December 2011 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: HDA: Realtek: Take vmaster dac from multiout dac list
by David Henningsson 09 Dec '11

09 Dec '11

With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++---------------------- 1 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..2bf08fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3039,6 +3039,8 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) @@ -3153,6 +3155,9 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + if (cfg->line_out_pins[0]) + spec->vmaster_nid = alc_look_for_out_vol_nid(codec, spec->multiout.dac_nids[0], + cfg->line_out_pins[0]); return 0; } @@ -4175,8 +4180,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4203,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4309,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4332,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4703,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4726,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4904,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5015,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5569,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5673,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5797,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6179,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6234,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- 1.7.5.4 --------------070204090502030907040600--

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[PATCH] ALSA: HDA: Realtek: Take vmaster dac from multiout dac list
by David Henningsson 09 Dec '11

09 Dec '11

With the auto-parser we can choose the dac nid for vmaster from the DACs we already know, instead of hard-coding it. This is more future-proof and was actually wrong on one machine. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 36 ++++++++++++++---------------------- 1 files changed, 14 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a74c1e..f394865 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3039,6 +3039,8 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ static int alc_auto_fill_dac_nids(struct hda_codec *codec) @@ -3153,6 +3155,9 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + i = cfg->line_outs ? cfg->line_out_pins[0] : + (cfg->speaker_outs ? cfg->speaker_pins[0] : cfg->hp_pins[0]); + spec->vmaster_nid = alc_look_for_out_vol_nid(codec, spec->multiout.dac_nids[0], i); return 0; } @@ -4175,8 +4180,10 @@ static int patch_alc880(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { + spec->vmaster_nid = 0x0c; setup_preset(codec, &alc880_presets[board_config]); + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4196,8 +4203,6 @@ static int patch_alc880(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4304,8 +4309,10 @@ static int patch_alc260(struct hda_codec *codec) #endif } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc260_presets[board_config]); + spec->vmaster_nid = 0x08; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4325,8 +4332,6 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x08; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4698,8 +4703,10 @@ static int patch_alc882(struct hda_codec *codec) goto error; } - if (board_config != ALC_MODEL_AUTO) + if (board_config != ALC_MODEL_AUTO) { setup_preset(codec, &alc882_presets[board_config]); + spec->vmaster_nid = 0x0c; + } if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4719,8 +4726,6 @@ static int patch_alc882(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4899,8 +4904,6 @@ static int patch_alc262(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x0c; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5012,8 +5015,6 @@ static int patch_alc268(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5568,8 +5569,6 @@ static int patch_alc269(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -5674,8 +5673,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - spec->vmaster_nid = 0x03; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -5800,8 +5797,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - spec->vmaster_nid = 0x02; - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; @@ -6184,7 +6179,6 @@ static int patch_alc662(struct hda_codec *codec) break; } } - spec->vmaster_nid = 0x02; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6240,8 +6234,6 @@ static int patch_alc680(struct hda_codec *codec) if (!spec->no_analog && !spec->cap_mixer) set_capture_mixer(codec); - spec->vmaster_nid = 0x02; - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -- 1.7.5.4 --------------000303040509040305040806--

1 0

Re: [alsa-devel] [PATCH v2] soc/lapis: add machine driver
by Mark Brown 09 Dec '11

