March 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3] ALSA: ASoC: cs4271: convert to direct regmap API usage
by Daniel Mack 15 Mar '13

15 Mar '13

By using the regmap API directly, we can make use of the .write_flag_mask for SPI, which allows us to drop the strange register hacks that were necessary so far. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v3: dropped one more (now misleading) comment v2: added a .write_flag_mask sound/soc/codecs/cs4271.c | 159 ++++++++++++++++++++++++++-------------------- 1 file changed, 91 insertions(+), 68 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 2415a41..ac0d3b4 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -39,17 +39,15 @@ /* * CS4271 registers - * High byte represents SPI chip address (0x10) + write command (0) - * Low byte - codec register address */ -#define CS4271_MODE1 0x2001 /* Mode Control 1 */ -#define CS4271_DACCTL 0x2002 /* DAC Control */ -#define CS4271_DACVOL 0x2003 /* DAC Volume & Mixing Control */ -#define CS4271_VOLA 0x2004 /* DAC Channel A Volume Control */ -#define CS4271_VOLB 0x2005 /* DAC Channel B Volume Control */ -#define CS4271_ADCCTL 0x2006 /* ADC Control */ -#define CS4271_MODE2 0x2007 /* Mode Control 2 */ -#define CS4271_CHIPID 0x2008 /* Chip ID */ +#define CS4271_MODE1 0x01 /* Mode Control 1 */ +#define CS4271_DACCTL 0x02 /* DAC Control */ +#define CS4271_DACVOL 0x03 /* DAC Volume & Mixing Control */ +#define CS4271_VOLA 0x04 /* DAC Channel A Volume Control */ +#define CS4271_VOLB 0x05 /* DAC Channel B Volume Control */ +#define CS4271_ADCCTL 0x06 /* ADC Control */ +#define CS4271_MODE2 0x07 /* Mode Control 2 */ +#define CS4271_CHIPID 0x08 /* Chip ID */ #define CS4271_FIRSTREG CS4271_MODE1 #define CS4271_LASTREG CS4271_MODE2 @@ -144,23 +142,27 @@ * Array do not include Chip ID, as codec driver does not use * registers read operations at all */ -static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = { - 0, - 0, - CS4271_DACCTL_AMUTE, - CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, - 0, - 0, - 0, - 0, +static const struct reg_default cs4271_reg_defaults[] = { + { CS4271_MODE1, 0, }, + { CS4271_DACCTL, CS4271_DACCTL_AMUTE, }, + { CS4271_DACVOL, CS4271_DACVOL_SOFT | CS4271_DACVOL_ATAPI_AL_BR, }, + { CS4271_VOLA, 0, }, + { CS4271_VOLB, 0, }, + { CS4271_ADCCTL, 0, }, + { CS4271_MODE2, 0, }, }; +static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == CS4271_CHIPID; +} + struct cs4271_private { /* SND_SOC_I2C or SND_SOC_SPI */ - enum snd_soc_control_type bus_type; unsigned int mclk; bool master; bool deemph; + struct regmap *regmap; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -210,14 +212,14 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_LEFT_J: val |= CS4271_MODE1_DAC_DIF_LJ; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_LJ); if (ret < 0) return ret; break; case SND_SOC_DAIFMT_I2S: val |= CS4271_MODE1_DAC_DIF_I2S; - ret = snd_soc_update_bits(codec, CS4271_ADCCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_ADCCTL, CS4271_ADCCTL_ADC_DIF_MASK, CS4271_ADCCTL_ADC_DIF_I2S); if (ret < 0) return ret; @@ -227,7 +229,7 @@ static int cs4271_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_DAC_DIF_MASK | CS4271_MODE1_MASTER, val); if (ret < 0) return ret; @@ -252,7 +254,7 @@ static int cs4271_set_deemph(struct snd_soc_codec *codec) val <<= 4; } - ret = snd_soc_update_bits(codec, CS4271_DACCTL, + ret = regmap_update_bits(cs4271->regmap, CS4271_DACCTL, CS4271_DACCTL_DEM_MASK, val); if (ret < 0) return ret; @@ -341,14 +343,14 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, !dai->capture_active) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !dai->playback_active)) { - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, + CS4271_MODE2_PDN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; } @@ -378,7 +380,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, val |= cs4271_clk_tab[i].ratio_mask; - ret = snd_soc_update_bits(codec, CS4271_MODE1, + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val); if (ret < 0) return ret; @@ -389,6 +391,7 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); int ret; int val_a = 0; int val_b = 0; @@ -398,10 +401,13 @@ static int cs4271_digital_mute(struct snd_soc_dai *dai, int mute) val_b = CS4271_VOLB_MUTE; } - ret = snd_soc_update_bits(codec, CS4271_VOLA, CS4271_VOLA_MUTE, val_a); + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLA, + CS4271_VOLA_MUTE, val_a); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_VOLB, CS4271_VOLB_MUTE, val_b); + + ret = regmap_update_bits(cs4271->regmap, CS4271_VOLB, + CS4271_VOLB_MUTE, val_b); if (ret < 0) return ret; @@ -463,25 +469,33 @@ static struct snd_soc_dai_driver cs4271_dai = { static int cs4271_soc_suspend(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Set power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, - CS4271_MODE2_PDN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, CS4271_MODE2_PDN); if (ret < 0) return ret; + return 0; } static int cs4271_soc_resume(struct snd_soc_codec *codec) { int ret; + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + /* Restore codec state */ - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(cs4271->regmap); if (ret < 0) return ret; + /* then disable the power-down bit */ - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; + return 0; } #else @@ -542,40 +556,22 @@ static int cs4271_probe(struct snd_soc_codec *codec) cs4271->gpio_nreset = gpio_nreset; - /* - * In case of I2C, chip address specified in board data. - * So cache IO operations use 8 bit codec register address. - * In case of SPI, chip address and register address - * passed together as 16 bit value. - * Anyway, register address is masked with 0xFF inside - * soc-cache code. - */ - if (cs4271->bus_type == SND_SOC_SPI) - ret = snd_soc_codec_set_cache_io(codec, 16, 8, - cs4271->bus_type); - else - ret = snd_soc_codec_set_cache_io(codec, 8, 8, - cs4271->bus_type); - if (ret) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ret = snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN, - CS4271_MODE2_PDN | CS4271_MODE2_CPEN); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN, + CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret < 0) return ret; - ret = snd_soc_update_bits(codec, CS4271_MODE2, CS4271_MODE2_PDN, 0); + ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_PDN, 0); if (ret < 0) return ret; /* Power-up sequence requires 85 uS */ udelay(85); if (amutec_eq_bmutec) - snd_soc_update_bits(codec, CS4271_MODE2, - CS4271_MODE2_MUTECAEQUB, - CS4271_MODE2_MUTECAEQUB); + regmap_update_bits(cs4271->regmap, CS4271_MODE2, + CS4271_MODE2_MUTECAEQUB, + CS4271_MODE2_MUTECAEQUB); return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); @@ -597,13 +593,24 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, - .reg_cache_default = cs4271_dflt_reg, - .reg_cache_size = ARRAY_SIZE(cs4271_dflt_reg), - .reg_word_size = sizeof(cs4271_dflt_reg[0]), - .compress_type = SND_SOC_FLAT_COMPRESSION, }; #if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; @@ -613,7 +620,9 @@ static int cs4271_spi_probe(struct spi_device *spi) return -ENOMEM; spi_set_drvdata(spi, cs4271); - cs4271->bus_type = SND_SOC_SPI; + cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); @@ -643,6 +652,18 @@ static const struct i2c_device_id cs4271_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4271_i2c_id); +static const struct regmap_config cs4271_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = CS4271_LASTREG, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { @@ -653,7 +674,9 @@ static int cs4271_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, cs4271); - cs4271->bus_type = SND_SOC_I2C; + cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); + if (IS_ERR(cs4271->regmap)) + return PTR_ERR(cs4271->regmap); return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271, &cs4271_dai, 1); -- 1.8.1.4

