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

[alsa-devel] dmix format S24_LE
by Bart De Vos 02 Mar '10

02 Mar '10

Hi, I've added support for the S24_LE format in the dmix plugin. This has been tested 'only' on i.MX31 (arm11) with sound DAC AK4367 over I2S, which supports this format natively. Patch is against alsa-lib-1.0.22. Please let me know your thoughts. diff -urN alsa-lib-1.0.22-orig/src/pcm/pcm_direct.c alsa-lib-1.0.22/src/pcm/pcm_direct.c --- alsa-lib-1.0.22-orig/src/pcm/pcm_direct.c 2009-12-16 16:18:51.000000000 +0100 +++ alsa-lib-1.0.22/src/pcm/pcm_direct.c 2010-03-02 10:01:00.000000000 +0100 @@ -888,6 +888,7 @@ SND_PCM_FORMAT_S32 ^ SND_PCM_FORMAT_S32_LE ^ SND_PCM_FORMAT_S32_BE, SND_PCM_FORMAT_S16, SND_PCM_FORMAT_S16 ^ SND_PCM_FORMAT_S16_LE ^ SND_PCM_FORMAT_S16_BE, + SND_PCM_FORMAT_S24_LE, SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_U8, }; diff -urN alsa-lib-1.0.22-orig/src/pcm/pcm_dmix.c alsa-lib-1.0.22/src/pcm/pcm_dmix.c --- alsa-lib-1.0.22-orig/src/pcm/pcm_dmix.c 2009-12-16 16:18:51.000000000 +0100 +++ alsa-lib-1.0.22/src/pcm/pcm_dmix.c 2010-03-02 10:02:39.000000000 +0100 @@ -174,6 +174,10 @@ sample_size = 4; do_mix_areas = (mix_areas_t *)dmix->u.dmix.mix_areas_32; break; + case SND_PCM_FORMAT_S24_LE: + sample_size = 4; + do_mix_areas = (mix_areas_t *)dmix->u.dmix.mix_areas_24; + break; case SND_PCM_FORMAT_S24_3LE: sample_size = 3; do_mix_areas = (mix_areas_t *)dmix->u.dmix.mix_areas_24; @@ -238,6 +242,10 @@ sample_size = 4; do_remix_areas = (mix_areas_t *)dmix->u.dmix.remix_areas_32; break; + case SND_PCM_FORMAT_S24_LE: + sample_size = 4; + do_remix_areas = (mix_areas_t *)dmix->u.dmix.remix_areas_24; + break; case SND_PCM_FORMAT_S24_3LE: sample_size = 3; do_remix_areas = (mix_areas_t *)dmix->u.dmix.remix_areas_24; diff -urN alsa-lib-1.0.22-orig/src/pcm/pcm_dmix_generic.c alsa-lib-1.0.22/src/pcm/pcm_dmix_generic.c --- alsa-lib-1.0.22-orig/src/pcm/pcm_dmix_generic.c 2009-12-16 16:18:51.000000000 +0100 +++ alsa-lib-1.0.22/src/pcm/pcm_dmix_generic.c 2010-03-02 10:37:16.000000000 +0100 @@ -122,7 +122,7 @@ #define generic_dmix_supported_format \ ((1ULL << SND_PCM_FORMAT_S16_LE) | (1ULL << SND_PCM_FORMAT_S32_LE) |\ (1ULL << SND_PCM_FORMAT_S16_BE) | (1ULL << SND_PCM_FORMAT_S32_BE) |\ - (1ULL << SND_PCM_FORMAT_S24_3LE) | \ + (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S24_3LE)|\ (1ULL << SND_PCM_FORMAT_U8)) #include <byteswap.h> -- -------------------------------------------------- Bart De Vos Development Engineer mailto:bart.devos@eia.be EIA Electronics Vluchtenburgstraat 3 B-2630 Aartselaar (Belgium) phone : +32(0)3 870 85 29 fax : +32(0)3 877 14 72 website : http://www.eia.be/ -------------------------------------------------- Basically, perfect development is impossible. Development can be fast, good and cheap. Pick two! — Larry Wall --------------------------------------------------

