December 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 9/9] sound/soc/codecs: Correct code taking the size of a pointer
by Julia Lawall 14 Dec '09

14 Dec '09

From: Julia Lawall <julia(a)diku.dk> sizeof(codec->reg_cache) is just the size of the pointer. Elsewhere in the file, codec->reg_cache is used with sizeof(wm8900_reg_defaults), so the code is changed to do the same here. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression *x; expression f; type T; @@ *f(...,(T)x,...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/codecs/wm8900.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index c9438dd..dbc368c 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec) snd_soc_write(codec, WM8900_REG_RESET, 0); memcpy(codec->reg_cache, wm8900_reg_defaults, - sizeof(codec->reg_cache)); + sizeof(wm8900_reg_defaults)); } static int wm8900_hp_event(struct snd_soc_dapm_widget *w,

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[alsa-devel] Two codecs using the same I2S bus
by Niels Langendorff 14 Dec '09

14 Dec '09

Hi, for a samsung based project we have chosen two codecs, one for input, the other for output. The two codecs are using the same I2S bus due to audio requirements (HF audio in combination with EC) When having two machine implementations at the same time for the two codecs we will run into problems attaching to the I2S bus. A solution is to combine the machine implementation into one file, but then I probaly have to modify generic kernel code. I there some solution for this and/or example code Thanks in advance, Niels Langendorff

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[alsa-devel] [PATCH] Recognise and use subdevice 0x3061 found in some HP dv6 notebooks.
by Kunal Gangakhedkar 14 Dec '09

14 Dec '09

Obvious patch. Recently bought an HP Pavilion dv6-1110ax notebook. The modparm 'model=hp-dv5' works, but I thought it was better if it were natively supported. Also, the 'hp_detect=1' hint (echo "hp_detect = 1" > /sys/class/sound/hwC0D0/hints) works nicely to mute the speakers when using the headphone jack(s). Is there any way to automatically enable this? I'm no alsa dev - just scratching my itch here ;) PS: I'm not subscribed to the list - so, please put me in Cc: in the replies. Thanks, Kunal Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f83..0ed9a09 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1725,6 +1725,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP Pavilion dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- 1.6.5.GIT

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[alsa-devel] [PATCH] sb_mixer: convert pointer tables to mixer control tables
by Krzysztof Helt 14 Dec '09