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Re: [alsa-devel] [PATCH 08/10] ASoC: dmaengine_pcm: Add open function for DT DMA request
by Mark Brown 15 Mar '13

15 Mar '13

On Sun, Mar 10, 2013 at 07:33:09PM +0100, Markus Pargmann wrote: > Add a function to open a DMA PCM substream using devicetree data > provided via the client device node. The patch introduces a public > function and a private subfunction that is called by both open > functions. Someone (I think it was Shawn) sent a very similar patch just recently which appears to have fallen out of my inbox unfortunately - can you please check what's going on there and coordinate with them? Let me know if you can't find the patch and I'll dig it out.

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Re: [alsa-devel] [Alsa-user] Master volume control
by Paul D. DeRocco 14 Mar '13

14 Mar '13

> From: Clemens Ladisch [mailto:cladisch@googlemail.com] > > #include <stdio.h> > #include <stdlib.h> > #include <alsa/asoundlib.h> > > static void check(int err, const char *f) > { > if (err < 0) { > fprintf(stderr, "%s failed: %s\n", f, > snd_strerror(err)); > exit(EXIT_FAILURE); > } > } > #define CHECK(f) check(f, #f) > > int main() > { > snd_ctl_t *ctl; > snd_ctl_elem_value_t *value; > > CHECK(snd_ctl_open(&ctl, "default", 0)); > > snd_ctl_elem_value_alloca(&value); > snd_ctl_elem_value_set_interface(value, > SND_CTL_ELEM_IFACE_MIXER); > snd_ctl_elem_value_set_name(value, "Master Playback Volume"); > snd_ctl_elem_value_set_integer(value, 0, 42); > CHECK(snd_ctl_elem_write(ctl, value)); > > snd_ctl_close(ctl); > return 0; > } Well, I finally got a chance to try this. It "works", but the snd_ctl_elem_write call takes a couple of milliseconds (that's around a million instructions on my Atom), and it apparently spends this time in kernel state, because it causes my sound thread to underrun, so it's not usable. If I use alsamixer, I don't have this problem; I can manipulate the master volume smoothly while I'm playing my synth app. So I went through the source, and found that it's using snd_mixer_selem_set_playback_volume. This is apparently part of something called the "Simple Mixer Interface", yet another section of ALSA which is completely undocumented, other than the Doxygen stuff (which is only meaningful as a reference for someone who already knows it inside out, but can't remember e.g. whether playback_pcm comes before or after capture_pcm in the snd_mixer_selem_regopt structure). So what's the minimum I need to do in order to get at the Master Playback Volume, via the Simple Mixer Interface? -- Ciao, Paul D. DeRocco Paul mailto:pderocco@ix.netcom.com

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[alsa-devel] [PATCH v2] ALSA: hda - Disable IDT eapd_switch if there are no internal speakers
by David Henningsson 14 Mar '13

14 Mar '13

If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) I'm okay with your suggestion. I only changed from "false" to "0" to be consistent with other usage of eapd_switch. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d57c81e..3566731 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -815,6 +815,29 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) return 0; } +/* check whether a built-in speaker is included in parsed pins */ +static bool has_builtin_speaker(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t *nid_pin; + int nids, i; + + if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) { + nid_pin = spec->gen.autocfg.line_out_pins; + nids = spec->gen.autocfg.line_outs; + } else { + nid_pin = spec->gen.autocfg.speaker_pins; + nids = spec->gen.autocfg.speaker_outs; + } + + for (i = 0; i < nids; i++) { + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]); + if (snd_hda_get_input_pin_attr(def_conf) == INPUT_PIN_ATTR_INT) + return true; + } + return false; +} + /* * PC beep controls */ @@ -3891,6 +3914,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; } + /* Don't GPIO-mute speakers if there are no internal speakers, because + * the GPIO might be necessary for Headphone + */ + if (spec->eapd_switch && !has_builtin_speaker(codec)) + spec->eapd_switch = 0; + codec->proc_widget_hook = stac92hd7x_proc_hook; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Disable IDT eapd_switch if there are no internal speakers
by David Henningsson 14 Mar '13

14 Mar '13

If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) Hi Takashi, I encountered the bug when working with a pre-release machine (so no alsa-info, unfortunately). Feel free to commit if you think this is a good idea. Or can you think of a use case where this would cause a regression? diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d57c81e..66200ea8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3524,6 +3524,31 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + /* Don't GPIO-mute speakers if there are no internal speakers, because + the GPIO might be necessary for Headphone */ + if (spec->eapd_switch) { + hda_nid_t *nid_pin; + int nids, i; + bool found = false; + if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) { + nid_pin = spec->gen.autocfg.line_out_pins; + nids = spec->gen.autocfg.line_outs; + } else { + nid_pin = spec->gen.autocfg.speaker_pins; + nids = spec->gen.autocfg.speaker_outs; + } + for (i = 0; i < nids; i++) { + unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid_pin[i]); + unsigned int attr = snd_hda_get_input_pin_attr(def_conf); + if (attr == INPUT_PIN_ATTR_INT) { + found = true; + break; + } + } + if (!found) + spec->eapd_switch = 0; + } + /* minimum value is actually mute */ spec->gen.vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ALSA: info: Avoid leaking kernel memory
by Takashi Iwai 14 Mar '13

