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[alsa-devel] Query regarding SND_PCM_PLUGIN_DEFINE_FUNC()
by Pharaoh . 18 Mar '08

18 Mar '08

Hi, What does the mode field in SND_PCM_PLUGIN_DEFINE_FUNC() used for? is it for determining blocking mode or non-blocking mode of opening? -pharaoh

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[alsa-devel] [PATCH] Removed deprecated sound/driver.h from Freescale MPC8610 drivers
by Timur Tabi 18 Mar '08

18 Mar '08

With commit 9004acc70e8c49c50c4c7b652f906f1e0ed5709d, include/sound/driver.h is deprecated. This patch removes the #include from fsl_ssi.c and fsl_dma.c. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 1 - sound/soc/fsl/fsl_ssi.c | 1 - 2 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 652514f..78de716 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -20,7 +20,6 @@ #include <linux/interrupt.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 145ad13..b2a11b0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -15,7 +15,6 @@ #include <linux/device.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.5.4

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[alsa-devel] [PATCH] usbmidi.c: Call usb_autopm functions from rawmidi open/close callbacks
by Karsten Wiese 18 Mar '08

18 Mar '08

Lets "usb auto suspend" work for devices that set their usb_driver::supports_autosuspend to 1. Has no effect for other devices. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- sound/usb/usbmidi.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6c616c7..842486f 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -860,11 +860,14 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; + usb_autopm_get_interface(umidi->iface); return 0; } static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct snd_usb_midi* umidi = substream->rmidi->private_data; + usb_autopm_put_interface(umidi->iface); return 0; } -- 1.5.3.3 -------------------------------------------------------

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[alsa-devel] /usr/share/alsa/alsa.conf :and file extention
by Alexander Indenbaum 18 Mar '08

18 Mar '08

Hello, I'm playing with embedded system without any hardware sound device. I'd like to grab sound from ALSA application, for instance Firefox displaying YouTube clip inside flash plugin and to store output PCM stream into file on local disk. I got an advice to use File extension. So I defined following /usr/share/alsa/alsa.conf pcm.default{ type file file /tmp/pcm.out } But when alsa lib still complains about "Unknown PCM plughw:0,0" "Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_file.so" I have a nasty feeling that syntax of my /usr/share/alsa/alsa.conf is broken beyond recognition. I wonder what is the correct way to define my situation: * No real audio devices * All sound is output goes to file as PCM stream Do you have a clue? Alexander Indenbaum

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[alsa-devel] ~ALSA Audio coordinator/manager
by Harsha priya gupta 18 Mar '08

18 Mar '08

Hi all, I am looking to see if anyone is working on an Audio manager kind of solution as a part of ALSA. 'By Audio manager, I mean a coordinator of all audio input and output streams. A coordinator which is application and devices aware . Example functionality of an audio manager would to direct different steams to different devices like (voice output to headset and music output to speakers). Or to pause an media player when a voice recording starts.' If anyone is doing any work on these lines, please let me know about the work and its integration into ALSA as well. Thanks in Advance, Harsha.

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[alsa-devel] basic questions
by Harsha priya gupta 18 Mar '08

18 Mar '08

Hi, I am ramping up on ALSA architecture and framework. I have a couple of basic questions. I apologize if these questions are very basic to be raised in this mailing list. If anyone can help me find answers, it would be great. Thanks in advance. 1. My assumption is that ALSA handles only PCM streams today. Does it handle encoded streams from application to sound card (Which can do hardware decoding)? 2. What are the hwdep files in ALSA used for? Where do the concerned IOCTLS land to? Thanks and Regards, Harsha.

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[alsa-devel] [PATCH] Add missing audio path between Mono Mixer and Mic PGAs.
by Mark Brown 18 Mar '08

18 Mar '08

From: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9713.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e2949c5..172ca1a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -500,6 +500,8 @@ static const char *audio_map[][3] = { {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"}, + {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"}, {"Mono Mixer", NULL, "Capture Mono Mux"}, /* DAC inv mux 1 */ -- 1.5.4.4

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[alsa-devel] [PATCH] Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267)
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:18:52.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:48:18.000000000 -0300 @@ -7763,6 +7763,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), -- []'s Herton

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[alsa-devel] [PATCH] Fix DAC assignment order in ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Actually clfe and surround DACs are inverted in alc883_dac_nids array (see ALC883 datasheet). I discovered this while testing multi-channel setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped in 6 channel mode (also in 4 channel mode you didn't get rear left/right output). Other models also were affected by this bug, as can be seen by the mixer layouts that "workaround" this (the real bug was not noticed, and some other models simply played with mixer and initial verbs). Thus along with fixing the order of dac nids, also change the models that relied on previous dac ordering properly. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:16:12.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:18:52.000000000 -0300 @@ -6450,7 +6450,7 @@ static int patch_alc882(struct hda_codec static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03, 0x05 + 0x02, 0x03, 0x04, 0x05 }; static hda_nid_t alc883_adc_nids[2] = { @@ -6958,12 +6958,12 @@ static struct snd_kcontrol_new alc883_me static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -6998,12 +6998,12 @@ static struct snd_kcontrol_new alc888_6s static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -7036,12 +7036,12 @@ static struct snd_kcontrol_new alc888_3s static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -7332,26 +7332,14 @@ static struct hda_verb alc883_haier_w66_ { } /* end */ }; -static struct hda_verb alc888_6st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ - { } -}; - static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; static struct hda_verb alc888_6st_dell_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -7985,7 +7973,7 @@ static struct alc_config_preset alc883_p }, [ALC888_6ST_HP] = { .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, + .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, -- []'s Herton

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[alsa-devel] query about snd_pcm_hw_refine
by Pharaoh . 18 Mar '08

18 Mar '08

Hi I get following error when trying to run aplay with my plugin: #aplay -D myplug test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, St ereo aplay: set_params:882: Broken configuration for this PCM: no configurations available. After looking at the code, I figured out that its snd_pcm_hw_refine which is causing problem. It looks like following in pcm_params.c in alsa lib 1.0.13, int snd_pcm_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { int res; #ifdef REFINE_DEBUG snd_output_t *log; snd_output_stdio_attach(&log, stderr, 0); #endif printf("#####%s \n", __FUNCTION__); assert(pcm && params); #ifdef REFINE_DEBUG snd_output_printf(log, "REFINE called:\n"); snd_pcm_hw_params_dump(params, log); #endif printf("#####%s calling hw_refine\n", __FUNCTION__); res = pcm->ops->hw_refine(pcm->op_arg, params); printf("#####% return val of hw_refine = %d\n", res); #ifdef REFINE_DEBUG snd_output_printf(log, "refine done - result = %i\n", res); snd_pcm_hw_params_dump(params, log); snd_output_close(log); #endif return res; } I have modified the alsa.conf for my plugin by adding following in it at the end, pcm.myplug { type myplug card 0 } What could be the problem? -pharaoh.

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