December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v5] sun4i-codec: add inputs
by Danny Milosavljevic 11 Dec '15

11 Dec '15

Hi, this is the fifth version of the patch that adds inputs to sun4i-codec. (Previous versions have been on the linux-sunxi mailing list only) The inputs added are: - FM-In Left and Right - Line-In Left and Right - Mic1-In - Mic2-In Changes compared to v4 are: - Mic preamplifier controls have more common names now. - Mic preamplifier scale has a 0 dB entry as well now, as documented in the A20 user manual. - Mic preamplifier has special cases for A20 and A10 now. - Gain controls have "Gain" in the name now. I successfully tested it on an A20 board using alsamixer, headphones, a radio and my ears. Note that because of missing capturing support I tested only the mixing, for Mic, Line, and FM. The patches are on top of <git://git.kernel.org/pub/scm/linux/kernel/git/mripard/linux.git>, branch "sunxi/for-next". Regards, Danny Danny (1): b/sound/soc/sunxi/sun4i-codec.c | 153 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 150 insertions(+), 3 deletions(-) Signed-off-by: Danny Milosavljevic <dannym+a(a)scratchpost.org> --- diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index bcbf4da..106b568 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -57,9 +57,20 @@ #define SUN4I_CODEC_DAC_ACTL_DACAENR (31) #define SUN4I_CODEC_DAC_ACTL_DACAENL (30) #define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LNG (26) +#define SUN4I_CODEC_DAC_ACTL_FMG (23) +#define SUN4I_CODEC_DAC_ACTL_MICG (20) +#define SUN4I_CODEC_DAC_ACTL_LLNS (19) +#define SUN4I_CODEC_DAC_ACTL_RLNS (18) +#define SUN4I_CODEC_DAC_ACTL_LFMS (17) +#define SUN4I_CODEC_DAC_ACTL_RFMS (16) #define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) #define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) #define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_MIC1LS (12) +#define SUN4I_CODEC_DAC_ACTL_MIC1RS (11) +#define SUN4I_CODEC_DAC_ACTL_MIC2LS (10) +#define SUN4I_CODEC_DAC_ACTL_MIC2RS (9) #define SUN4I_CODEC_DAC_ACTL_DACPAS (8) #define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) #define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) @@ -84,8 +95,11 @@ #define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) #define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) #define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_PREG1_A10 (25) +#define SUN4I_CODEC_ADC_ACTL_PREG2_A10 (23) #define SUN4I_CODEC_ADC_ACTL_VADCG (20) #define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_LNRDF (16) #define SUN4I_CODEC_ADC_ACTL_PA_EN (4) #define SUN4I_CODEC_ADC_ACTL_DDE (3) #define SUN4I_CODEC_ADC_DEBUG (0x2c) @@ -94,7 +108,18 @@ #define SUN4I_CODEC_DAC_TXCNT (0x30) #define SUN4I_CODEC_ADC_RXCNT (0x34) #define SUN4I_CODEC_AC_SYS_VERI (0x38) -#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +/* MIC_PHONE_CAL register offsets and bit fields (A20 only) */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL (0x3c) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1 (29) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2 (26) +/* note: no idea where the output pins for the following are. */ +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTG (5) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTEN (4) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS3 (3) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS2 (2) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS1 (1) +#define SUN7I_CODEC_AC_MIC_PHONE_CAL_PHONEOUTS0 (0) struct sun4i_codec { struct device *dev; @@ -402,16 +427,94 @@ static const struct snd_kcontrol_new sun4i_codec_pa_mute = SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(sun4i_codec_linein_loopback_gain_scale, -150, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_fmin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_SCALE(sun4i_codec_micin_loopback_gain_scale, -450, 150, 0); +static DECLARE_TLV_DB_RANGE(sun7i_codec_micin_preamp_gain_scale, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(2400, 300, 0) +); +static DECLARE_TLV_DB_RANGE(sun4i_codec_micin_preamp_gain_scale_a10, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_SCALE_ITEM(3500, 300, 0) +); + +struct snd_kcontrol_new sun7ionly_codec_widgets[] = { + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG1, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN7I_CODEC_AC_MIC_PHONE_CAL, + SUN7I_CODEC_AC_MIC_PHONE_CAL_PREG2, + 