From 2b1d9c8f87235f593826b9cf46ec10247741fff9 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 20 Mar 2019 16:15:24 -0500 Subject: [PATCH 01/11] ALSA: rawmidi: Fix potential Spectre v1 vulnerability info->stream is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap) Fix this by sanitizing info->stream before using it to index rmidi->streams. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index ee601d7f0926..c0690d1ecd55 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include @@ -601,6 +602,7 @@ static int __snd_rawmidi_info_select(struct snd_card *card, return -ENXIO; if (info->stream < 0 || info->stream > 1) return -EINVAL; + info->stream = array_index_nospec(info->stream, 2); pstr = &rmidi->streams[info->stream]; if (pstr->substream_count == 0) return -ENOENT; From c709f14f0616482b67f9fbcb965e1493a03ff30b Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Wed, 20 Mar 2019 18:42:01 -0500 Subject: [PATCH 02/11] ALSA: seq: oss: Fix Spectre v1 vulnerability dev is indirectly controlled by user-space, hence leading to a potential exploitation of the Spectre variant 1 vulnerability. This issue was detected with the help of Smatch: sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap) Fix this by sanitizing dev before using it to index dp->synths. Notice that given that speculation windows are large, the policy is to kill the speculation on the first load and not worry if it can be completed with a dependent load/store [1]. [1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/ Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_synth.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 278ebb993122..c93945917235 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -617,13 +617,14 @@ int snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_info *inf) { struct seq_oss_synth *rec; + struct seq_oss_synthinfo *info = get_synthinfo_nospec(dp, dev); - if (dev < 0 || dev >= dp->max_synthdev) + if (!info) return -ENXIO; - if (dp->synths[dev].is_midi) { + if (info->is_midi) { struct midi_info minf; - snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf); + snd_seq_oss_midi_make_info(dp, info->midi_mapped, &minf); inf->synth_type = SYNTH_TYPE_MIDI; inf->synth_subtype = 0; inf->nr_voices = 16; From 2733ccebf4a937a0858e7d05a4a003b89715033f Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Thu, 21 Mar 2019 16:39:04 +0800 Subject: [PATCH 03/11] ALSA: hda/realtek: Enable headset MIC of Acer Aspire Z24-890 with ALC286 The Acer Aspire Z24-890 cannot detect the headset MIC until ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied. Signed-off-by: Jian-Hong Pan Signed-off-by: Daniel Drake Cc: 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 191830d4fa40..916bb5a3f324 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6715,6 +6715,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), From c7531e31c8a440b5fe6bd62664def5bcb6262f96 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Thu, 21 Mar 2019 17:17:31 +0800 Subject: [PATCH 04/11] ALSA: hda/realtek - Add support for Acer Aspire E5-523G/ES1-432 headset mic The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu Signed-off-by: Daniel Drake Signed-off-by: Jian-Hong Pan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 916bb5a3f324..6042ddf2b2ae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6712,6 +6712,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), From e1037354a0a75acdea2b27043c0a371ed85cf262 Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Fri, 22 Mar 2019 11:37:18 +0800 Subject: [PATCH 05/11] ALSA: hda/realtek: Enable ASUS X441MB and X705FD headset MIC with ALC256 The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu Signed-off-by: Daniel Drake Signed-off-by: Jian-Hong Pan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6042ddf2b2ae..f2539007d09e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5688,6 +5688,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6696,6 +6697,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7334,6 +7344,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x1a, 0x90a70130}, + {0x1b, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x12, 0xb7a60130}, {0x13, 0xb8a61140}, From a806ef1cf3bbc0baadc6cdeb11f12b5dd27e91c2 Mon Sep 17 00:00:00 2001 From: Chris Chiu Date: Fri, 22 Mar 2019 11:37:20 +0800 Subject: [PATCH 06/11] ALSA: hda/realtek: Enable headset mic of ASUS P5440FF with ALC256 The ASUS laptop P5440FF with ALC256 can't detect the headset microphone until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Chris Chiu Signed-off-by: Daniel Drake Signed-off-by: Jian-Hong Pan Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2539007d09e..8ec0cd95341a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7344,6 +7344,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, {0x1a, 0x90a70130}, {0x1b, 0x90170110}, From 6ac371aa1a74240fb910c98aa3484d5ece8473d3 Mon Sep 17 00:00:00 2001 From: Jian-Hong Pan Date: Fri, 22 Mar 2019 11:37:22 +0800 Subject: [PATCH 07/11] ALSA: hda/realtek: Enable headset MIC of ASUS X430UN and X512DK with ALC256 The ASUS X430UN and X512DK with ALC256 cannot detect the headset MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied. Signed-off-by: Jian-Hong Pan Signed-off-by: Daniel