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[RESEND PATCH] ASoC: core: Exit all links before removing their components
by Cezary Rojewski 20 Nov '22

20 Nov '22

Flows leading to link->init() and link->exit() are not symmetric. Currently the relevant part of card probe sequence goes as: for_each_card_rtds(card, rtd) for_each_rtd_components(rtd, i, component) component->probe() for_each_card_rtds(card, rtd) for_each_rtd_dais(rtd, i, dai) dai->probe() for_each_card_rtds(card, rtd) rtd->init() On the other side, equivalent remove sequence goes as: for_each_card_rtds(card, rtd) for_each_rtd_dais(rtd, i, dai) dai->remove() for_each_card_rtds(card, rtd) for_each_rtd_components(rtd, i, component) component->remove() for_each_card_rtds(card, rtd) rtd->exit() what can lead to errors as link->exit() may still operate on resources owned by its components despite the probability of them being freed during the component->remove(). This change modifies the remove sequence to: for_each_card_rtds(card, rtd) rtd->exit() for_each_card_rtds(card, rtd) for_each_rtd_dais(rtd, i, dai) dai->remove() for_each_card_rtds(card, rtd) for_each_rtd_components(rtd, i, component) component->remove() so code found in link->exit() is safe to touch any component stuff as component->remove() has not been called yet. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- Original review for v1: https://lore.kernel.org/alsa-devel/20220621115758.3154933-1-cezary.rojewski… sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4a3fca50a536..638e781df3b0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -935,9 +935,6 @@ void snd_soc_remove_pcm_runtime(struct snd_soc_card *card, { lockdep_assert_held(&client_mutex); - /* release machine specific resources */ - snd_soc_link_exit(rtd); - /* * Notify the machine driver for extra destruction */ @@ -1888,6 +1885,9 @@ static void soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_shutdown(card); + /* release machine specific resources */ + for_each_card_rtds(card, rtd) + snd_soc_link_exit(rtd); /* remove and free each DAI */ soc_remove_link_dais(card); soc_remove_link_components(card); -- 2.25.1

2 2

es8336 HiFi.conf
by GitHub issues - opened 19 Nov '22

19 Nov '22

alsa-project/alsa-ucm-conf issue #244 was opened from junocomp: I believe the current HiFi.conf for es8336 is incorrect and does not enable the require volumes for the speakers. I have attached my conf file, this enables the speakers from startup, detects the HDMI audio and enables the internal microphone. I have tested this on 3 devices with Debian now and it seems to work. Can other people test it too please. If it works, it will be worth pushing it upstream. Make sure you backup your /usr/share/alsa/ucm2/Intel/sof-essx8336/HiFi.conf and replace it with the configuration below. Restart your machine to take effect. ``` SectionVerb { EnableSequence [ #disdevall "" # Disable all inputs / outputs # (may be duplicated with disdevall) cset "name='Left Headphone Mixer Left DAC Switch' on" cset "name='Right Headphone Mixer Right DAC Switch' on" ] } SectionDevice."Mic" { Comment "Analog Microphone" EnableSequence [ cset "name='Internal Mic Switch' on" cset "name='Headset Mic Switch' on" ] DisableSequence [ ] Value { CapturePriority 100 CapturePCM "hw:${CardId}" CaptureMixerElem "ADC PGA Gain" CaptureMasterElem "ADC" } } SectionDevice."Speaker" { Comment "Speakers" ConflictingDevice [ "Headphones" ] EnableSequence [ cset "name='Speaker Switch' on" cset "name='Differential Mux' lin1-rin1" ] DisableSequence [ cset "name='Speaker Switch' off" cset "name='Differential Mux' lin2-rin2" ] Value { PlaybackPriority 100 PlaybackPCM "hw:${CardId}" # The es8316 only has a HP-amp which is muxed to the speaker # or to the headpones output PlaybackMixerElem "Headphone Mixer" PlaybackMasterElem "DAC" } } SectionDevice."Headphones" { Comment "Headphones" ConflictingDevice [ "Speaker" ] EnableSequence [ cset "name='Headset Mic Switch' on" cset "name='Speaker Switch' off" cset "name='Differential Mux' lin2-rin2" ] DisableSequence [ cset "name='Differential Mux' lin1-rin1" ] Value { PlaybackPriority 300 PlaybackPCM "hw:${CardId}" PlaybackMixerElem "Headphone Mixer" PlaybackMasterElem "DAC" JackControl "Headphone Jack" JackHWMute "Speaker" } } Include.hdmi.File "/Intel/sof-essx8336/Hdmi.conf" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/244 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: seq: Fix function prototype mismatch in snd_seq_expand_var_event
by Kees Cook 19 Nov '22

