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[alsa-devel] [PATCH 1/5] ASoC: TWL4030: Enable/disable voice digital filters
by Mark Brown 18 May '09

18 May '09

From: Misael Lopez Cruz <x0052729(a)ti.com> Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 2 +- 2 files changed, 38 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index eaf91ab..e4d683d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1629,6 +1629,28 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1665,6 +1687,17 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, return 0; } +static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 0); +} + static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -1673,6 +1706,9 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u8 old_mode, mode; + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 1); + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~(TWL4030_CODECPDZ); @@ -1780,6 +1816,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, + .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 3441115..9668bdf 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -110,7 +110,7 @@ #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 #define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x04 +#define TWL4030_SEL_16K 0x08 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 #define TWL4030_OPTION_1 (1 << 0) -- 1.6.3.1

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[alsa-devel] [PATCH 1/1] Improved MacBook 3,1 support
by Torben Schulz 18 May '09

18 May '09

This patch adds support for MacBook 3,1 sound by adding a model new "mb31" with the appropriate init verbs, mixers and channel modes to the ALC883 configuration. patch_alc882() and patch_alc883() are modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A) correctly. Note: Diff'ed against the 2009-05-16 snapshot (Soory, I don't have regular net access atm.) Signed-off by: Torben Schulz <public(a)letorbi.de> --- alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-05-16 21:24:45.000000000 +0200 +++ alsa-driver-mod/alsa-kernel/pci/hda/patch_realtek.c 2009-05-17 19:45:48.000000000 +0200 @@ -240,6 +240,7 @@ ALC883_3ST_6ch_INTEL, ALC888_ASUS_M90V, ALC888_ASUS_EEE1601, + ALC889A_MB31, ALC1200_ASUS_P5Q, ALC883_AUTO, ALC883_MODEL_LAST, @@ -7290,7 +7291,7 @@ case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ - case 0x106b3600: /* Macbook 3.1 */ + /* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */ case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; @@ -7491,6 +7492,17 @@ }, }; +static struct hda_input_mux alc889A_mb31_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + /* Front Mic (0x01) unused */ + { "Line", 0x2 }, + /* Line 2 (0x03) unused */ + /* CD (0x04) unsused? */ + }, +}; + /* * 2ch mode */ @@ -7609,6 +7621,49 @@ { 8, alc883_sixstack_ch8_init }, }; +/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ +static struct hda_verb alc889A_mb31_ch2_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphones as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer output on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input*/ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line output off */ + { } /* end */ +}; + +/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ +static struct hda_verb alc889A_mb31_ch4_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphones as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer output on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line output on */ + { } /* end */ +}; + +/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ +static struct hda_verb alc889A_mb31_ch5_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphones as rear */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer output on */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input*/ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line output off */ + { } /* end */ +}; + +/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ +static struct hda_verb alc889A_mb31_ch6_init[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphones as front */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer output off */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line output on */ + { } /* end */ +}; + +static struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { + { 2, alc889A_mb31_ch2_init }, + { 4, alc889A_mb31_ch4_init }, + { 5, alc889A_mb31_ch5_init }, + { 6, alc889A_mb31_ch6_init }, +}; + static struct hda_verb alc883_medion_eapd_verbs[] = { /* eanable EAPD on medion laptop */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -7887,6 +7942,31 @@ { } /* end */ }; +static struct snd_kcontrol_new alc889A_mb31_mixer[] = { + /* Output mixers */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT), + /* Output switches */ + HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), + /* Boost mixers */ + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Boost", 0x1a, 0x00, HDA_INPUT), + /* Input mixers */ + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + { } /* end */ +}; + static struct hda_bind_ctls alc883_bind_cap_vol = { .ops = &snd_hda_bind_vol, .values = { @@ -8559,6 +8639,38 @@ alc_automute_pin(codec); } +static struct hda_verb alc889A_mb31_verbs[] = { + /* Init rear pin (used as headphone output) */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Init line pin (used as output in 4ch and 6ch mode) */ + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */ + /* Init line 2 pin (used as headphone out by default) */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */ + { } /* end */ +}; + +/* Mute speakers according to the headphone jack state */ +static void alc889A_mb31_automute(struct hda_codec *codec) +{ + unsigned int present; + + /* Mute only in 2ch or 4ch mode */ + if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) == 0x00) { + present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + } +} + +static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc889A_mb31_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc883_loopbacks alc880_loopbacks #endif @@ -8599,6 +8711,7 @@ [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", [ALC883_3ST_6ch_INTEL] = "3stack-6ch-intel", [ALC1200_ASUS_P5Q] = "asus-p5q", + [ALC889A_MB31] = "mb31", [ALC883_AUTO] = "auto", }; @@ -9050,6 +9163,20 @@ .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, }, + [ALC889A_MB31] = { + .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, + .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, alc880_gpio1_init_verbs }, + .adc_nids = alc883_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .dac_nids = alc883_dac_nids, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .channel_mode = alc889A_mb31_6ch_modes, + .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes), + .input_mux = &alc889A_mb31_capture_source, + .dig_out_nid = ALC883_DIGOUT_NID, + .unsol_event = alc889A_mb31_unsol_event, + .init_hook = alc889A_mb31_automute, + }, }; @@ -9195,11 +9322,19 @@ board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST, alc883_models, alc883_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: Unknown model for ALC883, " - "trying auto-probe from BIOS...\n"); - board_config = ALC883_AUTO; - } + if (board_config < 0 || board_config >= ALC883_MODEL_LAST) { + /* Pick up systems that don't supply PCI SSID */ + switch (codec->subsystem_id) { + case 0x106b3600: /* Macbook 3.1 */ + board_config = ALC889A_MB31; + break; + default: + printk(KERN_INFO "hda_codec: Unknown model for ALC883, " + "trying auto-probe from BIOS...\n"); + board_config = ALC883_AUTO; + } + } + if (board_config == ALC883_AUTO) { /* automatic parse from the BIOS config */