7, + 0, + sun7i_codec_micin_preamp_gain_scale), +}; -static const struct snd_kcontrol_new sun4i_codec_widgets[] = { +/* Note: this is modified by the sun4i_codec_probe() function. */ +static struct snd_kcontrol_new sun4i_codec_widgets[] = { SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, sun4i_codec_pa_volume_scale), + /* Line-In, FM-In, Mic1-In, Mic2-In */ + SOC_SINGLE_TLV("Line-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LNG, + 1, + 0, + sun4i_codec_linein_loopback_gain_scale), + SOC_SINGLE_TLV("FM-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_FMG, + 3, + 0, + sun4i_codec_fmin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic-In Playback Gain", + SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MICG, + 7, + 0, + sun4i_codec_micin_loopback_gain_scale), + SOC_SINGLE_TLV("Mic1-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic2-In Boost Gain", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2_A10, + 3, + 0, + sun4i_codec_micin_preamp_gain_scale_a10), + SOC_SINGLE_TLV("Mic1-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG1EN, + 1, + 0, + NULL), + SOC_SINGLE_TLV("Mic2-In Boost Switch", + SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PREG2EN, + 1, + 0, + NULL), }; static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), + SOC_DAPM_SINGLE("Left Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LLNS, 1, 0), + SOC_DAPM_SINGLE("Left FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1LS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2LS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { @@ -419,6 +522,14 @@ static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Right Line-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RLNS, 1, 0), + SOC_DAPM_SINGLE("Right FM-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RFMS, 1, 0), + SOC_DAPM_SINGLE("Mic1-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC1RS, 1, 0), + SOC_DAPM_SINGLE("Mic2-In Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIC2RS, 1, 0), }; static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { @@ -462,8 +573,16 @@ static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP Right"), SND_SOC_DAPM_OUTPUT("HP Left"), + + SND_SOC_DAPM_INPUT("Line-In Right"), + SND_SOC_DAPM_INPUT("Line-In Left"), + SND_SOC_DAPM_INPUT("FM-In Right"), + SND_SOC_DAPM_INPUT("FM-In Left"), + SND_SOC_DAPM_INPUT("Mic1-In"), + SND_SOC_DAPM_INPUT("Mic2-In"), }; +/* {sink, control, source} */ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { /* Left DAC Routes */ { "Left DAC", NULL, "DAC" }, @@ -490,6 +609,20 @@ static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, { "HP Right", NULL, "Pre-Amplifier Mute" }, { "HP Left", NULL, "Pre-Amplifier Mute" }, + + /* Line-In, FM-In */ + { "Right Mixer", "Right Line-In Playback Switch", "Line-In Right" }, + { "Left Mixer", "Left Line-In Playback Switch", "Line-In Left" }, + { "Right Mixer", "Right FM-In Playback Switch", "FM-In Right" }, + { "Left Mixer", "Left FM-In Playback Switch", "FM-In Left" }, + + /* Mic1-In */ + { "Right Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + { "Left Mixer", "Mic1-In Playback Switch", "Mic1-In" }, + + /* Mic2-In */ + { "Right Mixer", "Mic2-In Playback Switch", "Mic2-In" }, + { "Left Mixer", "Mic2-In Playback Switch", "Mic2-In" }, }; static struct snd_soc_codec_driver sun4i_codec_codec = { @@ -537,7 +670,7 @@ static const struct regmap_config sun4i_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, - .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, + .max_register = SUN7I_CODEC_AC_MIC_PHONE_CAL, }; static const struct of_device_id sun4i_codec_of_match[] = { @@ -594,6 +727,20 @@ static int sun4i_codec_probe(struct platform_device *pdev) void __iomem *base; int ret; + if (of_device_is_compatible(pdev->dev.of_node, "allwinner,sun7i-a20-codec")) { + size_t i, j; + for (i = 0; i < ARRAY_SIZE(sun7ionly_codec_widgets); ++i) { + struct snd_kcontrol_new *new_kcontrol; + new_kcontrol = &sun7ionly_codec_widgets[i]; + for (j = 0; j < ARRAY_SIZE(sun4i_codec_widgets); ++j) { + struct snd_kcontrol_new *kcontrol; + kcontrol = &sun4i_codec_widgets[j]; + if (strcmp(kcontrol->name, new_kcontrol->name) == 0) + memcpy(kcontrol, new_kcontrol, sizeof(*kcontrol)); + } + } + } + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); if (!scodec) return -ENOMEM;