09 Dec '11

On Fri, Dec 02, 2011 at 06:45:15PM +0900, Tomoya MORINAGA wrote: > +#define CODEC_DEV_ADDR 0x1A > +static struct i2c_board_info ioh_hwmon_info[] = { > + {I2C_BOARD_INFO("ioh_i2c-0", CODEC_DEV_ADDR + 1)}, > + {I2C_BOARD_INFO("ioh_i2c-1", CODEC_DEV_ADDR + 2)}, > + {I2C_BOARD_INFO("ioh_i2c-2", CODEC_DEV_ADDR + 3)}, > + {I2C_BOARD_INFO("ioh_i2c-3", CODEC_DEV_ADDR + 4)}, > + {I2C_BOARD_INFO("ioh_i2c-4", CODEC_DEV_ADDR + 5)}, > + {I2C_BOARD_INFO("ioh_i2c-5", CODEC_DEV_ADDR + 0)}, This looks completely wrong. I'd not expect to see any I2C_BOARD_INFO usage at all in a machine driver (that should be done by whatever enumerates the system as a whole) and I'm not clear what the intended effect of the above is - the names look like the names of I2C controllers... > +static int ml7213_ioh_init(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_codec *codec = rtd->codec; > + struct snd_soc_dapm_context *dapm = &codec->dapm; > + > + snd_soc_dapm_new_controls(dapm, ml7213_dapm_widgets, > + ARRAY_SIZE(ml7213_dapm_widgets)); > + > + snd_soc_dapm_add_routes(dapm, ml7213_routes, ARRAY_SIZE(ml7213_routes)); Use table based init for these. > +static int ml7213_startup(struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_codec *codec = rtd->codec; > + struct i2c_client *i2c; > + struct i2c_adapter *adapter; > + > + mutex_lock(&codec->mutex); > + > + adapter = i2c_get_adapter(1); > + if (!adapter) > + return -ENODEV; > + > + i2c = i2c_new_device(adapter, &ioh_hwmon_info[substream->number]); > + if (!i2c) { > + mutex_unlock(&codec->mutex); > + return -1; > + } > + i2c_put_adapter(adapter); > + mutex_unlock(&codec->mutex); No, this is definitely wrong - we certainly shouldn't be registering new devices whenever someone starts an audio stream. I'm struggling to understand what the intended effect is though. > + switch (params_rate(params)) { > + case 16000: > + case 32000: > + case 48000: > + clk = 12288800; > + break; You may as well just set the CODEC clock rate once on init() and let the CODEC driver worry about what it can do with the clock it's got, if the values aren't going to change there's no need to set them every time. > + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | > + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); > + if (ret < 0) > + return ret; Use the dai_fmt field in the dai_link to set this. > +MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm(a)gmail.com>"); > +MODULE_DESCRIPTION("LAPIS Semiconductor ML7213 IOH ALSA SoC machine driver"); > +MODULE_LICENSE("GPL"); Should have MODULE_ALIAS too.

3 13

[alsa-devel] [PATCH] ASoC: wm8960: Use snd_soc_update_bits for read-modify-write
by Axel Lin 08 Dec '11

08 Dec '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8960.c | 67 ++++++++++++++------------------------------- 1 files changed, 21 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 6a9c41d..2315b86 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -543,30 +543,24 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, static int wm8960_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8960_DACCTL1) & 0xfff7; if (mute) - snd_soc_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + snd_soc_update_bits(codec, WM8960_DACCTL1, 0x8, 0x8); else - snd_soc_write(codec, WM8960_DACCTL1, mute_reg); + snd_soc_update_bits(codec, WM8960_DACCTL1, 0x8, 0); return 0; } static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* Set VMID to 2x50k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg &= ~0x180; - reg |= 0x80; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_STANDBY: @@ -579,23 +573,19 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, WM8960_BUFDCOPEN | WM8960_BUFIOEN); /* Enable & ramp VMID at 2x50k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg |= 0x80; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x80, 0x80); msleep(100); /* Enable VREF */ - snd_soc_write(codec, WM8960_POWER1, reg | WM8960_VREF); + snd_soc_update_bits(codec, WM8960_POWER1, WM8960_VREF, + WM8960_VREF); /* Disable anti-pop features */ snd_soc_write(codec, WM8960_APOP1, WM8960_BUFIOEN); } /* Set VMID to 2x250k */ - reg = snd_soc_read(codec, WM8960_POWER1); - reg &= ~0x180; - reg |= 0x100; - snd_soc_write(codec, WM8960_POWER1, reg); + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x100); break; case SND_SOC_BIAS_OFF: @@ -787,10 +777,8 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Disable the PLL: even if we are changing the frequency the * PLL needs to be disabled while we do so. */ - snd_soc_write(codec, WM8960_CLOCK1, - snd_soc_read(codec, WM8960_CLOCK1) & ~1); - snd_soc_write(codec, WM8960_POWER2, - snd_soc_read(codec, WM8960_POWER2) & ~1); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, 0); + snd_soc_update_bits(codec, WM8960_POWER2, 0x1, 0); if (!freq_in || !freq_out) return 0; @@ -809,11 +797,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, snd_soc_write(codec, WM8960_PLL1, reg); /* Turn it on */ - snd_soc_write(codec, WM8960_POWER2, - snd_soc_read(codec, WM8960_POWER2) | 1); + snd_soc_update_bits(codec, WM8960_POWER2, 0x1, 0x1); msleep(250); - snd_soc_write(codec, WM8960_CLOCK1, - snd_soc_read(codec, WM8960_CLOCK1) | 1); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, 0x1); return 0; } @@ -913,7 +899,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); struct wm8960_data *pdata = dev_get_platdata(codec->dev); int ret; - u16 reg; wm8960->set_bias_level = wm8960_set_bias_level_out3; @@ -944,26 +929,16 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch the update bits */ - reg = snd_soc_read(codec, WM8960_LINVOL); - snd_soc_write(codec, WM8960_LINVOL, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RINVOL); - snd_soc_write(codec, WM8960_RINVOL, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LADC); - snd_soc_write(codec, WM8960_LADC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RADC); - snd_soc_write(codec, WM8960_RADC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LDAC); - snd_soc_write(codec, WM8960_LDAC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_RDAC); - snd_soc_write(codec, WM8960_RDAC, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LOUT1); - snd_soc_write(codec, WM8960_LOUT1, reg | 0x100); - reg = snd_soc_read(codec, WM8960_ROUT1); - snd_soc_write(codec, WM8960_ROUT1, reg | 0x100); - reg = snd_soc_read(codec, WM8960_LOUT2); - snd_soc_write(codec, WM8960_LOUT2, reg | 0x100); - reg = snd_soc_read(codec, WM8960_ROUT2); - snd_soc_write(codec, WM8960_ROUT2, reg | 0x100); + snd_soc_update_bits(codec, WM8960_LINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RINVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LADC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RADC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LDAC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_RDAC, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LOUT1, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_ROUT1, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); + snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); snd_soc_add_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2 0/5] ASoC: WM8903: DT bindings & related
by Stephen Warren 08 Dec '11

