September 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [ALSA hda_codec 1/1] NULL pointer deference fixes for slave_dig_outs
by robbat2＠gentoo.org 13 Sep '08

13 Sep '08

From: Robin H. Johnson <robbat2(a)gentoo.org> The new multiple digital output code forgot to check the slave_dig_outs for NULL in several places. This manifests as an OOPS when changing volume amongst other things. Signed-off-by: Robin H. Johnson <robbat2(a)gentoo.org> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 696d77e..b22061b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1462,14 +1462,15 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); - for (d = codec->slave_dig_outs; *d; d++) { - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_2, - val >> 8); - } + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_2, + val >> 8); + } } mutex_unlock(&codec->spdif_mutex); @@ -1507,10 +1508,12 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) @@ -1664,9 +1667,10 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, val); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); return change; @@ -2617,10 +2621,11 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); /* turn on again (if needed) */ @@ -2628,10 +2633,11 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); } } -- 1.6.0.1

2 1

[alsa-devel] [PATCH] Fix oopses in snd-hda-intel from alsa v1.0.18rc3
by Herton Ronaldo Krzesinski 13 Sep '08

13 Sep '08

Fixes the following oopses and other likewise possible ones: BUG: unable to handle kernel NULL pointer dereference at 00000000 IP: [<f8cc5917>] :snd_hda_intel:snd_hda_spdif_out_switch_put+0x97/0x130 *pde = 00000000 Oops: 0000 [#1] SMP Modules linked in: snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device ipv6 snd_pcm_oss snd_mixer_oss coretemp it87 hwmon_vid binfmt_misc loop nls_utf8 ntfs dm_mirror dm_log dm_mod floppy cpufreq_ondemand cpufreq_conservative cpufreq_powersave acpi_cpufreq freq_table snd_hda_intel snd_pcm ub ppdev snd_timer evdev snd_page_alloc r8169 snd_hwdep usbhid snd mii iTCO_wdt button thermal iTCO_vendor_support i2c_i801 processor parport_pc parport serio_raw i2c_core usb_storage intel_agp sr_mod rtc_cmos ff_memless agpgart soundcore sg ata_piix ahci libata dock sd_mod scsi_mod crc_t10dif raid456 async_xor async_memcpy async_tx xor raid0 ext3 jbd uhci_hcd ohci_hcd ehci_hcd usbcore [last unloaded: scsi_wait_scan] Pid: 3730, comm: alsactl Not tainted (2.6.27-desktop-0.rc6.2mnb #1) EIP: 0060:[<f8cc5917>] EFLAGS: 00010246 CPU: 3 EIP is at snd_hda_spdif_out_switch_put+0x97/0x130 [snd_hda_intel] EAX: 00000000 EBX: 00000000 ECX: 00000000 EDX: 00000003 ESI: f7b86200 EDI: 00000001 EBP: f6d8bd94 ESP: f6d8bd68 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process alsactl (pid: 3730, ti=f6d8a000 task=f731d6a0 task.ti=f6d8a000) Stack: 0000070d 00000001 00000000 00000001 00000006 f7b86380 00065a80 00000001 f6c35a80 f69d8c00 f8bd12d0 f6d8bda4 f8bd106a f6c35a80 f69d8c00 f6d8bdb4 f8bd1318 f69d8c00 f711da00 f6d8bf4c f8bcfa59 f6d8bf04 bf893e30 f6cdc8c0 Call Trace: [<f8bd12d0>] ? slave_put+0x0/0x60 [snd] [<f8bd106a>] ? slave_put_val+0x2a/0xa0 [snd] [<f8bd1318>] ? slave_put+0x48/0x60 [snd] [<f8bcfa59>] ? snd_ctl_ioctl+0xc99/0xe70 [snd] [<c01b0317>] ? dput+0x87/0x130 [<c0207e40>] ? ccs_capable+0x30/0x1f0 [<f8bcedc0>] ? snd_ctl_ioctl+0x0/0xe70 [snd] [<c01a928e>] ? vfs_ioctl+0x5e/0xa0 [<c01a4805>] ? putname+0x25/0x40 [<c01a932e>] ? do_vfs_ioctl+0x5e/0x2e0 [<c019a617>] ? do_sys_open+0xc7/0xf0 [<c01a9613>] ? sys_ioctl+0x63/0x70 [<c0103e1b>] ? sysenter_do_call+0x12/0x2f ======================= Code: b7 45 ee 0f b7 c9 0f b6 f9 89 4d e0 31 c9 89 7c 24 04 89 45 e4 89 c2 89 f0 c7 04 24 0d 07 00 00 e8 3f f7 ff ff 8b 9e 9c 01 00 00 <0f> b7 03 66 85 c0 74 23 90 0f b7 d0 31 c9 83 c3 02 89 f0 89 7c EIP: [<f8cc5917>] snd_hda_spdif_out_switch_put+0x97/0x130 [snd_hda_intel] SS:ESP 0068:f6d8bd68 ---[ end trace 1e63c3510a780bec ]--- BUG: unable to handle kernel NULL pointer dereference at 00000000 IP: [<f8cb07d5>] :snd_hda_intel:setup_dig_out_stream+0xf5/0x140 *pde = 00000000 Oops: 0000 [#1] SMP Modules linked in: nfs lockd nfs_acl sunrpc af_packet radeon drm snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device