From 39013bd60e79148961583402ed70bd105f95d260 Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Mon, 15 Oct 2012 14:13:25 +0100 Subject: ASoC: Ux500: Dispose of device nodes correctly When of_parse_phandle() is used to find a device node, its reference count is incremented by the helper. Once we're finished with them, it's our responsibly to ensure they are freed in the correct manor. Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/mop500.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 356611d9654d..54f7e25b6f7d 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -57,6 +57,20 @@ static struct snd_soc_card mop500_card = { .num_links = ARRAY_SIZE(mop500_dai_links), }; +static void mop500_of_node_put(void) +{ + int i; + + for (i = 0; i < 2; i++) { + if (mop500_dai_links[i].cpu_of_node) + of_node_put((struct device_node *) + mop500_dai_links[i].cpu_of_node); + if (mop500_dai_links[i].codec_of_node) + of_node_put((struct device_node *) + mop500_dai_links[i].codec_of_node); + } +} + static int __devinit mop500_of_probe(struct platform_device *pdev, struct device_node *np) { @@ -69,6 +83,7 @@ static int __devinit mop500_of_probe(struct platform_device *pdev, if (!(msp_np[0] && msp_np[1] && codec_np)) { dev_err(&pdev->dev, "Phandle missing or invalid\n"); + mop500_of_node_put(); return -EINVAL; } @@ -83,6 +98,7 @@ static int __devinit mop500_of_probe(struct platform_device *pdev, return 0; } + static int __devinit mop500_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; @@ -128,6 +144,7 @@ static int __devexit mop500_remove(struct platform_device *pdev) snd_soc_unregister_card(mop500_card); mop500_ab8500_remove(mop500_card); + mop500_of_node_put(); return 0; } -- cgit v1.2.3 From 05304949332c6d2c7b50f2d0f666a52369f09ced Mon Sep 17 00:00:00 2001 From: Lee Jones Date: Mon, 15 Oct 2012 14:13:26 +0100 Subject: ASoC: ux500_msp_i2s: Fix devm_* and return code merge error Some ux500_msp_i2s patches clashed with: b18e93a493626c1446f9788ebd5844d008bbf71c ASoC: ux500_msp_i2s: better use devm functions and fix error return code ... leaving the driver uncompilable. This patch fixes the issues encountered. Acked-by: Linus Walleij Signed-off-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index b7c996e77570..a26c6bf0a29b 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include @@ -697,14 +698,11 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, platform_data = devm_kzalloc(&pdev->dev, sizeof(struct msp_i2s_platform_data), GFP_KERNEL); if (!platform_data) - ret = -ENOMEM; + return -ENOMEM; } } else if (!platform_data) - ret = -EINVAL; - - if (ret) - goto err_res; + return -EINVAL; dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, pdev->name, platform_data->id); -- cgit v1.2.3 From 84f98fdf7865fbd35b312eb39ea91e5618c514c7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Stef=C3=A1n=20Freyr?= Date: Fri, 19 Oct 2012 22:46:00 +0200 Subject: ALSA: hda - add dock support for Thinkpad T430 I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking station. Without this patch, if I plug my headphones into the jack on the computer, everything works fine. The computer speakers mute and the audio is played in the headphones. However, if I plug into the docking station headphone jack the computer speakers are muted but there is no audio in the headphones. Addresses https://bugs.launchpad.net/bugs/1060372 Signed-off-by: Joseph Salisbury Cc: stable@vger.kernel.org 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 48d9d609f89b..2c62edd6d4fd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6188,6 +6188,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- cgit v1.2.3 From 41285a98f8f052b59f01bbbcb0064e5e41b0ca2d Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:22 +0200 Subject: ALSA: hdspm - Allow DDS/Varispeed to be set from userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The DDS value is the actual physical sample rate. We set it indirectly when selecting 44100, 48000 and so on via snd_hdspm_hw_params or hdspm_set_clock_source. This commit now allows the DDS value to be altered at runtime, thus speeding up or slowing down the physical sample rate. This is required for MADI's varispeed that allows for ±12.5% speed adjustment from the "selected" rate (32kHz, 44100kHz, 48kHz and so on). Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b12308b5ba2a..742bd5e430ab 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2003,8 +2003,10 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ - .access = SNDRV_CTL_ELEM_ACCESS_READ, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |\ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ .info = snd_hdspm_info_system_sample_rate, \ + .put = snd_hdspm_put_system_sample_rate, \ .get = snd_hdspm_get_system_sample_rate \ } @@ -2030,6 +2032,16 @@ static int snd_hdspm_get_system_sample_rate(struct snd_kcontrol *kcontrol, return 0; } +static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value * + ucontrol) +{ + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + + hdspm_set_dds_value(hdspm, ucontrol->value.enumerated.item[0]); + return 0; +} + /** * Returns the WordClock sample rate class for the given card. -- cgit v1.2.3 From 21a164df0c1c2e72d4bb78b2f67dc75a1c9d14fb Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:23 +0200 Subject: ALSA: hdspm - Report external rate in slave mode on PCI MADI As a follow-up to a97bda7d29d02a2e9c6609d0947b15e55f5200e5, report the external sample rate as system_sample_rate when in slave mode. For PCIe MADI cards, the DDS value automatically contains the external sample rate, but the PCI version needs this manual workaround. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 742bd5e430ab..1131a8ab3f73 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -971,6 +971,7 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm); static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm); static int hdspm_autosync_ref(struct hdspm *hdspm); static int snd_hdspm_set_defaults(struct hdspm *hdspm); +static int hdspm_system_clock_mode(struct hdspm *hdspm); static void hdspm_set_sgbuf(struct hdspm *hdspm, struct snd_pcm_substream *substream, unsigned int reg, int channels); @@ -1989,10 +1990,14 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) rate = hdspm_calc_dds_value(hdspm, period); if (rate > 207000) { - /* Unreasonable high sample rate as seen on PCI MADI cards. - * Use the cached value instead. - */ - rate = hdspm->system_sample_rate; + /* Unreasonable high sample rate as seen on PCI MADI cards. */ + if (0 == hdspm_system_clock_mode(hdspm)) { + /* master mode, return internal sample rate */ + rate = hdspm->system_sample_rate; + } else { + /* slave mode, return external sample rate */ + rate = hdspm_external_sample_rate(hdspm); + } } return rate; -- cgit v1.2.3 From fba30fd342beda8e7908f214d9e7f8a62dcdc57d Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:24 +0200 Subject: ALSA: hdspm - Fix sync check reporting on all RME HDSPM cards Due to missing breaks and the resulting fall-through, card subtype selection was effectively missing, thus causing the wrong sync check functions to be called. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 1131a8ab3f73..81d83faab510 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3959,6 +3959,7 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, default: val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); } + break; case AIO: switch (kcontrol->private_value) { @@ -3971,6 +3972,7 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, default: val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); } + break; case MADI: switch (kcontrol->private_value) { @@ -3983,6 +3985,7 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 3: /* SYNC_IN */ val = hdspm_sync_in_sync_check(hdspm); break; } + break; case MADIface: val = hdspm_madi_sync_check(hdspm); /* MADI */ @@ -4000,6 +4003,7 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, val = hdspm_aes_sync_check(hdspm, kcontrol->private_value-1); } + break; } -- cgit v1.2.3 From d681deaa74748ce1d4b1a862438c955b70f46690 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:25 +0200 Subject: ALSA: hdspm - Fix reported autosync_sample_rate Missing breaks lead to a fall-through, thus causing the wrong autosync_sample_rate to be reported. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 81d83faab510..976e3a6b0508 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2180,6 +2180,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, hdspm_get_s1_sample_rate(hdspm, kcontrol->private_value-1); } + break; case AIO: switch (kcontrol->private_value) { @@ -2200,6 +2201,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, hdspm_get_s1_sample_rate(hdspm, ucontrol->id.index-1); } + break; case AES32: @@ -2221,8 +2223,8 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, hdspm_get_s1_sample_rate(hdspm, kcontrol->private_value-1); break; - } + break; default: break; } -- cgit v1.2.3 From b8812c55c5e59596bbf4b1d3e5b4b3ef50a89cad Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:26 +0200 Subject: ALSA: hdspm - Also report autosync_sample_rate on MADI and MADIface MADI and MADIface used to report the autosync_sample_rate. This functionality was lost in commit 0dca1793063c28dde8f6c49c9c72203fe5cb6efc, this commit now adds it back. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 976e3a6b0508..5f243aedeef7 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2225,6 +2225,21 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, break; } break; + + case MADI: + case MADIface: + { + int rate = hdspm_external_sample_rate(hdspm); + int i, selected_rate = 0; + for (i = 1; i < 10; i++) + if (HDSPM_bit2freq(i) == rate) { + selected_rate = i; + break; + } + ucontrol->value.enumerated.item[0] = selected_rate; + } + break; + default: break; } @@ -4450,6 +4465,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_madi[] = { HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0), HDSPM_AUTOSYNC_REF("AutoSync Reference", 0), HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), + HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0), HDSPM_SYNC_CHECK("WC SyncCheck", 0), HDSPM_SYNC_CHECK("MADI SyncCheck", 1), HDSPM_SYNC_CHECK("TCO SyncCHeck", 2), -- cgit v1.2.3 From 2e0452f544758d798ba76016c00783f654b43fe3 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:27 +0200 Subject: ALSA: hdspm - Fix sync_in reporting on RME MADI cards In contrast to AES32, MADI uses the first status register to report the sync_in status. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 5f243aedeef7..5d70efec26c0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3885,6 +3885,11 @@ static int hdspm_sync_in_sync_check(struct hdspm *hdspm) break; case MADI: + status = hdspm_read(hdspm, HDSPM_statusRegister); + lock = (status & HDSPM_syncInLock) ? 1 : 0; + sync = (status & HDSPM_syncInSync) ? 1 : 0; + break; + case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister2); lock = (status & HDSPM_syncInLock) ? 1 : 0; -- cgit v1.2.3 From 9a215f47da029b04bf92f1f68abf978211155ceb Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:28 +0200 Subject: ALSA: hdspm - Fix sync_in detection on AES/AES32 According to the documentation, AES32 cards use a different bit position for reporting the sync_in status. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 5d70efec26c0..b8b15ee5eebb 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3892,8 +3892,8 @@ static int hdspm_sync_in_sync_check(struct hdspm *hdspm) case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister2); - lock = (status & HDSPM_syncInLock) ? 1 : 0; - sync = (status & HDSPM_syncInSync) ? 1 : 0; + lock = (status & 0x100000) ? 1 : 0; + sync = (status & 0x200000) ? 1 : 0; break; case MADIface: -- cgit v1.2.3 From 930f4ff078b9c4484b0e9383fc17a04d4f9253bf Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:29 +0200 Subject: ALSA: hdspm - Fix typo in kcontrol element on RME MADI cards Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b8b15ee5eebb..096580473529 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4473,7 +4473,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_madi[] = { HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0), HDSPM_SYNC_CHECK("WC SyncCheck", 0), HDSPM_SYNC_CHECK("MADI SyncCheck", 1), - HDSPM_SYNC_CHECK("TCO SyncCHeck", 2), + HDSPM_SYNC_CHECK("TCO SyncCheck", 2), HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 3), HDSPM_LINE_OUT("Line Out", 0), HDSPM_TX_64("TX 64 channels mode", 0), -- cgit v1.2.3 From f27a64f9973ff932ece576793c195de60c1c6c9b Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 19 Oct 2012 17:42:30 +0200 Subject: ALSA: hdspm - Fix coding style in CTL_ELEM macros checkpatch.pl discourages the use of spaces at the beginning of lines. Some of the CTL_ELEM defines were not properly indented. This patch replaces the leading spaces by tabs. No functionality is changed, the commit is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 180 +++++++++++++++++++++++----------------------- 1 file changed, 90 insertions(+), 90 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 096580473529..f1cd1e387801 