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

[alsa-devel] [PATCH v3] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 14 Jan '15

14 Jan '15

In soc_new_compress() when rtd->dai_link->daynamic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v3->v2: - fix a typo s/dynmic/dynamic/ in commit message - use if else if because either dpcm_playback or dpcm_capture should be on for compressed devices v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..025c38fbe3c0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

1 0

[alsa-devel] [PATCH v2] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 14 Jan '15

14 Jan '15

In soc_new_compress() when rtd->dai_link->daynmic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..27a668463ad7 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

4 6

[alsa-devel] "ak" vs "asahi-kasei" (was: Re: [PATCH 3/3 v3] ARM: shmobile: lager: add sound support on DT)
by Geert Uytterhoeven 13 Jan '15

13 Jan '15

Hi Morimoto-san, CC devicetree, alsa-devel On Wed, Jun 11, 2014 at 9:40 AM, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: >> On Wed, Jun 11, 2014 at 8:54 AM, Kuninori Morimoto >> <kuninori.morimoto.gx(a)renesas.com> wrote: >> > + compatible = "asahi-kasei,ak4643"; >> >> As Sergei already commented, >> Documentation/devicetree/bindings/vendor-prefixes.txt >> says >> >> ak Asahi Kasei Corp. > > I know, but, > > grep asahi-kasei sound/soc/codecs/* > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4554.c: { .compatible = "asahi-kasei,ak4554" }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > ak4642 driver can't probe by "ak"; That's bad. And all but one of these match entries have been added _after_ the addition of "ak" to vendor-prefixes.txt... Outside sound, there's also "asahi-kasei,ak8975" vs. "ak,ak8975". > My code is based on "code", not "document" Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

14 44

Re: [alsa-devel] randconfig build error with next-20150113, in sound/pci/hda/patch_realtek.c
by Takashi Iwai 13 Jan '15

13 Jan '15

At Tue, 13 Jan 2015 11:39:12 -0700, Jim Davis wrote: > > Building with the attached random configuration file, > > sound/built-in.o: In function `alc280_fixup_hp_gpio2_mic_hotkey': > /home/jim/linux/sound/pci/hda/patch_realtek.c:3512: undefined > reference to `input_allocate_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3520: undefined > reference to `input_register_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3522: undefined > reference to `input_free_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3550: undefined > reference to `input_unregister_device' > sound/built-in.o: In function `gpio2_mic_hotkey_event': > /home/jim/linux/include/linux/input.h:392: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:417: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:392: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:417: undefined reference to `input_event' > make: *** [vmlinux] Error 1 Thanks. So this is the case where the audio driver is built-in while the input driver is module. I guess the cleanest workaround is just to select INPUT from SND_HDA_CODEC_REALTEK in sound/pci/hda/Kconfig. Practically seen, CONFIG_INPUT is always set, so there must be no real drawback with it. (And we can reduce a few lines as a bonus.) David, care to prepare a patch? thanks, Takashi

1 0

[alsa-devel] [PATCH] pyalsa: alsaseq: fix memory leaks
by Hector Martin 13 Jan '15