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[alsa-devel] Problems with maintaining a separate alsa-driver in a distribution
by Ozan Çağlayan 18 May '09

18 May '09

Hi, I'm maintaining the ALSA stack as a separate alsa-driver using the supplied tarballs from www.alsa-project.org unlike some popular distros which uses the in-kernel sound/ stack. This has the advantage of quickly acting against latest quirk or driver support changes and pushing a driver update instead of whole the kernel update. But there is a big disadvantage that I'm not able to figure out how to solve: When we enable the SND Kconfig option in kernel, some other sound support for some DVB devices(e.g. SAA7134 sound support, etc.) are enabled as well. So if you're using the in-kernel ALSA, you can enable and use them happily but in my case, those additional sound supports can't be enabled in kernel. Is there a way to hack around this when a separate alsa-driver is used in a distro? Is it technically possible to patch those tiny drivers onto alsa-driver and compile them if the .config file from the kernel source has enabled those DVB devices? Or maybe I should enable ALSA in-kernel, compile those DVB sound stuff and remove the sound/ directory from the kernel package? (Even the idea sounds really *dirty*, dunno if it's technically possible too) Thanks, Ozan Caglayan http://www.pardus.org.tr/eng

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[alsa-devel] [PATCH 0/1] Improved MacBook 3,1 support
by Torben Schulz 18 May '09

18 May '09

Hi guys, this patch improves the support for the MacBook 3,1, which had been handled by the "mbp3" model before. The "mbp3" model is quite inappropriate for the sound-chip of the MacBook 3,1 (RealTek ALC889A). Especially the lack of subwoofer support is one of the biggest disadvantages of this model (See bug 0004086). In contrast to the other models this model is part of the ALC883 configuration, which reflects the ALC889A sound-chip far better than the ALC882 configuration, which all other Apple models are part of. I myself own only a MacBook 3,1, but as far as I know it should also work for the MacBook 4,1 - perhaps even for other models. It would be good if someone could check this out... The new MacBook 3,1 model is called "mb31" and has the following features: - sound from the third speaker (aka "subwoofer") - 5.1 surround sound - speakers and subwoofer are muted when headphones are plugged in. - 4 channel modes - 2ch: speakers:front, subwoofer:CLFE, headphones:front, line:input - 4ch: speakers:front, subwoofer:CLFE, headphones:front, line:CLFE - 5ch: speakers:front, subwoofer:CLFE, headphones:rear, line:input - 6ch: speakers:front, subwoofer:off, headphones:rear, line:CLFE However, there are also two "issues": Speaker will be unmuted, when channel-mode is changed while headphones are plugged in. But this can easily be solved, by plugging the headphones out and in again or using the speaker switches in the mixer. I haven't managed to get "real" surround sound. It seems that the front channel is mapped to all other channels before they get to the mixers (front-left=front-left, front-right=front-right, rear-left=front-left, rear-right=front-right, LFE=front-right, center=front-right). The volume of each channel can be set individually through the mixer. However, this could also be an high configuration issue on my side (please tell me ^^)... Bai Torben

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[alsa-devel] [PATCH 2/3] ASoC: TWL4030: Add control for selecting codec operation mode
by Lopez Cruz, Misael 18 May '09