Drake Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ec0cd95341a..01c71467600b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7348,6 +7348,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x14, 0x90170110}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, {0x1a, 0x90a70130}, {0x1b, 0x90170110}, From ca0214ee2802dd47239a4e39fb21c5b00ef61b22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Mar 2019 16:00:54 +0100 Subject: [PATCH 08/11] ALSA: pcm: Fix possible OOB access in PCM oss plugins The PCM OSS emulation converts and transfers the data on the fly via "plugins". The data is converted over the dynamically allocated buffer for each plugin, and recently syzkaller caught OOB in this flow. Although the bisection by syzbot pointed out to the commit 65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer allocations"), this is merely a commit to replace vmalloc() with kvmalloc(), hence it can't be the cause. The further debug action revealed that this happens in the case where a slave PCM doesn't support only the stereo channels while the OSS stream is set up for a mono channel. Below is a brief explanation: At each OSS parameter change, the driver sets up the PCM hw_params again in snd_pcm_oss_change_params_lock(). This is also the place where plugins are created and local buffers are allocated. The problem is that the plugins are created before the final hw_params is determined. Namely, two snd_pcm_hw_param_near() calls for setting the period size and periods may influence on the final result of channels, rates, etc, too, while the current code has already created plugins beforehand with the premature values. So, the plugin believes that channels=1, while the actual I/O is with channels=2, which makes the driver reading/writing over the allocated buffer size. The fix is simply to move the plugin allocation code after the final hw_params call. Reported-by: syzbot+d4503ae45b65c5bc1194@syzkaller.appspotmail.com Cc: Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_oss.c | 43 ++++++++++++++++++++-------------------- 1 file changed, 22 insertions(+), 21 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d5b0d7ba83c4..f6ae68017608 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -940,6 +940,28 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) oss_frame_size = snd_pcm_format_physical_width(params_format(params)) * params_channels(params) / 8; + err = snd_pcm_oss_period_size(substream, params, sparams); + if (err < 0) + goto failure; + + n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size); + err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL); + if (err < 0) + goto failure; + + err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS, + runtime->oss.periods, NULL); + if (err < 0) + goto failure; + + snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); + + err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams); + if (err < 0) { + pcm_dbg(substream->pcm, "HW_PARAMS failed: %i\n", err); + goto failure; + } + #ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_oss_plugin_clear(substream); if (!direct) { @@ -974,27 +996,6 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) } #endif - err = snd_pcm_oss_period_size(substream, params, sparams); - if (err < 0) - goto failure; - - n = snd_pcm_plug_slave_size(substream, runtime->oss.period_bytes / oss_frame_size); - err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, n, NULL); - if (err < 0) - goto failure; - - err = snd_pcm_hw_param_near(substream, sparams, SNDRV_PCM_HW_PARAM_PERIODS, - runtime->oss.periods, NULL); - if (err < 0) - goto failure; - - snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); - - if ((err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_HW_PARAMS, sparams)) < 0) { - pcm_dbg(substream->pcm, "HW_PARAMS failed: %i\n", err); - goto failure; - } - if (runtime->oss.trigger) { sw_params->start_threshold = 1; } else { From 4fc90fb883fcb72d6bfbf84d554a3e820a05ef62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Mar 2019 15:51:36 +0100 Subject: [PATCH 09/11] ALSA: hda/ca0132 - Simplify alt firmware loading code ca0132 codec driver loads the firmware selectively depending on the model in addition to the fallback of the default firmware. The code works good, but a minor problem is that the current code seems confusing for Clang where it spews a warning about uninitialized variable. This patch simplifies the code flow for such a false-positive warning. After this refactoring, the ca0132_spec.alt_firmware_present field is no longer used, hence it's eliminated as well. Reported-and-tested-by: Arnd Bergmann Reviewed-by: Nathan Chancellor Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 29882bda7632..e1ebc6d5f382 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1005,7 +1005,6 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; - bool alt_firmware_present; bool startup_check_entered; bool dsp_reload; @@ -7518,7 +7517,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) bool dsp_loaded = false; struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; /* * Alternate firmwares for different variants. The Recon3Di apparently * can use the default firmware, but I'll leave the option in case @@ -7529,33 +7528,26 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) case