08 Dec '11

v2: * Fixed default pdata logic so that 0 in pdata fields really means "do nothing". * Added a patch to fix pdata gpio_cfg value confusion. * Moved all the pdata default & DT parsing logic to i2c_probe from codec probe. * Split parsing logic out into separate functions so probe is easier to read. * Rebased on top of the WM8903 cleanup patches Mark posted today. Stephen Warren (5): ASoC: WM8903: Fix platform data gpio_cfg confusion ASoC: WM8903: Create default platform data structure ASoC: WM8903: Remove conditionals checking pdata != NULL ASoC: WM8903: Get default irq_active_low from IRQ controller ASoC: WM8903: Add device tree binding Documentation/devicetree/bindings/sound/wm8903.txt | 50 ++++++ arch/arm/mach-tegra/board-harmony.c | 8 +- arch/arm/mach-tegra/board-seaboard.c | 8 +- include/sound/wm8903.h | 7 +- sound/soc/codecs/wm8903.c | 181 +++++++++++++++----- 5 files changed, 205 insertions(+), 49 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt

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Re: [alsa-devel] [PATCH 5/6] compress: add the core file
by Nallasellan, Singaravelan 07 Dec '11

07 Dec '11

> > Even the thread which actually writes the data to ring buffer may not > > have the idea of frames So the start threshold does not require to be > > on frame boundaries. It is just to avoid START when the buffer is > > written with the data. If the write gives more data than the ring > > buffer, write will not unblock and the user space application may not > > be able to call START. > > This will be an issue if you go for non-blocking implementation. > > Otherwise, START should be called in another thread than the thread > > which writes the data. > > I did not understand a single line of this comment. > Why is this start threshold needed again and how would its value be defined for > compressed data? I will try to explain it with more details. Consider the case of blocking "write" call: If the user space app writes the buffer with the content of size greater than the ring buffer size, write call will block. How do you expect the user space to invoke START ioctl? I assume the write call will block until it writes the complete content into the buffer. To avoid this behavior, the app has to query only the empty space and write only that much data. Then it should call START.

1 0

[alsa-devel] [PATCH 1/3] ASoC: Fix error handling in e800_init to free gpios
by Axel Lin 07 Dec '11

07 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/e800_wm9712.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 6a8f38b..26e0232 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -136,8 +136,10 @@ static int __init e800_init(void) goto free_spk_amp_gpio; e800_snd_device = platform_device_alloc("soc-audio", -1); - if (!e800_snd_device) - return -ENOMEM; + if (!e800_snd_device) { + ret = -ENOMEM; + goto free_spk_amp_gpio; + } platform_set_drvdata(e800_snd_device, &e800); ret = platform_device_add(e800_snd_device); -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: Fix a typo in s3c24xx_simtec_tlv320aic23 driver
by Axel Lin 07 Dec '11