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[alsa-devel] HDMI Repoll after pm?
by David Henningsson 10 Dec '15

10 Dec '15

Hi Takashi, A recent change in sound.git seems to have introduced some repoll after pm. Is this an intentional change? I don't think it's wrong per se (it stops after 6 repolls), but it breaks my hda-emu tester when "list" does not return the expected input. To reproduce, try running: ./hda-emu codecs/canonical/92hd91bxx-dell-inspiron-5423-ccert-201208-11456 -i 1 > pm .... lots of data .... ** TEST_PM DONE ** HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > list 1: HDMI/DP Jack:0 (1) 2: IEC958 Playback Con Mask:0 (1) 3: IEC958 Playback Pro Mask:0 (1) 4: IEC958 Playback Default:0 (1) 5: IEC958 Playback Switch:0 (1) 6: ELD:0 (1) 7: Playback Channel Map:0 (1) HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 JACK report HDMI/DP, status 0 > ...notice how the last two lines are somewhat unexpected output for a "list" command, but I can make my tool more robust in case you confirm this is an expected change. // David

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[alsa-devel] [PATCH 1/2] ALSA: hda - Optimize audio component check in patch_hdmi.c
by Takashi Iwai 10 Dec '15

10 Dec '15

The audio component is enabled only when CONFIG_SND_HDA_I915 is set. Give a dummy macro for allowing the compiler optimize out the relevant codes when this Kconfig isn't set. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 85342d261043..44d0d2374bb6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -152,8 +152,12 @@ struct hdmi_spec { struct i915_audio_component_audio_ops i915_audio_ops; }; +#ifdef CONFIG_SND_HDA_I915 #define codec_has_acomp(codec) \ ((codec)->bus->core.audio_component != NULL) +#else +#define codec_has_acomp(codec) false +#endif struct hdmi_audio_infoframe { u8 type; /* 0x84 */ -- 2.6.3

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[alsa-devel] [RFC][PATCH 0/3] ALSA: fireface: new driver for RME Fireface series (MIDI only)
by Takashi Sakamoto 10 Dec '15

10 Dec '15

Hi, This patchset is to add a new driver for RME Fireface series. This is still working in progress but currently the new driver support MIDI functionality. Unfortunately, ffado library can disturb this functionality. In RME::Device::init_hardware() function, the library sends a write transaction to 0x0000801003f4 with invalid value as higher part of address in IEEE 1212 or ISO/IEC 13213. This is a worst case I describe in patch 03. A workaround is to load this module after running ffado library. This module sends write transactions with valid values and regain MIDI functionality. I think it better that FFADO developers fixes the bug as long as they doesn't support MIDI functionality. Takashi Sakamoto (3): fireface: add skeleton for RME Fireface series fireface: add transaction support fireface: add support for MIDI functionality sound/firewire/Kconfig | 7 + sound/firewire/Makefile | 1 + sound/firewire/fireface/Makefile | 2 + sound/firewire/fireface/fireface-midi.c | 129 ++++++++ sound/firewire/fireface/fireface-transaction.c | 388 +++++++++++++++++++++++++ sound/firewire/fireface/fireface.c | 151 ++++++++++ sound/firewire/fireface/fireface.h | 73 +++++ 7 files changed, 751 insertions(+) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/fireface-midi.c create mode 100644 sound/firewire/fireface/fireface-transaction.c create mode 100644 sound/firewire/fireface/fireface.c create mode 100644 sound/firewire/fireface/fireface.h -- 2.5.0

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[alsa-devel] A root cause of "unable to create IPC shm instance"
by IceBsi 10 Dec '15