ipv6 snd_pcm_oss snd_mixer_oss coretemp it87 hwmon_vid binfmt_misc loop nls_utf8 ntfs dm_mirror dm_log dm_mod floppy cpufreq_ondemand cpufreq_conservative cpufreq_powersave acpi_cpufreq freq_table ppdev evdev snd_hda_intel thermal snd_pcm snd_timer r8169 snd_page_alloc ub snd_hwdep intel_agp mii snd i2c_i801 button usbhid parport_pc agpgart i2c_core parport serio_raw processor soundcore iTCO_wdt rtc_cmos ff_memless iTCO_vendor_support sr_mod usb_storage sg ata_piix ahci libata dock sd_mod scsi_mod crc_t10dif raid456 async_xor async_memcpy async_tx xor raid0 ext3 jbd uhci_hcd ohci_hcd ehci_hcd usbcore [last unloaded: scsi_wait_scan] Pid: 5139, comm: artsd Not tainted (2.6.27-desktop-0.rc6.2mnb #1) EIP: 0060:[<f8cb07d5>] EFLAGS: 00210202 CPU: 2 EIP is at setup_dig_out_stream+0xf5/0x140 [snd_hda_intel] EAX: 00000000 EBX: 00000000 ECX: 00000000 EDX: 00000003 ESI: f6d54600 EDI: 00000006 EBP: f5afbcd8 ESP: f5afbcc0 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process artsd (pid: 5139, ti=f5afa000 task=f6b50000 task.ti=f5afa000) Stack: 0000070d 00000000 00000005 00000031 f7022050 f6d54600 f5afbd04 f8cb0a8d 00000031 0000e008 00000005 f6d54780 f8ceb39c 00000002 00000031 f6d55500 f5afbd60 f5afbd18 f8cba051 00000031 f7066d00 f7066d00 f5afbd70 f8cae925 Call Trace: [<f8cb0a8d>] ? snd_hda_multi_out_analog_prepare+0x1fd/0x240 [snd_hda_intel] [<f8cba051>] ? alc880_playback_pcm_prepare+0x21/0x30 [snd_hda_intel] [<f8cae925>] ? azx_pcm_prepare+0x345/0x3c0 [snd_hda_intel] [<f8c17d5c>] ? snd_pcm_do_prepare+0xc/0x20 [snd_pcm] [<f8c178ab>] ? snd_pcm_action_single+0x2b/0x60 [snd_pcm] [<f8c18a8b>] ? snd_pcm_action_nonatomic+0x6b/0x70 [snd_pcm] [<f8c1b437>] ? snd_pcm_common_ioctl1+0x607/0x1720 [snd_pcm] [<c0182f6f>] ? handle_mm_fault+0x11f/0x7e0 [<c012a0c0>] ? default_wake_function+0x0/0x10 [<c03820f8>] ? unlock_kernel+0x28/0x2f [<f8bc8688>] ? snd_open+0xd8/0x150 [snd] [<c011cef0>] ? do_page_fault+0x330/0x870 [<c017d7a2>] ? vma_prio_tree_insert+0x22/0xc0 [<c0185f89>] ? vma_link+0xa9/0xf0 [<f8c1c988>] ? snd_pcm_playback_ioctl1+0x38/0x3e0 [snd_pcm] [<c018699e>] ? mmap_region+0x2be/0x4b0 [<c0207e40>] ? ccs_capable+0x30/0x1f0 [<f8c1cd76>] ? snd_pcm_playback_ioctl+0x46/0x50 [snd_pcm] [<f8c1cd30>] ? snd_pcm_playback_ioctl+0x0/0x50 [snd_pcm] [<c01a928e>] ? vfs_ioctl+0x5e/0xa0 [<c01a932e>] ? do_vfs_ioctl+0x5e/0x2e0 [<c01a9613>] ? sys_ioctl+0x63/0x70 [<c0103e1b>] ? sysenter_do_call+0x12/0x2f [<c0380000>] ? schedule+0x760/0x940 ======================= Code: b4 26 00 00 00 00 81 e2 fe 00 00 00 0f b7 f8 89 54 24 04 31 c9 89 fa 89 f0 c7 04 24 0d 07 00 00 e8 91 eb ff ff 8b 9e 9c 01 00 00 <0f> b7 03 66 85 c0 0f 84 2e ff ff ff 8d b4 26 00 00 00 00 0f b7 EIP: [<f8cb07d5>] setup_dig_out_stream+0xf5/0x140 [snd_hda_intel] SS:ESP 0068:f5afbcc0 ---[ end trace d44a3a67f9b19615 ]--- --- Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 696d77e..531364d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1462,14 +1462,15 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); - for (d = codec->slave_dig_outs; *d; d++) { - snd_hda_codec_write_cache(codec, *d, 0, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) { + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - snd_hda_codec_write_cache(codec, *d, 0, + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); - } + } } mutex_unlock(&codec->spdif_mutex); @@ -1507,8 +1508,9 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); /* unmute amp switch (if any) */ @@ -1664,8 +1666,9 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write_cache(codec, *d, 0, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write_cache(codec, *d, 0, AC_VERB_SET_DIGI_CONVERT_1, val); } mutex_unlock(&codec->spdif_mutex); @@ -2617,9 +2620,10 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); } snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -2628,9 +2632,10 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); - for (d = codec->slave_dig_outs; *d; d++) - snd_hda_codec_write(codec, *d, 0, - AC_VERB_SET_DIGI_CONVERT_1, + if (codec->slave_dig_outs) + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_write(codec, *d, 0, + AC_VERB_SET_DIGI_CONVERT_1, codec->spdif_ctls & 0xff); } -- []'s Herton