19 Nov '22

With clang's kernel control flow integrity (kCFI, CONFIG_CFI_CLANG), indirect call targets are validated against the expected function pointer prototype to make sure the call target is valid to help mitigate ROP attacks. If they are not identical, there is a failure at run time, which manifests as either a kernel panic or thread getting killed. seq_copy_in_user() and seq_copy_in_kernel() did not have prototypes matching snd_seq_dump_func_t. Adjust this and remove the casts. There are not resulting binary output differences. This was found as a result of Clang's new -Wcast-function-type-strict flag, which is more sensitive than the simpler -Wcast-function-type, which only checks for type width mismatches. Reported-by: kernel test robot <lkp(a)intel.com> Link: https://lore.kernel.org/lkml/202211041527.HD8TLSE1-lkp@intel.com Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: "Gustavo A. R. Silva" <gustavoars(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/core/seq/seq_memory.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index b7aee23fc387..47ef6bc30c0e 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -113,15 +113,19 @@ EXPORT_SYMBOL(snd_seq_dump_var_event); * expand the variable length event to linear buffer space. */ -static int seq_copy_in_kernel(char **bufptr, const void *src, int size) +static int seq_copy_in_kernel(void *ptr, void *src, int size) { + char **bufptr = ptr; + memcpy(*bufptr, src, size); *bufptr += size; return 0; } -static int seq_copy_in_user(char __user **bufptr, const void *src, int size) +static int seq_copy_in_user(void *ptr, void *src, int size) { + char __user **bufptr = ptr; + if (copy_to_user(*bufptr, src, size)) return -EFAULT; *bufptr += size; @@ -151,8 +155,7 @@ int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char return newlen; } err = snd_seq_dump_var_event(event, - in_kernel ? (snd_seq_dump_func_t)seq_copy_in_kernel : - (snd_seq_dump_func_t)seq_copy_in_user, + in_kernel ? seq_copy_in_kernel : seq_copy_in_user, &buf); return err < 0 ? err : newlen; } -- 2.34.1

2 1

[PATCH AUTOSEL 5.15 25/27] ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue
by Sasha Levin 19 Nov '22

19 Nov '22

From: Ai Chao <aichao(a)kylinos.cn> [ Upstream commit bf990c10231937c0f51e5da5558e08cf5adc6a78 ] For Hamedal C20, the current rate is different from the runtime rate, snd_usb_endpoint stop and close endpoint to resetting rate. if snd_usb_endpoint close the endpoint, sometimes usb will disconnect the device. Signed-off-by: Ai Chao <aichao(a)kylinos.cn> Link: https://lore.kernel.org/r/20221110063452.295110-1-aichao@kylinos.cn Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/endpoint.c | 3 ++- sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 3 +++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 3bbc227769d0..092350eb5f4e 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -887,7 +887,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->iface_ref->opened) + if (!--ep->iface_ref->opened && + !(chip->quirk_flags & QUIRK_FLAG_IFACE_SKIP_CLOSE)) endpoint_set_interface(chip, ep, false); if (!--ep->opened) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 879d8b1f301c..2ae9ad993ff4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1884,6 +1884,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0525, 0xa4ad, /* Hamedal C20 usb camero */ + QUIRK_FLAG_IFACE_SKIP_CLOSE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 39c3c61a7e49..ec06f441e890 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -169,6 +169,8 @@ extern bool snd_usb_skip_validation; * Apply the generic implicit feedback sync mode (same as implicit_fb=1 option) * QUIRK_FLAG_SKIP_IMPLICIT_FB * Don't apply implicit feedback sync mode + * QUIRK_FLAG_IFACE_SKIP_CLOSE + * Don't closed interface during setting sample rate */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -190,5 +192,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) #define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) #define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) +#define QUIRK_FLAG_IFACE_SKIP_CLOSE (1U << 19) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 41/44] ALSA: usb-audio: add quirk to fix Hamedal C20 disconnect issue
by Sasha Levin 19 Nov '22

