yahoo.co.jp Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 385fede6d77f..ea173bdab1d5 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21JN"), + } + }, { .driver_data = &acp6x_card, .matches = { From 8d78e4f906cf0a1984cf3c852653727835e1a9e0 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Thu, 17 Sep 2026 15:58:51 +0200 Subject: [PATCH 22/30] ASoC: rt766: fix HID dependency for SND_SOC_SDCA_HID Selecting SND_SOC_SDCA_HID only works if HID is enabled: WARNING: unmet direct dependencies detected for SND_SOC_SDCA_HID Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_SOC_SDCA [=y] && (HID [=n]=y [=y] || HID [=n]=SND_SOC_SDCA [=y]) Selected by [y]: - SND_SOC_RT766_SDCA_SDW [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SOUNDWIRE [=y] && SND_SOC_SDCA [=y] Change the dependency to require SND_SOC_SDCA_CLASS instead, which solves the problem without extra selects. Fixes: b50ecf5873df ("ASoC: rt766: add RT766/RT767 SDCA driver") Signed-off-by: Arnd Bergmann Link: https://patch.msgid.link/20260917135950.405423-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d88593c2bac8..8e251e1d84b4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1989,9 +1989,7 @@ config SND_SOC_RT715_SDCA_SDW config SND_SOC_RT766_SDCA_SDW tristate "Realtek RT766 SDCA Codec - SDW" depends on SOUNDWIRE - depends on SND_SOC_SDCA - select SND_SOC_SDCA_HID - select SND_SOC_SDCA_IRQ + depends on SND_SOC_SDCA_CLASS select REGMAP_SOUNDWIRE select REGMAP_SOUNDWIRE_MBQ From 28114992dd2f03724eb3d024839b3f9f299656d3 Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Thu, 17 Sep 2026 22:21:49 +0530 Subject: [PATCH 23/30] ASoC: amd: acp: enable TAS2783 and add RT712-VB SoundWire machine The ACP7.0 SoundWire machine table did not cover boards that combine TI TAS2783 smart amplifiers on SoundWire link 0 with a Realtek RT712-VB codec on link 1. In addition, the TAS2783 SoundWire codec driver was not selected by the AMD legacy SoundWire machine config, so it was not built for these platforms. Select SND_SOC_TAS2783_SDW from SND_SOC_AMD_LEGACY_SDW_MACH so the TAS2783 SoundWire codec driver is built together with the AMD legacy (no-DSP) SoundWire machine. Add a new link/address map, acp70_tas2783a_l0_rt712_vb_l1, describing the TAS2783 amplifiers on link 0 and the RT712-VB codec on link 1, and register a corresponding entry in snd_soc_acpi_amd_acp70_sdw_machines[]. The entry uses snd_soc_acpi_amd_sdca_is_device_rt712_vb() as its machine_check callback so it only matches platforms that actually expose the RT712-VB device. Signed-off-by: Syed Saba Kareem Link: https://patch.msgid.link/20260917165314.1880855-1-syed.sabakareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 20 ++++++++++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index a04c301df4d6..e880e44b9763 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_AMD_LEGACY_SDW_MACH select SND_SOC_RT715_SDW select SND_SOC_RT715_SDCA_SDW select SND_SOC_RT722_SDCA_SDW + select SND_SOC_TAS2783_SDW help This option enables Legacy(No DSP) sound card support for SoundWire enabled AMD platforms along with ACP PDM controller. diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 815e088b437d..2a9da6de7bbc 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -748,7 +748,27 @@ static const struct snd_soc_acpi_link_adr acp70_rt721_only[] = { {} }; +static const struct snd_soc_acpi_link_adr acp70_tas2783a_l0_rt712_vb_l1[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(tas2783_2_adr), + .adr_d = tas2783_2_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt712_vb_1_group1_adr), + .adr_d = rt712_vb_1_group1_adr, + }, + {} +}; + struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { + { + .link_mask = BIT(0) | BIT(1), + .links = acp70_tas2783a_l0_rt712_vb_l1, + .machine_check = snd_soc_acpi_amd_sdca_is_device_rt712_vb, + .drv_name = "amd_sdw", + }, { .link_mask = BIT(0), .links = acp70_tas2783_2, From bcd61aa247c28b721ac21dda6e4135f75a2de29f Mon Sep 17 00:00:00 2001 From: Jiangshan Yi Date: Mon, 14 Sep 2026 18:44:56 +0800 Subject: [PATCH 24/30] ASoC: codecs: max98363: fix uninitialized stream_config->type stream_config is not initialized before being passed to sdw_stream_add_slave(). The type field may contain garbage and is later copied to stream->type by sdw_config_stream(). Zero-initialize stream_config so type defaults to SDW_STREAM_PCM. Fixes: 18c0af945fa3 ("ASoC: max98363: add soundwire amplifier driver") Cc: stable@vger.kernel.org Signed-off-by: Jiangshan Yi Link: https://patch.msgid.link/20260914104456.376666-1-yijiangshan@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/codecs/max98363.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98363.c b/sound/soc/codecs/max98363.c index 0e32b7a94cb0..4a6d7b163acf 100644 --- a/sound/soc/codecs/max98363.c +++ b/sound/soc/codecs/max98363.c @@ -200,7 +200,7 @@ static int max98363_sdw_dai_hw_params(struct snd_pcm_substream *substream, struct max98363_priv *max98363 = snd_soc_component_get_drvdata(component); - struct sdw_stream_config stream_config; + struct sdw_stream_config stream_config = {0}; struct sdw_port_config port_config; enum sdw_data_direction direction; struct sdw_stream_runtime *stream; From 174d160884199cad97413f33fe1b56027dc0d3e1 Mon Sep 17 00:00:00 2001 From: Jiangshan Yi Date: Mon, 14 Sep 2026 18:45:51 +0800 Subject: [PATCH 25/30] ASoC: codecs: rt1017-sdca-sdw: fix uninitialized stream_config->type stream_config is not initialized before being passed to sdw_stream_add_slave(). The type field may contain garbage and is later copied to stream->type by sdw_config_stream(). Zero-initialize stream_config so type defaults to SDW_STREAM_PCM. While at it, use snd_sdw_params_to_config() helper instead of open-coding the same logic. Fixes: 2b7aecd58528 ("ASoC: rt1017: Add RT1017 SDCA amplifier driver") Cc: stable@vger.kernel.org Signed-off-by: Jiangshan Yi Link: https://patch.msgid.link/20260914104551.377880-1-yijiangshan@kylinos.cn Signed-off-by: Mark Brown --- sound/soc/codecs/rt1017-sdca-sdw.