10 Dec '15

Dear ALSA developers: I found a root cause of the following issue by using strace: unable to create IPC shm instance Firstly, please see the test code below: 1 #include <stdio.h> 2 #include <unistd.h> 3 #include <sys/ipc.h> 4 #include <sys/shm.h> 5 #include <errno.h> 6 #include <string.h> 7 8 int main(int argc, char *argv[]) 9 { 10 int r = 0; 11 struct shmid_ds buf; 12 key_t ipc_key = (key_t)325559; 13 size_t shmsize = 488; 14 int shmid = -1; 15 mode_t ipc_perm = 0666; 16 void *shmptr = NULL; 17 pid_t pid; 18 19 printf("___%d:%d call shmget(%d,%u,%d)\n", __LINE__, getpid(), (int)(ipc_key), shmsize, IPC_CREAT | ipc_perm); 20 shmid = shmget(ipc_key, shmsize, IPC_CREAT | ipc_perm); 21 printf("___%d:%d ret shmget=%d\n", __LINE__, getpid(), shmid); 22 23 printf("___%d:%d call shmat(%d,0,0)\n", __LINE__, getpid(), shmid); 24 shmptr = shmat(shmid, 0, 0); 25 printf("___%d:%d ret shmat=%p\n", __LINE__, getpid(), shmptr); 26 27 printf("+++%d:%d call fork\n", __LINE__, getpid()); 28 pid = fork(); 29 printf("+++%d:%d ret fork=%d\n", __LINE__, getpid(), (int)pid); 30 31 printf("___%d:%d call shmctl(%d,IPC_STAT,%p)\n", __LINE__, getpid(), shmid, &buf); 32 r = shmctl(shmid, IPC_STAT, &buf); 33 printf("___%d:%d ret shmctl=%d\n", __LINE__, getpid(), r); 34 printf("___%d:%d nattch=%lu\n", __LINE__, getpid(), buf.shm_nattch); 35 36 printf("___%d:%d call shmdt(%p)\n", __LINE__, getpid(), shmptr); 37 r = shmdt(shmptr); 38 printf("___%d:%d ret shmdt=%d\n", __LINE__, getpid(), r); 39 40 printf("___%d:%d call shmctl(%d,IPC_RMID,NULL)\n", __LINE__, getpid(), shmid); 41 r = shmctl(shmid, IPC_RMID, NULL); 42 printf("___%d:%d ret shmctl=%d, %s\n", __LINE__, getpid(), r, strerror(errno)); 43 44 if ((pid_t)0 != pid) { 45 printf("+++%d:%d call wait(NULL)\n", __LINE__, getpid()); 46 pid = wait(NULL); 47 printf("+++%d:%d ret wait=%d\n", __LINE__, getpid(), (int)pid); 48 } 49 50 return 0; 51 } The output of the test code is: ___19:11782 call shmget(325559,488,950) ___21:11782 ret shmget=2850820 ___23:11782 call shmat(2850820,0,0) ___25:11782 ret shmat=0x7f452c223000 +++27:11782 call fork +++29:11782 ret fork=11783 ___31:11782 call shmctl(2850820,IPC_STAT,0x7fffdaf0a750) ___33:11782 ret shmctl=0 ___34:11782 nattch=2 ___36:11782 call shmdt(0x7f452c223000) +++29:11783 ret fork=0 ___38:11782 ret shmdt=0 ___40:11782 call shmctl(2850820,IPC_RMID,NULL) ___31:11783 call shmctl(2850820,IPC_STAT,0x7fffdaf0a750) ___42:11782 ret shmctl=0, Success +++45:11782 call wait(NULL) ___33:11783 ret shmctl=0 ___34:11783 nattch=1 ___36:11783 call shmdt(0x7f452c223000) ___38:11783 ret shmdt=0 ___40:11783 call shmctl(2850820,IPC_RMID,NULL) ___42:11783 ret shmctl=-1, Invalid argument +++47:11782 ret wait=11783 And sometimes output is: ___19:11798 call shmget(325559,488,950) ___21:11798 ret shmget=3112964 ___23:11798 call shmat(3112964,0,0) ___25:11798 ret shmat=0x7f6b39378000 +++27:11798 call fork +++29:11798 ret fork=11799 ___31:11798 call shmctl(3112964,IPC_STAT,0x7fff3a6472a0) ___33:11798 ret shmctl=0 +++29:11799 ret fork=0 ___31:11799 call shmctl(3112964,IPC_STAT,0x7fff3a6472a0) ___34:11798 nattch=2 ___36:11798 call shmdt(0x7f6b39378000) ___33:11799 ret shmctl=0 ___38:11798 ret shmdt=0 ___34:11799 nattch=2 ___40:11798 call shmctl(3112964,IPC_RMID,NULL) ___36:11799 call shmdt(0x7f6b39378000) ___42:11798 ret shmctl=0, Success +++45:11798 call wait(NULL) ___38:11799 ret shmdt=0 ___40:11799 call shmctl(3112964,IPC_RMID,NULL) ___42:11799 ret shmctl=-1, Invalid argument +++47:11798 ret wait=11799 >From the outputs the parent process always get nattch=2, the child process get nattch=1 or sometimes get nattch=2. Then please see the following lines of src/pcm/pcm_direct.c (from alsa-lib-1.1.0.tar.bz2): 110 dmix->shmptr = shmat(dmix->shmid, 0, 0); 111 if (dmix->shmptr == (void *) -1) { 112 err = -errno; 113 snd_pcm_direct_shm_discard(dmix); 114 return err; 115 } 116 mlock(dmix->shmptr, sizeof(snd_pcm_direct_share_t)); 117 if (shmctl(dmix->shmid, IPC_STAT, &buf) < 0) { 118 err = -errno; 119 snd_pcm_direct_shm_discard(dmix); 120 return err; 121 } 122 if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ 123 memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); 124 if (dmix->ipc_gid>= 0) { 125 buf.shm_perm.gid = dmix->ipc_gid; 126 shmctl(dmix->shmid, IPC_SET, &buf); 127 } 128 dmix->shmptr->magic = SND_PCM_DIRECT_MAGIC; 129 return 1; 130 } else { 131 if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { 132 snd_pcm_direct_shm_discard(dmix); 133 return -EINVAL; 134 } 135 } If an applicatioin that uses alsa-lib invokes fork() and the fork() is invoked just after shmat(line 110) and just before shmctl(line 117), buf.shm_nattch will be 2, so "buf.shm_nattch == 1" will be false. But while buf.shm_nattch is 2, we're still the first user this time, and we should executes line 123 to line 129 of the codes above. There is the following statement in `man shmget`: When a new shared memory segment is created, its contents are initialized to zero values According to the statement above, dmix->shmptr->magic is 0 on shm first created. So, to fix this issue, we should change the following line: 122 if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ to: 122 if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { /* we're the first user, clear the segment */ Sincerely, Sisrnb

