From 74f73476c3755664504b7d266b5e8f91050ffed9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Feb 2020 12:14:19 +0100 Subject: [PATCH 1/7] ALSA: usb-audio: Apply 48kHz fixed rate playback for Jabra Evolve 65 headset Jabra Evolve 65 headset appears as if supporting lower rates than 48kHz, but it actually doesn't work but with 48kHz for playback. This patch applies a workaround to enforce the 48kHz like LINE6 devices already did. The workaround is put in a unified helper function, set_fixed_rate(), to be called from both places now. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206149 Link: https://lore.kernel.org/r/20200211111419.5895-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/format.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 9260136e4c9b..50cb183958bf 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -151,6 +151,19 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, return pcm_formats; } +static int set_fixed_rate(struct audioformat *fp, int rate, int rate_bits) +{ + kfree(fp->rate_table); + fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); + if (!fp->rate_table) + return -ENOMEM; + fp->nr_rates = 1; + fp->rate_min = rate; + fp->rate_max = rate; + fp->rates = rate_bits; + fp->rate_table[0] = rate; + return 0; +} /* * parse the format descriptor and stores the possible sample rates @@ -223,6 +236,14 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof fp->rate_min = combine_triple(&fmt[offset + 1]); fp->rate_max = combine_triple(&fmt[offset + 4]); } + + /* Jabra Evolve 65 headset */ + if (chip->usb_id == USB_ID(0x0b0e, 0x030b)) { + /* only 48kHz for playback while keeping 16kHz for capture */ + if (fp->nr_rates != 1) + return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); + } + return 0; } @@ -299,17 +320,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ - /* supported rates: 48Khz */ - kfree(fp->rate_table); - fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); - if (!fp->rate_table) - return -ENOMEM; - fp->nr_rates = 1; - fp->rate_min = 48000; - fp->rate_max = 48000; - fp->rates = SNDRV_PCM_RATE_48000; - fp->rate_table[0] = 48000; - return 0; + return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } return -ENODEV; From d75a170fd848f037a1e28893ad10be7a4c51f8a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 11 Feb 2020 17:05:21 +0100 Subject: [PATCH 2/7] ALSA: usb-audio: Fix UAC2/3 effect unit parsing We've got a regression report about M-Audio Fast Track C400 device, and the git bisection resulted in the commit e0ccdef92653 ("ALSA: usb-audio: Clean up check_input_term()"). This commit was about the rewrite of the input terminal parser, and it's not too obvious from the change what really broke. The answer is: it's the interpretation of UAC2/3 effect units. In the original code, UAC2 effect unit is as if through UAC1 processing unit because both UAC1 PU and UAC2/3 EU share the same number (0x07). The old code went through a complex switch-case fallthrough, finally bailing out in the middle: if (protocol == UAC_VERSION_2 && hdr[2] == UAC2_EFFECT_UNIT) { /* UAC2/UAC1 unit IDs overlap here in an * uncompatible way. Ignore this unit for now. */ return 0; } ... and this special handling was missing in the new code; the new code treats UAC2/3 effect unit as if it were equivalent with the processing unit. Actually, the old code was too confusing. The effect unit has an incompatible unit description with the processing unit, so we shouldn't have dealt with EU in the same way. This patch addresses the regression by changing the effect unit handling to the own parser function. The own parser function makes the clear distinct with PU, so it improves the readability, too. The EU parser just sets the type and the id like the old kernels. Once when the proper effect unit support is added, we can revisit this parser function, but for now, let's keep this simple setup as is. Fixes: e0ccdef92653 ("ALSA: usb-audio: Clean up check_input_term()") Cc: BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206147 Link: https://lore.kernel.org/r/20200211160521.31990-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d659fdb475e2..81b2db0edd5f 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -897,6 +897,15 @@ static int parse_term_proc_unit(struct mixer_build *state, return 0; } +static int parse_term_effect_unit(struct mixer_build *state, + struct usb_audio_term *term, + void *p1, int id) +{ + term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ + term->id = id; + return 0; +} + static int parse_term_uac2_clock_source(struct mixer_build *state, struct usb_audio_term *term, void *p1, int id) @@ -981,8 +990,7 @@ static int __check_input_term(struct mixer_build *state, int id, UAC3_PROCESSING_UNIT); case PTYPE(UAC_VERSION_2, UAC2_EFFECT_UNIT): case PTYPE(UAC_VERSION_3, UAC3_EFFECT_UNIT): - return parse_term_proc_unit(state, term, p1, id, - UAC3_EFFECT_UNIT); + return parse_term_effect_unit(state, term, p1, id); case PTYPE(UAC_VERSION_1, UAC1_EXTENSION_UNIT): case PTYPE(UAC_VERSION_2, UAC2_EXTENSION_UNIT_V2): case PTYPE(UAC_VERSION_3, UAC3_EXTENSION_UNIT): From 93f9d1a4ac5930654c17412e3911b46ece73755a Mon Sep 17 00:00:00 2001 From: Arvind Sankar Date: Tue, 11 Feb 2020 11:22:35 -0500 Subject: [PATCH 3/7] ALSA: usb-audio: Apply sample rate quirk for Audioengine D1 The Audioengine D1 (0x2912:0x30c8) does support reading the sample rate, but it returns the rate in byte-reversed order. When setting sampling rate, the driver produces these warning messages: [168840.944226] usb 3-2.2: current rate 4500480 is different from the runtime rate 44100 [168854.930414] usb 3-2.2: current rate 8436480 is different from the runtime rate 48000 [168905.185825] usb 3-2.1.2: current rate 30465 is different from the runtime rate 96000 As can be seen from the hexadecimal conversion, the current rate read back is byte-reversed from the rate that was set. 44100 == 0x00ac44, 4500480 == 0x44ac00 48000 == 0x00bb80, 8436480 == 0x80bb00 96000 == 0x017700, 30465 == 0x007701 Rather than implementing a new quirk to reverse the order, just skip checking the rate to avoid spamming the log. Signed-off-by: Arvind Sankar Cc: Link: https://lore.kernel.org/r/20200211162235.1639889-1-nivedita@alum.mit.edu Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 3a5242e383b2..7f558f4b4520 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1440,6 +1440,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x1395, 0x740a): /* Sennheiser DECT */ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ return true; } From 2b3b6497c38d123934de68ea82a247b557d95290 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Mon, 10 Feb 2020 16:15:14 +0800 Subject: [PATCH 4/7] ALSA: hda/realtek - Add more codec supported Headset Button Add supported Headset Button for ALC215/ALC285/ALC289. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/948f70b4488f4cc2b629a39ce4e4be33@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4770fb3f51fb..3ee88adf57e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5701,8 +5701,11 @@ static void alc_fixup_headset_jack(struct hda_codec *codec, break; case HDA_FIXUP_ACT_INIT: switch (codec->core.vendor_id) { + case 0x10ec0215: case 0x10ec0225: + case 0x10ec0285: case 0x10ec0295: + case 0x10ec0289: case 0x10ec0299: alc_write_coef_idx(codec, 0x48, 0xd011); alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); From 7dafba3762d6c0083ded00a48f8c1a158bc86717 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 12 Feb 2020 09:10:47 +0100 Subject: [PATCH 5/7] ALSA: hda/realtek - Fix silent output on MSI-GL73 MSI-GL73 laptop with ALC1220 codec requires a similar workaround for Clevo laptops to enforce the DAC/mixer connection path. Set up a quirk entry for that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204159 Cc: Link: https://lore.kernel.org/r/20200212081047.27727-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ee88adf57e7..6c8cb4ce517e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), From 9f35a31283775e6f6af73fb2c95c686a4c0acac7 Mon Sep 17 00:00:00 2001 From: Alexander Tsoy Date: Thu, 13 Feb 2020 02:54:50 +0300 Subject: [PATCH 6/7] ALSA: usb-audio: Add clock validity quirk for Denon MC7000/MCX8000 It should be safe to ignore clock validity check result if the following conditions are met: - only one single sample rate is supported; - the terminal is directly connected to the clock source; - the clock type is internal. This is to deal with some Denon DJ controllers that always reports that clock is invalid. Tested-by: Tobias Oszlanyi Signed-off-by: Alexander Tsoy Cc: Link: https://lore.kernel.org/r/20200212235450.697348-1-alexander@tsoy.me Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 91 ++++++++++++++++++++++++++++++++-------------- sound/usb/clock.h | 4 +- sound/usb/format.c | 3 +- 3 files changed, 66 insertions(+), 32 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 018b1ecb5404..a48313dfa967 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,8 +151,34 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } +/* + * Assume the clock is valid if clock source supports only one single sample + * rate, the terminal is connected directly to it (there is no clock selector) + * and clock type is internal. This is to deal with some Denon DJ controllers + * that always reports that clock is invalid. + */ +static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, + struct audioformat *fmt, + int source_id) +{ + if (fmt->protocol == UAC_VERSION_2) { + struct uac_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source(chip->ctrl_intf, source_id); + + if (!cs_desc) + return false; + + return (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT); + } + + return false; +} + static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, - int protocol, + struct audioformat *fmt, int source_id) { int err; @@ -160,7 +186,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; u32 bmControls; - if (protocol == UAC_VERSION_3) { + if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); @@ -194,10 +220,14 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, return false; } - return data ? true : false; + if (data) + return true; + else + return uac_clock_source_is_valid_quirk(chip, fmt, source_id); } -static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, unsigned long *visited, bool validate) { struct uac_clock_source_descriptor *source; @@ -217,7 +247,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_2, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -248,8 +278,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac_clock_find_source(chip, selector->baCSourceID[ret - 1], - visited, validate); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], + visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -260,8 +291,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -281,14 +313,16 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, /* FIXME: multipliers only act as pass-thru element for now */ multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id); if (multiplier) - return __uac_clock_find_source(chip, multiplier->bCSourceID, - visited, validate); + return __uac_clock_find_source(chip, fmt, + multiplier->bCSourceID, + visited, validate); return -EINVAL; } -static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, - unsigned long *visited, bool validate) +static int __uac3_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, + unsigned long *visited, bool validate) { struct uac3_clock_source_descriptor *source; struct uac3_clock_selector_descriptor *selector; @@ -307,7 +341,7 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source_v3(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_3, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -338,7 +372,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[ret - 1], + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -350,8 +385,9 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -372,7 +408,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, multiplier = snd_usb_find_clock_multiplier_v3(chip->ctrl_intf, entity_id); if (multiplier) - return __uac3_clock_find_source(chip, multiplier->bCSourceID, + return __uac3_clock_find_source(chip, fmt, + multiplier->bCSourceID, visited, validate); return -EINVAL; @@ -389,18 +426,18 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, * * Returns the clock source UnitID (>=0) on success, or an error. */ -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate) +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - switch (protocol) { + switch (fmt->protocol) { case UAC_VERSION_2: - return __uac_clock_find_source(chip, entity_id, visited, + return __uac_clock_find_source(chip, fmt, fmt->clock, visited, validate); case UAC_VERSION_3: - return __uac3_clock_find_source(chip, entity_id, visited, + return __uac3_clock_find_source(chip, fmt, fmt->clock, visited, validate); default: return -EINVAL; @@ -501,8 +538,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * automatic clock selection if the current clock is not * valid. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, true); + clock = snd_usb_clock_find_source(chip, fmt, true); if (clock < 0) { /* We did not find a valid clock, but that might be * because the current sample rate does not match an @@ -510,8 +546,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * and we will do another validation after setting the * rate. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, false); + clock = snd_usb_clock_find_source(chip, fmt, false); if (clock < 0) return clock; } @@ -577,7 +612,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, validation: /* validate clock after rate change */ - if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + if (!uac_clock_source_is_valid(chip, fmt, clock)) return -ENXIO; return 0; } diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 076e31b79ee0..68df0fbe09d0 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -6,7 +6,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 50cb183958bf..9f5cb4ed3a0c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -336,8 +336,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0, ret_l6; - int clock = snd_usb_clock_find_source(chip, fp->protocol, - fp->clock, false); + int clock = snd_usb_clock_find_source(chip, fp, false); if (clock < 0) { dev_err(&dev->dev, From 0fbb027b44e79700da80e4b8bd1c1914d4796af6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 13 Feb 2020 07:03:49 +0100 Subject: [PATCH 7/7] ALSA: pcm: Fix double hw_free calls The commit 66f2d19f8116 ("ALSA: pcm: Fix memory leak at closing a stream without hw_free") tried to fix the regression wrt the missing hw_free call at closing without SNDRV_PCM_IOCTL_HW_FREE ioctl. However, the code change dropped mistakenly the state check, resulting in calling hw_free twice when SNDRV_PCM_IOCTL_HW_FRE got called beforehand. For most drivers, this is almost harmless, but the drivers like SOF show another regression now. This patch adds the state condition check before calling do_hw_free() at releasing the stream for avoiding the double hw_free calls. Fixes: 66f2d19f8116 ("ALSA: pcm: Fix memory leak at closing a stream without hw_free") Reported-by: Bard Liao Reported-by: Pierre-Louis Bossart Tested-by: Pierre-Louis Bossart Cc: Link: https://lore.kernel.org/r/s5hd0ajyprg.wl-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 336406bcb59e..d5443eeb8b63 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2594,7 +2594,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - do_hw_free(substream); + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; }