14 Dec '09

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Convert table of pointers to mixer controls into tables of the mixer controls. It saves about 20% of the snd-sb-common module size reported by lsmod. The als4000 uses part of sb16's control table. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sb/sb_mixer.c | 330 ++++++++++++++++------------------------------ 1 files changed, 115 insertions(+), 215 deletions(-) diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 318ff0c..8cfc41f 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -528,20 +528,11 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty * SB 2.0 specific mixer elements */ -static struct sbmix_elem snd_sb20_ctl_master_play_vol = - SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_pcm_play_vol = - SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3); -static struct sbmix_elem snd_sb20_ctl_synth_play_vol = - SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7); -static struct sbmix_elem snd_sb20_ctl_cd_play_vol = - SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7); - -static struct sbmix_elem *snd_sb20_controls[] = { - &snd_sb20_ctl_master_play_vol, - &snd_sb20_ctl_pcm_play_vol, - &snd_sb20_ctl_synth_play_vol, - &snd_sb20_ctl_cd_play_vol +static struct sbmix_elem snd_sb20_controls[] = { + SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7), + SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3), + SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7), + SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7) }; static unsigned char snd_sb20_init_values[][2] = { @@ -552,41 +543,24 @@ static unsigned char snd_sb20_init_values[][2] = { /* * SB Pro specific mixer elements */ -static struct sbmix_elem snd_sbpro_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_pcm_play_filter = - SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7); -static struct sbmix_elem snd_sbpro_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3); -static struct sbmix_elem snd_sbpro_ctl_capture_source = +static struct sbmix_elem snd_sbpro_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7), + SB_DOUBLE("PCM Playback Volume", + SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7), + SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7), + SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7), + SB_DOUBLE("Line Playback Volume", + SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7), + SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3), { .name = "Capture Source", .type = SB_MIX_CAPTURE_PRO - }; -static struct sbmix_elem snd_sbpro_ctl_capture_filter = - SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1); -static struct sbmix_elem snd_sbpro_ctl_capture_low_filter = - SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1); - -static struct sbmix_elem *snd_sbpro_controls[] = { - &snd_sbpro_ctl_master_play_vol, - &snd_sbpro_ctl_pcm_play_vol, - &snd_sbpro_ctl_pcm_play_filter, - &snd_sbpro_ctl_synth_play_vol, - &snd_sbpro_ctl_cd_play_vol, - &snd_sbpro_ctl_line_play_vol, - &snd_sbpro_ctl_mic_play_vol, - &snd_sbpro_ctl_capture_source, - &snd_sbpro_ctl_capture_filter, - &snd_sbpro_ctl_capture_low_filter + }, + SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1), + SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1) }; static unsigned char snd_sbpro_init_values[][2] = { @@ -598,68 +572,42 @@ static unsigned char snd_sbpro_init_values[][2] = { /* * SB16 specific mixer elements */ -static struct sbmix_elem snd_sb16_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_3d_enhance_switch = - SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1); -static struct sbmix_elem snd_sb16_ctl_tone_bass = - SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_tone_treble = - SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15); -static struct sbmix_elem snd_sb16_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_synth_capture_route = - SB16_INPUT_SW("Synth Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5); -static struct sbmix_elem snd_sb16_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_cd_capture_route = - SB16_INPUT_SW("CD Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_switch = - SB_DOUBLE("CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1); -static struct sbmix_elem snd_sb16_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_line_capture_route = - SB16_INPUT_SW("Line Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3); -static struct sbmix_elem snd_sb16_ctl_line_play_switch = - SB_DOUBLE("Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1); -static struct sbmix_elem snd_sb16_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31); -static struct sbmix_elem snd_sb16_ctl_mic_capture_route = - SB16_INPUT_SW("Mic Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0); -static struct sbmix_elem snd_sb16_ctl_mic_play_switch = - SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1); -static struct sbmix_elem snd_sb16_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); -static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); -static struct sbmix_elem snd_sb16_ctl_capture_vol = - SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_play_vol = - SB_DOUBLE("Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3); -static struct sbmix_elem snd_sb16_ctl_auto_mic_gain = - SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1); - -static struct sbmix_elem *snd_sb16_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_sb16_ctl_3d_enhance_switch, - &snd_sb16_ctl_tone_bass, - &snd_sb16_ctl_tone_treble, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_sb16_ctl_auto_mic_gain +static struct sbmix_elem snd_sb16_controls[] = { + SB_DOUBLE("Master Playback Volume", + SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31), + SB_DOUBLE("PCM Playback Volume", + SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Synth Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5), + SB_DOUBLE("Synth Playback Volume", + SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("CD Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31), + SB16_INPUT_SW("Mic Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + SB_DOUBLE("Capture Volume", + SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), + SB_DOUBLE("Playback Volume", + SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), + SB16_INPUT_SW("Line Capture Route", + SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), + SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), + SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1), + SB_DOUBLE("Tone Control - Bass", + SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15), + SB_DOUBLE("Tone Control - Treble", + SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15) }; static unsigned char snd_sb16_init_values[][2] = { @@ -678,46 +626,34 @@ static unsigned char snd_sb16_init_values[][2] = { /* * DT019x specific mixer elements */ -static struct sbmix_elem snd_dt019x_ctl_master_play_vol = - SB_DOUBLE("Master Playback Volume", SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_vol = - SB_DOUBLE("PCM Playback Volume", SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_synth_play_vol = - SB_DOUBLE("Synth Playback Volume", SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = - SB_DOUBLE("CD Playback Volume", SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = - SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); -static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); -static struct sbmix_elem snd_dt019x_ctl_line_play_vol = - SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); -static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = - SB_DOUBLE("PCM Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2,1, 1); -static struct sbmix_elem