13 Jan '15

Fix a bunch of dangling reference issues. Also switch to Py_BuildValue where appropriate, since that makes it easier to build complex dictionary/tuple structures without leaking integer/string references. Signed-off-by: Hector Martin <marcan(a)marcan.st> --- I haven't looked at the other parts of pyalsa; those could probably use some looking over too. I went through alsaseq since I'm writing a long-running MIDI client and I noticed it was leaking all event objects. pyalsa/alsaseq.c | 156 +++++++++++++++++++++---------------------------------- 1 file changed, 58 insertions(+), 98 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index e34d3b4..ce332ed 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -124,7 +124,9 @@ if (PyModule_AddObject(module, name, tmp) < 0) { \ return; \ } \ - PyDict_SetItem(TDICT(subtype), PyInt_FromLong(value), tmp); \ + PyObject *key = PyInt_FromLong(value); \ + PyDict_SetItem(TDICT(subtype), key, tmp); \ + Py_DECREF(key); \ } #define TCONSTRETURN(subtype, value) { \ @@ -199,8 +201,11 @@ PyDict_SetItemString(dict, name, object) /* sets a integer into the dict */ -#define SETDICTINT(name, value) \ - PyDict_SetItemString(dict, name, PyInt_FromLong(value)) +#define SETDICTINT(name, value) { \ + PyObject *val = PyInt_FromLong(value); \ + PyDict_SetItemString(dict, name, val); \ + Py_DECREF(val); \ + } /* sets note info dict (used by SeqEvent_get_data) */ #define SETDICT_NOTE3 { \ @@ -288,6 +293,7 @@ PyList_SetItem(list, i, PyInt_FromLong(t[i])); \ } \ SETDICTOBJ("ext", list); \ + Py_DECREF(list); \ } /* gets integer from python param */ @@ -1767,7 +1773,7 @@ SeqEvent_repr(SeqEventObject *self) { return PyString_FromFormat("<alsaseq.SeqEvent type=%s(%d) flags=%d tag=%d " "queue=%d time=%s(%u.%u) from=%d:%d to=%d:%d " - "at 0x%p>", + "at %p>", typestr, self->event->type, self->event->flags, self->event->tag, self->event->queue, @@ -2367,34 +2373,23 @@ _query_connections_list(snd_seq_t *handle, PyObject *list = PyList_New(0); int index = 0; - long tmplong; snd_seq_query_subscribe_set_type(query, type); snd_seq_query_subscribe_set_index(query, index); while (snd_seq_query_port_subscribers(handle, query) >= 0) { const snd_seq_addr_t *addr = snd_seq_query_subscribe_get_addr(query); - PyObject *dict = PyDict_New(); - - tmplong = snd_seq_query_subscribe_get_queue(query); - PyDict_SetItemString(dict, "queue", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_exclusive(query); - PyDict_SetItemString(dict, "exclusive", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_time_update(query); - PyDict_SetItemString(dict, "time_update", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_time_real(query); - PyDict_SetItemString(dict, "time_real", PyInt_FromLong(tmplong)); - - - PyObject *tuple = PyTuple_New(3); - PyTuple_SetItem(tuple, 0, PyInt_FromLong(addr->client)); - PyTuple_SetItem(tuple, 1, PyInt_FromLong(addr->port)); - PyTuple_SetItem(tuple, 2, dict); + PyObject *tuple = Py_BuildValue( + "(ii{sisisisi})", + (int)addr->client, + (int)addr->port, + "queue", (int)snd_seq_query_subscribe_get_queue(query), + "exclusive", (int)snd_seq_query_subscribe_get_exclusive(query), + "time_update", (int)snd_seq_query_subscribe_get_time_update(query), + "time_real", (int)snd_seq_query_subscribe_get_time_real(query)); PyList_Append(list, tuple); + Py_DECREF(tuple); snd_seq_query_subscribe_set_index(query, ++index); } return list; @@ -2466,11 +2461,13 @@ Sequencer_connection_list(SequencerObject *self, PyTuple_SetItem(porttuple, 2, conntuple); PyList_Append(portlist, porttuple); + Py_DECREF(porttuple); } PyTuple_SetItem(tuple, 2, portlist); /* append list of port tuples */ PyList_Append(list, tuple); + Py_DECREF(tuple); } return list; @@ -2506,9 +2503,6 @@ Sequencer_get_client_info(SequencerObject *self, snd_seq_client_info_t *cinfo; int client_id = -1; int ret; - PyObject *tmpobj; - long tmplong; - const char * tmpchar; char *kwlist[] = { "client_id", NULL}; if (!PyArg_ParseTupleAndKeywords(args, kwds, "|i", kwlist, @@ -2532,37 +2526,21 @@ Sequencer_get_client_info(SequencerObject *self, } } - PyObject *dict = PyDict_New(); - if (dict == NULL) { - return NULL; - } - - TCONSTASSIGN(ADDR_CLIENT, client_id, tmpobj); - PyDict_SetItemString(dict, "id", tmpobj); - - tmplong = snd_seq_client_info_get_type(cinfo); - TCONSTASSIGN(CLIENT_TYPE, tmplong, tmpobj); - PyDict_SetItemString(dict, "type", tmpobj); - - tmpchar = snd_seq_client_info_get_name(cinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "name", PyString_FromString(tmpchar)); - - tmplong = snd_seq_client_info_get_broadcast_filter(cinfo); - PyDict_SetItemString(dict, "broadcast_filter", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_client_info_get_error_bounce(cinfo); - PyDict_SetItemString(dict, "error_bounce", PyInt_FromLong(tmplong)); - - tmpchar = (const char *)snd_seq_client_info_get_event_filter(cinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "event_filter", PyString_FromString(tmpchar)); + PyObject *d_id, *d_type; + TCONSTASSIGN(ADDR_CLIENT, client_id, d_id); + TCONSTASSIGN(CLIENT_TYPE, snd_seq_client_info_get_type(cinfo), d_type); + const char *d_name = snd_seq_client_info_get_name(cinfo); - tmplong = snd_seq_client_info_get_num_ports(cinfo); - PyDict_SetItemString(dict, "num_ports", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_client_info_get_event_lost(cinfo); - PyDict_SetItemString(dict, "event_lost", PyInt_FromLong(tmplong)); + PyObject *dict = Py_BuildValue( + "{sNsNsssisiss#sisi}", + "id", d_id, + "type", d_type, + "name", d_name == NULL ? "" : d_name, + "broadcast_filter", (int)snd_seq_client_info_get_broadcast_filter(cinfo), + "error_bounce", (int)snd_seq_client_info_get_error_bounce(cinfo), + "event_filter", snd_seq_client_info_get_event_filter(cinfo), 32, + "num_ports", (int)snd_seq_client_info_get_num_ports(cinfo), + "event_lost", (int)snd_seq_client_info_get_event_lost(cinfo)); return dict; } @@ -2594,8 +2572,6 @@ Sequencer_get_port_info(SequencerObject *self, snd_seq_client_info_t *cinfo; int port_id; int client_id; - const char *tmpchar; - long tmplong; int ret; char *kwlist[] = { "port_id", "client_id", NULL }; @@ -2612,11 +2588,6 @@ Sequencer_get_port_info(SequencerObject *self, return NULL; } - PyObject *dict = PyDict_New(); - if (dict == NULL) { - return NULL; - } - snd_seq_port_info_alloca(&pinfo); ret = snd_seq_get_any_port_info(self->handle, client_id, port_id, pinfo); @@ -2625,17 +2596,11 @@ Sequencer_get_port_info(SequencerObject *self, return NULL; } - tmpchar = snd_seq_port_info_get_name(pinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "name", PyString_FromString(tmpchar)); - - tmplong = snd_seq_port_info_get_capability(pinfo); - PyDict_SetItemString(dict, "capability", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_info_get_type(pinfo); - PyDict_SetItemString(dict, "type", PyInt_FromLong(tmplong)); - - return dict; + return Py_BuildValue( + "{sssIsI}", + "name", snd_seq_port_info_get_name(pinfo), + "capability", (unsigned int)snd_seq_port_info_get_capability(pinfo), + "type", (unsigned int)snd_seq_port_info_get_type(pinfo)); } /** alsaseq.Sequencer connect_ports() method: __doc__ */ @@ -2768,7 +2733,6 @@ Sequencer_get_connect_info(SequencerObject *self, snd_seq_addr_t sender, dest; snd_seq_port_subscribe_t *sinfo; int ret; - long tmplong; if (!PyArg_ParseTuple(args, "(BB)(BB)", &(sender.client), &(sender.port), &(dest.client), &(dest.port))) { @@ -2786,21 +2750,12 @@ Sequencer_get_connect_info(SequencerObject *self, return NULL; } - PyObject *dict = PyDict_New(); - - tmplong = snd_seq_port_subscribe_get_queue(sinfo); - PyDict_SetItemString(dict, "queue", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_exclusive(sinfo); - PyDict_SetItemString(dict, "exclusive", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_time_update(sinfo); - PyDict_SetItemString(dict, "time_update", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_time_real(sinfo); - PyDict_SetItemString(dict, "time_real", PyInt_FromLong(tmplong)); - - return dict; + return Py_BuildValue( + "{sisisisi}", + "queue", (int)snd_seq_port_subscribe_get_queue(sinfo), + "exclusive", (int)snd_seq_port_subscribe_get_exclusive(sinfo), + "time_update", (int)snd_seq_port_subscribe_get_time_update(sinfo), + "time_real", (int)snd_seq_port_subscribe_get_time_real(sinfo)); } /** alsaseq.Sequencer receive_events() method: __doc__ */ @@ -2878,6 +2833,7 @@ Sequencer_receive_events(SequencerObject *self, } PyList_Append(list, SeqEventObject); + Py_DECREF(SeqEventObject); maxevents --; @@ -3263,19 +3219,23 @@ Sequencer_registerpoll(SequencerObject *self, PyObject *args, PyObject *kwds) if (count <= 0) Py_RETURN_NONE; - reg = PyObject_GetAttr(pollObj, PyString_InternFromString("register")); + reg = PyObject_GetAttrString(pollObj, "register"); + if (!reg) + return NULL; for (i = 0; i < count; i++) { t = PyTuple_New(2); - if (t) { - PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); - PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); - Py_XDECREF(PyObject_CallObject(reg, t)); - Py_DECREF(t); + if (!t) { + Py_DECREF(reg); + return NULL; } + PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); + PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); + Py_XDECREF(PyObject_CallObject(reg, t)); + Py_DECREF(t); } - Py_XDECREF(reg); + Py_DECREF(reg); Py_RETURN_NONE; } -- 2.2.1

