From 417b9c51f59734d852e47252476fadc293ad994a Mon Sep 17 00:00:00 2001 From: Callum Osmotherly Date: Wed, 5 Oct 2022 17:44:16 +1030 Subject: ALSA: hda/realtek: remove ALC289_FIXUP_DUAL_SPK for Dell 5530 After some feedback from users with Dell Precision 5530 machines, this patch reverts the previous change to add ALC289_FIXUP_DUAL_SPK. While it improved the speaker output quality, it caused the headphone jack to have an audible "pop" sound when power saving was toggled. Fixes: 1885ff13d4c4 ("ALSA: hda/realtek: Enable 4-speaker output Dell Precision 5530 laptop") Signed-off-by: Callum Osmotherly Cc: Link: https://lore.kernel.org/r/Yz0uyN1zwZhnyRD6@piranha Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bce82b834cec..d89f95ae0efc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9198,7 +9198,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB), - SND_PCI_QUIRK(0x1028, 0x087d, "Dell Precision 5530", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), -- cgit v1.2.3 From 9902b303b5ade208b58f0dd38a09831813582211 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 9 Oct 2022 12:42:09 +0200 Subject: ALSA: usb-audio: Avoid unnecessary interface change at EP close We toggle USB interface at PCM prepare and reset at close. When the PCM isn't prepared, resetting again makes little sense. Check the current altset and avoid unnecessary interface reset at EP close. Link: https://lore.kernel.org/r/20221009104212.18877-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 48a3843a08f1..f21acbc9f4f4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -32,6 +32,7 @@ struct snd_usb_iface_ref { unsigned char iface; bool need_setup; int opened; + int altset; struct list_head list; }; @@ -899,6 +900,9 @@ static int endpoint_set_interface(struct snd_usb_audio *chip, int altset = set ? ep->altsetting : 0; int err; + if (ep->iface_ref->altset == altset) + return 0; + usb_audio_dbg(chip, "Setting usb interface %d:%d for EP 0x%x\n", ep->iface, altset, ep->ep_num); err = usb_set_interface(chip->dev, ep->iface, altset); @@ -910,6 +914,7 @@ static int endpoint_set_interface(struct snd_usb_audio *chip, if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) msleep(50); + ep->iface_ref->altset = altset; return 0; } -- cgit v1.2.3 From a74f8d0aa902ca494676b79226e0b5a1747b81d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 9 Oct 2022 12:42:10 +0200 Subject: ALSA: usb-audio: Apply mutex around snd_usb_endpoint_set_params() The protection with chip->mutex was lost after splitting snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(). Apply the same mutex again to the former function. Fixes: 2be79d586454 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)") Link: https://lore.kernel.org/r/20221009104212.18877-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index f21acbc9f4f4..da378e565ef8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1337,10 +1337,11 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, const struct audioformat *fmt = ep->cur_audiofmt; int err; + mutex_lock(&chip->mutex); /* release old buffers, if any */ err = release_urbs(ep, false); if (err < 0) - return err; + goto unlock; ep->datainterval = fmt->datainterval; ep->maxpacksize = fmt->maxpacksize; @@ -1378,13 +1379,16 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, usb_audio_dbg(chip, "Set up %d URBS, ret=%d\n", ep->nurbs, err); if (err < 0) - return err; + goto unlock; /* some unit conversions in runtime */ ep->maxframesize = ep->maxpacksize / ep->cur_frame_bytes; ep->curframesize = ep->curpacksize / ep->cur_frame_bytes; - return update_clock_ref_rate(chip, ep); + err = update_clock_ref_rate(chip, ep); + unlock: + mutex_unlock(&chip->mutex); + return err; } static int init_sample_rate(struct snd_usb_audio *chip, -- cgit v1.2.3 From 9355b60e401d825590d37f04ea873c58efe9b7bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 9 Oct 2022 12:42:11 +0200 Subject: ALSA: usb-audio: Correct the return code from snd_usb_endpoint_set_params() snd_usb_endpoint_set_params() should return zero for a success, but currently it returns the sample rate. Correct it. Fixes: 2be79d586454 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)") Link: https://lore.kernel.org/r/20221009104212.18877-4-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index da378e565ef8..44cce6cec9da 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1386,6 +1386,8 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, ep->curframesize = ep->curpacksize / ep->cur_frame_bytes; err = update_clock_ref_rate(chip, ep); + if (err >= 0) + err = 0; unlock: mutex_unlock(&chip->mutex); return err; -- cgit v1.2.3 From 1045f5f1ff0751423aeb65648e5e1abd7a7a8672 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 9 Oct 2022 12:42:12 +0200 Subject: ALSA: usb-audio: Avoid superfluous endpoint setup