2 1

[alsa-devel] [PATCH 1/1] sound/pci/hda/hda_codec.c: various coding style fixes
by Norberto Lopes 02 Mar '10

02 Mar '10

Signed-off-by: Norberto Lopes <nlopes.ml(a)gmail.com> --- sound/pci/hda/hda_codec.c | 69 ++++++++++++++++++++++++-------------------- 1 files changed, 38 insertions(+), 31 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f98b47c..444b989 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -955,8 +955,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * * Returns 0 if successful, or a negative error code. */ -int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp) +int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, + unsigned int codec_addr, + struct hda_codec **codecp) { struct hda_codec *codec; char component[31]; @@ -1162,7 +1163,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); */ /* FIXME: more better hash key? */ -#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_KEY(nid, dir, idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) #define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) @@ -1332,7 +1333,8 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) if (!codec->no_trigger_sense) { pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); } return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); @@ -1348,8 +1350,8 @@ EXPORT_SYMBOL_HDA(snd_hda_pin_sense); */ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return !!(sense & AC_PINSENSE_PRESENCE); + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); } EXPORT_SYMBOL_HDA(snd_hda_jack_detect); @@ -1891,7 +1893,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - + for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; int i = 0; @@ -2378,27 +2380,27 @@ static struct snd_kcontrol_new dig_mixes[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_cmask_get, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, PRO_MASK), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_pmask_get, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_default_get, .put = snd_hda_spdif_default_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), .info = snd_hda_spdif_out_switch_info, .get = snd_hda_spdif_out_switch_get, .put = snd_hda_spdif_out_switch_put, @@ -2549,7 +2551,7 @@ static int snd_hda_spdif_in_status_get(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new dig_in_ctls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, SWITCH), .info = snd_hda_spdif_in_switch_info, .get = snd_hda_spdif_in_switch_get, .put = snd_hda_spdif_in_switch_put, @@ -2557,7 +2559,7 @@ static struct snd_kcontrol_new dig_in_ctls[] = { { .access = SNDRV_CTL_ELEM_ACCESS_READ, .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT), + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), .info = snd_hda_spdif_mask_info, .get = snd_hda_spdif_in_status_get, }, @@ -2822,7 +2824,7 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) int err = snd_hda_codec_build_controls(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build controls" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -2918,8 +2920,12 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, val |= channels - 1; switch (snd_pcm_format_width(format)) { - case 8: val |= 0x00; break; - case 16: val |= 0x10; break; + case 8: + val |= 0x00; + break; + case 16: + val |= 0x10; + break; case 20: case 24: case 32: @@ -3235,7 +3241,8 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) return audio_idx[type][i]; - snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); + snd_printk(KERN_WARNING "Too many %s devices\n", + snd_hda_pcm_type_name[type]); return -EAGAIN; } @@ -3273,7 +3280,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) err = codec->patch_ops.build_pcms(codec); if (err < 0) { printk(KERN_ERR "hda_codec: cannot build PCMs" - "for #%d (error %d)\n", codec->addr, err); + "for #%d (error %d)\n", codec->addr, err); err = snd_hda_codec_reset(codec); if (err < 0) { printk(KERN_ERR @@ -3403,8 +3410,8 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** * snd_hda_check_board_codec_sid_config - compare the current codec - subsystem ID with the - config table + subsystem ID with the + config table This is important for Gateway notebooks with SB450 HDA Audio where the vendor ID of the PCI device is: @@ -3542,7 +3549,7 @@ void snd_hda_update_power_acct(struct hda_codec *codec) * * Increment the power-up counter and power up the hardware really when * not turned on yet. - */ + */ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; @@ -3571,7 +3578,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); * * Decrement the power-up counter and schedules the power-off work if * the counter rearches to zero. - */ + */ void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; @@ -3597,7 +3604,7 @@ EXPORT_SYMBOL_HDA(snd_hda_power_down); * * This function is supposed to be set or called from the check_power_status * patch ops. - */ + */ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) @@ -3765,7 +3772,7 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, { /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) - set_dig_out_convert(codec, nid, + set_dig_out_convert(codec, nid, codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff, -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); @@ -4024,13 +4031,13 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) /* * Sort an associated group of pins according to their sequence numbers. */ -static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, +static void sort_pins_by_sequence(hda_nid_t *pins, short *sequences, int num_pins) { int i, j; short seq; hda_nid_t nid; - + for (i = 0; i < num_pins; i++) { for (j = i + 1; j < num_pins; j++) { if (sequences[i] > sequences[j]) { @@ -4058,7 +4065,7 @@ static void sort_pins_by_sequence(hda_nid_t * pins, short * sequences, * is detected, one of speaker of HP pins is assigned as the primary * output, i.e. to line_out_pins[0]. So, line_outs is always positive * if any analog output exists. - * + * * The analog input pins are assigned to input_pins array. * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, * respectively. @@ -4121,9 +4128,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, case AC_JACK_SPEAKER: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); - if (! assoc) + if (!assoc) continue; - if (! assoc_speaker) + if (!assoc_speaker) assoc_speaker = assoc; else if (assoc_speaker != assoc) continue; @@ -4221,7 +4228,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->speaker_outs); sort_pins_by_sequence(cfg->hp_pins, sequences_hp, cfg->hp_outs); - + /* if we have only one mic, make it AUTO_PIN_MIC */ if (!cfg->input_pins[AUTO_PIN_MIC] && cfg->input_pins[AUTO_PIN_FRONT_MIC]) { @@ -4371,7 +4378,7 @@ EXPORT_SYMBOL_HDA(snd_hda_resume); /** * snd_array_new - get a new element from the given array * @array: the array object - * + * * Get a new element from the given array. If it exceeds the * pre-allocated array size, re-allocate the array. * -- 1.6.4

