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

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Add compatible strings for specific devices
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> The driver supports a range of devices but currently doesn't allow those device names to be used for enumeration on DT. Add the currently listed I2C IDs as compatible strings. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 8 +++++++- sound/soc/codecs/tlv320aic3x.c | 2 ++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index f47c3f5..26f65f9 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -3,7 +3,13 @@ Texas Instruments - tlv320aic3x Codec module The tlv320aic3x serial control bus communicates through I2C protocols Required properties: -- compatible - "string" - "ti,tlv320aic3x" + +- compatible - "string" - One of: + "ti,tlv320aic3x" - Generic TLV320AIC3x device + "ti,tlv320aic33" - TLV320AIC33 + "ti,tlv320aic3007" - TLV320AIC3007 + + - reg - <int> - I2C slave address diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b9268..c9bb760 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1582,6 +1582,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) #if defined(CONFIG_OF) static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, + { .compatible = "ti,tlv320aic33" }, + { .compatible = "ti,tlv320aic3007" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: imx: Enable COMPILE_TEST builds
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Since DT based boards don't have any dependency on arch/arm enable them if COMPILE_TEST is enabled. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index aa43854..87e28bc 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -98,7 +98,7 @@ endif # SND_POWERPC_SOC menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC + depends on ARCH_MXC || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the i.MX CPUs. -- 1.8.3.2

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[alsa-devel] [PATCH v2] ASoC: kirkwood-i2s: fix a compilation warning
by Jean-Francois Moine 16 Jul '13

16 Jul '13

In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function The ALSA subsystem should never gives such a rate because: - the rates with the internal clock are limited to 44.1, 48 and 96 kHz as specified by the kirkwood_i2s_dai structure, - the other rates are proposed in the structure kirkwood_i2s_dai_extclk only when the external clock is present. In case of programming error (bad rate for internal clock and no external clock), the function will simply cause a backtrace. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- v2 - add more explanations --- sound/soc/kirkwood/kirkwood-i2s.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..c9daf3b 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -105,14 +105,16 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, uint32_t clks_ctrl; if (rate == 44100 || rate == 48000 || rate == 96000) { - /* use internal dco for supported rates */ + /* use internal dco for the supported rates + * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", __func__, rate); kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { - /* use optional external clk for other rates */ + } else { + /* use the external clock for the other rates + * defined in kirkwood_i2s_dai_extclk */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate); clk_set_rate(priv->extclk, 256 * rate);

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[alsa-devel] [RFC 1/2] ASoC: dmaengine-pcm: Add support for querying DMA capabilities
by Lars-Peter Clausen 16 Jul '13

16 Jul '13

Currently each platform making use the the generic dmaengine PCM driver still needs to provide a custom snd_pcm_hardware struct which specifies the capabilities of the DMA controller, e.g. the maximum period size that can be supported. This patch adds code which uses the newly introduced dma_get_slave_caps() API to query this information from the dmaengine driver. The new code path will only be taken if the 'pcm_hardware' field of the snd_dmaengine_pcm_config struct is NULL. The patch also introduces a new 'fifo_size' field to the snd_dmaengine_dai_dma_data struct which is used to initialize the snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- For review only for now. The dma_get_slave_caps() API has not been merged yet, the patch adding support for it can be found here: http://permalink.gmane.org/gmane.linux.kernel/1524880 --- include/sound/dmaengine_pcm.h | 2 ++ sound/soc/soc-generic-dmaengine-pcm.c | 46 +++++++++++++++++++++++++++++++++-- 2 files changed, 46 insertions(+), 2 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35c..83b2c3e 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. + * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -68,6 +69,7 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + unsigned int fifo_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3c..98044a7 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -92,6 +92,43 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } +static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data; + struct snd_pcm_hardware hw; + struct dma_slave_caps dma_caps; + int ret; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret) + return ret; + + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = dma_caps.max_sg_nr; + hw.period_bytes_min = 16; + hw.period_bytes_max = dma_caps.max_sg_len; + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = dma_data->fifo_size; + + if (hw.periods_max == 0) + hw.periods_max = UINT_MAX; + if (hw.period_bytes_max == 0) + hw.period_bytes_max = SIZE_MAX; + + if (dma_caps.cmd_pause) + hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + + return snd_soc_set_runtime_hwparams(substream, &hw); +} + static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -99,8 +136,13 @@ static int dmaengine_pcm_open(struct snd_pcm_substream *substream) struct dma_chan *chan = pcm->chan[substream->stream]; int ret; - ret = snd_soc_set_runtime_hwparams(substream, - pcm->config->pcm_hardware); + if (pcm->config->pcm_hardware) { + ret = snd_soc_set_runtime_hwparams(substream, + pcm->config->pcm_hardware); + } else { + ret = dmaengine_pcm_set_runtime_hwparams(substream); + } + if (ret) return ret; -- 1.8.0