snd_dt019x_ctl_synth_play_switch = - SB_DOUBLE("Synth Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4,3, 1); -static struct sbmix_elem snd_dt019x_ctl_capture_source = +static struct sbmix_elem snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ + SB_DOUBLE("Master Playback Volume", + SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4, 0, 15), + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("PCM Playback Volume", + SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4, 0, 15), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_DOUBLE("Synth Playback Volume", + SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4, 0, 15), + SB_DOUBLE("CD Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1), + SB_DOUBLE("CD Playback Volume", + SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4, 0, 15), + SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), + SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7), + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7), + SB_DOUBLE("Line Playback Switch", + SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1), + SB_DOUBLE("Line Playback Volume", + SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4, 0, 15), { .name = "Capture Source", .type = SB_MIX_CAPTURE_DT019X - }; - -static struct sbmix_elem *snd_dt019x_controls[] = { - /* ALS4000 below has some parts which we might be lacking, - * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ - &snd_dt019x_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_vol, - &snd_dt019x_ctl_synth_play_vol, - &snd_dt019x_ctl_cd_play_vol, - &snd_dt019x_ctl_mic_play_vol, - &snd_dt019x_ctl_pc_speaker_vol, - &snd_dt019x_ctl_line_play_vol, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_line_play_switch, - &snd_dt019x_ctl_pcm_play_switch, - &snd_dt019x_ctl_synth_play_switch, - &snd_dt019x_ctl_capture_source + } }; static unsigned char snd_dt019x_init_values[][2] = { @@ -735,82 +671,37 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = - SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { +static struct sbmix_elem snd_als4000_controls[] = { + SB_DOUBLE("PCM Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1), + SB_DOUBLE("Synth Playback Switch", + SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1), + SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03), + SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1), + { .name = "Master Mono Capture Route", .type = SB_MIX_MONO_CAPTURE_ALS4K - }; -static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = - SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); -static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = - SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = - SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); -static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + }, + SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1), + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01), + SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01), SB_SINGLE("Digital Loopback Switch", - SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); -/* FIXME: functionality of 3D controls might be swapped, I didn't find - * a description of how to identify what is supposed to be what */ -static struct sbmix_elem snd_als4000_3d_control_switch = - SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01); -static struct sbmix_elem snd_als4000_3d_control_ratio = - SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07); -static struct sbmix_elem snd_als4000_3d_control_freq = + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01), + /* FIXME: functionality of 3D controls might be swapped, I didn't find + * a description of how to identify what is supposed to be what */ + SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07), /* FIXME: maybe there's actually some standard 3D ctrl name for it?? */ - SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03); -static struct sbmix_elem snd_als4000_3d_control_delay = + SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03), /* FIXME: ALS4000a.pdf mentions BBD (Bucket Brigade Device) time delay, * but what ALSA 3D attribute is that actually? "Center", "Depth", * "Wide" or "Space" or even "Level"? Assuming "Wide" for now... */ - SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); -static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = - SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); -static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f), + SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01), SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", - SB_ALS4000_FMDAC, 5, 0x01); + SB_ALS4000_FMDAC, 5, 0x01), #ifdef NOT_AVAILABLE -static struct sbmix_elem snd_als4000_ctl_fmdac = - SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); -static struct sbmix_elem snd_als4000_ctl_qsound = - SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f); -#endif - -static struct sbmix_elem *snd_als4000_controls[] = { - /* ALS4000a.PDF regs page */ - &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ - &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ - &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ - &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ - &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ - &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ - &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ - &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ - &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ - &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ - &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ - &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ - &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ - &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ - &snd_sb16_ctl_play_vol, /* MX41/42 15 */ - &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ - &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ - &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ - &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ - &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ - &snd_als4000_3d_control_switch, /* MX50 17 */ - &snd_als4000_3d_control_ratio, /* MX50 17 */ - &snd_als4000_3d_control_freq, /* MX50 17 */ - &snd_als4000_3d_control_delay, /* MX51 18 */ - &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ - &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ -#ifdef NOT_AVAILABLE - &snd_als4000_ctl_fmdac, - &snd_als4000_ctl_qsound, + SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01), + SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f), #endif }; @@ -829,11 +720,10 @@ static unsigned char snd_als4000_init_values[][2] = { { SB_ALS4000_MIC_IN_GAIN, 0 }, }; - /* */ static int snd_sbmixer_init(struct snd_sb *chip, - struct sbmix_elem **controls, + struct sbmix_elem *controls, int controls_count, unsigned char map[][2], int map_count, @@ -856,7 +746,8 @@ static int snd_sbmixer_init(struct snd_sb *chip, } for (idx = 0; idx < controls_count; idx++) { - if ((err = snd_sbmixer_add_ctl_elem(chip, controls[idx])) < 0) + err = snd_sbmixer_add_ctl_elem(chip, &controls[idx]); + if (err < 0) return err; } snd_component_add(card, name); @@ -908,6 +799,15 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_ALS4000: + /* use only the first 16 controls from SB16 */ + err = snd_sbmixer_init(chip, + snd_sb16_controls, + 16, + snd_sb16_init_values, + ARRAY_SIZE(snd_sb16_init_values), + "ALS4000"); + if (err < 0) + return err; if ((err = snd_sbmixer_init(chip, snd_als4000_controls, ARRAY_SIZE(snd_als4000_controls), -- 1.6.4 ---------------------------------------------------------------------- Wybierz samochod roku i wygraj! Sprawdz >>> http://link.interia.pl/f24e6