2 1

Re: [alsa-devel] HD-Audio device unsuppored (Nvidia G2xx-series GPU implemented)
by Stefan Ringel 02 Mar '10

02 Mar '10

I forgot this attachment 1.patch and alsa-info.txt.

4 20

[alsa-devel] [PATCHv4 7/7] ASoC: TWL6030: Detect power-up sequence completion
by Olaya, Margarita 02 Mar '10

02 Mar '10

From: Misael Lopez Cruz <x0052729(a)ti.com> When the codec is powered-up through external AUDPWRON line it starts its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. If NAUDINT irq is provided, CODEC driver starts a wait_for_completion just after AUDPWRON line transitions from low to high. It's signaled as complete when servicing READYINT interrupt. If AUDPWRON gpio line is provided but NAUDINT irq is not, then CODEC driver enables READYINT and polls on INTID register. If none of them are provided, then CODEC uses manual power sequences and disables all audio interrupts. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6030.c | 61 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/twl6030.h | 1 + 2 files changed, 53 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index b8dd5ae..2847f1b 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -52,6 +52,7 @@ struct twl6030_data { int non_lp; unsigned int sysclk; struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct completion ready; }; /* @@ -372,6 +373,7 @@ static int twl6030_power_mode_event(struct snd_soc_dapm_widget *w, static irqreturn_t twl6030_naudint_handler(int irq, void *data) { struct snd_soc_codec *codec = data; + struct twl6030_data *priv = codec->private_data; u8 intid; twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6030_REG_INTID); @@ -391,7 +393,7 @@ static irqreturn_t twl6030_naudint_handler(int irq, void *data) dev_alert(codec->dev, "vib drivers over current detection\n"); break; case TWL6030_READYINT: - dev_alert(codec->dev, "codec is ready\n"); + complete(&priv->ready); break; default: dev_err(codec->dev, "unknown audio interrupt %d\n", intid); @@ -626,11 +628,45 @@ static int twl6030_add_widgets(struct snd_soc_codec *codec) return 0; } +static int twl6030_power_up_completion(struct snd_soc_codec *codec, + int naudint) +{ + struct twl6030_data *priv = codec->private_data; + int time_left; + u8 intid; + + if (naudint) { + /* wait for ready interrupt with 48 ms timeout */ + time_left = wait_for_completion_timeout(&priv->ready, + msecs_to_jiffies(48)); + } else { + /* retry 3 times only */ + for (time_left = 3; time_left > 0; time_left--) { + mdelay(16); + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + TWL6030_REG_INTID); + if (intid & TWL6030_READYINT) + break; + } + } + + if (!time_left) { + dev_err(codec->dev, "timeout waiting for READYINT\n"); + return -ETIMEDOUT; + } + + priv->codec_powered = 1; + + return 0; +} + static int twl6030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct twl6030_data *priv = codec->private_data; int audpwron = priv->audpwron; + int naudint = priv->naudint; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -643,8 +679,10 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, /* use AUDPWRON line */ gpio_set_value(audpwron, 1); - /* power-up sequence latency */ - mdelay(16); + /* wait for power-up completion */ + ret = twl6030_power_up_completion(codec, naudint); + if (ret) + return ret; /* sync registers updated during power-up sequence */ twl6030_read(codec, TWL6030_REG_NCPCTL); @@ -653,12 +691,11 @@ static int twl6030_set_bias_level(struct snd_soc_codec *codec, } else { /* use manual power-up sequence */ twl6030_power_up(codec); + priv->codec_powered = 1; } /* initialize vdd/vss registers with reg_cache */ twl6030_init_vdd_regs(codec); - - priv->codec_powered = 1; break; case SND_SOC_BIAS_OFF: if (!priv->codec_powered) @@ -1067,6 +1104,7 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + init_completion(&priv->ready); if (gpio_is_valid(audpwron)) { ret = gpio_request(audpwron, "audpwron"); @@ -1089,10 +1127,15 @@ static int __devinit twl6030_codec_probe(struct platform_device *pdev) if (ret) goto gpio2_err; } else { - dev_warn(codec->dev, - "no naudint irq, audio interrupts disabled\n"); - twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, - TWL6030_ALLINT_MSK); + if (gpio_is_valid(audpwron)) { + /* enable only codec ready interrupt */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + ~TWL6030_READYMSK & TWL6030_ALLINT_MSK); + } else { + /* no interrupts at all */ + twl6030_write_reg_cache(codec, TWL6030_REG_INTMR, + TWL6030_ALLINT_MSK); + } } /* init vio registers */ diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 90b8a44..d591be1 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -79,6 +79,7 @@ /* INTMR (0x04) fields */ +#define TWL6030_READYMSK 0x40 #define TWL6030_ALLINT_MSK 0x7B /* NCPCTL (0x05) fields */ -- 1.6.1.3