18 May '09

Add a control for selecting the codec operation mode. TWL4030 codec has two modes: - Option 1. Audio only (4 audio DACs) - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC) Control is restricted when a stream is ongoing, since codec's operation mode cannot be changed on-the-fly. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 47 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e4d683d..671f777 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -814,6 +814,48 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +/* Codec operation modes */ +static const char *twl4030_op_modes_texts[] = { + "Option 2 (voice/audio)", "Option 1 (audio)" +}; + +static const struct soc_enum twl4030_op_modes_enum = + SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, + ARRAY_SIZE(twl4030_op_modes_texts), + twl4030_op_modes_texts); + +int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct twl4030_priv *twl4030 = codec->private_data; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned short val; + unsigned short mask, bitmask; + + for (bitmask = 1; bitmask < e->max; bitmask <<= 1) + ; + if (ucontrol->value.enumerated.item[0] > e->max - 1) + return -EINVAL; + + if (twl4030->configured) { + printk(KERN_ERR "twl4030 operation mode cannot be " + "changed on-the-fly\n"); + return -EBUSY; + } + + val = ucontrol->value.enumerated.item[0] << e->shift_l; + mask = (bitmask - 1) << e->shift_l; + if (e->shift_l != e->shift_r) { + if (ucontrol->value.enumerated.item[1] > e->max - 1) + return -EINVAL; + val |= ucontrol->value.enumerated.item[1] << e->shift_r; + mask |= (bitmask - 1) << e->shift_r; + } + + return snd_soc_update_bits(codec, e->reg, mask, val); +} + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -895,6 +937,11 @@ static const struct soc_enum twl4030_vibradir_enum = twl4030_vibradir_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + /* Codec operation mode control */ + SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, + snd_soc_get_enum_double, + snd_soc_put_twl4030_opmode_enum_double), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, -- 1.5.4.3

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[alsa-devel] [PATCH 1/3] ASoC: TWL4030: Enable/disable voice digital filters
by Lopez Cruz, Misael 18 May '09

18 May '09

Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink and downlink paths, respectively. This patch also corrects voice 8/16kHz mode selection bit (SEL_16K) of CODEC_MODE register. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 2 +- 2 files changed, 38 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index eaf91ab..e4d683d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1629,6 +1629,28 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R + * (VTXL, VTXR) for uplink has to be enabled/disabled. */ +static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction, + int enable) +{ + u8 reg, mask; + + reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION); + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + mask = TWL4030_ARXL1_VRX_EN; + else + mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN; + + if (enable) + reg |= mask; + else + reg &= ~mask; + + twl4030_write(codec, TWL4030_REG_OPTION, reg); +} + static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1665,6 +1687,17 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, return 0; } +static void twl4030_voice_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 0); +} + static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -1673,6 +1706,9 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; u8 old_mode, mode; + /* Enable voice digital filters */ + twl4030_voice_enable(codec, substream->stream, 1); + /* bit rate */ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) & ~(TWL4030_CODECPDZ); @@ -1780,6 +1816,7 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { static struct snd_soc_dai_ops twl4030_dai_voice_ops = { .startup = twl4030_voice_startup, + .shutdown = twl4030_voice_shutdown, .hw_params = twl4030_voice_hw_params, .set_sysclk = twl4030_voice_set_dai_sysclk, .set_fmt = twl4030_voice_set_dai_fmt, diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 3441115..9668bdf 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -110,7 +110,7 @@ #define TWL4030_APLL_RATE_44100 0x90 #define TWL4030_APLL_RATE_48000 0xA0 #define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x04 +#define TWL4030_SEL_16K 0x08 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 #define TWL4030_OPTION_1 (1 << 0) -- 1.5.4.3

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[alsa-devel] [PATCH 0/1] ASoC: TWL4030: Headset ramp down fix
by Peter Ujfalusi 18 May '09

18 May '09

Hello, The following patch should fix (or at least makes it less intrusive) the 'tuck' phenomena on observed on the twl4030 Headset outputs when the codec is going to standby mode. I have debugged this with the scope and if we are not waiting RAMP_DELAY time, the VMID is cut before it reaching 0, which generates the 'tuck'. Note that I have not removed the control for the RAMP_DELAY configuration, the code will use that to calculate the needed delay (also the sysfreq is needed, which also stored). However I do think that one should not touch the RAMP_DELAY... It would be nice to hear from you guys, if it solves/softens the 'tuck'. --- Peter Ujfalusi (1): ASoC: TWL4030: Wait RAMP_DELAY on headsetl_event: POST_PMD sound/soc/codecs/twl4030.c | 20 ++++++++++++++++++++ 1 files changed, 20 insertions(+), 0 deletions(-)

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Re: [alsa-devel] AC97 reset fail after suspend
by Marek Vasut 18 May '09