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c index caf75e5657ef..01cc2511a7d0 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.c +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include @@ -573,11 +574,10 @@ static int rt1017_sdca_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_component *component = dai->component; struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); - struct sdw_stream_config stream_config; + struct sdw_stream_config stream_config = {0}; struct sdw_port_config port_config; - enum sdw_data_direction direction; struct sdw_stream_runtime *sdw_stream; - int retval, port, num_channels, ch_mask; + int retval, port; unsigned int sampling_rate; dev_dbg(dai->dev, "%s %s", __func__, dai->name); @@ -591,28 +591,17 @@ static int rt1017_sdca_pcm_hw_params(struct snd_pcm_substream *substream, /* SoundWire specific configuration */ /* port 1 for playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - direction = SDW_DATA_DIR_RX; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) port = 1; - } else { - direction = SDW_DATA_DIR_TX; + else port = 2; - } - num_channels = params_channels(params); - ch_mask = (1 << num_channels) - 1; - - stream_config.frame_rate = params_rate(params); - stream_config.ch_count = num_channels; - stream_config.bps = snd_pcm_format_width(params_format(params)); - stream_config.direction = direction; - - port_config.ch_mask = ch_mask; + snd_sdw_params_to_config(substream, params, &stream_config, &port_config); port_config.num = port; dev_dbg(dai->dev, "frame_rate %d, ch_count %d, bps %d, direction %d, ch_mask %d, port: %d\n", - params_rate(params), num_channels, snd_pcm_format_width(params_format(params)), - direction, ch_mask, port); + stream_config.frame_rate, stream_config.ch_count, stream_config.bps, + stream_config.direction, port_config.ch_mask, port); retval = sdw_stream_add_slave(rt1017->sdw_slave, &stream_config, &port_config, 1, sdw_stream); From 78da571393daf0ece9067fe040018a2c02c081a7 Mon Sep 17 00:00:00 2001 From: Christian Marangi Date: Tue, 8 Sep 2026 11:04:43 +0200 Subject: [PATCH 26/30] ASoC: dt-bindings: Add Airoha AN7581 AFE Sound card Add YAML schema for Airoha AN7581 AFE SoC sound card. The AFE handling on this SoC is very basic with clock always enabled and all the power saving feature not available. Signed-off-by: Christian Marangi Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20260908090448.3332-2-ansuelsmth@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/airoha,an7581-afe.yaml | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/airoha,an7581-afe.yaml diff --git a/Documentation/devicetree/bindings/sound/airoha,an7581-afe.yaml b/Documentation/devicetree/bindings/sound/airoha,an7581-afe.yaml new file mode 100644 index 000000000000..80d9e87f1470 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/airoha,an7581-afe.yaml @@ -0,0 +1,41 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/airoha,an7581-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Airoha AFE PCM controller for AN7581 + +maintainers: + - Christian Marangi + +properties: + compatible: + const: airoha,an7581-afe + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + +additionalProperties: false + +examples: + - | + #include + #include + + afe@1fbe2200 { + compatible = "airoha,an7581-afe"; + reg = <0x1fbe2200 0x9000>; + + interrupts = ; + }; + +... From 8fcf0497e6da049a3ff19e6baa958bbb146da996 Mon Sep 17 00:00:00 2001 From: Christian Marangi Date: Tue, 8 Sep 2026 11:04:44 +0200 Subject: [PATCH 27/30] ASoC: dt-bindings: Add Airoha AN7581 AFE with WM8960 Codec schema Add DT schema for Airoha AN7581 Sound Card with the specific WM8960 i2c Codec. Reviewed-by: Krzysztof Kozlowski Signed-off-by: Christian Marangi Link: https://patch.msgid.link/20260908090448.3332-3-ansuelsmth@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/airoha,an7581-wm8960.yaml | 71 +++++++++++++++++++ 1 file changed, 71 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/airoha,an7581-wm8960.yaml diff --git a/Documentation/devicetree/bindings/sound/airoha,an7581-wm8960.yaml b/Documentation/devicetree/bindings/sound/airoha,an7581-wm8960.yaml new file mode 100644 index 000000000000..8886fbb6b43c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/airoha,an7581-wm8960.yaml @@ -0,0 +1,71 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/airoha,an7581-wm8960.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Airoha AN7581 sound card with WM8960 codec + +maintainers: + - Christian Marangi + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: airoha,an7581-wm8960-sound + + platform: + type: object + + additionalProperties: false + + properties: + sound-dai: + items: + - description: The phandle of AN7581 platform. + + required: + - sound-dai + + codec: + type: object + + additionalProperties: false + + properties: + sound-dai: + items: + - description: The phandle of WM8960 i2c codec. + + required: + - sound-dai + +required: + - compatible + - audio-routing + - platform + - codec + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "airoha,an7581-wm8960-sound"; + model = "an7581-wm8960"; + audio-routing = + "Headphone", "HP_L", + "Headphone", "HP_R", + "LINPUT1", "AMIC", + "RINPUT1", "AMIC"; + + platform { + sound-dai = <&afe>; + }; + + codec { + sound-dai = <&wm8960>; + }; + }; From b49799e403137b12fb9e57671901fe9d56f73d99 Mon Sep 17 00:00:00 2001 From: Christian Marangi Date: Tue, 8 Sep 2026 11:04:45 +0200 Subject: [PATCH 28/30] ASoC: mediatek: common: permit to provide dedicated regmap for irq Some SoC might require dedicated regmap to configure some specific IRQ register. Add an extra entry in the base_afe_irq struct and use the specific regmap if defined. If not defined then the global AFE regmap is used instead. Signed-off-by: Christian Marangi Link: https://patch.msgid.link/20260908090448.3332-4-ansuelsmth@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 14 +++++++++----- sound/soc/mediatek/common/mtk-base-afe.h | 2 ++ 2 files changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c index 2a20fa5dba49..948f606cbaf2 100644 --- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c +++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c @@ -204,11 +204,15 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, struct mtk_base_afe_irq *irqs = &afe->irqs[memif->irq_usage]; const struct mtk_base_irq_data *irq_data = irqs->irq_data; unsigned int counter = runtime->period_size; + struct regmap *regmap = afe->regmap; int fs; int ret; dev_dbg(afe->dev, "%s %s cmd=%d\n", __func__, memif->data->name, cmd); + if (irqs->regmap) + regmap = irqs->regmap; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -220,7 +224,7 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, } /* set irq counter */ - mtk_regmap_update_bits(afe->regmap, irq_data->irq_cnt_reg, + mtk_regmap_update_bits(regmap, irq_data->irq_cnt_reg, irq_data->irq_cnt_maskbit, counter, irq_data->irq_cnt_shift); @@ -230,12 +234,12 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, if (fs < 0) return -EINVAL; - mtk_regmap_update_bits(afe->regmap, irq_data->irq_fs_reg, + mtk_regmap_update_bits(regmap, irq_data->irq_fs_reg, irq_data->irq_fs_maskbit, fs, irq_data->irq_fs_shift); /* enable interrupt */ - mtk_regmap_update_bits(afe->regmap, irq_data->irq_en_reg, + mtk_regmap_update_bits(regmap, irq_data->irq_en_reg, 1, 1, irq_data->irq_en_shift); return 0; @@ -248,10 +252,10 @@ int mtk_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, } /* disable interrupt */ - mtk_regmap_update_bits(afe->regmap, irq_data->irq_en_reg, + mtk_regmap_update_bits(regmap, irq_data->irq_en_reg, 1, 0, irq_data->irq_en_shift); /* and clear pending IRQ */ - mtk_regmap_write(afe->regmap, irq_data->irq_clr_reg, + mtk_regmap_write(regmap, irq_data->irq_clr_reg, 1 << irq_data->irq_clr_shift); return ret; default: diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index a406f2e3e7a8..b5300c9ed415 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -153,6 +153,8 @@ struct mtk_base_afe_memif { struct mtk_base_afe_irq { const struct mtk_base_irq_data *irq_data; int irq_occupyed; + + struct regmap *regmap; }; struct mtk_base_afe_dai { From 4974ffa0d6c28dfaba838a29503c3af02080a8a1 Mon Sep 17 00:00:00 2001 From: Christian Marangi Date: Tue, 8 Sep 2026 11:04:46 +0200 Subject: [PATCH 29/30] ASoC: airoha: Add AFE driver for Airoha AN7581 Add support for the Sound system present on Airoha AN7581 SoC. This is based on the Mediatek AFE drivers and adds the PCM and the ETDM driver. Signed-off-by: Christian Marangi Link: https://patch.msgid.link/20260908090448.3332-5-ansuelsmth@gmail.com Signed-off-by: Mark Brown --- MAINTAINERS | 8 + sound/soc/mediatek/Kconfig | 17 +- sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/an7581/Makefile | 8 + sound/soc/mediatek/an7581/an7581-afe-common.h | 49 ++ sound/soc/mediatek/an7581/an7581-afe-pcm.c | 515 ++++++++++++++++++ sound/soc/mediatek/an7581/an7581-dai-etdm.c | 433 +++++++++++++++ sound/soc/mediatek/an7581/an7581-reg.h | 114 ++++ 8 files changed, 1144 insertions(+), 1 deletion(-) create mode 100644 sound/soc/mediatek/an7581/Makefile create mode 100644 sound/soc/mediatek/an7581/an7581-afe-common.h create mode 100644 sound/soc/mediatek/an7581/an7581-afe-pcm.c create mode 100644 sound/soc/mediatek/an7581/an7581-dai-etdm.c create mode 100644 sound/soc/mediatek/an7581/an7581-reg.h diff --git a/MAINTAINERS b/MAINTAINERS index c2414447892c..679da67ce63d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -767,6 +767,14 @@ F: Documentation/devicetree/bindings/phy/airoha,en7581-pcie-phy.yaml F: drivers/phy/phy-airoha-pcie-regs.h F: drivers/phy/phy-airoha-pcie.c +AIROHA SOUND DRIVER +M: Christian Marangi +L: linux-sound@vger.kernel.org +S: Maintained +F: Documentation/devicetree/bindings/sound/airoha,an7581-afe.yaml +F: Documentation/devicetree/bindings/sound/airoha,an7581-wm8960.yaml +F: sound/soc/mediatek/an7581/* + AIROHA SPI SNFI DRIVER M: Lorenzo Bianconi M: Ray Liu diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 224746e7664f..eb6f2a443b23 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -1,10 +1,25 @@ # SPDX-License-Identifier: GPL-2.0-only -menu "Mediatek" config SND_SOC_MEDIATEK tristate select REGMAP_MMIO +menu "Airoha" + +config SND_SOC_AN7581 + tristate "ASoC support for Airoha AN7581 chip" + depends on ARCH_AIROHA || COMPILE_TEST + select SND_SOC_MEDIATEK + help + This adds ASoC platform driver support for Airoha AN7581 chip + that can be used with other codecs. + Select Y if you have such device. + If unsure select "N". + +endmenu + +menu "Mediatek" + config SND_SOC_MT2701 tristate "ASoC support for Mediatek MT2701 chip" depends on ARCH_MEDIATEK || COMPILE_TEST diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index a6815a3c5988..54522e1fa016 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0 obj-$(CONFIG_SND_SOC_MEDIATEK) += common/ +obj-$(CONFIG_SND_SOC_AN7581) += an7581/ obj-$(CONFIG_SND_SOC_MT2701) += mt2701/ obj-$(CONFIG_SND_SOC_MT6797) += mt6797/ obj-$(CONFIG_SND_SOC_MT7986) += mt7986/ diff --git a/sound/soc/mediatek/an7581/Makefile b/sound/soc/mediatek/an7581/Makefile new file mode 100644 index 000000000000..d6de5ee1668e --- /dev/null +++ b/sound/soc/mediatek/an7581/Makefile @@ -0,0 +1,8 @@ +# SPDX-License-Identifier: GPL-2.0 + +# platform driver +snd-soc-an7581-afe-y := \ + an7581-afe-pcm.o \ + an7581-dai-etdm.o + +obj-$(CONFIG_SND_SOC_AN7581) += snd-soc-an7581-afe.o diff --git a/sound/soc/mediatek/an7581/an7581-afe-common.h b/sound/soc/mediatek/an7581/an7581-afe-common.h new file mode 100644 index 000000000000..ebc8e2f7fec4 --- /dev/null +++ b/sound/soc/mediatek/an7581/an7581-afe-common.h @@ -0,0 +1,49 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * an7581-afe-common.h -- Airoha AN7581 audio driver definitions + */ + +#ifndef _AN7581_AFE_COMMON_H_ +#define _AN7581_AFE_COMMON_H_ + +#include +#include +#include +#include + +#include "../../mediatek/common/mtk-base-afe.h" + +enum { + AN7581_MEMIF_DL1, + AN7581_MEMIF_UL1, + AN7581_MEMIF_NUM, + AN7581_DAI_ETDM = AN7581_MEMIF_NUM, + AN7581_DAI_NUM, +}; + +enum { + AN7581_IRQ_0, + AN7581_IRQ_1, + AN7581_IRQ_NUM, +}; + +struct an7581_memif_irq_desc { + u8 irq; + u32 status_bit; + u32 clear_reg; +}; + +struct an7581_afe_private { + unsigned int users; + /* protect concurrent ETDM user tracking */ + struct mutex user_lock; + + void *dai_priv[AN7581_DAI_NUM]; +}; + +unsigned int an7581_afe_rate_transform(struct device *dev, + unsigned int rate); + +int an7581_dai_etdm_register(struct mtk_base_afe *afe); + +#endif diff --git a/sound/soc/mediatek/an7581/an7581-afe-pcm.c b/sound/soc/mediatek/an7581/an7581-afe-pcm.c new file mode 100644 index 000000000000..a1d742362538 --- /dev/null +++ b/sound/soc/mediatek/an7581/an7581-afe-pcm.c @@ -0,0 +1,515 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Airoha ALSA SoC AFE platform driver for AN7581 + * + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "an7581-afe-common.h" +#include "an7581-reg.h" +#include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-afe-fe-dai.h" + +enum { + ARH_AFE_RATE_8K = 0, + ARH_AFE_RATE_12K = 1, + ARH_AFE_RATE_16K = 2, + ARH_AFE_RATE_24K = 3, + ARH_AFE_RATE_32K = 