14 Mar '13

Make sure that the allocated buffer for reading the proc file won't expose the uncleared kernel memory. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/info.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index db308db..58e97b3 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -89,7 +89,7 @@ static int resize_info_buffer(struct snd_info_buffer *buffer, char *nbuf; nsize = PAGE_ALIGN(nsize); - nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL); + nbuf = krealloc(buffer->buffer, nsize, GFP_KERNEL | __GFP_ZERO); if (! nbuf) return -ENOMEM; @@ -353,7 +353,7 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) goto __nomem; data->rbuffer = buffer; buffer->len = PAGE_SIZE; - buffer->buffer = kmalloc(buffer->len, GFP_KERNEL); + buffer->buffer = kzalloc(buffer->len, GFP_KERNEL); if (buffer->buffer == NULL) goto __nomem; } -- 1.8.1.4

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[alsa-devel] [PATCH] ASoC: arizona: Ensure we clock two channels for I2S mode
by Mark Brown 13 Mar '13

13 Mar '13

I2S requires stereo clocking even for mono data. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c70d503..2b0803e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -818,7 +818,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct arizona *arizona = priv->arizona; int base = dai->driver->base; const int *rates; - int i, ret; + int i, ret, val; int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; @@ -834,6 +834,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, bclk_target *= chan_limit; } + /* Force stereo for I2S mode */ + val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); + if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + arizona_aif_dbg(dai, "Forcing stereo mode\n"); + bclk_target *= 2; + } + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { if (rates[i] >= bclk_target && rates[i] % params_rate(params) == 0) { -- 1.7.10.4

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[alsa-devel] [PATCH 1/3] ALSA: hda - Drop explicit memset() by reallocation with __GFP_ZERO
by Takashi Iwai 13 Mar '13

13 Mar '13

Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..a1c5053 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5539,10 +5539,9 @@ void *snd_array_new(struct snd_array *array) void *nlist; if (snd_BUG_ON(num >= 4096)) return NULL; - nlist = krealloc(array->list, size, GFP_KERNEL); + nlist = krealloc(array->list, size, GFP_KERNEL | __GFP_ZERO); if (!nlist) return NULL; - memset(nlist + oldsize, 0, size - oldsize); array->list = nlist; array->alloced = num; } -- 1.8.1.4

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[alsa-devel] How to send non-SysEx system messages via ALSA?
by Joerg-Cyril.Hoehle＠t-systems.com 13 Mar '13

13 Mar '13

Hi, seqmid.h contains function that declare the major event types like snd_seq_ev_set_noteon and snd_seq_ev_set_sysex. However, seq_event.h knows many more types, e.g. SND_SEQ_EVENT_START and SND_SEQ_EVENT_CLOCK/TICK. Now, consider an app playing a MIDI file or a debug tool that allows to send arbitrary messages. How should such an app send e.g. the 2-byte MTC Quarter Frame sequence via ALSA? A. Use a 2 byte buffer F1 xx and snd_seq_ev_set_sysex(&event, 2, buf); B. Use SND_SEQ_EVENT_TICK and figure out some copy&paste code from seqmidi.h, perhaps snd_seq_ev_set_queue_control(ev,SND_SEQ_EVENT_TICK,SND_SEQ_QUEUE_DIRECT,xx) But why does that alter my destination port via snd_seq_ev_set_dest(ev, SND_SEQ_CLIENT_SYSTEM, SND_SEQ_PORT_SYSTEM_TIMER)? C. It's SND_SEQ_EVENT_CLOCK actually, TICK being the one byte F8 real-time message? Background: Wine on Linux needs to generate such messages on behalf of applications and users with real MIDI HW. So far Wine assumes that it can encapsulate all system messages in a SysEx, IOW Wine does A. Is that usage of the set_sysex API correct? More generally, can Wine transparently stuff system messages into a byte buffer and send them off via snd_seq_ev_set_sysex? Thank you, Jörg Höhle

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[alsa-devel] [patch] ALSA: ASoC: tas5086: signedness bug in tas5086_hw_params()
by Dan Carpenter 13 Mar '13

13 Mar '13

"val" has to be signed for the error handling to work. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 008bea4..41d03ae 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -251,7 +251,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - unsigned int val; + int val; int ret; priv->rate = params_rate(params);

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