After splitting to snd_usb_endpoint_set_params() and *_prepare(), the skip of each function should be checked with different flags, while we still use ep->need_setup as the single one. Introduce ep->need_prepare for indicating the need of prepare, and also add the missing check of ep->need_setup at the set_params. Fixes: 2be79d586454 ("ALSA: usb-audio: Split endpoint setups for hw_params and prepare (take#2)") Link: https://lore.kernel.org/r/20221009104212.18877-5-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/card.h | 3 ++- sound/usb/endpoint.c | 17 ++++++++++++----- 2 files changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index ca75f2206170..40061550105a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -129,7 +129,8 @@ struct snd_usb_endpoint { in a stream */ bool implicit_fb_sync; /* syncs with implicit feedback */ bool lowlatency_playback; /* low-latency playback mode */ - bool need_setup; /* (re-)need for configure? */ + bool need_setup; /* (re-)need for hw_params? */ + bool need_prepare; /* (re-)need for prepare? */ /* for hw constraints */ const struct audioformat *cur_audiofmt; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 44cce6cec9da..d0b8d61d1d22 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -824,6 +824,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->implicit_fb_sync = fp->implicit_fb; ep->need_setup = true; + ep->need_prepare = true; usb_audio_dbg(chip, " channels=%d, rate=%d, format=%s, period_bytes=%d, periods=%d, implicit_fb=%d\n", ep->cur_channels, ep->cur_rate, @@ -952,7 +953,7 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, /* Prepare for suspening EP, called from the main suspend handler */ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) { - ep->need_setup = true; + ep->need_prepare = true; if (ep->iface_ref) ep->iface_ref->need_setup = true; if (ep->clock_ref) @@ -1335,9 +1336,12 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; - int err; + int err = 0; mutex_lock(&chip->mutex); + if (!ep->need_setup) + goto unlock; + /* release old buffers, if any */ err = release_urbs(ep, false); if (err < 0) @@ -1386,8 +1390,11 @@ int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, ep->curframesize = ep->curpacksize / ep->cur_frame_bytes; err = update_clock_ref_rate(chip, ep); - if (err >= 0) + if (err >= 0) { + ep->need_setup = false; err = 0; + } + unlock: mutex_unlock(&chip->mutex); return err; @@ -1437,7 +1444,7 @@ int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, mutex_lock(&chip->mutex); if (WARN_ON(!ep->iface_ref)) goto unlock; - if (!ep->need_setup) + if (!ep->need_prepare) goto unlock; /* If the interface has been already set up, just set EP parameters */ @@ -1491,7 +1498,7 @@ int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, ep->iface_ref->need_setup = false; done: - ep->need_setup = false; + ep->need_prepare = false; err = 1; unlock: -- cgit v1.2.3 From 66ba7c88507344dee68ad1acbdb630473ab36114 Mon Sep 17 00:00:00 2001 From: "Luke D. Jones" Date: Mon, 10 Oct 2022 19:57:02 +1300 Subject: ALSA: hda/realtek: Correct pin configs for ASUS G533Z The initial fix for ASUS G533Z was based on faulty information. This fixes the pincfg to values that have been verified with no existing module options or other hacks enabled. Enables headphone jack, and 5.1 surround. [ corrected the indent level by tiwai ] Fixes: bc2c23549ccd ("ALSA: hda/realtek: Add pincfg for ASUS G533Z HP jack") Signed-off-by: Luke D. Jones Cc: Link: https://lore.kernel.org/r/20221010065702.35190-1-luke@ljones.dev Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d89f95ae0efc..77a308a71cd4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8449,11 +8449,13 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_ASUS_G533Z_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x90170120 }, + { 0x14, 0x90170152 }, /* Speaker Surround Playback Switch */ + { 0x19, 0x03a19020 }, /* Mic Boost Volume */ + { 0x1a, 0x03a11c30 }, /* Mic Boost Volume */ + { 0x1e, 0x90170151 }, /* Rear jack, IN OUT EAPD Detect */ + { 0x21, 0x03211420 }, { } }, - .chained = true, - .chain_id = ALC294_FIXUP_ASUS_G513_PINS, }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, -- cgit v1.2.3 From 2ea8e1297801f7b0220ebf6ae61a5b74ca83981e Mon Sep 17 00:00:00 2001 From: "Luke D. Jones" Date: Mon, 10 Oct 2022 20:03:47 +1300 Subject: ALSA: hda/realtek: Add quirk for ASUS GV601R laptop The ASUS ROG X16 (GV601R) series laptop has the same node-to-DAC pairs as early models and the G14, this includes bass speakers which are by default mapped incorrectly to the 0x06 node. Add a quirk to use the same DAC pairs as the G14. Signed-off-by: Luke D. Jones Cc: Link: https://lore.kernel.org/r/20221010070347.36883-1-luke@ljones.dev Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 