2 8

[alsa-devel] Audio4DJ (snd-usb-caiaq) noise when using DVB usb adap
by Pedro Ribeiro 02 Mar '10

02 Mar '10

Hi all, I'm encountering a problem with the Audio4DJ audio card with a DVB USB adapter (Win-TV NOVA-T USB2 STICK). The problem is strange because it only happens after I disconnect and reconnect the audio card. Procedure 1: - boot from fresh - plug the DVB adapter - plug the Audio4DJ Everything works correctly - meaning I can watch TV and my audio is fine. Procedure 2: - boot from fresh - plug the Audio4DJ - plug the DVB adapter The audio starts cracking every few seconds, and constantly if I use jackd. Procedure 3: - boot from fresh - plug the DVB adapter - plug the Audio4DJ (everything is correct, as above in Procedure 1) - unplug the Audio4DJ - plug it again Cracking audio. I'm insisting in "boot from fresh" because from now on (that is, from the moment that I unplug the Audio4DJ) I always get cracking audio. rmmod'ing the snd-usb-caiaq and the usb-dvb modules produces the same result. I have to boot from fresh and do Procedure 1 for it to work OK. Can anyone please give a hint on how to debug this? Regards, Pedro

1 0

[alsa-devel] Audio4DJ (snd-usb-caiaq) noise when using DVB usb adapter
by Pedro Ribeiro 02 Mar '10

02 Mar '10

Hi all, I'm encountering a problem with the Audio4DJ audio card with a DVB USB adapter (Win-TV NOVA-T USB2 STICK). The problem is strange because it only happens after I disconnect and reconnect the audio card. Procedure 1: - boot from fresh - plug the DVB adapter - plug the Audio4DJ Everything works correctly - meaning I can watch TV and my audio is fine. Procedure 2: - boot from fresh - plug the Audio4DJ - plug the DVB adapter The audio starts cracking every few seconds, and constantly if I use jackd. Procedure 3: - boot from fresh - plug the DVB adapter - plug the Audio4DJ (everything is correct, as above in Procedure 1) - unplug the Audio4DJ - plug it again Cracking audio. I'm insisting in "boot from fresh" because from now on (that is, from the moment that I unplug the Audio4DJ) I always get cracking audio. rmmod'ing the snd-usb-caiaq and the usb-dvb modules produces the same result. I have to boot from fresh and do Procedure 1 for it to work OK. Can anyone please give a hint on how to debug this? (Please CC me directly as I'm not in both lists). Regards, Pedro