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2005,14 +2005,14 @@ static int hdspm_get_system_sample_rate(struct hdspm *hdspm) #define HDSPM_SYSTEM_SAMPLE_RATE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |\ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .info = snd_hdspm_info_system_sample_rate, \ - .put = snd_hdspm_put_system_sample_rate, \ - .get = snd_hdspm_get_system_sample_rate \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |\ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .info = snd_hdspm_info_system_sample_rate, \ + .put = snd_hdspm_put_system_sample_rate, \ + .get = snd_hdspm_get_system_sample_rate \ } static int snd_hdspm_info_system_sample_rate(struct snd_kcontrol *kcontrol, @@ -2464,7 +2464,7 @@ static int snd_hdspm_put_clock_source(struct snd_kcontrol *kcontrol, #define HDSPM_PREF_SYNC_REF(xname, xindex) \ -{.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .index = xindex, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |\ @@ -2800,12 +2800,12 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol, #define HDSPM_AUTOSYNC_REF(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .access = SNDRV_CTL_ELEM_ACCESS_READ, \ - .info = snd_hdspm_info_autosync_ref, \ - .get = snd_hdspm_get_autosync_ref, \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READ, \ + .info = snd_hdspm_info_autosync_ref, \ + .get = snd_hdspm_get_autosync_ref, \ } static int hdspm_autosync_ref(struct hdspm *hdspm) @@ -2889,12 +2889,12 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol, #define HDSPM_LINE_OUT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_line_out, \ - .get = snd_hdspm_get_line_out, \ - .put = snd_hdspm_put_line_out \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_line_out, \ + .get = snd_hdspm_get_line_out, \ + .put = snd_hdspm_put_line_out \ } static int hdspm_line_out(struct hdspm * hdspm) @@ -2946,12 +2946,12 @@ static int snd_hdspm_put_line_out(struct snd_kcontrol *kcontrol, #define HDSPM_TX_64(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_tx_64, \ - .get = snd_hdspm_get_tx_64, \ - .put = snd_hdspm_put_tx_64 \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_tx_64, \ + .get = snd_hdspm_get_tx_64, \ + .put = snd_hdspm_put_tx_64 \ } static int hdspm_tx_64(struct hdspm * hdspm) @@ -3002,12 +3002,12 @@ static int snd_hdspm_put_tx_64(struct snd_kcontrol *kcontrol, #define HDSPM_C_TMS(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_c_tms, \ - .get = snd_hdspm_get_c_tms, \ - .put = snd_hdspm_put_c_tms \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_c_tms, \ + .get = snd_hdspm_get_c_tms, \ + .put = snd_hdspm_put_c_tms \ } static int hdspm_c_tms(struct hdspm * hdspm) @@ -3058,12 +3058,12 @@ static int snd_hdspm_put_c_tms(struct snd_kcontrol *kcontrol, #define HDSPM_SAFE_MODE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_safe_mode, \ - .get = snd_hdspm_get_safe_mode, \ - .put = snd_hdspm_put_safe_mode \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_safe_mode, \ + .get = snd_hdspm_get_safe_mode, \ + .put = snd_hdspm_put_safe_mode \ } static int hdspm_safe_mode(struct hdspm * hdspm) @@ -3114,12 +3114,12 @@ static int snd_hdspm_put_safe_mode(struct snd_kcontrol *kcontrol, #define HDSPM_EMPHASIS(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_emphasis, \ - .get = snd_hdspm_get_emphasis, \ - .put = snd_hdspm_put_emphasis \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_emphasis, \ + .get = snd_hdspm_get_emphasis, \ + .put = snd_hdspm_put_emphasis \ } static int hdspm_emphasis(struct hdspm * hdspm) @@ -3170,12 +3170,12 @@ static int snd_hdspm_put_emphasis(struct snd_kcontrol *kcontrol, #define HDSPM_DOLBY(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_dolby, \ - .get = snd_hdspm_get_dolby, \ - .put = snd_hdspm_put_dolby \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_dolby, \ + .get = snd_hdspm_get_dolby, \ + .put = snd_hdspm_put_dolby \ } static int hdspm_dolby(struct hdspm * hdspm) @@ -3226,12 +3226,12 @@ static int snd_hdspm_put_dolby(struct snd_kcontrol *kcontrol, #define HDSPM_PROFESSIONAL(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_professional, \ - .get = snd_hdspm_get_professional, \ - .put = snd_hdspm_put_professional \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_professional, \ + .get = snd_hdspm_get_professional, \ + .put = snd_hdspm_put_professional \ } static int