2 2

[alsa-devel] [PATCH] hda: fixed hp_nid DAC for DELL_M6
by Matthew Ranostay 13 Sep '08

13 Sep '08

This patch sets the HP out not used by the "Headphone to Line Out" switch to the hp_nid. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f41ba17..0218aa7 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -747,8 +747,8 @@ static struct hda_verb dell_m6_core_init[] = { { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, /* setup audio connections */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},

2 1

[alsa-devel] [PATCH] hda: Added a STAC_HP_M4 quirk
by Matthew Ranostay 13 Sep '08

13 Sep '08

Added a new quirk for STAC_HP_M4 series for an 92hd75xxx family laptop. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f00334f..3000ecf 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -80,6 +80,7 @@ enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, + STAC_HP_M4, STAC_92HD71BXX_MODELS }; @@ -1527,12 +1528,14 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, + [STAC_HP_M4] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", + [STAC_HP_M4] = "hp-m4", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { @@ -4204,10 +4207,24 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + switch (spec->board_config) { + case STAC_HP_M4: + spec->num_dmics = 0; + spec->num_smuxes = 1; + spec->num_dmuxes = 0; + + /* enable internal microphone */ + snd_hda_codec_write_cache(codec, 0x0e, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80); + break; + default: + spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + }; + spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids;