2 1

[alsa-devel] [PATCH] hdajackretask: Add dock hp/mic/line to simple options
by David Henningsson 13 Jan '15

13 Jan '15

This allows for simpler setting of dock headphone, dock mic, dock line out and dock line in. Also improve detection a bit so that an existing dock headphone shows up as such and not as regular headphone. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/sysfs-pin-configs.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index 592ffe2..693842f 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -289,6 +289,8 @@ unsigned long actual_pin_config(pin_configs_t* pins) #define AC_DEFCFG_DEVICE_SHIFT 20 #define AC_DEFCFG_LOCATION (0x3f<<24) #define AC_DEFCFG_LOCATION_SHIFT 24 +#define AC_DEFCFG_GROSSLOC (0x3<<28) +#define AC_DEFCFG_GROSSLOC_SHIFT 28 #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 @@ -594,21 +596,27 @@ static gboolean disabled_caps(unsigned long pin_caps, unsigned long wid_caps) return TRUE; } +#define DEFAULT_MASK (AC_DEFCFG_PORT_CONN | AC_DEFCFG_GROSSLOC | AC_DEFCFG_DEVICE) + static typical_pins_t simple_typical_pins[] = { - {"Headphone", headphone_caps, 0x0321403f, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE, }, - {"Microphone", extmic_caps, 0x03a19020, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"Line out (Front)", lineout_caps, 0x01014410, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Center/LFE)", lineout_caps, 0x01014411, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Back)", lineout_caps, 0x01014412, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Side)", lineout_caps, 0x01014413, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line in", linein_caps, 0x0181344f, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"Internal speaker", lineout_caps, 0x90170150, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal speaker (LFE)", lineout_caps, 0x90170151, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal speaker (Back)", lineout_caps, 0x90170152, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal mic", intmic_caps, 0x90a60160, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"HDMI", hdmi_caps, 0x18560070, AC_DEFCFG_PORT_CONN | AC_DEFCFG_LOCATION,}, + {"Headphone", headphone_caps, 0x0321403f, DEFAULT_MASK, }, + {"Microphone", extmic_caps, 0x03a19020, DEFAULT_MASK,}, + {"Line out (Front)", lineout_caps, 0x01014010, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Center/LFE)", lineout_caps, 0x01014011, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Back)", lineout_caps, 0x01014012, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Side)", lineout_caps, 0x01014013, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line in", linein_caps, 0x0181304f, DEFAULT_MASK,}, + {"Internal speaker", lineout_caps, 0x90170150, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal speaker (LFE)", lineout_caps, 0x90170151, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal speaker (Back)", lineout_caps, 0x90170152, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal mic", intmic_caps, 0x90a60160, DEFAULT_MASK,}, + {"HDMI / DisplayPort", hdmi_caps, 0x18560070, AC_DEFCFG_PORT_CONN | AC_DEFCFG_LOCATION,}, {"SPDIF out", spdifout_caps, 0x014b1180, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, {"SPDIF in", spdifin_caps, 0x01cb6190, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, + {"Dock Headphone", headphone_caps, 0x222140af, DEFAULT_MASK, }, + {"Dock Microphone", extmic_caps, 0x22a190a0, DEFAULT_MASK,}, + {"Dock Line out", lineout_caps, 0x220140b0, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Dock Line in", linein_caps, 0x228130bf, DEFAULT_MASK,}, {"Not connected", disabled_caps, 0x40f000f0, AC_DEFCFG_PORT_CONN,}, {} }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: compress_driver.h: include sound/core.h explicitly
by Qais Yousef 13 Jan '15