2 1

[alsa-devel] [PATCH v3 0/4] Add get_eld audio component for i915/HD-audio
by Takashi Iwai 10 Dec '15

10 Dec '15

Hi, this is the third revision of the patchset to add get_eld op to audio component for communicating more directly between i915 and HD-audio. Currently, the HDMI/DP audio status and ELD are notified and obtained via the hardware-level communication over HD-audio unsolicited event and verbs although the graphics driver holds the exactly same information. As we already have a notification via audio component, this is another step forward; the audio driver fetches directly the audio status and ELD via the new component op. A few patches have been dropped from this patchset as they were already applied individually beforehand. Also, two cleanup patches for i915 were squashed into one patch. The current patchset is found in sound git tree test/hdmi-jack branch. thanks, Takashi === v2->v3: * Track drm_connector for easier ELD retrieval, remove superfluous audio_enabled flag instead * Back to av_mutex * Adapt the earlier applied preliminary patches * Direct invocation of get_eld now instead of deferred call, as we have no longer deadlocks * Rebase to drm-next, update get_eld documentation for new kernel doc v1->v2: * Use modeset lock for get_eld lock, drop av mutex * Return the expected size from get_eld, not the copied size * Add reverse map from a port number to the encoder * Deferred invocation of get_eld from the eld_notify in HDA side * A bit more code refactoring in HDA side * Move audio component accessors to hdac_i915.c Takashi Iwai (4): drm/i915: Add get_eld audio component drm/i915: Add reverse mapping between port and intel_encoder ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling ALSA: hda - Move audio component accesses to hdac_i915.c drivers/gpu/drm/i915/i915_drv.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 59 +++++++++++++++------ drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 1 + drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_hdmi.c | 2 + include/drm/i915_component.h | 14 +++++ include/sound/hda_i915.h | 14 +++++ sound/hda/hdac_i915.c | 66 +++++++++++++++++++++++ sound/pci/hda/patch_hdmi.c | 104 ++++++++++++++++++++++++++++++------- 10 files changed, 229 insertions(+), 36 deletions(-) -- 2.6.3