4, + ARH_AFE_RATE_48K = 5, + ARH_AFE_RATE_96K = 6, + ARH_AFE_RATE_192K = 7, + ARH_AFE_RATE_384K = 8, + ARH_AFE_RATE_7K = 16, + ARH_AFE_RATE_11K = 17, + ARH_AFE_RATE_14K = 18, + ARH_AFE_RATE_22K = 19, + ARH_AFE_RATE_29K = 20, + ARH_AFE_RATE_44K = 21, + ARH_AFE_RATE_88K = 22, + ARH_AFE_RATE_176K = 23, + ARH_AFE_RATE_352K = 24, +}; + +unsigned int an7581_afe_rate_transform(struct device *dev, unsigned int rate) +{ + switch (rate) { + case 7350: + return ARH_AFE_RATE_7K; + case 8000: + return ARH_AFE_RATE_8K; + case 11025: + return ARH_AFE_RATE_11K; + case 12000: + return ARH_AFE_RATE_12K; + case 14700: + return ARH_AFE_RATE_14K; + case 16000: + return ARH_AFE_RATE_16K; + case 22050: + return ARH_AFE_RATE_22K; + case 24000: + return ARH_AFE_RATE_24K; + case 29400: + return ARH_AFE_RATE_29K; + case 32000: + return ARH_AFE_RATE_32K; + case 44100: + return ARH_AFE_RATE_44K; + case 48000: + return ARH_AFE_RATE_48K; + case 88200: + return ARH_AFE_RATE_88K; + case 96000: + return ARH_AFE_RATE_96K; + case 176400: + return ARH_AFE_RATE_176K; + case 192000: + return ARH_AFE_RATE_192K; + case 352800: + return ARH_AFE_RATE_352K; + case 384000: + return ARH_AFE_RATE_384K; + default: + dev_warn_ratelimited(dev, "%s(), rate %u invalid, using %d!\n", + __func__, rate, ARH_AFE_RATE_48K); + return ARH_AFE_RATE_48K; + } +} + +static const struct an7581_memif_irq_desc an7581_memif_irq_descs[AN7581_MEMIF_NUM] = { + [AN7581_MEMIF_DL1] = { + .irq = AN7581_IRQ_0, + .status_bit = AFE_IRQ_STS_PLAY, + .clear_reg = AFE_IRQ_CON0, + }, + [AN7581_MEMIF_UL1] = { + .irq = AN7581_IRQ_1, + .status_bit = AFE_IRQ_STS_RECORD, + .clear_reg = AFE_IRQ1_CON0, + }, +}; + +static const struct snd_pcm_hardware an7581_afe_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = 512, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 256 * 1024, +}; + +static int an7581_memif_irq_fs(struct snd_pcm_substream *substream, unsigned int rate) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, AFE_PCM_NAME); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + + return an7581_afe_rate_transform(afe->dev, rate); +} + +static int an7581_afe_fe_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = mtk_afe_fe_startup(substream, dai); + if (ret < 0) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 0x2000, UINT_MAX); + if (ret < 0) + dev_err(afe->dev, "hw_constraint_minmax failed\n"); + } + + return ret; +} + +const struct snd_soc_dai_ops an7581_afe_fe_ops = { + .startup = an7581_afe_fe_startup, + .shutdown = mtk_afe_fe_shutdown, + .hw_params = mtk_afe_fe_hw_params, + .hw_free = mtk_afe_fe_hw_free, + .prepare = mtk_afe_fe_prepare, + .trigger = mtk_afe_fe_trigger, +}; + +#define ARH_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver an7581_memif_dai_driver[] = { + /* FE DAIs: memory intefaces to CPU */ + { + .name = "DL1", + .id = AN7581_MEMIF_DL1, + .playback = { + .stream_name = "DL1", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = ARH_PCM_FORMATS, + }, + .ops = &an7581_afe_fe_ops, + }, + { + .name = "UL1", + .id = AN7581_MEMIF_UL1, + .capture = { + .stream_name = "UL1", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = ARH_PCM_FORMATS, + }, + .ops = &an7581_afe_fe_ops, + }, +}; + +static const struct snd_soc_dapm_widget an7581_memif_widgets[] = { + /* DL */ + SND_SOC_DAPM_MIXER("I032", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I033", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* UL */ + SND_SOC_DAPM_MIXER("O018", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("O019", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route an7581_memif_routes[] = { + {"I032", NULL, "DL1"}, + {"I033", NULL, "DL1"}, + {"UL1", NULL, "O018"}, + {"UL1", NULL, "O019"}, + {"O018", NULL, "I150"}, + {"O019", NULL, "I151"}, +}; + +static const struct snd_soc_component_driver an7581_afe_pcm_dai_component = { + .name = "an7581-afe-pcm-dai", +}; + +static const struct mtk_base_memif_data memif_data[AN7581_MEMIF_NUM] = { + [AN7581_MEMIF_DL1] = { + .name = "DL1", + .id = AN7581_MEMIF_DL1, + .reg_ofs_base = AFE_DL1_BASE, + .reg_ofs_cur = AFE_DL1_CUR, + .reg_ofs_end = AFE_DL1_END, + .fs_reg = -1, + .fs_shift = -1, + .fs_maskbit = -1, + .mono_reg = -1, + .mono_shift = -1, + .hd_reg = AFE_DL1_CON0, + .hd_shift = AFE_HD_SHIFT, + .hd_align_reg = -1, + .hd_align_mshift = -1, + .enable_reg = AFE_DAC_CON0, + .enable_shift = AFE_DL1_ENABLE_SHIFT, + .msb_reg = -1, + .msb_shift = -1, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, + [AN7581_MEMIF_UL1] = { + .name = "UL1", + .id = AN7581_MEMIF_UL1, + .reg_ofs_base = AFE_UL1_BASE, + .reg_ofs_cur = AFE_UL1_CUR, + .reg_ofs_end = AFE_UL1_END, + .fs_reg = -1, + .fs_shift = -1, + .fs_maskbit = -1, + .mono_reg = -1, + .mono_shift = -1, + .hd_reg = AFE_UL1_CON0, + .hd_shift = AFE_HD_SHIFT, + .hd_align_reg = AFE_UL1_CON0, + .hd_align_mshift = AFE_HD_ALIGN_SHIFT, + .enable_reg = AFE_DAC_CON0, + .enable_shift = AFE_UL1_ENABLE_SHIFT, + .msb_reg = -1, + .msb_shift = -1, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, +}; + +static const struct mtk_base_irq_data irq_data[AN7581_IRQ_NUM] = { + [AN7581_IRQ_0] = { + .id = AN7581_IRQ_0, + .irq_cnt_reg = AFE_IRQ_CNT, + .irq_cnt_shift = AFE_IRQ_CNT_SHIFT, + .irq_cnt_maskbit = AFE_IRQ_CNT_MASK, + .irq_en_reg = AFE_IRQ_CON0, + .irq_en_shift = AFE_IRQ_ON_SHIFT, + .irq_fs_reg = -1, + .irq_fs_shift = -1, + .irq_fs_maskbit = -1, + .irq_clr_reg = -1, + .irq_clr_shift = -1, + }, + [AN7581_IRQ_1] = { + .id = AN7581_IRQ_1, + .irq_cnt_reg = AFE_IRQ1_CNT, + .irq_cnt_shift = AFE_IRQ_CNT_SHIFT, + .irq_cnt_maskbit = AFE_IRQ_CNT_MASK, + .irq_en_reg = AFE_IRQ1_CON0, + .irq_en_shift = AFE_IRQ_ON_SHIFT, + .irq_fs_reg = -1, + .irq_fs_shift = -1, + .irq_fs_maskbit = -1, + .irq_clr_reg = -1, + .irq_clr_shift = -1, + }, +}; + +static const struct regmap_config an7581_afe_regmap_config = { + .name = "afe", + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AFE_MAX_REGISTER, +}; + +static const struct regmap_config an7581_afe_irq1_regmap_config = { + .name = "afe_irq1", + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AFE_IRQ1_MAX_REGISTER, +}; + +static irqreturn_t an7581_afe_irq_handler(int irq_id, void *dev) +{ + const struct an7581_memif_irq_desc *irq_desc; + struct mtk_base_afe *afe = dev; + u32 status = 0; + int i; + + regmap_read(afe->regmap, AFE_IRQ_STS, &status); + + for (i = 0; i < AN7581_MEMIF_NUM; i++) { + struct mtk_base_afe_memif *memif = &afe->memif[i]; + + if (!memif->substream) + continue; + + if (memif->irq_usage < 0) + continue; + + irq_desc = &an7581_memif_irq_descs[i]; + if (status & irq_desc->status_bit) + snd_pcm_period_elapsed(memif->substream); + } + + for (i = 0; i < AN7581_MEMIF_NUM; i++) { + struct regmap *irq_regmap; + + irq_desc = &an7581_memif_irq_descs[i]; + if (!