19 Nov '22

From: Ai Chao <aichao(a)kylinos.cn> [ Upstream commit bf990c10231937c0f51e5da5558e08cf5adc6a78 ] For Hamedal C20, the current rate is different from the runtime rate, snd_usb_endpoint stop and close endpoint to resetting rate. if snd_usb_endpoint close the endpoint, sometimes usb will disconnect the device. Signed-off-by: Ai Chao <aichao(a)kylinos.cn> Link: https://lore.kernel.org/r/20221110063452.295110-1-aichao@kylinos.cn Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/endpoint.c | 3 ++- sound/usb/quirks.c | 2 ++ sound/usb/usbaudio.h | 3 +++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 2420dc994632..4c9ea13f72d4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -923,7 +923,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n", ep->ep_num, ep->opened); - if (!--ep->iface_ref->opened) + if (!--ep->iface_ref->opened && + !(chip->quirk_flags & QUIRK_FLAG_IFACE_SKIP_CLOSE)) endpoint_set_interface(chip, ep, false); if (!--ep->opened) { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index eadac586bcc8..a50e15be1229 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2185,6 +2185,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x0525, 0xa4ad, /* Hamedal C20 usb camero */ + QUIRK_FLAG_IFACE_SKIP_CLOSE), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 2c6575029b1c..e97141ef730a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -170,6 +170,8 @@ extern bool snd_usb_skip_validation; * Apply the generic implicit feedback sync mode (same as implicit_fb=1 option) * QUIRK_FLAG_SKIP_IMPLICIT_FB * Don't apply implicit feedback sync mode + * QUIRK_FLAG_IFACE_SKIP_CLOSE + * Don't closed interface during setting sample rate */ #define QUIRK_FLAG_GET_SAMPLE_RATE (1U << 0) @@ -191,5 +193,6 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) #define QUIRK_FLAG_GENERIC_IMPLICIT_FB (1U << 17) #define QUIRK_FLAG_SKIP_IMPLICIT_FB (1U << 18) +#define QUIRK_FLAG_IFACE_SKIP_CLOSE (1U << 19) #endif /* __USBAUDIO_H */ -- 2.35.1

1 0

[PATCH] ASoC: Intel: avs: Initialize private data for subsequent HDA FEs
by Cezary Rojewski 18 Nov '22

18 Nov '22

HDAudio implementation found in sound/pci/hda expects a valid stream pointer in substream->runtime->private_data location. For ASoC users, that should point to a valid link stream which is assigned when BE opens. As BE borrows its runtime from FE, the information may be lost when reparenting comes into picture - see dpcm_be_reparent(). To support the DPCM reparenting functionality for HDAudio scenarios while still fulfilling expectations of HDAudio common code, have all FEs point to the same private data. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- This is a tough one and spawned a range of patches. Some of the discussion is present at thesofproject/linux [1]. In the end decided to drop any soc-core/pcm.c modifications for now, instead addressing the problem directly in the avs-driver. Long term - perhaps BE should have a separate runtime so there is no ambiguity between FE and BE. [1]: https://github.com/thesofproject/linux/pull/3987 sound/soc/intel/avs/pcm.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index e237ab23fef9..41030aaae02d 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -1507,9 +1507,29 @@ static int avs_component_hda_open(struct snd_soc_component *component, if (!rtd->dai_link->no_pcm) { struct snd_pcm_hardware hwparams = avs_pcm_hardware; + struct snd_soc_pcm_runtime *be; + struct snd_soc_dpcm *dpcm; + int dir = substream->stream; + + /* + * Support the DPCM reparenting while still fulfilling expectations of HDAudio + * common code - a valid stream pointer at substream->runtime->private_data - + * by having all FEs point to the same private data. + */ + for_each_dpcm_be(rtd, dir, dpcm) { + struct snd_pcm_substream *be_substream; + + be = dpcm->be; + if (be->dpcm[dir].users == 1) + break; + + be_substream = snd_soc_dpcm_get_substream(be, dir); + substream->runtime->private_data = be_substream->runtime->private_data; + break; + } /* RESUME unsupported for de-coupled HD-Audio capture. */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + if (dir == SNDRV_PCM_STREAM_CAPTURE) hwparams.info &= ~SNDRV_PCM_INFO_RESUME; return snd_soc_set_runtime_hwparams(substream, &hwparams); -- 2.25.1

2 1

[PATCH] ASoC: Intel: sof_sdw_amp: mark coeff tables with __maybe_unused
by Bard Liao 18 Nov '22

18 Nov '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The same file provides two tables used in separate drivers, make them as __maybe_unused to avoid errors: sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h:163:17: error: ‘dell_0b00_bq_params’ defined but not used [-Werror=unused-const-variable=] 163 | static const u8 dell_0b00_bq_params[] = { Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h b/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h index 82230fbb5b05..4a3e6fdbd623 100644 --- a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h +++ b/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h @@ -11,7 +11,7 @@ #define RT1308_MAX_BQ_REG 480 #define RT1316_MAX_BQ_REG 580 -static const u8 dell_0a5d_bq_params[] = { +static const u8 __maybe_unused dell_0a5d_bq_params[] = { 0xb0, 0xc5, 0x00, /* address: 0xc5b0; data: 0x00 */ 0xb1, 0xc5, 0x32, 0xb2, 0xc5, 0x44, @@ -160,7 +160,7 @@ static const u8 dell_0a5d_bq_params[] = { 0x50, 0xc5, 0x01, }; -static const u8 dell_0b00_bq_params[] = { +static const u8 __maybe_unused dell_0b00_bq_params[] = { 0x03, 0xc2, 0x00, 0x04, 0xc2, 0xb2, 0x05, 0xc2, 0xe0, -- 2.25.1