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[alsa-devel] [PATCH] ALSA: hda: Fix max PCM level to 0 dB for AD1981_HP
by Dan Chen 14 Dec '09

14 Dec '09

Hi, Takashi, Attached is a patch against sound-2.6 (master) HEAD capping PCM at 0 dB for the AD1981_HP model. Thanks, Dan

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[alsa-devel] [PATCH] ALSA: hda: Use ALC260_WILL quirk for another Acer model (0x1025007f)
by Dan Chen 14 Dec '09

14 Dec '09

Hi, Takashi, Here's a patch against sound-2.6 (master) HEAD to use the appropriate quirk for another Acer model. Thanks, Dan

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[alsa-devel] [PATCH 8/9] sound/isa/gus: Correct code taking the size of a pointer
by Julia Lawall 14 Dec '09

14 Dec '09

From: Julia Lawall <julia(a)diku.dk> sizeof(share_id) is just the size of the pointer. On the other hand, block->share_id is an array, so its size seems more appropriate. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression *x; expression f; type T; @@ *f(...,(T)x,...) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/isa/gus/gus_mem.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 661205c..af888a0 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -127,7 +127,8 @@ static struct snd_gf1_mem_block *snd_gf1_mem_share(struct snd_gf1_mem * alloc, !share_id[2] && !share_id[3]) return NULL; for (block = alloc->first; block; block = block->next) - if (!memcmp(share_id, block->share_id, sizeof(share_id))) + if (!memcmp(share_id, block->share_id, + sizeof(block->share_id))) return block; return NULL; }

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[alsa-devel] [PATCH 2/2 v2] ASoC: Add FSI-DA7210 sound support for SuperH
by Kuninori Morimoto 14 Dec '09