(status & irq_desc->status_bit)) + continue; + + irq_regmap = afe->irqs[irq_desc->irq].regmap; + regmap_set_bits(irq_regmap, irq_desc->clear_reg, + BIT(AFE_IRQ_CLR_SHIFT)); + regmap_clear_bits(irq_regmap, irq_desc->clear_reg, + BIT(AFE_IRQ_CLR_SHIFT)); + + regmap_set_bits(irq_regmap, irq_desc->clear_reg, + BIT(AFE_IRQ_MISS_FLG_CLR_SHIFT)); + regmap_clear_bits(irq_regmap, irq_desc->clear_reg, + BIT(AFE_IRQ_MISS_FLG_CLR_SHIFT)); + } + + return IRQ_HANDLED; +} + +static int an7581_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = an7581_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(an7581_memif_dai_driver); + + dai->dapm_widgets = an7581_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(an7581_memif_widgets); + dai->dapm_routes = an7581_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(an7581_memif_routes); + + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + an7581_dai_etdm_register, + an7581_dai_memif_register, +}; + +static int an7581_afe_pcm_dev_probe(struct platform_device *pdev) +{ + struct an7581_afe_private *afe_priv; + struct device *dev = &pdev->dev; + struct reset_control *reset; + struct mtk_base_afe *afe; + int i, irq_id, ret; + void *base; + + afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL); + if (!afe) + return -ENOMEM; + platform_set_drvdata(pdev, afe); + + afe->platform_priv = devm_kzalloc(dev, sizeof(*afe_priv), + GFP_KERNEL); + if (!afe->platform_priv) + return -ENOMEM; + + afe->irqs_size = AN7581_IRQ_NUM; + afe->irqs = devm_kcalloc(dev, afe->irqs_size, sizeof(*afe->irqs), + GFP_KERNEL); + if (!afe->irqs) + return -ENOMEM; + + afe->memif_size = AN7581_MEMIF_NUM; + afe->memif = devm_kcalloc(dev, afe->memif_size, sizeof(*afe->memif), + GFP_KERNEL); + if (!afe->memif) + return -ENOMEM; + + afe_priv = afe->platform_priv; + mutex_init(&afe_priv->user_lock); + afe->dev = &pdev->dev; + + reset = devm_reset_control_get_exclusive(dev, NULL); + if (IS_ERR(reset)) + return PTR_ERR(reset); + + /* Global reset I2S */ + reset_control_assert(reset); + usleep_range(10, 20); + reset_control_deassert(reset); + + afe->base_addr = devm_platform_ioremap_resource(pdev, 0); + if (IS_ERR(afe->base_addr)) + return PTR_ERR(afe->base_addr); + + base = devm_platform_ioremap_resource(pdev, 1); + if (IS_ERR(base)) + return PTR_ERR(base); + + ret = devm_pm_runtime_enable(dev); + if (ret) + return ret; + + afe->regmap = devm_regmap_init_mmio(&pdev->dev, afe->base_addr, + &an7581_afe_regmap_config); + if (IS_ERR(afe->regmap)) + return PTR_ERR(afe->regmap); + + mutex_init(&afe->irq_alloc_lock); + + /* irq initialize */ + for (i = 0; i < afe->irqs_size; i++) + afe->irqs[i].irq_data = &irq_data[i]; + + afe->irqs[AN7581_IRQ_0].regmap = afe->regmap; + afe->irqs[AN7581_IRQ_1].regmap = devm_regmap_init_mmio(&pdev->dev, base, + &an7581_afe_irq1_regmap_config); + if (IS_ERR(afe->irqs[AN7581_IRQ_1].regmap)) + return PTR_ERR(afe->irqs[AN7581_IRQ_1].regmap); + + /* request irq */ + irq_id = platform_get_irq(pdev, 0); + if (irq_id < 0) + return irq_id; + + /* init memif */ + for (i = 0; i < afe->memif_size; i++) { + int sel_irq = an7581_memif_irq_descs[i].irq; + + afe->memif[i].data = &memif_data[i]; + afe->memif[i].irq_usage = sel_irq; + afe->memif[i].const_irq = 1; + afe->irqs[sel_irq].irq_occupyed = true; + } + + /* init sub_dais */ + INIT_LIST_HEAD(&afe->sub_dais); + + ret = devm_request_irq(dev, irq_id, an7581_afe_irq_handler, + IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); + if (ret) + return dev_err_probe(dev, ret, "Failed to request irq for asys-isr\n"); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) + return dev_err_probe(dev, ret, "DAI register failed, i: %d\n", i); + } + + /* init dai_driver and component_driver */ + ret = mtk_afe_combine_sub_dai(afe); + if (ret) + return dev_err_probe(dev, ret, "mtk_afe_combine_sub_dai fail\n"); + + afe->mtk_afe_hardware = &an7581_afe_hardware; + afe->memif_fs = an7581_memif_irq_fs; + afe->irq_fs = an7581_memif_irq_fs; + + /* register component */ + ret = devm_snd_soc_register_component(&pdev->dev, + &mtk_afe_pcm_platform, + NULL, 0); + if (ret) + return dev_err_probe(dev, ret, "Cannot register AFE component\n"); + + ret = devm_snd_soc_register_component(afe->dev, + &an7581_afe_pcm_dai_component, + afe->dai_drivers, + afe->num_dai_drivers); + if (ret) + return dev_err_probe(dev, ret, "Cannot register PCM DAI component\n"); + + return 0; +} + +static const struct of_device_id an7581_afe_pcm_dt_match[] = { + { .compatible = "airoha,an7581-afe" }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, an7581_afe_pcm_dt_match); + +static struct platform_driver an7581_afe_pcm_driver = { + .driver = { + .name = "an7581-audio", + .of_match_table = an7581_afe_pcm_dt_match, + }, + .probe = an7581_afe_pcm_dev_probe, +}; +module_platform_driver(an7581_afe_pcm_driver); + +MODULE_DESCRIPTION("Airoha SoC AFE platform driver for ALSA AN7581"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/mediatek/an7581/an7581-dai-etdm.c b/sound/soc/mediatek/an7581/an7581-dai-etdm.c new file mode 100644 index 000000000000..7fbfd8fe1836 --- /dev/null +++ b/sound/soc/mediatek/an7581/an7581-dai-etdm.c @@ -0,0 +1,433 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Airoha ALSA SoC Audio DAI eTDM Control + * + */ + +#include +#include +#include +#include +#include "an7581-afe-common.h" +#include "an7581-reg.h" + +#define HOPPING_CLK 0 +#define APLL_CLK 1 +#define MTK_DAI_ETDM_FORMAT_I2S 0 +#define MTK_DAI_ETDM_FORMAT_DSPA 4 +#define MTK_DAI_ETDM_FORMAT_DSPB 5 + +enum { + MTK_ETDM_RATE_8K = 0, + MTK_ETDM_RATE_12K = 1, + MTK_ETDM_RATE_16K = 2, + MTK_ETDM_RATE_24K = 3, + MTK_ETDM_RATE_32K = 4, + MTK_ETDM_RATE_48K = 5, + MTK_ETDM_RATE_96K = 6, + MTK_ETDM_RATE_192K = 7, + MTK_ETDM_RATE_384K = 8, + MTK_ETDM_RATE_7K = 16, + MTK_ETDM_RATE_11K = 17, + MTK_ETDM_RATE_14K = 18, + MTK_ETDM_RATE_22K = 19, + MTK_ETDM_RATE_29K = 20, + MTK_ETDM_RATE_44K = 21, + MTK_ETDM_RATE_88K = 22, + MTK_ETDM_RATE_176K = 23, + MTK_ETDM_RATE_352K = 24, +}; + +struct mtk_dai_etdm_priv { + bool bck_inv; + bool lrck_inv; + bool