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[alsa-devel] using all channels of RME Babyface Pro in class compliant mode
by Be 10 Dec '15

10 Dec '15

Hello, I just got an RME Babyface Pro USB sound card. It can operate either as a USB audio class compliant device (with mixing controlled by the device's physical controls) or make use of a proprietary driver on Windows and Mac OS X. The proprietary driver allows for control of the mixing capabilities of the device and using onboard DSP EQ. It would be great if someone wrote a Linux driver to make use of these. I'd be happy to send debugging info or whatever about the device and test kernel patches. I could set up a Windows VM to intercept USB communication. In the meantime, I am trying to use its class compliant mode to use with Mixxx for DJing. However, I cannot figure out how to use channels 3-4 for my headphones. The device has 4 analog channels out and in plus 8 digital out and in (via ADAT). Its manual says, regarding class compliant mode, "in Mac OS X, either the first two or all channels will work, depending on the choice in the Audio MIDI Setup." I believe the two channel operation exists so when the Babyface Pro is plugged into an iPad, the iPad sends stereo output from iTunes and other programs to the Babyface Pro's main output. However, I do not know how to select between this 2 channel mode and the 12 channel mode with Linux. PulseAudio only shows it as a stereo device. Mixxx, which uses the PortAudio library, displays all 12 channels as available but doesn't work when any channels other than 1-2 are selected. Here is debugging output of Mixxx: Debug [Main]: SoundManager::setupDevices() Debug [Main]: SoundDevicePortAudio::open() "2, Babyface Pro (70785713): USB Audio (hw:1,0)" Debug [Main]: framesPerBuffer: 256 Debug [Main]: Requested sample rate: 44100 Hz, latency: 5.80499 ms Debug [Main]: Output channels: 4 | Input channels: 0 Debug [Main]: Opening stream with id 2 Expression 'alsa_snd_pcm_hw_params_set_channels( pcm, hwParams, self->numHostChannels )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2051 Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2717 Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/hostapi/alsa/pa_linux_alsa.c', line: 2838 Warning [Main]: Error opening stream: Invalid number of channels aplay -l output: **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC272 Analog [ALC272 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Pro70785713 [Babyface Pro (70785713)], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L output: null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default ALSA Output (currently PulseAudio Sound Server) sysdefault:CARD=PCH HDA Intel PCH, ALC272 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC272 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output sysdefault:CARD=Pro70785713 Babyface Pro (70785713), USB Audio Default Audio Device front:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio Front speakers surround21:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Pro70785713,DEV=0 Babyface Pro (70785713), USB Audio IEC958 (S/PDIF) Digital Audio Output /proc/asound/card1/stream0 when playing to channels 1-2 with PulseAudio: RME Babyface Pro (70785713) at usb-0000:00:1d.0-1.2, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 47 Momentary freq = 44118 Hz (0x5.83c9) Feedback Format = 16.16 Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Interface 1 Altset 2 Format: S24_3LE Channels: 12 Endpoint: 3 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Capture: Status: Stop Interface 2 Altset 1 Format: S24_3LE Channels: 12 Endpoint: 4 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Interface 2 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 4 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us So I guess the question is how to get PortAudio to use the "Altset 2" option. Could this be accomplished with some asoundrc sorcery? Does something need to change in snd-usb-audio? Is the issue with PortAudio and PulseAudio? How can I test channels 3-4 with aplay or another program?

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[alsa-devel] [PATCH] ALSA: hda - Implement loopback control switch for Realtek and other codecs
by Takashi Iwai 10 Dec '15