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[alsa-devel] [PATCH] ALSA: hda - Headphone mic support for an Asus/Conexant device
by David Henningsson 16 Jul '13

16 Jul '13

This Conexant codec has a single jack that can be used as either headphone or mic (but not headset). The existing hp_mic functionality does not apply here, because the mic and the HP are on separate pins. Hence make a lighter version of what has been earlier done for Realtek codecs. BugLink: https://bugs.launchpad.net/bugs/1198030 Tested-by: Franz Hsieh <franz.hsieh(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 79 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 78 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index de00ce1..4edd2d0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -66,6 +66,8 @@ struct conexant_spec { hda_nid_t eapds[4]; bool dynamic_eapd; + unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + #ifdef ENABLE_CXT_STATIC_QUIRKS const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec) snd_hda_gen_init(codec); if (!spec->dynamic_eapd) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + return 0; } @@ -3224,6 +3229,8 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_HEADPHONE_MIC_PIN, + CXT_FIXUP_HEADPHONE_MIC, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec, (0 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_update_headset_mode(struct hda_codec *codec) +{ + /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */ + int i; + bool mic_mode = false; + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == mux_pin) { + mic_mode = !!cfg->inputs[i].is_headphone_mic; + break; + } + + if (mic_mode) { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */ + spec->gen.hp_jack_present = false; + } else { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */ + spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]); + } + + snd_hda_gen_update_outputs(codec); +} + +static void cxt_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + cxt_update_headset_mode(codec); +} + +static void cxt_fixup_headphone_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; + spec->gen.automute_hook = cxt_update_headset_mode; + break; + case HDA_FIXUP_ACT_INIT: + cxt_update_headset_mode(codec); + break; + } +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_HEADPHONE_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADPHONE_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { } + } + }, + [CXT_FIXUP_HEADPHONE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headphone_mic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), @@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, + spec->parse_flags); if (err < 0) goto error; @@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->bus->allow_bus_reset = 1; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: kirkwood-i2s: Remove empty remove()
by Mark Brown 16 Jul '13

16 Jul '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/kirkwood/kirkwood-i2s.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 44412ea..becc082 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -398,11 +398,6 @@ static int kirkwood_i2s_probe(struct snd_soc_dai *dai) } -static int kirkwood_i2s_remove(struct snd_soc_dai *dai) -{ - return 0; -} - static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .startup = kirkwood_i2s_startup, .trigger = kirkwood_i2s_trigger, @@ -413,7 +408,6 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { static struct snd_soc_dai_driver kirkwood_i2s_dai = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, @@ -431,7 +425,6 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .probe = kirkwood_i2s_probe, - .remove = kirkwood_i2s_remove, .playback = { .channels_min = 1, .channels_max = 2, -- 1.8.3.2

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[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID to snd-hda
by Aaron Plattner 16 Jul '13

16 Jul '13

Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip. Reviewed-by: Andy Ritger <aritger(a)nvidia.com> Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..486def7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2536,6 +2536,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2588,6 +2589,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- 1.7.12