77a308a71cd4..54a0f6b4ffc7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9423,6 +9423,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- cgit v1.2.3 From 4f2e56a59b9947b3e698d3cabcb858765c12b1e8 Mon Sep 17 00:00:00 2001 From: Saranya Gopal Date: Tue, 11 Oct 2022 10:19:16 +0530 Subject: ALSA: hda/realtek: Add Intel Reference SSID to support headset keys This patch fixes the issue with 3.5mm headset keys on RPL-P platform. [ Rearranged the entry in SSID order by tiwai ] Signed-off-by: Saranya Gopal Signed-off-by: Ninad Naik Cc: Link: https://lore.kernel.org/r/20221011044916.2278867-1-saranya.gopal@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 54a0f6b4ffc7..4b076912bbf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9445,6 +9445,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_HEADSET_MODE), -- cgit v1.2.3 From a70aef7982b012e86dfd39fbb235e76a21ae778a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Oct 2022 09:01:46 +0200 Subject: ALSA: rawmidi: Drop register_mutex in snd_rawmidi_free() The register_mutex taken around the dev_unregister callback call in snd_rawmidi_free() may potentially lead to a mutex deadlock, when OSS emulation and a hot unplug are involved. Since the mutex doesn't protect the actual race (as the registration itself is already protected by another means), let's drop it. Link: https://lore.kernel.org/r/CAB7eexJP7w1B0mVgDF0dQ+gWor7UdkiwPczmL7pn91xx8xpzOA@mail.gmail.com Cc: Link: https://lore.kernel.org/r/20221011070147.7611-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6963d5a487b3..d8edb6055072 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1899,10 +1899,8 @@ static int snd_rawmidi_free(struct snd_rawmidi *rmidi) snd_info_free_entry(rmidi->proc_entry); rmidi->proc_entry = NULL; - mutex_lock(®ister_mutex); if (rmidi->ops && rmidi->ops->dev_unregister) rmidi->ops->dev_unregister(rmidi); - mutex_unlock(®ister_mutex); snd_rawmidi_free_substreams(&rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]); snd_rawmidi_free_substreams(&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]); -- cgit v1.2.3 From 97d917879d7f92df09c3f21fd54609a8bcd654b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Oct 2022 09:01:47 +0200 Subject: ALSA: oss: Fix potential deadlock at unregistration We took sound_oss_mutex around the calls of unregister_sound_special() at unregistering OSS devices. This may, however, lead to a deadlock, because we manage the card release via the card's device object, and the release may happen at unregister_sound_special() call -- which will take sound_oss_mutex again in turn. Although the deadlock might be fixed by relaxing the rawmidi mutex in the previous commit, it's safer to move unregister_sound_special() calls themselves out of the sound_oss_mutex, too. The call is race-safe as the function has a spinlock protection by itself. Link: https://lore.kernel.org/r/CAB7eexJP7w1B0mVgDF0dQ+gWor7UdkiwPczmL7pn91xx8xpzOA@mail.gmail.com Cc: Link: https://lore.kernel.org/r/20221011070147.7611-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/sound_oss.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 7ed0a2a91035..2751bf2ff61b 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -162,7 +162,6 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) mutex_unlock(&sound_oss_mutex); return -ENOENT; } - unregister_sound_special(minor); switch (SNDRV_MINOR_OSS_DEVICE(minor)) { case SNDRV_MINOR_OSS_PCM: track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_AUDIO); @@ -174,12 +173,18 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) track2 = SNDRV_MINOR_OSS(cidx, SNDRV_MINOR_OSS_DMMIDI1); break; } - if (track2 >= 0) { - unregister_sound_special(track2); + if (track2 >= 0) snd_oss_minors[track2] = NULL; - } snd_oss_minors[minor] = NULL; mutex_unlock(&sound_oss_mutex); + + /* call unregister_sound_special() outside sound_oss_mutex; + * otherwise may deadlock, as it can trigger the release of a card + */ + unregister_sound_special(minor); + if (track2 >= 0) + unregister_sound_special(track2); + kfree(mptr); return 0; } -- cgit v1.2.3 From 49b0dea1eb5e0fd5e498a2c2ce50d2e036494072 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 11 Oct 2022 15:35:48 +0100 Subject: ALSA: hda: hda_cs_dsp_ctl: Minor clean and redundant code removal The cs_dsp core will return an error if passed a NULL cs_dsp struct so there is no need for the hda_cs_dsp_write|read_ctl functions to manually check that. The cs_dsp core will also check the data is within bounds of the control so the additional bounds check is redundant too. Simplify things a bit by removing said code. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20221011143552.621792-2-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_cs_dsp_ctl.c | 17 ++++------------- 1 file changed, 4 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 89ee549cb7d5..41d3e8fd289d 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -199,16 +199,10 @@ EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg, const void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; + struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); struct hda_cs_dsp_coeff_ctl *ctl; int ret; - cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; - - ctl = cs_ctl->priv; - ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); if (ret) return ret; @@ -216,6 +210,8 @@ int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) return 0; + ctl = cs_ctl->priv; + snd_ctl_notify(ctl->card, SNDRV_CTL_EVENT_MASK_VALUE, &ctl->kctl->id); return 0; @@ -225,13 +221,8 @@ EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_write_ctl, SND_HDA_CS_DSP_CONTROLS); int hda_cs_dsp_read_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl; - - cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); - if (!cs_ctl) - return -EINVAL; + return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(dsp, name, type, alg), 0, buf, len); - return cs_dsp_coeff_read_ctrl(cs_ctl, 0, buf, len); } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_read_ctl, SND_HDA_CS_DSP_CONTROLS); -- cgit v1.2.3 From 06f3a0a758c4246dc644e22fb33f85c6e5f92af6 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 11 Oct 2022 15:35:49 +0100 Subject: ALSA: hda: hda_cs_dsp_ctl: Ensure pwr_lock is held before reading/writing controls These apis require the pwr_lock to be held. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20221011143552.621792-3-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_cs_dsp_ctl.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 41d3e8fd289d..75fb69185817 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -199,11 +199,14 @@ EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_remove, SND_HDA_CS_DSP_CONTROLS); int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg, const void *buf, size_t len) { - struct cs_dsp_coeff_ctl *cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); + struct cs_dsp_coeff_ctl *cs_ctl; struct hda_cs_dsp_coeff_ctl *ctl; int ret; + mutex_lock(&dsp->pwr_lock); + cs_ctl = cs_dsp_get_ctl(dsp, name, type, alg); ret = cs_dsp_coeff_write_ctrl(cs_ctl, 0, buf, len); + mutex_unlock(&dsp->pwr_lock); if (ret) return ret; @@ -221,7 +224,13 @@ EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_write_ctl, SND_HDA_CS_DSP_CONTROLS); int hda_cs_dsp_read_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg, void *buf, size_t len) { - return cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(dsp, name, type, alg), 0, buf, len); + int ret; + + mutex_lock(&dsp->pwr_lock); + ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(dsp, name, type, alg), 0, buf, len); + mutex_unlock(&dsp->pwr_lock); + + return ret; } EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_read_ctl, SND_HDA_CS_DSP_CONTROLS); -- cgit v1.2.3 From 2176c6b599dba55a640cffec0182c0b6bab680d1 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 11 Oct 2022 15:35:50 +0100 Subject: ALSA: hda/cs_dsp_ctl: Fix mutex inversion when creating controls Redesign the creation of ALSA controls so that the cs_dsp pwr_lock is not held when calling snd_ctl_add(). Instead of creating the ALSA control from the cs_dsp control_add callback, do it after cs_dsp_power_up() has completed. The existing functions are changed to return void instead of passing errors back - this duplicates the original behaviour, as cs_dsp does not abort firmware load if creation of a control fails. It is safe to walk the control list without taking any mutex provided that the caller is not trying to load a new firmware or remove the driver in parallel. There is no other situation that the list can change. So the caller can trigger creation of ALSA controls after cs_dsp_power_up() has returned. A cs_dsp control will have a non-NULL priv pointer if we have created an ALSA control. With the previous code the ALSA controls were created from the cs_dsp control_add callback. But this is called with pwr_lock held (as it is part of the DSP power-up sequence). The kernel lock checking will show a mutex inversion between this and the control creation path: control_add pwr_lock held, takes controls_rwsem (in snd_ctl_add) get/put controls_rwsem held, takes pwr_lock to call cs_dsp. This is not completely theoretical. Although the time window is very small, it is possible for these to run in parallel and deadlock the old implementation. Signed-off-by: Richard Fitzgerald Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20221011143552.621792-4-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 8 +++--- sound/pci/hda/hda_cs_dsp_ctl.c | 59 +++++++++++++++++++++++++----------------- sound/pci/hda/hda_cs_dsp_ctl.h | 2 +- 3 files changed, 40 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 