1 0

[alsa-devel] [PATCH 2.6.34] ASoC: Fix S3C64xx IIS driver for Samsung header reorg
by Mark Brown 01 Mar '10

01 Mar '10

The reorgs of the Samsung headers have moved the GPIO bank definitions from plat/ to mach/ - the IIS driver needs to be updated to take care of this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Ideally this could also go in via the same path as the ARM/Samsung changes which do the moves. If those get merged without this fix then it may as well go via ASoC. sound/soc/s3c24xx/s3c64xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 93ed3aa..a72c251 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -22,8 +22,8 @@ #include <sound/soc.h> #include <plat/regs-s3c2412-iis.h> -#include <plat/gpio-bank-d.h> -#include <plat/gpio-bank-e.h> +#include <mach/gpio-bank-d.h> +#include <mach/gpio-bank-e.h> #include <plat/gpio-cfg.h> #include <mach/map.h> -- 1.7.0

1 0

[alsa-devel] [Regression] With the new 2.6.33 when I plug headphones, the speaker doen't get off anymore
by giggzounet 01 Mar '10

01 Mar '10

Hi, I have installed debian stable lenny + backports on an eeepc 1201n. With the 2.6.30 or 2.6.32 from lenny-backport I don't have problem with sound : when I plug the headphones, speakers get off. I have tested with 2.6.33 and when I plug headphones, speakers don't get off anymore. so I have sound in headphones and speakers. 22:42 giggz@baal ~ % cat /proc/asound/card0/codec#* | grep Codec Codec: Realtek ALC269 Codec: Nvidia MCP7A HDMI I attach the output of alsa-info with the 2.6.32.9 and the 2.6.33 (I'm in a Uni Campus and it's quite difficult to have internet on a external laptop...so I attach them at this mail and I don't upload them on pastbin). I have opened a bug on the kernel bugzilla (bug 15399). Should I provide anything more ? Thx a lot, Best regards Guillaume

4 14

[alsa-devel] [PATCH 3/3] ALSA: usb-audio: support multiple formats with audio class v2 devices
by Clemens Ladisch 01 Mar '10