14 Dec '09

Signed-off-by: Kuninori Morimoto <morimoto.kuninori(a)renesas.com> --- v1 -> v2 o remove strange return o remove add_i2c_device function sound/soc/sh/Kconfig | 8 ++++ sound/soc/sh/Makefile | 2 + sound/soc/sh/fsi-da7210.c | 83 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 93 insertions(+), 0 deletions(-) create mode 100644 sound/soc/sh/fsi-da7210.c diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9e69765..8072a6d 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -47,4 +47,12 @@ config SND_FSI_AK4642 This option enables generic sound support for the FSI - AK4642 unit +config SND_FSI_DA7210 + bool "FSI-DA7210 sound support" + depends on SND_SOC_SH4_FSI + select SND_SOC_DA7210 + help + This option enables generic sound support for the + FSI - DA7210 unit + endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile index a699787..1d0ec0a 100644 --- a/sound/soc/sh/Makefile +++ b/sound/soc/sh/Makefile @@ -13,6 +13,8 @@ obj-$(CONFIG_SND_SOC_SH4_FSI) += snd-soc-fsi.o ## boards snd-soc-sh7760-ac97-objs := sh7760-ac97.o snd-soc-fsi-ak4642-objs := fsi-ak4642.o +snd-soc-fsi-da7210-objs := fsi-da7210.o obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o obj-$(CONFIG_SND_FSI_AK4642) += snd-soc-fsi-ak4642.o +obj-$(CONFIG_SND_FSI_DA7210) += snd-soc-fsi-da7210.o diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c new file mode 100644 index 0000000..33b4d17 --- /dev/null +++ b/sound/soc/sh/fsi-da7210.c @@ -0,0 +1,83 @@ +/* + * fsi-da7210.c + * + * Copyright (C) 2009 Renesas Solutions Corp. + * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/io.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <sound/sh_fsi.h> +#include "../codecs/da7210.h" + +static int fsi_da7210_init(struct snd_soc_codec *codec) +{ + return snd_soc_dai_set_fmt(&da7210_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); +} + +static struct snd_soc_dai_link fsi_da7210_dai = { + .name = "DA7210", + .stream_name = "DA7210", + .cpu_dai = &fsi_soc_dai[1], /* FSI B */ + .codec_dai = &da7210_dai, + .init = fsi_da7210_init, +}; + +static struct snd_soc_card fsi_soc_card = { + .name = "FSI", + .platform = &fsi_soc_platform, + .dai_link = &fsi_da7210_dai, + .num_links = 1, +}; + +static struct snd_soc_device fsi_da7210_snd_devdata = { + .card = &fsi_soc_card, + .codec_dev = &soc_codec_dev_da7210, +}; + +static struct platform_device *fsi_da7210_snd_device; + +static int __init fsi_da7210_sound_init(void) +{ + int ret; + + fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); + if (!fsi_da7210_snd_device) + return -ENOMEM; + + platform_set_drvdata(fsi_da7210_snd_device, &fsi_da7210_snd_devdata); + fsi_da7210_snd_devdata.dev = &fsi_da7210_snd_device->dev; + ret = platform_device_add(fsi_da7210_snd_device); + if (ret) + platform_device_put(fsi_da7210_snd_device); + + return ret; +} + +static void __exit fsi_da7210_sound_exit(void) +{ + platform_device_unregister(fsi_da7210_snd_device); +} + +module_init(fsi_da7210_sound_init); +module_exit(fsi_da7210_sound_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC FSI DA2710"); +MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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Re: [alsa-devel] Q: Why I do have to restart alsa everytime after a resume from disk/ram ? Is this alsa- or kernel or config related??
by Robert Hancock 13 Dec '09

13 Dec '09

On 12/11/2009 03:21 AM, Christian Hartmann wrote: > hello list, > > everytime I suspended to disk or to ram, regardless of the used kernel > (git or stable from distri), I must restart alsa to hear sound again. > searched google and tried a lot but still the same awful behaviour. > Has anybode some suggestions for me? is this alsa related ? or is this > a kernel issue which could be dealt here? I'm not sure what you mean by restarting ALSA, do you mean reloading the audio device module or something? You didn't mention which kind of sound hardware you were using either..

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[alsa-devel] [PATCH] Overwrite pin config on intel DG45ID board.
by Alexey Fisher 12 Dec '09

12 Dec '09

The pin config provided by BIOS have some problems: 0x0221401f: [Jack] HP Out at Ext Front <-- other association and sequence 0x02a19020: [Jack] Mic at Ext Front <-- other association 0x01113014: [Jack] Speaker at Ext Rear <-- line out (not speaker) 0x01114010: [Jack] Speaker at Ext Rear <-- line out 0x01a19030: [Jack] Mic at Ext Rear <-- other association 0x01111012: [Jack] Speaker at Ext Rear <-- line out 0x01116011: [Jack] Speaker at Ext Rear <-- line out 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x40f000f0: [N/A] Other at Ext N/A 0x01451140: [Jack] SPDIF Out at Ext Rear 0x40f000f0: [N/A] Other at Ext N/A just overwrite it. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/pci/hda/patch_sigmatel.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6b0bc04..395c174 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1538,6 +1538,13 @@ static unsigned int alienware_m17x_pin_configs[13] = { 0x904601b0, }; +static unsigned int intel_dg45id_pin_configs[14] = { + 0x02214230, 0x02A19240, 0x01013214, 0x01014210, + 0x01A19250, 0x01011212, 0x01016211, 0x40f000f0, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x014510A0, + 0x074510B0, 0x40f000f0 +}; + static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, [STAC_DELL_M6_AMIC] = dell_m6_pin_configs, @@ -1545,6 +1552,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { [STAC_DELL_M6_BOTH] = dell_m6_pin_configs, [STAC_DELL_EQ] = dell_m6_pin_configs, [STAC_ALIENWARE_M17X] = alienware_m17x_pin_configs, + [STAC_92HD73XX_INTEL] = intel_dg45id_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { -- 1.6.3.3

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