3952f2853703..102ac4a94a9d 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -91,20 +91,18 @@ static const struct reg_sequence cs35l41_hda_mute[] = { { CS35L41_AMP_DIG_VOL_CTRL, 0x0000A678 }, // AMP_VOL_PCM Mute }; -static int cs35l41_control_add(struct cs_dsp_coeff_ctl *cs_ctl) +static void cs35l41_add_controls(struct cs35l41_hda *cs35l41) { - struct cs35l41_hda *cs35l41 = container_of(cs_ctl->dsp, struct cs35l41_hda, cs_dsp); struct hda_cs_dsp_ctl_info info; info.device_name = cs35l41->amp_name; info.fw_type = cs35l41->firmware_type; info.card = cs35l41->codec->card; - return hda_cs_dsp_control_add(cs_ctl, &info); + hda_cs_dsp_add_controls(&cs35l41->cs_dsp, &info); } static const struct cs_dsp_client_ops client_ops = { - .control_add = cs35l41_control_add, .control_remove = hda_cs_dsp_control_remove, }; @@ -435,6 +433,8 @@ static int cs35l41_init_dsp(struct cs35l41_hda *cs35l41) if (ret) goto err_release; + cs35l41_add_controls(cs35l41); + ret = cs35l41_save_calibration(cs35l41); err_release: diff --git a/sound/pci/hda/hda_cs_dsp_ctl.c b/sound/pci/hda/hda_cs_dsp_ctl.c index 75fb69185817..1622a22f96f6 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.c +++ b/sound/pci/hda/hda_cs_dsp_ctl.c @@ -97,7 +97,7 @@ static unsigned int wmfw_convert_flags(unsigned int in) return out; } -static int hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name) +static void hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char *name) { struct cs_dsp_coeff_ctl *cs_ctl = ctl->cs_ctl; struct snd_kcontrol_new kcontrol = {0}; @@ -107,7 +107,7 @@ static int hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char if (cs_ctl->len > ADSP_MAX_STD_CTRL_SIZE) { dev_err(cs_ctl->dsp->dev, "KControl %s: length %zu exceeds maximum %d\n", name, cs_ctl->len, ADSP_MAX_STD_CTRL_SIZE); - return -EINVAL; + return; } kcontrol.name = name; @@ -120,24 +120,21 @@ static int hda_cs_dsp_add_kcontrol(struct hda_cs_dsp_coeff_ctl *ctl, const char /* Save ctl inside private_data, ctl is owned by cs_dsp, * and will be freed when cs_dsp removes the control */ kctl = snd_ctl_new1(&kcontrol, (void *)ctl); - if (!kctl) { - ret = -ENOMEM; - return ret; - } + if (!kctl) + return; ret = snd_ctl_add(ctl->card, kctl); if (ret) { dev_err(cs_ctl->dsp->dev, "Failed to add KControl %s = %d\n", kcontrol.name, ret); - return ret; + return; } dev_dbg(cs_ctl->dsp->dev, "Added KControl: %s\n", kcontrol.name); ctl->kctl = kctl; - - return 0; } -int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ctl_info *info) +static void hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, + const struct hda_cs_dsp_ctl_info *info) { struct cs_dsp *cs_dsp = cs_ctl->dsp; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -145,13 +142,10 @@ int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ct const char *region_name; int ret; - if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) - return 0; - region_name = cs_dsp_mem_region_name(cs_ctl->alg_region.type); if (!region_name) { - dev_err(cs_dsp->dev, "Unknown region type: %d\n", cs_ctl->alg_region.type); - return -EINVAL; + dev_warn(cs_dsp->dev, "Unknown region type: %d\n", cs_ctl->alg_region.type); + return; } ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %.12s %x", info->device_name, @@ -171,22 +165,39 @@ int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ct ctl = kzalloc(sizeof(*ctl), GFP_KERNEL); if (!ctl) - return -ENOMEM; + return; ctl->cs_ctl = cs_ctl; ctl->card = info->card; cs_ctl->priv = ctl; - ret = hda_cs_dsp_add_kcontrol(ctl, name); - if (ret) { - dev_err(cs_dsp->dev, "Error (%d) adding control %s\n", ret, name); - kfree(ctl); - return ret; - } + hda_cs_dsp_add_kcontrol(ctl, name); +} - return 0; +void hda_cs_dsp_add_controls(struct cs_dsp *dsp, const struct hda_cs_dsp_ctl_info *info) +{ + struct cs_dsp_coeff_ctl *cs_ctl; + + /* + * pwr_lock would cause mutex inversion with ALSA control lock compared + * to the get/put functions. + * It is safe to walk the list without holding a mutex because entries + * are persistent and only cs_dsp_power_up() or cs_dsp_remove() can + * change the list. + */ + lockdep_assert_not_held(&dsp->pwr_lock); + + list_for_each_entry(cs_ctl, &dsp->ctl_list, list) { + if (cs_ctl->flags & WMFW_CTL_FLAG_SYS) + continue; + + if (cs_ctl->priv) + continue; + + hda_cs_dsp_control_add(cs_ctl, info); + } } -EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_control_add, SND_HDA_CS_DSP_CONTROLS); +EXPORT_SYMBOL_NS_GPL(hda_cs_dsp_add_controls, SND_HDA_CS_DSP_CONTROLS); void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl) { diff --git a/sound/pci/hda/hda_cs_dsp_ctl.h b/sound/pci/hda/hda_cs_dsp_ctl.h index 4babc69cf2f0..2cf93359c4f2 100644 --- a/sound/pci/hda/hda_cs_dsp_ctl.h +++ b/sound/pci/hda/hda_cs_dsp_ctl.h @@ -29,7 +29,7 @@ struct hda_cs_dsp_ctl_info { extern const char * const hda_cs_dsp_fw_ids[HDA_CS_DSP_NUM_FW]; -int hda_cs_dsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl, struct hda_cs_dsp_ctl_info *info); +void hda_cs_dsp_add_controls(struct cs_dsp *dsp, const struct hda_cs_dsp_ctl_info *info); void hda_cs_dsp_control_remove(struct cs_dsp_coeff_ctl *cs_ctl); int hda_cs_dsp_write_ctl(struct cs_dsp *dsp, const char *name, int type, unsigned int alg, const void *buf, size_t len); -- cgit v1.2.3 From 23904f7b2518e9b6bbfe2ac7bbe9e284bcdda18e Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 11 Oct 2022 15:35:51 +0100 Subject: ALSA: hda: cs35l41: Remove suspend/resume hda hooks The current code uses calls from the HDA Codec driver to determine when to suspend/resume by calling hooks via the hda_component binding. However, this means the cs35l41 driver relies on the HDA Codec driver to tell it when to suspend or resume, creating an additional external dependency, and potentially creating race conditions in the future. It is better for the cs35l41 hda driver to decide for itself when the part should be suspended or resumed. This makes supporting system suspend easier. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20221011143552.621792-5-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 31 ++++++++++++------------------- sound/pci/hda/hda_component.h | 2 -- sound/pci/hda/patch_realtek.c | 19 +------------------ 3 files changed, 13 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 102ac4a94a9d..89f6b4a28d3d 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -487,10 +487,10 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) struct regmap *reg = cs35l41->regmap; int ret = 0; - mutex_lock(&cs35l41->fw_mutex); - switch (action) { case HDA_GEN_PCM_ACT_OPEN: + pm_runtime_get_sync(dev); + mutex_lock(&cs35l41->fw_mutex); cs35l41->playback_started = true; if (cs35l41->firmware_running) { regmap_multi_reg_write(reg, cs35l41_hda_config_dsp, @@ -508,15 +508,21 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) CS35L41_AMP_EN_MASK, 1 << CS35L41_AMP_EN_SHIFT); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) regmap_write(reg, CS35L41_GPIO1_CTRL1, 0x00008001); + mutex_unlock(&cs35l41->fw_mutex); break; case HDA_GEN_PCM_ACT_PREPARE: + mutex_lock(&cs35l41->fw_mutex); ret = cs35l41_global_enable(reg, cs35l41->hw_cfg.bst_type, 1); + mutex_unlock(&cs35l41->fw_mutex); break; case HDA_GEN_PCM_ACT_CLEANUP: + mutex_lock(&cs35l41->fw_mutex); regmap_multi_reg_write(reg, cs35l41_hda_mute, ARRAY_SIZE(cs35l41_hda_mute)); ret = cs35l41_global_enable(reg, cs35l41->hw_cfg.bst_type, 0); + mutex_unlock(&cs35l41->fw_mutex); break; case HDA_GEN_PCM_ACT_CLOSE: + mutex_lock(&cs35l41->fw_mutex); ret = regmap_update_bits(reg, CS35L41_PWR_CTRL2, CS35L41_AMP_EN_MASK, 0 << CS35L41_AMP_EN_SHIFT); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) @@ -530,14 +536,16 @@ static void cs35l41_hda_playback_hook(struct device *dev, int action) } cs35l41_irq_release(cs35l41); cs35l41->playback_started = false; + mutex_unlock(&cs35l41->fw_mutex); + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); break; default: dev_warn(cs35l41->dev, "Playback action not supported: %d\n", action); break; } - mutex_unlock(&cs35l41->fw_mutex); - if (ret) dev_err(cs35l41->dev, "Regmap access fail: %d\n", ret); } @@ -618,19 +626,6 @@ static int cs35l41_runtime_resume(struct device *dev) return 0; } -static int cs35l41_hda_suspend_hook(struct device *dev) -{ - dev_dbg(dev, "Request Suspend\n"); - pm_runtime_mark_last_busy(dev); - return pm_runtime_put_autosuspend(dev); -} - -static int cs35l41_hda_resume_hook(struct device *dev) -{ - dev_dbg(dev, "Request Resume\n"); - return pm_runtime_get_sync(dev); -} - static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) { int halo_sts; @@ -863,8 +858,6 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas ret = cs35l41_create_controls(cs35l41); comps->playback_hook = cs35l41_hda_playback_hook; - comps->suspend_hook = cs35l41_hda_suspend_hook; - comps->resume_hook = cs35l41_hda_resume_hook; pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index 1223621bd62c..534e845b9cd1 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -16,6 +16,4 @@ struct hda_component { char name[HDA_MAX_NAME_SIZE]; struct hda_codec *codec; void (*playback_hook)(struct device *dev, int action); - int (*suspend_hook)(struct device *dev); - int (*resume_hook)(struct device *dev); }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4b076912bbf4..e6c4bb5fa041 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4022,22 +4022,16 @@ static void alc5505_dsp_init(struct hda_codec *codec) static int alc269_suspend(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i; if (spec->has_alc5505_dsp) alc5505_dsp_suspend(codec); - for (i = 0; i < HDA_MAX_COMPONENTS; i++) - if (spec->comps[i].suspend_hook) - spec->comps[i].suspend_hook(spec->comps[i].dev); - return alc_suspend(codec); } static int alc269_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i; if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); @@ -4068,10 +4062,6 @@ static int alc269_resume(struct hda_codec *codec) if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); - for (i = 0; i < HDA_MAX_COMPONENTS; i++) - if (spec->comps[i].resume_hook) - spec->comps[i].resume_hook(spec->comps[i].dev); - return 0; } #endif /* CONFIG_PM */ @@ -6664,19 +6654,12 @@ static int comp_bind(struct device *dev) { struct hda_codec *cdc = dev_to_hda_codec(dev); struct alc_spec *spec = cdc->spec; - int ret, i; + int ret; ret = component_bind_all(dev, spec->comps); if (ret) return ret; - if (snd_hdac_is_power_on(&cdc->core)) { - codec_dbg(cdc, "Resuming after bind.\n"); - for (i = 0; i < HDA_MAX_COMPONENTS; i++) - if (spec->comps[i].resume_hook) - spec->comps[i].resume_hook(spec->comps[i].dev); - } - return 0; } -- cgit v1.2.3 From 88672826e2a465d2f4c0a50fb5ced2956f4ffcbc Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Tue, 11 Oct 2022 15:35:52 +0100 Subject: ALSA: hda: cs35l41: Support System Suspend Add support for system suspend into the CS35L41 HDA Driver. Since S4 suspend may power off the system, it is required that the driver ensure the part is safe to be shutdown before system suspend, as well as ensuring that the firmware is unloaded before shutdown. The part must then be restored on system resume, including re-downloading the firmware. Signed-off-by: Stefan Binding Link: https://lore.kernel.org/r/20221011143552.621792-6-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 160 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 136 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 89f6b4a28d3d..e5f0549bf06d 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -461,9 +461,12 @@ static void cs35l41_remove_dsp(struct cs35l41_hda *cs35l41) struct cs_dsp *dsp = &cs35l41->cs_dsp; cancel_work_sync(&cs35l41->fw_load_work); + + mutex_lock(&cs35l41->fw_mutex); cs35l41_shutdown_dsp(cs35l41); cs_dsp_remove(dsp); cs35l41->halo_initialized = false; + mutex_unlock(&cs35l41->fw_mutex); } /* Protection release cycle to get the speaker out of Safe-Mode */ @@ -570,45 +573,148 @@ static int cs35l41_hda_channel_map(struct device *dev, unsigned int tx_num, unsi rx_slot); } +static void cs35l41_ready_for_reset(struct cs35l41_hda *cs35l41) +{ + mutex_lock(&cs35l41->fw_mutex); + if (cs35l41->firmware_running) { + + regcache_cache_only(cs35l41->regmap, false); + + cs35l41_exit_hibernate(cs35l41->dev, cs35l41->regmap); + cs35l41_shutdown_dsp(cs35l41); + cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type); + + regcache_cache_only(cs35l41->regmap, true); + regcache_mark_dirty(cs35l41->regmap); + } + mutex_unlock(&cs35l41->fw_mutex); +} + +static int cs35l41_system_suspend(struct device *dev) +{ + struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + int ret; + + dev_dbg(cs35l41->dev, "System Suspend\n"); + + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { + dev_err(cs35l41->dev, "System Suspend not supported\n"); + return -EINVAL; + } + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown DSP before system suspend */ + cs35l41_ready_for_reset(cs35l41); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int cs35l41_system_resume(struct device *dev) +{ + struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + int ret; + + dev_dbg(cs35l41->dev, "System Resume\n"); + + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { + dev_err(cs35l41->dev, "System Resume not supported\n"); + return -EINVAL; + } + + if (cs35l41->reset_gpio) { + usleep_range(2000, 2100); + gpiod_set_value_cansleep(cs35l41->reset_gpio, 1); + } + + usleep_range(2000, 2100); + + ret = pm_runtime_force_resume(dev); + + mutex_lock(&cs35l41->fw_mutex); + if (!ret && cs35l41->request_fw_load && !cs35l41->fw_request_ongoing) { + cs35l41->fw_request_ongoing = true; + schedule_work(&cs35l41->fw_load_work); + } + mutex_unlock(&cs35l41->fw_mutex); + + return ret; +} + static int cs35l41_runtime_suspend(struct device *dev) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); + int ret = 0; - dev_dbg(cs35l41->dev, "Suspend\n"); + dev_dbg(cs35l41->dev, "Runtime Suspend\n"); - if (!cs35l41->firmware_running) + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { + dev_dbg(cs35l41->dev, "Runtime Suspend not supported\n"); return 0; + } - if (cs35l41_enter_hibernate(cs35l41->dev, cs35l41->regmap, cs35l41->hw_cfg.bst_type) < 0) - return 0; + mutex_lock(&cs35l41->fw_mutex); + + if (cs35l41->playback_started) { + regmap_multi_reg_write(cs35l41->regmap, cs35l41_hda_mute, + ARRAY_SIZE(cs35l41_hda_mute)); + cs35l41_global_enable(cs35l41->regmap, cs35l41->hw_cfg.bst_type, 0); + regmap_update_bits(cs35l41->regmap, CS35L41_PWR_CTRL2, + CS35L41_AMP_EN_MASK, 