01 Mar '10

Change the parser to correctly handle v2 descriptors with multiple format bits set. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- linux/sound/usb/format.c +++ linux/sound/usb/format.c @@ -37,19 +37,20 @@ * @format: the format tag (wFormatTag) * @fmt: the format type descriptor */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, +static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, int format, void *_fmt, int protocol) { - int pcm_format, i; int sample_width, sample_bytes; + u64 pcm_formats; switch (protocol) { case UAC_VERSION_1: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; + format = 1 << format; break; } @@ -57,24 +58,7 @@ static int parse_audio_format_i_type(str struct uac_format_type_i_ext_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubslotSize; - - /* - * FIXME - * USB audio class v2 devices specify a bitmap of possible - * audio formats rather than one fix value. For now, we just - * pick one of them and report that as the only possible - * value for this setting. - * The bit allocation map is in fact compatible to the - * wFormatTag of the v1 AS streaming descriptors, which is why - * we can simply map the matrix. - */ - - for (i = 0; i < 5; i++) - if (format & (1UL << i)) { - format = i + 1; - break; - } - + format <<= 1; break; } @@ -82,15 +66,15 @@ static int parse_audio_format_i_type(str return -EINVAL; } - /* FIXME: correct endianess and sign? */ - pcm_format = -1; + pcm_formats = 0; - switch (format) { - case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ + if (format == 0 || format == (1 << UAC_FORMAT_TYPE_I_UNDEFINED)) { + /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); - /* fall-through */ - case UAC_FORMAT_TYPE_I_PCM: + format = 1 << UAC_FORMAT_TYPE_I_PCM; + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) { if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -99,22 +83,22 @@ static int parse_audio_format_i_type(str /* check the format byte size */ switch (sample_bytes) { case 1: - pcm_format = SNDRV_PCM_FORMAT_S8; + pcm_formats |= SNDRV_PCM_FMTBIT_S8; break; case 2: if (snd_usb_is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S16_BE; /* grrr, big endian!! */ + pcm_formats |= SNDRV_PCM_FMTBIT_S16_BE; /* grrr, big endian!! */ else - pcm_format = SNDRV_PCM_FORMAT_S16_LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S16_LE; break; case 3: if (snd_usb_is_big_endian_format(chip, fp)) - pcm_format = SNDRV_PCM_FORMAT_S24_3BE; /* grrr, big endian!! */ + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3BE; /* grrr, big endian!! */ else - pcm_format = SNDRV_PCM_FORMAT_S24_3LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S24_3LE; break; case 4: - pcm_format = SNDRV_PCM_FORMAT_S32_LE; + pcm_formats |= SNDRV_PCM_FMTBIT_S32_LE; break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", @@ -122,30 +106,29 @@ static int parse_audio_format_i_type(str sample_width, sample_bytes); break; } - break; - case UAC_FORMAT_TYPE_I_PCM8: - pcm_format = SNDRV_PCM_FORMAT_U8; - + } + if (format & (1 << UAC_FORMAT_TYPE_I_PCM8)) { /* Dallas DS4201 workaround: it advertises U8 format, but really supports S8. */ if (chip->usb_id == USB_ID(0x04fa, 0x4201)) - pcm_format = SNDRV_PCM_FORMAT_S8; - break; - case UAC_FORMAT_TYPE_I_IEEE_FLOAT: - pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; - break; - case UAC_FORMAT_TYPE_I_ALAW: - pcm_format = SNDRV_PCM_FORMAT_A_LAW; - break; - case UAC_FORMAT_TYPE_I_MULAW: - pcm_format = SNDRV_PCM_FORMAT_MU_LAW; - break; - default: - snd_printk(KERN_INFO "%d:%u:%d : unsupported format type %d\n", + pcm_formats |= SNDRV_PCM_FMTBIT_S8; + else + pcm_formats |= SNDRV_PCM_FMTBIT_U8; + } + if (format & (1 << UAC_FORMAT_TYPE_I_IEEE_FLOAT)) { + pcm_formats |= SNDRV_PCM_FMTBIT_FLOAT_LE; + } + if (format & (1 << UAC_FORMAT_TYPE_I_ALAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_A_LAW; + } + if (format & (1 << UAC_FORMAT_TYPE_I_MULAW)) { + pcm_formats |= SNDRV_PCM_FMTBIT_MU_LAW; + } + if (format & ~0x3f) { + snd_printk(KERN_INFO "%d:%u:%d : unsupported format bits %#x\n", chip->dev->devnum, fp->iface, fp->altsetting, format); - break; } - return pcm_format; + return pcm_formats; } @@ -313,14 +296,14 @@ static int parse_audio_format_i(struct s default: pcm_format = SNDRV_PCM_FORMAT_S16_LE; } + fp->formats = 1uLL << pcm_format; } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); - if (pcm_format < 0) + fp->formats = parse_audio_format_i_type(chip, fp, format, + fmt, protocol); + if (!fp->formats) return -1; } - fp->formats = 1uLL << pcm_format; - /* gather possible sample rates */ /* audio class v1 reports possible sample rates as part of the * proprietary class specific descriptor.

2 2

[PATCH] ALSA: usb-audio: fix quirk formats
by Daniel Mack 01 Mar '10

01 Mar '10

'struct audioformat' takes a bit field of supported rates now rather than one fixed value. Adopt that change at two more places. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/quirks-table.h | 2 +- sound/usb/quirks.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index ff2df4e..2107d00 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2233,7 +2233,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { - .format = SNDRV_PCM_FORMAT_S24_3BE, + .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 1, .altsetting = 1, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 00e8a97..0119fa0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -89,7 +89,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, const struct snd_usb_audio_quirk *quirk) { static const struct audioformat ua1000_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_S32_LE, .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, -- 1.6.6.2

1 0