2 1

[alsa-devel] [PATCH] hda: Input port AMP controls
by Matthew Ranostay 13 Sep '08

13 Sep '08

Added support for controlling hardware gain amps on input ports using a volume control mixer with a mux selecting the port being controlled. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/pci/hda/hda_codec.c b/pci/hda/hda_codec.c index 696d77e..e12626b 100644 --- a/pci/hda/hda_codec.c +++ b/pci/hda/hda_codec.c @@ -961,15 +961,6 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } #endif /* SND_HDA_NEEDS_RESUME */ -/* - * AMP control callbacks - */ -/* retrieve parameters from private_value */ -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) -#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) -#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) -#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) - /* volume */ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) diff --git a/pci/hda/hda_local.h b/pci/hda/hda_local.h index 5c9e578..d688f50 100644 --- a/pci/hda/hda_local.h +++ b/pci/hda/hda_local.h @@ -418,4 +418,13 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, hda_nid_t nid); #endif /* CONFIG_SND_HDA_POWER_SAVE */ +/* + * AMP control callbacks + */ +/* retrieve parameters from private_value */ +#define get_amp_nid(kc) ((kc)->private_value & 0xffff) +#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) +#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) +#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index f00334f..ac56481 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -156,10 +156,13 @@ struct sigmatel_spec { unsigned int num_pwrs; unsigned int *pwr_mapping; hda_nid_t *pwr_nids; + hda_nid_t *amp_nids; hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; + struct hda_input_mux *amp_mux; + unsigned int cur_amux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -215,6 +218,7 @@ struct sigmatel_spec { struct hda_input_mux private_dimux; struct hda_input_mux private_imux; struct hda_input_mux private_smux; + struct hda_input_mux private_amp_mux; struct hda_input_mux private_mono_mux; }; @@ -243,6 +247,10 @@ static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; +static hda_nid_t stac92hd73xx_amp_nids[4] = { + 0x0b, 0x0c, 0x0e, 0 +}; + #define STAC92HD73XX_NUM_DMICS 2 static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -448,6 +456,34 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x21, 0x22, }; +#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info + +static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->amp_nids[spec->cur_amux]; + + kcontrol->private_value ^= get_amp_nid(kcontrol); + kcontrol->private_value |= nid; + + return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); +} + +static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->amp_nids[spec->cur_amux]; + + kcontrol->private_value ^= get_amp_nid(kcontrol); + kcontrol->private_value |= nid; + + return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); +} + static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -563,6 +599,41 @@ static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, spec->mono_nid, &spec->cur_mmux); } +static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->amp_mux, uinfo); +} + +static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_amux; + return 0; +} + +static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + struct snd_kcontrol *ctl = + snd_hda_find_mixer_ctl(codec, "Amp Capture Volume"); + if (!ctl) + return -EINVAL; + + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + + return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol, + 0, &spec->cur_amux); +} + #define stac92xx_aloopback_info snd_ctl_boolean_mono_info static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, @@ -837,6 +908,31 @@ static struct hda_verb stac9205_core_init[] = { .put = stac92xx_mono_mux_enum_put, \ } +#define STAC_AMP_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Amp Selector Capture Switch", \ + .count = 1, \ + .info = stac92xx_amp_mux_enum_info, \ + .get = stac92xx_amp_mux_enum_get, \ + .put = stac92xx_amp_mux_enum_put, \ + } + +#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .info = stac92xx_amp_volume_info, \ + .get = stac92xx_amp_volume_get, \ + .put = stac92xx_amp_volume_put, \ + .tlv = { .c = snd_hda_mixer_amp_tlv }, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -2418,6 +2514,8 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_AMP_MUX, + STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH @@ -2427,6 +2525,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_AMP_MUX, + STAC_AMP_VOL(NULL, 0, 0, 0, 0), STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), @@ -2844,6 +2944,35 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) "Mono Mux", spec->mono_nid); } +/* labels for amp mux outputs */ +static const char *stac92xx_amp_labels[3] = { + "Front Microphone", "Microphone", "Line In" +}; + +/* create amp out controls mux on capable codecs */ +static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *amp_mux = &spec->private_amp_mux; + int i, err; + + for (i = 0; i < ARRAY_SIZE(stac92xx_amp_labels); i++) { + amp_mux->items[amp_mux->num_items].label = + stac92xx_amp_labels[i]; + amp_mux->items[amp_mux->num_items].index = i; + amp_mux->num_items++; + } + + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, + "Amp Selector Capture Switch", 0); + if (err < 0) + return err; + return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, + "Amp Capture Volume", + HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); +} + + /* create PC beep volume controls */ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, hda_nid_t nid) @@ -3213,7 +3342,11 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - + if (spec->amp_nids) { + err = stac92xx_auto_create_amp_output_ctls(codec); + if (err < 0) + return err; + } if (spec->num_dmics > 0) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) @@ -3246,7 +3379,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->dinput_mux = &spec->private_dimux; spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; - + spec->amp_mux = &spec->private_amp_mux; return 1; } @@ -3914,6 +4047,7 @@ again: spec->dmic_nids = stac92hd73xx_dmic_nids; spec->dmux_nids = stac92hd73xx_dmux_nids; spec->smux_nids = stac92hd73xx_smux_nids; + spec->amp_nids = stac92hd73xx_amp_nids; spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);