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Re: [alsa-devel] Alsa USB Audio regression
by Raymond Yau 16 Jul '13

16 Jul '13

>>> >>> After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't work at >>> all, Ubuntu 11.10 and newer have the bug), the headset only plays sound >>> at maximum volume level if the volume is turned down just a little no >>> sound is being played. >>> >>> I have filled a Ubuntu bug for 12.04 LTS at: >>> https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . >>> I was advised to contact this mailing list. >>> >>> AlsaInfo for Ubuntu 12.04 LTS with the bug: >>> http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 . >>> AlsaInfo for Ubuntu 10.04 LTS without the bug: >>> http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc >> >> >> >> It is strange that dB range of your usb audio is 0 to 0.14 (only small gain and no attenuation) >> >> In 10.04 LTS pulseaudio provide software volume control >> >> >> ( 1.095| 0.000) I: alsa-sink.c: Driver does not support hardware volume control, falling back to software volume control. >> >> >> control.6 { >> comment.access 'read write' >> comment.type INTEGER >> comment.count 1 comment.range '0 - 37' >> comment.dbmin 0 comment.dbmax 14 >> iface MIXER >> name 'Speaker Playback Volume' value 30 } >> >> >> In 12.04 pulseaudio did not provide software volume attenuation and use the speaker playback volume control > > > So does that mean that the bug is in pulseaudio, or could it be even deeper like in the kernel or something? Are there any significant change in volume when you use "alsamixer -c0" to change speaker playback volume from min 0dB to 0.14dB ? Did pulseaudio provide software attuentation -1.61dB when the range of hardware volume control is only 0.14dB ? * index: 0 name: <alsa_output.usb-0d8c_USB_Audio-00-Audio.analog-surround-71> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE priority: 9049 volume: 0: 94% 1: 94% 2: 94% 3: 94% 4: 94% 5: 94% 6: 94% 7: 94% 0: -1,61 dB 1: -1,61 dB 2: -1,61 dB 3: -1,61 dB 4: -1,61 dB 5: -1,61 dB 6: -1,61 dB 7: -1,61 dB balance 0,00 base volume: 99% -0,14 dB volume steps: 65537 muted: no It is strange that 5.1 headset only support 8 channels playback

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[alsa-devel] [PATCH] ASoC: kirkwood-i2s: fix a compilation warning
by Jean-Francois Moine 16 Jul '13

16 Jul '13

In the function kirkwood_set_rate, when the rate cannot be satisfied by the internal nor by an external clock, the clock source in undefined: warning: ‘clks_ctrl’ may be used uninitialized in this function As the ALSA subsystem should never gives such a rate, this patch removes the check of the external clock pointer. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 4c9dad3..8da5cdb 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -111,7 +111,7 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, kirkwood_set_dco(priv->io, rate); clks_ctrl = KIRKWOOD_MCLK_SOURCE_DCO; - } else if (!IS_ERR(priv->extclk)) { + } else { /* use optional external clk for other rates */ dev_dbg(dai->dev, "%s: extclk set rate = %lu -> %lu\n", __func__, rate, 256 * rate);

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[alsa-devel] [PATCH] ALSA: hda - Fix error accessing non-existent node for Realtek codecs
by David Henningsson 15 Jul '13

15 Jul '13

A recent commit caused my hda-emu based test tool to complain about accessing nodes that don't exist. This simple test works around this. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) I don't have ALC5505 hardware here, no alsa-info to look at, and I don't know what it is. Therefore it would be good with some confirmation that this patch does not break such hardware, which could be the case if the Realtek codec breaks the HDA spec by having "hidden" nodes. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..ca73f10 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3844,7 +3844,8 @@ static int patch_alc269(struct hda_codec *codec) break; } - if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { + if (codec->num_nodes > 0x51 && + snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { spec->has_alc5505_dsp = true; spec->init_hook = alc5505_dsp_init; } -- 1.7.9.5

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