10 Dec '15

Many codecs, typically found on Realtek codecs, have the analog loopback path merged to the secondary input of the middle of the output paths. Currently, we don't offer the dynamic switching in such configuration but let each loopback path mute by itself. This should work well in theory, but in reality, we often see that such a dead loopback path causes some background noises even if all the elements get muted. Such a problem has been fixed by adding the quirk accordingly to disable aamix, and it's the right fix, per se. The only problem is that it's not so trivial to achieve it; user needs to pass a hint string via patch module option or sysfs. This patch gives a bit improvement on the situation: it adds "Loopback Mixing" control element for such codecs like other codecs (e.g. IDT or VIA codecs) with the individual loopback paths. User can turn on/off the loopback path simply via a mixer app. For keeping the compatibility, the loopback is still enabled on these codecs. But user can try to turn it off if experiencing a suspicious background or click noise on the fly, then build a static fixup later once after the problem is addressed. Other than the addition of the loopback enable/disablement control, there should be no changes. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 87 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 68 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f3c058f6c831..30c8efe0f80a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -754,9 +754,6 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps; unsigned int mask, val; - if (!enable && is_active_nid(codec, nid, dir, idx_to_check)) - return; - caps = query_amp_caps(codec, nid, dir); val = get_amp_val_to_activate(codec, nid, dir, caps, enable); mask = get_amp_mask_to_modify(codec, nid, dir, idx_to_check, caps); @@ -767,12 +764,22 @@ static void activate_amp(struct hda_codec *codec, hda_nid_t nid, int dir, update_amp(codec, nid, dir, idx, mask, val); } +static void check_and_activate_amp(struct hda_codec *codec, hda_nid_t nid, + int dir, int idx, int idx_to_check, + bool enable) +{ + /* check whether the given amp is still used by others */ + if (!enable && is_active_nid(codec, nid, dir, idx_to_check)) + return; + activate_amp(codec, nid, dir, idx, idx_to_check, enable); +} + static void activate_amp_out(struct hda_codec *codec, struct nid_path *path, int i, bool enable) { hda_nid_t nid = path->path[i]; init_amp(codec, nid, HDA_OUTPUT, 0); - activate_amp(codec, nid, HDA_OUTPUT, 0, 0, enable); + check_and_activate_amp(codec, nid, HDA_OUTPUT, 0, 0, enable); } static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, @@ -800,9 +807,16 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, * when aa-mixer is available, we need to enable the path as well */ for (n = 0; n < nums; n++) { - if (n != idx && (!add_aamix || conn[n] != spec->mixer_merge_nid)) - continue; - activate_amp(codec, nid, HDA_INPUT, n, idx, enable); + if (n != idx) { + if (conn[n] != spec->mixer_merge_nid) + continue; + /* when aamix is disabled, force to off */ + if (!add_aamix) { + activate_amp(codec, nid, HDA_INPUT, n, n, false); + continue; + } + } + check_and_activate_amp(codec, nid, HDA_INPUT, n, idx, enable); } } @@ -1563,6 +1577,12 @@ static bool map_singles(struct hda_codec *codec, int outs, return found; } +static inline bool has_aamix_out_paths(struct hda_gen_spec *spec) +{ + return spec->aamix_out_paths[0] || spec->aamix_out_paths[1] || + spec->aamix_out_paths[2]; +} + /* create a new path including aamix if available, and return its index */ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) { @@ -2405,25 +2425,51 @@ static void update_aamix_paths(struct hda_codec *codec, bool do_mix, } } +/* re-initialize the output paths; only called from loopback_mixing_put() */ +static void update_output_paths(struct hda_codec *codec, int num_outs, + const int *paths) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + int i; + + for (i = 0; i < num_outs; i++) { + path = snd_hda_get_path_from_idx(codec, paths[i]); + if (path) + snd_hda_activate_path(codec, path, path->active, + spec->aamix_mode); + } +} + static int loopback_mixing_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; + const struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int val = ucontrol->value.enumerated.item[0]; if (val == spec->aamix_mode) return 0; spec->aamix_mode = val; - update_aamix_paths(codec, val, spec->out_paths[0], - spec->aamix_out_paths[0], - spec->autocfg.line_out_type); - update_aamix_paths(codec, val, spec->hp_paths[0], - spec->aamix_out_paths[1], - AUTO_PIN_HP_OUT); - update_aamix_paths(codec, val, spec->speaker_paths[0], - spec->aamix_out_paths[2], - AUTO_PIN_SPEAKER_OUT); + if (has_aamix_out_paths(spec)) { + update_aamix_paths(codec, val, spec->out_paths[0], + spec->aamix_out_paths[0], + cfg->line_out_type); + update_aamix_paths(codec, val, spec->hp_paths[0], + spec->aamix_out_paths[1], + AUTO_PIN_HP_OUT); + update_aamix_paths(codec, val, spec->speaker_paths[0], + spec->aamix_out_paths[2], + AUTO_PIN_SPEAKER_OUT); + } else { + update_output_paths(codec, cfg->line_outs, spec->out_paths); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + update_output_paths(codec, cfg->hp_outs, spec->hp_paths); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + update_output_paths(codec, cfg->speaker_outs, + spec->speaker_paths); + } return 1; } @@ -2441,12 +2487,13 @@ static int create_loopback_mixing_ctl(struct hda_codec *codec) if (!spec->mixer_nid) return 0; - if (!(spec->aamix_out_paths[0] || spec->aamix_out_paths[1] || - spec->aamix_out_paths[2])) - return 0; if (!snd_hda_gen_add_kctl(spec, NULL, &loopback_mixing_enum)) return -ENOMEM; spec->have_aamix_ctl = 1; + /* if no explicit aamix path is present (e.g. for Realtek codecs), + * enable aamix as default -- just for compatibility + */ + spec->aamix_mode = !has_aamix_out_paths(spec); return 0; } @@ -5647,6 +5694,8 @@ static void init_aamix_paths(struct hda_codec *codec) if (!spec->have_aamix_ctl) return; + if (!has_aamix_out_paths(spec)) + return; update_aamix_paths(codec, spec->aamix_mode, spec->out_paths[0], spec->aamix_out_paths[0], spec->autocfg.line_out_type); -- 2.6.3