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[alsa-devel] bluetooth headset doesn't work well(Motorola HS820)
by Galaha Shine 13 Sep '08

13 Sep '08

Hi, I have a blue-tooth headset connected to Ubuntu 8.04 but it does't work well. I've created ~/.asoundrc as below: pcm.bluetooth { type bluetooth device 20:04:07:80:BC:89 profile voice } ctl.bluetooth { type bluetooth } now I can use command `aplay -Dbluetooth -v 2.wav` to play sound to the headset, but the sound is little bit wired due to inaccurate rate. Playing WAVE '2.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono Warning: rate is not accurate (requested = 16000Hz, got = 8000Hz) please, try the plug plugin (-Dplug:bluetooth) Bluetooth Audio Device Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 So I input command according to the aplay's recommendation `aplay -Dplug:bluetooth -v 2.wav`. The problem comes out. aplay -Dplug:bluetooth -v 2.wav Playing WAVE '2.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 8000 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Bluetooth Audio Device Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 It seems that ALSA plugs rate plugin. But I hear NOTHING from my headset and aplay seems suspended. I turn on bluez debug flag and rebuild it. I find the bluez suspending at 'bluetooth_playback_poll_descriptors'. Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 930 bluetooth_hsp_write: count=0 frames_to_read=24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 993 bluetooth_hsp_write: returning 24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 912 bluetooth_hsp_write: areas->step=16 areas->first=0 offset=0, size=1000 io->nonblock=0 io->hw_ptr=0 io->appl_ptr=0 stream.fd=8 link_mtu=48 data->count=0 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 930 bluetooth_hsp_write: count=0 frames_to_read=24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 993 bluetooth_hsp_write: returning 24 Sep 7 23:18:59 alex-laptop aplay: DEBUG: pcm_bluetooth.c 802 bluetooth_playback_poll_descriptors: Space =2 pfd[0].fd=5 pfd[1].fd=6 Sep 7 23:19:01 alex-laptop audio[7979]: Received AT+VGS=11 Now I go deep into ALSA then I find ALSA suspending at: int snd_pcm_wait_nocheck(snd_pcm_t *pcm, int timeout) { struct pollfd *pfd; unsigned short *revents; int i, npfds, pollio, err, err_poll; npfds = snd_pcm_poll_descriptors_count(pcm); if (npfds <= 0 || npfds >= 16) { SNDERR("Invalid poll_fds %d\n", npfds); return -EIO; } pfd = alloca(sizeof(*pfd) * npfds); revents = alloca(sizeof(*revents) * npfds); syslog( LOG_INFO, "%s", __FUNCTION__ ); err = snd_pcm_poll_descriptors(pcm, pfd, npfds); if (err < 0) return err; if (err != npfds) { SNDMSG("invalid poll descriptors %d\n", err); return -EIO; } do { pollio = 0; err_poll = poll(pfd, npfds, timeout); <------------- It seems the bluez doesn't feed ALSA some signals. But why? I notice 'bluetooth_playback_start' haven't been invoked. Is it the point? Then I keep tracing ALSA. I find in snd_pcm_rate_commit_area result = snd_pcm_mmap_commit(rate->gen.slave, slave_offset, slave_size); if (result < (snd_pcm_sframes_t)slave_size) { if (result < 0) return result; result = snd_pcm_rewind(rate->gen.slave, result); <----------- if (result < 0) return result; return 0; } The snd_pcm_mmap_commit returns with 24 which indicates bluetooth SCO link mtu(48bytes). And it causes snd_pcm_rewind being called after every package sent by bluetooth SCO link. Look at this piece of code in snd_pcm_write_areas: if (state == SND_PCM_STATE_PREPARED) { snd_pcm_sframes_t hw_avail = pcm->buffer_size - avail; hw_avail += frames; /* some plugins might automatically start the stream */ state = snd_pcm_state(pcm); if (state == SND_PCM_STATE_PREPARED && hw_avail >= (snd_pcm_sframes_t) pcm->start_threshold) { err = snd_pcm_start(pcm); if (err < 0) goto _end; } } the 'avail' will be equal to pcm->buffer due to snd_pcm_rewind, therefore the condition hw_avail>=pcm->start_threshold will not be satisfied forever. That's why snd_pcm_start can't be called. I don't know the mechanism of ALSA and I just assume ALSA wanna bluez plugin sending whole period_size instead of only one 48bytes package. I revise the bluez plugin: #if 0 // this section is original bluez frame_size = areas->step / 8; if ((data->count + size * frame_size) <= data->link_mtu) frames_to_read = size; else frames_to_read = (data->link_mtu - data->count) / frame_size; DBG("count=%d frames_to_read=%lu", data->count, frames_to_read); /* Ready for more data */ buff = (uint8_t *) areas->addr + (areas->first + areas->step * offset) / 8; memcpy(data->buffer + data->count, buff, frame_size * frames_to_read); /* Remember we have some frames in the pipe now */ data->count += frames_to_read * frame_size; if (data->count != data->link_mtu) { ret = frames_to_read; goto done; } rsend = send(data->stream.fd, data->buffer, data->link_mtu, io->nonblock ? MSG_DONTWAIT : 0); if (rsend > 0) { /* Reset count pointer */ data->count = 0; ret = frames_to_read; } else if (rsend < 0) ret = (errno == EPIPE) ? -EIO : -errno; else ret = -EIO; #else frame_size = areas->step / 8; buff = (uint8_t *) areas->addr + (areas->first + areas->step * offset) / 8; tobesent = frame_size*size; while( rsend<tobesent ) { if( tobesent-rsend>data->link_mtu ) sent = send(data->stream.fd, buff +counter*data->link_mtu, data->link_mtu, io->nonblock ? MSG_DONTWAIT : 0); else sent = send(data->stream.fd, buff +counter*data->link_mtu, tobesent-rsend, io->nonblock ? MSG_DONTWAIT : 0); if( sent > 0 ) { if( sent != data->link_mtu && tobesent-rsend > data->link_mtu ) DBG("got problem!"); counter++; rsend+=sent; } else if (rsend < 0) { ret = (errno == EPIPE) ? -EIO : -errno; break; } else { ret = -EIO; break; } } ret = size; #endif After this revision, the headset works but the performance isn't as good as playing MONO 8000Hz wav directly to headset with 'aplay -Dbluetooth 1.wav'. Any hints would be appreciated. Regards, Galaha