slave_mode; + unsigned int format; +}; + +static unsigned int an7581_etdm_rate_transform(struct device *dev, unsigned int rate) +{ + switch (rate) { + case 7350: + return MTK_ETDM_RATE_7K; + case 8000: + return MTK_ETDM_RATE_8K; + case 11025: + return MTK_ETDM_RATE_11K; + case 12000: + return MTK_ETDM_RATE_12K; + case 14700: + return MTK_ETDM_RATE_14K; + case 16000: + return MTK_ETDM_RATE_16K; + case 22050: + return MTK_ETDM_RATE_22K; + case 24000: + return MTK_ETDM_RATE_24K; + case 29400: + return MTK_ETDM_RATE_29K; + case 32000: + return MTK_ETDM_RATE_32K; + case 44100: + return MTK_ETDM_RATE_44K; + case 48000: + return MTK_ETDM_RATE_48K; + case 88200: + return MTK_ETDM_RATE_88K; + case 96000: + return MTK_ETDM_RATE_96K; + case 176400: + return MTK_ETDM_RATE_176K; + case 192000: + return MTK_ETDM_RATE_192K; + case 352800: + return MTK_ETDM_RATE_352K; + case 384000: + return MTK_ETDM_RATE_384K; + default: + dev_warn_ratelimited(dev, "%s(), rate %u invalid, using %d!\n", + __func__, rate, MTK_ETDM_RATE_48K); + return MTK_ETDM_RATE_48K; + } +} + +static int get_etdm_wlen(unsigned int bitwidth) +{ + return bitwidth <= 16 ? 16 : 32; +} + +static const struct snd_soc_dapm_widget mtk_dai_etdm_widgets[] = { + /* DL */ + SND_SOC_DAPM_MIXER("I150", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I151", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* UL */ + SND_SOC_DAPM_MIXER("O124", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("O125", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route mtk_dai_etdm_routes[] = { + {"I150", NULL, "ETDM Capture"}, + {"I151", NULL, "ETDM Capture"}, + {"ETDM Playback", NULL, "O124"}, + {"ETDM Playback", NULL, "O125"}, + {"O124", NULL, "I032"}, + {"O125", NULL, "I033"}, +}; + +/* dai ops */ +static int mtk_dai_etdm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct an7581_afe_private *afe_priv = afe->platform_priv; + + guard(mutex)(&afe_priv->user_lock); + + if (!afe_priv->users) + regmap_set_bits(afe->regmap, AFE_DAC_CON0, + BIT(AFE_AFE_ENABLE_SHIFT)); + afe_priv->users++; + + return 0; +} + +static void mtk_dai_etdm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct an7581_afe_private *afe_priv = afe->platform_priv; + + guard(mutex)(&afe_priv->user_lock); + + afe_priv->users--; + if (!afe_priv->users) + regmap_clear_bits(afe->regmap, AFE_DAC_CON0, + BIT(AFE_AFE_ENABLE_SHIFT)); +} + +static unsigned int get_etdm_ch_fixup(unsigned int channels) +{ + if (channels > 16) + return 24; + + if (channels > 8) + return 16; + + if (channels > 4) + return 8; + + if (channels > 2) + return 4; + + return 2; +} + +static int mtk_dai_etdm_config(struct mtk_base_afe *afe, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai, + int stream) +{ + struct an7581_afe_private *afe_priv = afe->platform_priv; + struct mtk_dai_etdm_priv *etdm_data = afe_priv->dai_priv[dai->id]; + unsigned int rate = params_rate(params); + unsigned int etdm_rate = an7581_etdm_rate_transform(afe->dev, rate); + unsigned int bit_width = params_width(params); + unsigned int mask, mask1; + unsigned int val, val1; + + dev_dbg(afe->dev, "%s(), stream %d, rate %u, bitwidth %u\n", + __func__, stream, rate, params_width(params)); + + /* CON0 */ + mask = ETDM_SLAVE_MODE | ETDM_BIT_LEN | ETDM_WRD_LEN | + ETDM_FMT | ETDM_CH_NUM; + val = FIELD_PREP(ETDM_BIT_LEN, params_width(params) - 1) | + FIELD_PREP(ETDM_WRD_LEN, get_etdm_wlen(bit_width) - 1) | + FIELD_PREP(ETDM_FMT, etdm_data->format) | + FIELD_PREP(ETDM_CH_NUM, + get_etdm_ch_fixup(params_channels(params)) - 1); + if (etdm_data->slave_mode) + val |= ETDM_SLAVE_MODE; + + /* CON1 */ + mask1 = EDTM_LRCK_AUTO_MODE | EDTM_CKEN_SEL | EDTM_LRCK_AUTO_OFF | + EDTM_INITIAL_POINT | EDTM_INITIAL_COUNT; + val1 = EDTM_LRCK_AUTO_MODE | EDTM_CKEN_SEL | EDTM_LRCK_AUTO_OFF | + FIELD_PREP(EDTM_INITIAL_POINT, 14) | FIELD_PREP(EDTM_INITIAL_COUNT, 14); + + switch (stream) { + case SNDRV_PCM_STREAM_PLAYBACK: + regmap_update_bits(afe->regmap, ETDM_OUT1_CON0, mask, val); + + mask1 |= EDTM_DIRECT_INPUT_MASTER_BCK; + val1 |= EDTM_DIRECT_INPUT_MASTER_BCK; + + regmap_update_bits(afe->regmap, ETDM_OUT1_CON1, mask1, val1); + + regmap_update_bits(afe->regmap, ETDM_OUT1_CON4, OUT_SEL_FS, + FIELD_PREP(OUT_SEL_FS, etdm_rate)); + + regmap_update_bits(afe->irqs[AN7581_IRQ_0].regmap, + afe->irqs[AN7581_IRQ_0].irq_data->irq_en_reg, + AFE_IRQ_EN_SEL, AFE_IRQ_EN_SEL_I2SOUT); + break; + case SNDRV_PCM_STREAM_CAPTURE: + regmap_update_bits(afe->regmap, ETDM_IN1_CON0, mask, val); + + regmap_update_bits(afe->regmap, ETDM_IN1_CON1, mask1, val1); + + regmap_update_bits(afe->regmap, ETDM_IN1_CON3, IN_SEL_FS, + FIELD_PREP(IN_SEL_FS, etdm_rate)); + + regmap_update_bits(afe->irqs[AN7581_IRQ_1].regmap, + afe->irqs[AN7581_IRQ_1].irq_data->irq_en_reg, + AFE_IRQ_EN_SEL, AFE_IRQ_EN_SEL_I2SIN); + break; + default: + break; + } + + return 0; +} + +static int mtk_dai_etdm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + unsigned int rate = params_rate(params); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + regmap_update_bits(afe->regmap, ETDM_COWORK_CON0, + EDTM_IN1_SLAVE_SEL, + EDTM_IN1_SLAVE_FROM_ETDMIN1_SLAVE); + regmap_update_bits(afe->regmap, ETDM_COWORK_CON0, + EDTM_OUT1_SLAVE_SEL, + EDTM_OUT1_SLAVE_FROM_ETDMOUT1_SLAVE); + regmap_update_bits(afe->regmap, ETDM_COWORK_CON1, + EDTM_IN1_SDATA0_SEL, + EDTM_IN1_SDATA0_FROM_PAD); + + switch (rate) { + case 7350: + case 8000: + case 11025: + case 12000: + case 14700: + case 16000: + case 22050: + case 24000: + case 29400: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + case 176400: + case 192000: + case 352800: + case 384000: + return mtk_dai_etdm_config(afe, params, dai, substream->stream); + default: + dev_err(afe->dev, "Sample rate %d invalid\n", rate); + return -EINVAL; + } +} + +static int mtk_dai_etdm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + dev_dbg(afe->dev, "%s(), cmd %d, dai id %d\n", __func__, cmd, dai->id); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_set_bits(afe->regmap, ETDM_OUT1_CON0, + ETDM_OUT_EN); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + regmap_set_bits(afe->regmap, ETDM_IN1_CON0, + ETDM_IN_EN); + + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_clear_bits(afe->regmap, ETDM_OUT1_CON0, + ETDM_OUT_EN); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + regmap_clear_bits(afe->regmap, ETDM_IN1_CON0, + ETDM_IN_EN); + + break; + default: + break; + } + + return 