18 May '09

On Monday 18 of May 2009 00:14:01 Dmitry Eremin-Solenikov wrote: > Tomas 'Sleep_Walker' Cech wrote: > > Hi, > > > > I've got following error on Palm LifeDrive and PalmTX after suspend & > > resume: > > > > pxa2xx_ac97_try_warm_reset: warm reset timeout (GSR=0x0) > > pxa2xx_ac97_read: read error (ac97_reg=0 GSR=0x0) > > pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44) > > pxa2xx_ac97_read: read error (ac97_reg=0 GSR=0x44) WM9712 AC97 reset > > failed > > could not reset AC97 codec > > Could you please check if changing reset_gpio to 113 inside > palmXX_ac97_pdata declarations in palmld and palmtx board files helps. No it doesn't, nRESET is on GPIO95, it's verified by hardware probing. See previous mails in this thread for the reason of these problems.

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[alsa-devel] [PATCH 3/3] sound: use dev_set_drvdata
by Julia Lawall 17 May '09

17 May '09

From: Julia Lawall <julia(a)diku.dk> Eliminate direct accesses to the driver_data field. cf 82ab13b26f15f49be45f15ccc96bfa0b81dfd015 The semantic patch that makes this change is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @@ struct device *dev; expression E; type T; @@ - dev->driver_data = (T)E + dev_set_drvdata(dev, E) @@ struct device *dev; type T; @@ - (T)dev->driver_data + dev_get_drvdata(dev) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -u -p a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -674,7 +674,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(s ssi_private->dev = ssi_info->dev; ssi_private->asynchronous = ssi_info->asynchronous; - ssi_private->dev->driver_data = fsl_ssi_dai; + dev_set_drvdata(ssi_private->dev, fsl_ssi_dai); /* Initialize the the device_attribute structure */ dev_attr->attr.name = "ssi-stats";

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Re: [alsa-devel] [RFC] ALSA vs. dedicated char device for a USB Audio Class gadget driver
by Laurent Pinchart 17 May '09

17 May '09

Hi Hal, first of all, thanks for your answer. On Thursday 14 May 2009 20:18:07 Hal Murray wrote: > > I need an API to transfer audio data from userspace to kernelspace. I > > initially thought about ALSA, but it turns out some assumptions made > > by ALSA are not fulfilled by my system. One of the most serious > > problems is that the UAC gadget driver doesn't have any audio clock. > > The only hardware clock available is the USB device controller > > interrupts generated at the USB transfer rate, and those are much > > faster than the audio sample rate. This will cause buffer underruns > > that I need to handle. > > You don't need a clock. The data will come to you at the right rate. All > you need to do is pass it on when you have enough to fill up a buffer. The > buffer size is fixed. It's part of the spec for the device you are > emulating. I'm not emulating any device. The buffer size is up to me, and I actually have a fixed number of small buffers, but that shouldn't make a difference. > Assume that you get samples one at a time each time the source clock > ticked. The normal state of your system would be to have a buffer that is > partially filled. When a new sample fills up the buffer, you would move it > from the input side to the ready-for-USB queue and setup a new buffer for > future input samples. > > Soon the USB side will read the queued buffer, you free that buffer and you > are now back to the normal state of collecting input data. > > You are in trouble if that doesn't happen before the nest buffer is ready, > that is the ready-for-USB queue should normally be empty. If you can't > keep it empty it will eventually overflow. Short chunks of time where the > queue builds up might be OK. For debugging, you should probably count > them. > > Now consider the case where you get several samples at a time rather than > one. Logically, copy them over one at a time. That turns into a burst of > source clocks, but the average will work out right. Your ready-for-USB > queue should still be empty most of the time. > > If that doesn't make sense, I'll try again. Consider something like audio > over a network. There is no audio clock on the network. The receiver can > derive the source clock by watching the data stream. This is more or less what I'm trying to do, with the difference that I don't move samples from the ALSA ring buffer when they arrive but when the USB layer asks for more data. My trouble is that I don't get notified when new samples are written to the ALSA ring buffer, and I'm not sure to find out how many samples are present in the buffer. If I could get some kind of "sample written" notification, or better, compute the size of data present in the ring buffer, my problem would be (mostly) solved. From what I understand, such a notification isn't possible when the ALSA ring buffer is mmap'ed, as ALSA itself doesn't get notified. This makes the second approach (reading the number of available samples) better, except that some kind of "free-wheeling" mode makes it impossible under some circumstances. From what Takashi Iwai explained in a mail to Jon Smirl on alsa-devel ("appl_ptr and DMA overrun at end of stream"), "we have also mode when appl_ptr is not updated at all (when stop_threshold == boundary)". This free-wheeling mode seems to be used by dmix. Best regards, Laurent Pinchart

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