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Re: [alsa-devel] [PATCH] ASoC: OMAP: Fix Gumstix Overo machine driver by using snd_soc_dai struct (rev 2)
by Steve Sakoman 13 Sep '08

13 Sep '08

From: Jarkko Nikula <jarkko.nikula(a)nokia.com> Structs snd_soc_codec_dai and snd_soc_cpu_dai are obsolete and are merged to snd_soc_dai by the commit 8687eb8bded8c3c5842a85bd0c30e43fc5a3e0e0. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Acked-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/overo.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index 542ff6b..1a086ce 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -39,8 +39,8 @@ static int overo_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ -- 1.5.6.5

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Re: [alsa-devel] [PATCH] ASoC: OMAP: Fix Gumstix Overo machine driver by using snd_soc_dai struct
by Steve Sakoman 12 Sep '08

12 Sep '08

On Fri, Sep 12, 2008 at 1:46 PM, Felipe Balbi <me(a)felipebalbi.com> wrote: > should this be From: Jarkko ??? Indeed. Will resubmit. Steve

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[alsa-devel] [PATCH] ASoC: OMAP: Fix Gumstix Overo machine driver by using snd_soc_dai struct
by Steve Sakoman 12 Sep '08

12 Sep '08

Structs snd_soc_codec_dai and snd_soc_cpu_dai are obsolete and are merged to snd_soc_dai by the commit 8687eb8bded8c3c5842a85bd0c30e43fc5a3e0e0. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Acked-by: Steve Sakoman <steve(a)sakoman.com> --- sound/soc/omap/overo.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index 542ff6b..1a086ce 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -39,8 +39,8 @@ static int overo_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ -- 1.5.6.5

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[alsa-devel] alsa callback not async
by William Estrada 12 Sep '08

12 Sep '08

Hi guys, I am cross posting this because I am not sure which list it belongs to. I have a test program I wrote to help me learn ALSA APIs, It work to a degree. It is a simple Capture/Playback interface. My problem is that when the Callback starts, it does not allow any cycles back to the main program. The main program waits for console input to tell the callback when to stop. Well it does not accept any console input when the callback is running. Any ideas?? The source is here: http://64.124.13.3/_ALSA_/Echo/ -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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