QUIRK_R3D: case QUIRK_AE5: if (request_firmware(&fw_entry, DESKTOP_EFX_FILE, - codec->card->dev) != 0) { + codec->card->dev) != 0) codec_dbg(codec, "Desktop firmware not found."); - spec->alt_firmware_present = false; - } else { + else codec_dbg(codec, "Desktop firmware selected."); - spec->alt_firmware_present = true; - } break; case QUIRK_R3DI: if (request_firmware(&fw_entry, R3DI_EFX_FILE, - codec->card->dev) != 0) { + codec->card->dev) != 0) codec_dbg(codec, "Recon3Di alt firmware not detected."); - spec->alt_firmware_present = false; - } else { + else codec_dbg(codec, "Recon3Di firmware selected."); - spec->alt_firmware_present = true; - } break; default: - spec->alt_firmware_present = false; break; } /* * Use default ctefx.bin if no alt firmware is detected, or if none * exists for your particular codec. */ - if (!spec->alt_firmware_present) { + if (!fw_entry) { codec_dbg(codec, "Default firmware selected."); if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) From 113ce08109f8e3b091399e7cc32486df1cff48e7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 25 Mar 2019 10:38:58 +0100 Subject: [PATCH 10/11] ALSA: pcm: Don't suspend stream in unrecoverable PCM state Currently PCM core sets each opened stream forcibly to SUSPENDED state via snd_pcm_suspend_all() call, and the user-space is responsible for re-triggering the resume manually either via snd_pcm_resume() or prepare call. The scheme works fine usually, but there are corner cases where the stream can't be resumed by that call: the streams still in OPEN state before finishing hw_params. When they are suspended, user-space cannot perform resume or prepare because they haven't been set up yet. The only possible recovery is to re-open the device, which isn't nice at all. Similarly, when a stream is in DISCONNECTED state, it makes no sense to change it to SUSPENDED state. Ditto for in SETUP state; which you can re-prepare directly. So, this patch addresses these issues by filtering the PCM streams to be suspended by checking the PCM state. When a stream is in either OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend action is skipped. To be noted, this problem was originally reported for the PCM runtime PM on HD-audio. And, the runtime PM problem itself was already addressed (although not intended) by the code refactoring commits 3d21ef0b49f8 ("ALSA: pcm: Suspend streams globally via device type PM ops") and 17bc4815de58 ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls"). These commits eliminated the snd_pcm_suspend*() calls from the runtime PM suspend callback code path, hence the racy OPEN state won't appear while runtime PM. (FWIW, the race window is between snd_pcm_open_substream() and the first power up in azx_pcm_open().) Although the runtime PM issue was already "fixed", the same problem is still present for the system PM, hence this patch is still needed. And for stable trees, this patch alone should suffice for fixing the runtime PM problem, too. Reported-and-tested-by: Jon Hunter Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f731f904e8cc..1d8452912b14 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1445,8 +1445,15 @@ static int snd_pcm_pause(struct snd_pcm_substream *substream, int push) static int snd_pcm_pre_suspend(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) + switch (runtime->status->state) { + case SNDRV_PCM_STATE_SUSPENDED: return -EBUSY; + /* unresumable PCM state; return -EBUSY for skipping suspend */ + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_DISCONNECTED: + return -EBUSY; + } runtime->trigger_master = substream; return 0; } From e2a829b3da01b9b32c4d0291d042b8a6e2a98ca3 Mon Sep 17 00:00:00 2001 From: Bernhard Rosenkraenzer Date: Tue, 5 Mar 2019 00:38:19 +0100 Subject: [PATCH 11/11] ALSA: hda/realtek - Fix speakers on Acer Predator Helios 500 Ryzen laptops On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers don't work out of the box. The problem can be worked around with hdajackretask, remapping the "Black Headphone, Right side" pin (0x21) to the Internal speaker. This patch adds a quirk to change this mapping by default. [ corrected ALC299_FIXUP_PREDATOR_SPK definition and adapted for the latest tree by tiwai ] Signed-off-by: Bernhard Rosenkraenzer Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01c71467600b..a3fb3d4c5730 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5689,6 +5689,7 @@ enum { ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, + ALC299_FIXUP_PREDATOR_SPK, }; static const struct hda_fixup alc269_fixups[] = { @@ -6706,6 +6707,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE }, + [ALC299_FIXUP_PREDATOR_SPK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x21, 0x90170150 }, /* use as headset mic, without its own jack detect */ + { } + } + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6724,6 +6732,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), @@ -7124,6 +7133,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_DELL_HEADSET_MIC, .name = "alc255-dell-headset"}, {.id = ALC295_FIXUP_HP_X360, .name = "alc295-hp-x360"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-sense-combo"}, + {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {} }; #define ALC225_STANDARD_PINS \