0; +} + +static int mtk_dai_etdm_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct an7581_afe_private *afe_priv = afe->platform_priv; + struct mtk_dai_etdm_priv *etdm_data; + void *priv_data; + + priv_data = devm_kzalloc(afe->dev, sizeof(struct mtk_dai_etdm_priv), + GFP_KERNEL); + if (!priv_data) + return -ENOMEM; + + afe_priv->dai_priv[dai->id] = priv_data; + etdm_data = afe_priv->dai_priv[dai->id]; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + etdm_data->format = MTK_DAI_ETDM_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + etdm_data->format = MTK_DAI_ETDM_FORMAT_DSPA; + break; + case SND_SOC_DAIFMT_DSP_B: + etdm_data->format = MTK_DAI_ETDM_FORMAT_DSPB; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + etdm_data->bck_inv = false; + etdm_data->lrck_inv = false; + break; + case SND_SOC_DAIFMT_NB_IF: + etdm_data->bck_inv = false; + etdm_data->lrck_inv = true; + break; + case SND_SOC_DAIFMT_IB_NF: + etdm_data->bck_inv = true; + etdm_data->lrck_inv = false; + break; + case SND_SOC_DAIFMT_IB_IF: + etdm_data->bck_inv = true; + etdm_data->lrck_inv = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_BP_FP: + etdm_data->slave_mode = false; + break; + case SND_SOC_DAIFMT_BC_FC: + etdm_data->slave_mode = true; + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops mtk_dai_etdm_ops = { + .startup = mtk_dai_etdm_startup, + .shutdown = mtk_dai_etdm_shutdown, + .hw_params = mtk_dai_etdm_hw_params, + .trigger = mtk_dai_etdm_trigger, + .set_fmt = mtk_dai_etdm_set_fmt, +}; + +/* dai driver */ +#define MTK_ETDM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver mtk_dai_etdm_driver[] = { + { + .name = "ETDM", + .id = AN7581_DAI_ETDM, + .capture = { + .stream_name = "ETDM Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = MTK_ETDM_FORMATS, + }, + .playback = { + .stream_name = "ETDM Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = MTK_ETDM_FORMATS, + }, + .ops = &mtk_dai_etdm_ops, + .symmetric_rate = 1, + .symmetric_sample_bits = 1, + }, +}; + +int an7581_dai_etdm_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_etdm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_etdm_driver); + + dai->dapm_widgets = mtk_dai_etdm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_etdm_widgets); + dai->dapm_routes = mtk_dai_etdm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_etdm_routes); + + return 0; +} diff --git a/sound/soc/mediatek/an7581/an7581-reg.h b/sound/soc/mediatek/an7581/an7581-reg.h new file mode 100644 index 000000000000..a6ada3ca97ab --- /dev/null +++ b/sound/soc/mediatek/an7581/an7581-reg.h @@ -0,0 +1,114 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * an7581-reg.h -- Airoha AN7581 audio driver reg definition + */ + +#ifndef _AN7581_REG_H_ +#define _AN7581_REG_H_ + +#define AFE_DAC_CON0 0x0 +#define AFE_DL1_ENABLE_SHIFT 17 +#define AFE_UL1_ENABLE_SHIFT 1 +#define AFE_AFE_ENABLE_SHIFT 0 + +#define ETDM_COWORK_CON0 0x4c +#define EDTM_IN1_SLAVE_SEL GENMASK(27, 24) +#define EDTM_IN1_SLAVE_FROM_ETDMIN1_SLAVE FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0x1) +#define EDTM_IN1_SLAVE_FROM_ETDMIN2_MASTER FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0x2) +#define EDTM_IN1_SLAVE_FROM_ETDMIN2_SLAVE FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0x3) +#define EDTM_IN1_SLAVE_FROM_ETDMOUT1_MASTER FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0x8) +#define EDTM_IN1_SLAVE_FROM_ETDMOUT1_SLAVE FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0x9) +#define EDTM_IN1_SLAVE_FROM_ETDMOUT2_MASTER FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0xa) +#define EDTM_IN1_SLAVE_FROM_ETDMOUT2_SLAVE FIELD_PREP_CONST(EDTM_IN1_SLAVE_SEL, 0xb) +#define EDTM_OUT1_SLAVE_SEL GENMASK(11, 8) +#define EDTM_OUT1_SLAVE_FROM_ETDMIN1_MASTER FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0x0) +#define EDTM_OUT1_SLAVE_FROM_ETDMIN1_SLAVE FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0x1) +#define EDTM_OUT1_SLAVE_FROM_ETDMIN2_MASTER FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0x2) +#define EDTM_OUT1_SLAVE_FROM_ETDMIN2_SLAVE FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0x3) +#define EDTM_OUT1_SLAVE_FROM_ETDMOUT1_SLAVE FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0x9) +#define EDTM_OUT1_SLAVE_FROM_ETDMOUT2_MASTER FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0xa) +#define EDTM_OUT1_SLAVE_FROM_ETDMOUT2_SLAVE FIELD_PREP_CONST(EDTM_OUT1_SLAVE_SEL, 0xb) +#define ETDM_COWORK_CON1 0x50 +#define EDTM_IN1_SDATA0_SEL GENMASK(3, 0) +#define EDTM_IN1_SDATA0_FROM_PAD FIELD_PREP_CONST(EDTM_IN1_SDATA0_SEL, 0x0) +#define EDTM_IN1_SDATA0_FROM_ETDMIN2 FIELD_PREP_CONST(EDTM_IN1_SDATA0_SEL, 0x2) +#define EDTM_IN1_SDATA0_FROM_ETDMOUT1 FIELD_PREP_CONST(EDTM_IN1_SDATA0_SEL, 0x8) +#define EDTM_IN1_SDATA0_FROM_ETDMOUT2 FIELD_PREP_CONST(EDTM_IN1_SDATA0_SEL, 0xa) +#define ETDM_IN1_CON0 0x5c +#define ETDM_CH_NUM GENMASK(26, 23) +#define ETDM_WRD_LEN GENMASK(20, 16) +#define ETDM_BIT_LEN GENMASK(15, 11) +#define ETDM_FMT GENMASK(8, 6) +#define ETDM_SLAVE_MODE BIT(5) +#define ETDM_IN_EN BIT(0) +#define ETDM_IN1_CON1 0x60 +#define EDTM_LRCK_AUTO_MODE BIT(29) +#define EDTM_DIRECT_INPUT_MASTER_BCK BIT(30) +#define EDTM_CKEN_SEL BIT(12) +#define EDTM_LRCK_AUTO_OFF BIT(10) +#define EDTM_INITIAL_POINT GENMASK(9, 5) +#define EDTM_INITIAL_COUNT GENMASK(4, 0) +#define ETDM_IN1_CON2 0x64 +#define IN_CLK_SRC GENMASK(12, 10) +#define ETDM_IN1_CON3 0x68 +#define IN_SEL_FS GENMASK(30, 26) +#define ETDM_IN1_CON4 0x6c +#define IN_RELATCH GENMASK(24, 20) +#define IN_CLK_INV BIT(18) +#define ETDM_IN1_CON5 0x70 +#define ETDM_IN1_CON6 0x74 +#define ETDM_OUT1_CON0 0x7c +#define ETDM_SYNC_MODE BIT(1) +#define ETDM_OUT_EN BIT(0) +#define ETDM_OUT1_CON1 0x80 +#define ETDM_OUT1_CON2 0x84 +#define ETDM_OUT1_CON3 0x88 +#define ETDM_OUT1_CON4 0x8c +#define OUT_RELATCH GENMASK(28, 24) +#define OUT_CLK_SRC GENMASK(8, 6) +#define OUT_SEL_FS GENMASK(4, 0) +#define ETDM_OUT1_CON5 0x90 +#define ETDM_CLK_DIV BIT(12) +#define OUT_CLK_INV BIT(9) +#define ETDM_OUT1_CON6 0x94 +#define ETDM_OUT1_CON7 0x98 + +#define AFE_DL1_BASE 0xa8 +#define AFE_DL1_END 0xb0 +#define AFE_DL1_CUR 0xac +#define AFE_DL1_CON0 0xb4 +#define AFE_PBUF_SIZE_SHIFT 16 +#define AFE_PBUF_SIZE_MASK GENMASK(1, 0) +#define AFE_MINLEN_SHIFT 8 +#define AFE_MINLEN_MASK GENMASK(3, 0) +#define AFE_HD_SHIFT 5 + +#define AFE_UL1_BASE 0xc4 +#define AFE_UL1_END 0xc8 +#define AFE_UL1_CUR 0xcc +#define