07 Dec '11

Fix a typo introduced by commit e00c3f55 "ASoC: Convert Samsung directory to module_platform_driver". This fixes the build error: CC sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_init': sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function) sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: (Each undeclared identifier is reported only once sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: for each function it appears in.) sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c: In function 'simtec_audio_tlv320aic32_driver_exit': sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c:105: error: 'simtec_audio_tlv320aic32_driver' undeclared (first use in this function) make[3]: *** [sound/soc/samsung/s3c24xx_simtec_tlv320aic23.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 I think we had better naming it with *driver, thus I change it to simtec_audio_tlv320aic23_driver. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c index 7324609..89b57b5 100644 --- a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -92,7 +92,7 @@ static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) return simtec_audio_core_probe(pd, &snd_soc_machine_simtec_aic23); } -static struct platform_driver simtec_audio_tlv320aic23_platdrv = { +static struct platform_driver simtec_audio_tlv320aic23_driver = { .driver = { .owner = THIS_MODULE, .name = "s3c24xx-simtec-tlv320aic23", @@ -102,7 +102,7 @@ static struct platform_driver simtec_audio_tlv320aic23_platdrv = { .remove = __devexit_p(simtec_audio_remove), }; -module_platform_driver(simtec_audio_tlv320aic32_driver); +module_platform_driver(simtec_audio_tlv320aic23_driver); MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Include linux/module.h for smdk2443_wm9710
by Axel Lin 07 Dec '11

07 Dec '11

Include linux/module.h to fix below build error: CC sound/soc/samsung/smdk2443_wm9710.o sound/soc/samsung/smdk2443_wm9710.c:64: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk2443_wm9710.c:64: warning: data definition has no type or storage class sound/soc/samsung/smdk2443_wm9710.c:64: warning: type defaults to 'int' in declaration of 'MODULE_AUTHOR' sound/soc/samsung/smdk2443_wm9710.c:64: warning: function declaration isn't a prototype sound/soc/samsung/smdk2443_wm9710.c:65: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk2443_wm9710.c:65: warning: data definition has no type or storage class sound/soc/samsung/smdk2443_wm9710.c:65: warning: type defaults to 'int' in declaration of 'MODULE_DESCRIPTION' sound/soc/samsung/smdk2443_wm9710.c:65: warning: function declaration isn't a prototype sound/soc/samsung/smdk2443_wm9710.c:66: error: expected declaration specifiers or '...' before string constant sound/soc/samsung/smdk2443_wm9710.c:66: warning: data definition has no type or storage class sound/soc/samsung/smdk2443_wm9710.c:66: warning: type defaults to 'int' in declaration of 'MODULE_LICENSE' sound/soc/samsung/smdk2443_wm9710.c:66: warning: function declaration isn't a prototype make[3]: *** [sound/soc/samsung/smdk2443_wm9710.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/smdk2443_wm9710.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c index 3a0dbfc7..8bd1dc5 100644 --- a/sound/soc/samsung/smdk2443_wm9710.c +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -12,6 +12,7 @@ * */ +#include <linux/module.h> #include <sound/soc.h> static struct snd_soc_card smdk2443; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix a typo in jive_wm8750
by Axel Lin 07 Dec '11

07 Dec '11

Fix a typo in jive_wm8750 that introduces below build error. Also removes an unused err variable. CC sound/soc/samsung/jive_wm8750.o sound/soc/samsung/jive_wm8750.c: In function 'jive_wm8750_init': sound/soc/samsung/jive_wm8750.c:104: warning: unused variable 'err' sound/soc/samsung/jive_wm8750.c: At top level: sound/soc/samsung/jive_wm8750.c:134: error: unknown field 'dapm_widgtets' specified in initializer sound/soc/samsung/jive_wm8750.c:134: warning: initialization from incompatible pointer type make[3]: *** [sound/soc/samsung/jive_wm8750.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/samsung/jive_wm8750.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 1826acf..8e523fd 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -101,7 +101,6 @@ static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* These endpoints are not being used. */ snd_soc_dapm_nc_pin(dapm, "LINPUT2"); @@ -131,7 +130,7 @@ static struct snd_soc_card snd_soc_machine_jive = { .dai_link = &jive_dai, .num_links = 1, - .dapm_widgtets = wm8750_dapm_widgets, + .dapm_widgets = wm8750_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), -- 1.7.5.4

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