hdspm_professional(struct hdspm * hdspm) @@ -3281,12 +3281,12 @@ static int snd_hdspm_put_professional(struct snd_kcontrol *kcontrol, } #define HDSPM_INPUT_SELECT(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_input_select, \ - .get = snd_hdspm_get_input_select, \ - .put = snd_hdspm_put_input_select \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_input_select, \ + .get = snd_hdspm_get_input_select, \ + .put = snd_hdspm_put_input_select \ } static int hdspm_input_select(struct hdspm * hdspm) @@ -3353,12 +3353,12 @@ static int snd_hdspm_put_input_select(struct snd_kcontrol *kcontrol, #define HDSPM_DS_WIRE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_ds_wire, \ - .get = snd_hdspm_get_ds_wire, \ - .put = snd_hdspm_put_ds_wire \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_ds_wire, \ + .get = snd_hdspm_get_ds_wire, \ + .put = snd_hdspm_put_ds_wire \ } static int hdspm_ds_wire(struct hdspm * hdspm) @@ -3425,12 +3425,12 @@ static int snd_hdspm_put_ds_wire(struct snd_kcontrol *kcontrol, #define HDSPM_QS_WIRE(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = xindex, \ - .info = snd_hdspm_info_qs_wire, \ - .get = snd_hdspm_get_qs_wire, \ - .put = snd_hdspm_put_qs_wire \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_hdspm_info_qs_wire, \ + .get = snd_hdspm_get_qs_wire, \ + .put = snd_hdspm_put_qs_wire \ } static int hdspm_qs_wire(struct hdspm * hdspm) @@ -3597,15 +3597,15 @@ static int snd_hdspm_put_madi_speedmode(struct snd_kcontrol *kcontrol, } #define HDSPM_MIXER(xname, xindex) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ - .name = xname, \ - .index = xindex, \ - .device = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .info = snd_hdspm_info_mixer, \ - .get = snd_hdspm_get_mixer, \ - .put = snd_hdspm_put_mixer \ +{ .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, \ + .name = xname, \ + .index = xindex, \ + .device = 0, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .info = snd_hdspm_info_mixer, \ + .get = snd_hdspm_get_mixer, \ + .put = snd_hdspm_put_mixer \ } static int snd_hdspm_info_mixer(struct snd_kcontrol *kcontrol, @@ -3704,12 +3704,12 @@ static int snd_hdspm_put_mixer(struct snd_kcontrol *kcontrol, */ #define HDSPM_PLAYBACK_MIXER \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | \ - SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ - .info = snd_hdspm_info_playback_mixer, \ - .get = snd_hdspm_get_playback_mixer, \ - .put = snd_hdspm_put_playback_mixer \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_WRITE | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .info = snd_hdspm_info_playback_mixer, \ + .get = snd_hdspm_get_playback_mixer, \ + .put = snd_hdspm_put_playback_mixer \ } static int snd_hdspm_info_playback_mixer(struct snd_kcontrol *kcontrol, -- cgit v1.2.3 From cb766404e6b8c566569eb9ada02ea45d28729864 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 20 Oct 2012 10:55:21 +0200 Subject: ALSA: hda - Fix silent headphone output from Toshiba P200 By some reason, Toshiba laptop doesn't like the EAPD turned up for the headphone pin. Add a fix up code to force to turn down EAPD for NID 0x15. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=569991 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2c62edd6d4fd..f7397ad02a0d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5677,6 +5677,7 @@ static const struct hda_verb alc268_beep_init_verbs[] = { enum { ALC268_FIXUP_INV_DMIC, + ALC268_FIXUP_HP_EAPD, }; static const struct alc_fixup alc268_fixups[] = { @@ -5684,10 +5685,26 @@ static const struct alc_fixup alc268_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC268_FIXUP_HP_EAPD] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0}, + {} + } + }, }; static const struct alc_model_fixup alc268_fixup_models[] = { {.