2 1

[PATCH 0/3] ASoC: mchp-spdiftx: add power saving features
by Claudiu Beznea 18 Nov '22

18 Nov '22

Hi, This series adds support for runtime PM and system suspend/resume for Microchip SPDIFTX (patches 2/3, 3/3). Along with it I took the chance and added a minor cleanup (patch 1/3). Thank you, Claudiu Beznea Claudiu Beznea (3): ASoC: mchp-spdiftx: simplify locking around ctrl->ch_stat ASoC: mchp-spdiftx: add runtime pm support ASoC: mchp-spdiftx: add support for system suspend/resume sound/soc/atmel/mchp-spdiftx.c | 165 ++++++++++++++++++++++----------- 1 file changed, 113 insertions(+), 52 deletions(-) -- 2.34.1

2 4

[PATCH v2] ASoC: amd: acp: Fix possible UAF in acp_dma_open
by Gaosheng Cui 18 Nov '22

18 Nov '22

Smatch report warning as follows: sound/soc/amd/acp/acp-platform.c:199 acp_dma_open() warn: '&stream->list' not removed from list If snd_pcm_hw_constraint_integer() fails in acp_dma_open(), stream will be freed, but stream->list will not be removed from adata->stream_list, then list traversal may cause UAF. Fix by adding the newly allocated stream to the list once it's fully initialised. Fixes: 7929985cfe36 ("ASoC: amd: acp: Initialize list to store acp_stream during pcm_open") Signed-off-by: Gaosheng Cui <cuigaosheng1(a)huawei.com> --- v2: - Fix by adding the newly allocated stream to the list once it's fully initialised. v1: - Fix by removeing it from adata->stream_list before free(). sound/soc/amd/acp/acp-platform.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 85a81add4ef9..447612a7a762 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -184,10 +184,6 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs stream->substream = substream; - spin_lock_irq(&adata->acp_lock); - list_add_tail(&stream->list, &adata->stream_list); - spin_unlock_irq(&adata->acp_lock); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; else @@ -203,6 +199,10 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs writel(1, ACP_EXTERNAL_INTR_ENB(adata)); + spin_lock_irq(&adata->acp_lock); + list_add_tail(&stream->list, &adata->stream_list); + spin_unlock_irq(&adata->acp_lock); + return ret; } -- 2.25.1

2 1

[PATCH] ASoC: Intel: sof_nau8825: support rt1015p speaker amplifier
by Pierre-Louis Bossart 18 Nov '22

18 Nov '22

From: Brent Lu <brent.lu(a)intel.com> Add rt1015p speaker amplifier support with a new board info 'adl_rt1015p_nau8825' which supports NAU8825 on SSP0 and ALC1015Q on SSP1. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_nau8825.c | 16 ++++++++++++++++ .../soc/intel/common/soc-acpi-intel-adl-match.c | 12 ++++++++++++ 2 files changed, 28 insertions(+) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 5585c217f78d3..27880224359d5 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -47,6 +47,7 @@ #define SOF_RT1019P_SPEAKER_AMP_PRESENT BIT(14) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) +#define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(17) static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); @@ -483,6 +484,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_nau8825_quirk & SOF_MAX98360A_SPEAKER_AMP_PRESENT) { max_98360a_dai_link(&links[id]); + } else if (sof_nau8825_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) { + sof_rt1015p_dai_link(&links[id]); } else { goto devm_err; } @@ -576,6 +579,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) max_98373_set_codec_conf(&sof_audio_card_nau8825); + else if (sof_nau8825_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) + sof_rt1015p_codec_conf(&sof_audio_card_nau8825); if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_nau8825.num_links++; @@ -642,6 +647,16 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .name = "adl_rt1015p_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015P_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); @@ -663,3 +678,4 @@ MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_REALTEK_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 3ca0a96d94de8..60aee56f94bd4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -430,6 +430,11 @@ static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs adl_rt1015p_amp = { + .num_codecs = 1, + .codecs = {"RTL1015"} +}; + static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { .num_codecs = 1, .codecs = {"RTL1019"} @@ -495,6 +500,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .quirk_data = &adl_rt1019p_amp, .sof_tplg_filename = "sof-adl-rt1019-rt5682.tplg", }, + { + .id = "10508825", + .drv_name = "adl_rt1015p_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_rt1015p_amp, + .sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg", + }, { .id = "10508825", .drv_name = "sof_nau8825", -- 2.34.1

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