AFE_UL1_CON0 0xd0 +#define AFE_HD_ALIGN_SHIFT 6 + +#define AFE_IRQ_CON0 0xe4 +#define AFE_IRQ_EN_SEL BIT(4) +#define AFE_IRQ_EN_SEL_I2SIN FIELD_PREP_CONST(AFE_IRQ_EN_SEL, 0x0) +#define AFE_IRQ_EN_SEL_I2SOUT FIELD_PREP_CONST(AFE_IRQ_EN_SEL, 0x1) +#define AFE_IRQ_MISS_FLG_CLR_SHIFT 3 +#define AFE_IRQ_CLR_SHIFT 2 +#define AFE_IRQ_ON_SHIFT 0 +#define AFE_IRQ_CNT 0xe8 +#define AFE_IRQ_CNT_SHIFT 0 +#define AFE_IRQ_CNT_MASK GENMASK(31, 0) + +#define AFE_IRQ_STS 0xf8 +#define AFE_IRQ_STS_PLAY BIT(1) +#define AFE_IRQ_STS_RECORD BIT(0) + +#define AFE_MAX_REGISTER AFE_IRQ_STS + +#define AFE_IRQ1_CON0 0x0 +#define AFE_IRQ1_CNT 0x4 + +#define AFE_IRQ1_MAX_REGISTER AFE_IRQ1_CNT + +#endif From 272d6e51bc1f7db9ee58bb4029c95118a587c1c9 Mon Sep 17 00:00:00 2001 From: Christian Marangi Date: Tue, 8 Sep 2026 11:04:47 +0200 Subject: [PATCH 30/30] ASoC: airoha: Add machine driver for Airoha AN7581 Add support for the machine driver for Airoha AN7581 paired with the WM8960 i2c codec. This driver creates the sound card for the AFE driver. Signed-off-by: Christian Marangi Link: https://patch.msgid.link/20260908090448.3332-6-ansuelsmth@gmail.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 10 ++ sound/soc/mediatek/an7581/Makefile | 1 + sound/soc/mediatek/an7581/an7581-wm8960.c | 161 ++++++++++++++++++++++ 3 files changed, 172 insertions(+) create mode 100644 sound/soc/mediatek/an7581/an7581-wm8960.c diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index eb6f2a443b23..ff2869641a35 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -16,6 +16,16 @@ config SND_SOC_AN7581 Select Y if you have such device. If unsure select "N". +config SND_SOC_AN7581_WM8960 + tristate "ASoc Audio driver for Airoha AN7581 with WM8960 codec" + depends on SND_SOC_AN7581 && I2C + select SND_SOC_WM8960 + help + This adds support for ASoC machine driver for Airoha AN7581 + boards with the WM8960 codecs. + Select Y if you have such device. + If unsure select "N". + endmenu menu "Mediatek" diff --git a/sound/soc/mediatek/an7581/Makefile b/sound/soc/mediatek/an7581/Makefile index d6de5ee1668e..f48cd0269bea 100644 --- a/sound/soc/mediatek/an7581/Makefile +++ b/sound/soc/mediatek/an7581/Makefile @@ -6,3 +6,4 @@ snd-soc-an7581-afe-y := \ an7581-dai-etdm.o obj-$(CONFIG_SND_SOC_AN7581) += snd-soc-an7581-afe.o +obj-$(CONFIG_SND_SOC_AN7581_WM8960) += an7581-wm8960.o diff --git a/sound/soc/mediatek/an7581/an7581-wm8960.c b/sound/soc/mediatek/an7581/an7581-wm8960.c new file mode 100644 index 000000000000..03ea5732563c --- /dev/null +++ b/sound/soc/mediatek/an7581/an7581-wm8960.c @@ -0,0 +1,161 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Airoha ALSA SoC I2S platform driver for AN7581 + * + */ + +#include +#include + +#include "an7581-afe-common.h" + +SND_SOC_DAILINK_DEFS(playback, + DAILINK_COMP_ARRAY(COMP_CPU("DL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(capture, + DAILINK_COMP_ARRAY(COMP_CPU("UL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(codec, + DAILINK_COMP_ARRAY(COMP_CPU("ETDM")), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8960-hifi")), + DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); + +static struct snd_soc_dai_link an7581_wm8960_dai_links[] = { + /* FE */ + { + .name = "wm8960-playback", + .stream_name = "wm8960-playback", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .playback_only = 1, + SND_SOC_DAILINK_REG(playback), + }, + { + .name = "wm8960-capture", + .stream_name = "wm8960-capture", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .capture_only = 1, + SND_SOC_DAILINK_REG(capture), + }, + /* BE */ + { + .name = "wm8960-codec", + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBP_CFP | + SND_SOC_DAIFMT_GATED, + SND_SOC_DAILINK_REG(codec), + }, +}; + +static struct snd_soc_card an7581_wm8960_card = { + .name = "an7581-wm8960", + .owner = THIS_MODULE, + .dai_link = an7581_wm8960_dai_links, + .num_links = ARRAY_SIZE(an7581_wm8960_dai_links), +}; + +static int an7581_wm8960_machine_probe(struct platform_device *pdev) +{ + struct device_node *platform_dai_node, *codec_dai_node; + struct snd_soc_card *card = &an7581_wm8960_card; + struct device_node *of_platform, *codec; + struct snd_soc_dai_link *dai_link; + int i, d, ret; + + card->dev = &pdev->dev; + + of_platform = of_get_child_by_name(pdev->dev.of_node, "platform"); + + if (of_platform) { + platform_dai_node = of_parse_phandle(of_platform, "sound-dai", 0); + of_node_put(of_platform); + + if (!platform_dai_node) { + dev_err(&pdev->dev, "Failed to parse platform/sound-dai property\n"); + return -EINVAL; + } + } else { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for_each_card_prelinks(card, i, dai_link) { + struct snd_soc_dai_link_component *platform; + + dai_link->num_platforms = 2; + for_each_link_platforms(dai_link, d, platform) { + if (platform->name) + continue; + platform->of_node = platform_dai_node; + } + } + + codec = of_get_child_by_name(pdev->dev.of_node, "codec"); + + if (codec) { + codec_dai_node = of_parse_phandle(codec, "sound-dai", 0); + of_node_put(codec); + + if (!codec_dai_node) { + of_node_put(platform_dai_node); + dev_err(&pdev->dev, "Failed to parse codec/sound-dai property\n"); + return -EINVAL; + } + } else { + of_node_put(platform_dai_node); + dev_err(&pdev->dev, "Property 'codec' missing or invalid\n"); + return -EINVAL; + } + + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->codecs->name) + continue; + dai_link->codecs->of_node = codec_dai_node; + } + + ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Failed to parse audio-routing: %d\n", ret); + goto err_of_node_put; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) { + dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__); + goto err_of_node_put; + } + + return 0; + +err_of_node_put: + of_node_put(platform_dai_node); + of_node_put(codec_dai_node); + return ret; +} + +static const struct of_device_id an7581_wm8960_machine_dt_match[] = { + { .compatible = "airoha,an7581-wm8960-sound" }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, an7581_wm8960_machine_dt_match); + +static struct platform_driver an7581_wm8960_driver = { + .driver = { + .name = "an7581-wm8960", + .of_match_table = an7581_wm8960_machine_dt_match, + }, + .probe = an7581_wm8960_machine_probe, +}; +module_platform_driver(an7581_wm8960_driver); + +MODULE_DESCRIPTION("Airoha SoC I2S platform driver for ALSA AN7581"); +MODULE_LICENSE("GPL");