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[alsa-devel] [PATCH] ASoC: es8328: Fix deemphasis values
by John Keeping 09 Dec '15

09 Dec '15

This is using completely the wrong mask and value when updating the register. Since the correct values are already defined in the header, switch to using a table with explicit constants rather than shifting the array index. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + 2 files changed, 18 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 84f5eb0..afa6c5d 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -85,7 +85,15 @@ static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0); -static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; +static const struct { + int rate; + unsigned int val; +} deemph_settings[] = { + { 0, ES8328_DACCONTROL6_DEEMPH_OFF }, + { 32000, ES8328_DACCONTROL6_DEEMPH_32k }, + { 44100, ES8328_DACCONTROL6_DEEMPH_44_1k }, + { 48000, ES8328_DACCONTROL6_DEEMPH_48k }, +}; static int es8328_set_deemph(struct snd_soc_codec *codec) { @@ -97,21 +105,22 @@ static int es8328_set_deemph(struct snd_soc_codec *codec) * rate. */ if (es8328->deemph) { - best = 1; - for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - es8328->playback_fs) < - abs(deemph_settings[best] - es8328->playback_fs)) + best = 0; + for (i = 1; i < ARRAY_SIZE(deemph_settings); i++) { + if (abs(deemph_settings[i].rate - es8328->playback_fs) < + abs(deemph_settings[best].rate - es8328->playback_fs)) best = i; } - val = best << 1; + val = deemph_settings[best].val; } else { - val = 0; + val = ES8328_DACCONTROL6_DEEMPH_OFF; } dev_dbg(codec->dev, "Set deemphasis %d\n", val); - return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val); + return snd_soc_update_bits(codec, ES8328_DACCONTROL6, + ES8328_DACCONTROL6_DEEMPH_MASK, val); } static int es8328_get_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index cb36afe..156c748 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -153,6 +153,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_DACCONTROL6_CLICKFREE (1 << 3) #define ES8328_DACCONTROL6_DAC_INVR (1 << 4) #define ES8328_DACCONTROL6_DAC_INVL (1 << 5) +#define ES8328_DACCONTROL6_DEEMPH_MASK (3 << 6) #define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6) #define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6) #define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6) -- 2.6.3.462.gbe2c914

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[alsa-devel] [PATCH V3 0/2] Add support for Pistachio internal DAC
by Damien Horsley 09 Dec '15

09 Dec '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add binding document and driver for Pistachio internal DAC Changes in V3: Mute control definition changed to SOC_SINGLE Codec driver probe function added to associate regmap with codec Damien.Horsley (2): ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC .../bindings/sound/img,pistachio-internal-dac.txt | 18 ++ sound/soc/img/Kconfig | 8 + sound/soc/img/Makefile | 2 + sound/soc/img/pistachio-internal-dac.c | 287 +++++++++++++++++++++ 4 files changed, 315 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 sound/soc/img/pistachio-internal-dac.c -- 2.1.4

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