13 Jan '15

Fixes the following compilation error: include/sound/compress_driver.h: In function ‘snd_compr_drain_notify’: include/sound/compress_driver.h:177:2: error: implicit declaration of function ‘snd_BUG_ON’ [-Werror=implicit-function-declaration] if (snd_BUG_ON(!stream)) snd_BUG_ON() is defined in sound/core.h but the file is not included explicitly, so include it. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- include/sound/compress_driver.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 396e8f73670a..1e2531058b7e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -27,6 +27,7 @@ #include <linux/types.h> #include <linux/sched.h> +#include <sound/core.h> #include <sound/compress_offload.h> #include <sound/asound.h> #include <sound/pcm.h> -- 2.1.0

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[alsa-devel] Configuring slave output for ALC889
by Dheeraj CVR 13 Jan '15

13 Jan '15

Hi, I am using an Intel Desktop Board DX58SO. I am using S/PDIF for HDMI Audio and there is no sound output via HDMI. I have checked with alsamixer and the volumes for the S/PDIF are stuck at 0 and I can't increase them. By looking at the commit history, I have found this commit which enables 0x10 as a slave output for 0x6 on ALC889 for the exact same board that I am using. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/015901.html However, it appears that with the commit "ALSA: hda/realtek - Drop ALC882 desktop model quirks", the older workaround has been removed, which caused a regression. I have checked the output of alsainfo and it seems that 0x06 is still being used instead of 0x010. Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x99430130: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 I was planning to edit the "patch_alc882" function to configure the "spec->gen.slave_dig_outs" and set the values to "{0x10, 0, 0}". However, I was not sure if this was the right way of doing. Please advice. I am running Arch-Linux with 3.17 kernel. -- Regards, Dheeraj CVR.

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[alsa-devel] [PATCH v2] ASoC: wm8904: fix runtime warning
by Bo Shen 13 Jan '15

13 Jan '15

Correct the path name for mux to get rid of the following warning: --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Correct the mux path name. - Route the mux to the pins. sound/soc/codecs/wm8904.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..110d0e6 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,14 +1092,20 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "AIFINL" }, + { "DACL", NULL, "AIFINR" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "AIFINL" }, + { "DACR", NULL, "AIFINR" }, { "DACR", NULL, "CLK_DSP" }, + { "DACL Mux", "Left", "DACL" }, + { "DACL Mux", "Right", "DACR" }, + + { "DACR Mux", "Left", "DACL" }, + { "DACR Mux", "Right", "DACR" }, + { "Charge pump", NULL, "SYSCLK" }, { "Headphone Output", NULL, "HPL PGA" }, -- 2.1.0.24.g4109c28

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[alsa-devel] [PATCH] ASoC: wm8904: fix runtime warning
by Bo Shen 13 Jan '15

13 Jan '15

As the commit 5fe5b76 (ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets) remove the fake control. Then, the wm8904 throw out the following warning at runtime. --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- This patch is used to remove these warning. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..38582d7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,10 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTL", NULL, "ADCR" }, + { "AIFOUTR", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1089,12 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "AIFINR" }, + { "DACL", NULL, "AIFINL" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "AIFINR" }, + { "DACR", NULL, "AIFINL" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, -- 2.1.0.24.g4109c28

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