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC268_FIXUP_HP_EAPD, .name = "hp-eapd"}, + {} +}; + +static const struct snd_pci_quirk alc268_fixup_tbl[] = { + /* below is codec SSID since multiple Toshiba laptops have the + * same PCI SSID 1179:ff00 + */ + SND_PCI_QUIRK(0x1179, 0xff06, "Toshiba P200", ALC268_FIXUP_HP_EAPD), {} }; @@ -5722,7 +5739,7 @@ static int patch_alc268(struct hda_codec *codec) spec = codec->spec; - alc_pick_fixup(codec, alc268_fixup_models, NULL, alc268_fixups); + alc_pick_fixup(codec, alc268_fixup_models, alc268_fixup_tbl, alc268_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); /* automatic parse from the BIOS config */ -- cgit v1.2.3 From edb66893860dbc0631afc2807efdb209a03d57a2 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Sun, 21 Oct 2012 12:52:03 +0200 Subject: ALSA: sound/isa/opti9xx/miro.c: eliminate possible double free snd_miro_probe is a static function that is only called twice in the file that defines it. At each call site, its argument is freed using snd_card_free. Thus, there is no need for snd_miro_probe to call snd_card_free on its argument on any of its error exit paths. Because snd_card_free both reads the fields of its argument and kfrees its argments, the results of the second snd_card_free should be unpredictable. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @r@ identifier f,free,a; parameter list[n] ps; type T; expression e; @@ f(ps,T a,...) { ... when any when != a = e if(...) { ... free(a); ... return ...; } ... when any } @@ identifier r.f,r.free; expression x,a; expression list[r.n] xs; @@ * x = f(xs,a,...); if (...) { ... free(a); ... return ...; } // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 3d1afb612b35..4a7ff4e8985b 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1286,7 +1286,6 @@ static int __devinit snd_miro_probe(struct snd_card *card) error = snd_card_miro_aci_detect(card, miro); if (error < 0) { - snd_card_free(card); snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; } -- cgit v1.2.3 From 21b3de881b38a84002c07b1b4bfb91892644e83f Mon Sep 17 00:00:00 2001 From: Denis Kirjanov Date: Mon, 22 Oct 2012 17:05:53 +0400 Subject: ALSA: als3000: check for the kzalloc return value Signed-off-by: Denis Kirjanov Signed-off-by: Takashi Iwai --- sound/pci/als300.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 00f157a2cf64..5af3cb6b0c18 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -394,6 +394,8 @@ static int snd_als300_playback_open(struct snd_pcm_substream *substream) struct snd_als300_substream_data *data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; snd_als300_dbgcallenter(); chip->playback_substream = substream; runtime->hw = snd_als300_playback_hw; @@ -425,6 +427,8 @@ static int snd_als300_capture_open(struct snd_pcm_substream *substream) struct snd_als300_substream_data *data = kzalloc(sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; snd_als300_dbgcallenter(); chip->capture_substream = substream; runtime->hw = snd_als300_capture_hw; -- cgit v1.2.3 From 79748cdb39dbf914bc5f26c75cfd5f91d84d82c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Oct 2012 15:28:30 +0100 Subject: ASoC: wm8994: Only enable extra BCLK cycles when required Rather than always assuming the maximum possible BCLK rate will be required generate BCLKs for stereo if either one or two channels is enabled. In order to support this we also need to ensure that only the relevant channels are enabled. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 18 +++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2b2dadc54dac..3fddc7ad1127 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1045,6 +1045,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; @@ -1063,6 +1064,10 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: + /* Don't enable timeslot 2 if not in use */ + if (wm8994->channels[0] <= 2) + mask &= ~(WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA); + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); if ((val & WM8994_AIF1ADCL_SRC) && (val & WM8994_AIF1ADCR_SRC)) @@ -2687,7 +2692,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bclk_rate = params_rate(params) * 4; + bclk_rate = params_rate(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: bclk_rate *= 16; @@ -2708,6 +2713,17 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8994->channels[id] = params_channels(params); + switch (params_channels(params)) { + case 1: + case 2: + bclk_rate *= 2; + break; + default: + bclk_rate *= 4; + break; + } + /* Try to find an appropriate sample rate; look for an exact match. */ for (i = 0; i < ARRAY_SIZE(srs); i++) if (srs[i].rate == params_rate(params)) diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f142ec198db3..ccbce5791e95 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -77,6 +77,7 @@ struct wm8994_priv { int sysclk_rate[2]; int mclk[2]; int aifclk[2]; + int channels[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; bool fll_locked_irq; -- cgit v1.2.3