0 << CS35L41_AMP_EN_SHIFT); + if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) + regmap_write(cs35l41->regmap, CS35L41_GPIO1_CTRL1, 0x00000001); + regmap_update_bits(cs35l41->regmap, CS35L41_PWR_CTRL2, + CS35L41_VMON_EN_MASK | CS35L41_IMON_EN_MASK, + 0 << CS35L41_VMON_EN_SHIFT | 0 << CS35L41_IMON_EN_SHIFT); + cs35l41->playback_started = false; + } + + if (cs35l41->firmware_running) { + ret = cs35l41_enter_hibernate(cs35l41->dev, cs35l41->regmap, + cs35l41->hw_cfg.bst_type); + if (ret) + goto err; + } else { + cs35l41_safe_reset(cs35l41->regmap, cs35l41->hw_cfg.bst_type); + } regcache_cache_only(cs35l41->regmap, true); regcache_mark_dirty(cs35l41->regmap); - return 0; +err: + mutex_unlock(&cs35l41->fw_mutex); + + return ret; } static int cs35l41_runtime_resume(struct device *dev) { struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev); - int ret; + int ret = 0; - dev_dbg(cs35l41->dev, "Resume.\n"); + dev_dbg(cs35l41->dev, "Runtime Resume\n"); if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) { - dev_dbg(cs35l41->dev, "System does not support Resume\n"); + dev_dbg(cs35l41->dev, "Runtime Resume not supported\n"); return 0; } - if (!cs35l41->firmware_running) - return 0; + mutex_lock(&cs35l41->fw_mutex); regcache_cache_only(cs35l41->regmap, false); - ret = cs35l41_exit_hibernate(cs35l41->dev, cs35l41->regmap); - if (ret) { - regcache_cache_only(cs35l41->regmap, true); - return ret; + if (cs35l41->firmware_running) { + ret = cs35l41_exit_hibernate(cs35l41->dev, cs35l41->regmap); + if (ret) { + dev_warn(cs35l41->dev, "Unable to exit Hibernate."); + goto err; + } } /* Test key needs to be unlocked to allow the OTP settings to re-apply */ @@ -617,13 +723,16 @@ static int cs35l41_runtime_resume(struct device *dev) cs35l41_test_key_lock(cs35l41->dev, cs35l41->regmap); if (ret) { dev_err(cs35l41->dev, "Failed to restore register cache: %d\n", ret); - return ret; + goto err; } if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST) cs35l41_init_boost(cs35l41->dev, cs35l41->regmap, &cs35l41->hw_cfg); - return 0; +err: + mutex_unlock(&cs35l41->fw_mutex); + + return ret; } static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) @@ -673,8 +782,6 @@ clean_dsp: static void cs35l41_load_firmware(struct cs35l41_hda *cs35l41, bool load) { - pm_runtime_get_sync(cs35l41->dev); - if (cs35l41->firmware_running && !load) { dev_dbg(cs35l41->dev, "Unloading Firmware\n"); cs35l41_shutdown_dsp(cs35l41); @@ -684,9 +791,6 @@ static void cs35l41_load_firmware(struct cs35l41_hda *cs35l41, bool load) } else { dev_dbg(cs35l41->dev, "Unable to Load firmware.\n"); } - - pm_runtime_mark_last_busy(cs35l41->dev); - pm_runtime_put_autosuspend(cs35l41->dev); } static int cs35l41_fw_load_ctl_get(struct snd_kcontrol *kcontrol, @@ -702,16 +806,21 @@ static void cs35l41_fw_load_work(struct work_struct *work) { struct cs35l41_hda *cs35l41 = container_of(work, struct cs35l41_hda, fw_load_work); + pm_runtime_get_sync(cs35l41->dev); + mutex_lock(&cs35l41->fw_mutex); /* Recheck if playback is ongoing, mutex will block playback during firmware loading */ if (cs35l41->playback_started) - dev_err(cs35l41->dev, "Cannot Load/Unload firmware during Playback\n"); + dev_err(cs35l41->dev, "Cannot Load/Unload firmware during Playback. Retrying...\n"); else cs35l41_load_firmware(cs35l41, cs35l41->request_fw_load); cs35l41->fw_request_ongoing = false; mutex_unlock(&cs35l41->fw_mutex); + + pm_runtime_mark_last_busy(cs35l41->dev); + pm_runtime_put_autosuspend(cs35l41->dev); } static int cs35l41_fw_load_ctl_put(struct snd_kcontrol *kcontrol, @@ -835,6 +944,8 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas pm_runtime_get_sync(dev); + mutex_lock(&cs35l41->fw_mutex); + comps->dev = dev; if (!cs35l41->acpi_subsystem_id) cs35l41->acpi_subsystem_id = kasprintf(GFP_KERNEL, "%.8x", @@ -847,10 +958,8 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas if (firmware_autostart) { dev_dbg(cs35l41->dev, "Firmware Autostart.\n"); cs35l41->request_fw_load = true; - mutex_lock(&cs35l41->fw_mutex); if (cs35l41_smart_amp(cs35l41) < 0) dev_warn(cs35l41->dev, "Cannot Run Firmware, reverting to dsp bypass...\n"); - mutex_unlock(&cs35l41->fw_mutex); } else { dev_dbg(cs35l41->dev, "Firmware Autostart is disabled.\n"); } @@ -859,6 +968,8 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas comps->playback_hook = cs35l41_hda_playback_hook; + mutex_unlock(&cs35l41->fw_mutex); + pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); @@ -1426,6 +1537,7 @@ EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41); const struct dev_pm_ops cs35l41_hda_pm_ops = { RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(cs35l41_system_suspend, cs35l41_system_resume) }; EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41); -- cgit v1.2.3