August 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - More pop noise fixes for Dell XPS 13 9333
by Gabriele Mazzotta 08 Aug '14

08 Aug '14

On init, mic-in is always set as input source, indipendently on what is plugged in. Since setting/unsetting mic-in as input source causes a pop noise, make sure the internal microphone is selected as input source on boot. On shutdown, make sure the codec is not suspended as that would cause a pop noise. Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 3 files changed, 15 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4c20277..92d8292 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5348,6 +5348,8 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus) if (!bus) return; list_for_each_entry(codec, &bus->codec_list, list) { + if (codec->resume_at_reboot) + hda_call_codec_resume(codec); if (hda_codec_is_power_on(codec) && codec->patch_ops.reboot_notify) codec->patch_ops.reboot_notify(codec); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5825aa1..5c3c66e 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -366,6 +366,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ + unsigned int resume_at_reboot:1; /* resume codec at reboot */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b60824e..defaa2a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4018,8 +4018,20 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, { if (action == HDA_FIXUP_ACT_PROBE) { struct alc_spec *spec = codec->spec; + struct hda_input_mux *imux = &spec->gen.input_mux; + int i; + spec->shutup = alc_no_shutup; codec->power_filter = alc_power_filter_xps13; + codec->resume_at_reboot = 1; + + /* Make the internal mic the default input source. */ + for (i = 0; i < imux->num_items; i++) { + if (spec->gen.imux_pins[i] == 0x12) { + spec->gen.cur_mux[0] = i; + break; + } + } } } -- 2.1.0.rc1

2 5

[alsa-devel] [PATCH] ASoC: pcm512x: Correct Digital Playback control names
by Mark Brown 08 Aug '14

08 Aug '14

From: Mark Brown <broonie(a)linaro.org> The source type should come before the direction specifier according to ControlNames.txt. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 163ec3855fd4..0c8aefab404c 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds = pcm512x_ramp_step_text); static const struct snd_kcontrol_new pcm512x_controls[] = { -SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, +SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, +SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), -- 2.0.1

1 0

[alsa-devel] [PATCH] ASoC: wm8962: Correct the bit offset to enable mono speaker output
by Nicolin Chen 08 Aug '14

08 Aug '14

As WM8962 datasheet describes for SPK_MONO bit of R51: When SPK_MONO is set to '1', both speakers output the signal from the left channel. So for mono speaker widget, we shall enable Left Channel whose enable bit is 6 instead of 7 (Right Channel). This patches just simply corrects the bit offset. Signed-off-by: Peter Chan <B18700(a)freescale.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- @Peter I cced you just to let you know that I'm merging your fix to upstream. And because you patch can't be applied to the upstream tree directly, I used my own commit with your signed-off. Thank you for the fix. sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1098ae3..105deec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2185,7 +2185,7 @@ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8962_MIXER_ENABLES, 1, 0, spkmixl, ARRAY_SIZE(spkmixl)), SND_SOC_DAPM_MUX_E("Speaker PGA", WM8962_PWR_MGMT_2, 4, 0, &spkoutl_mux, out_pga_event, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA("Speaker Output", WM8962_CLASS_D_CONTROL_1, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Speaker Output", WM8962_CLASS_D_CONTROL_1, 6, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUT"), }; -- 1.8.4

3 2

[alsa-devel] [PATCH V3 0/2] refine esai for tdm support
by Shengjiu Wang 08 Aug '14

08 Aug '14

These patchs is to refine esai for tdm support. Changes for V3 - update the comments for patch 2 Changes for V2 - update the comments according the reviewer's suggestion - add init value for slots and change pin to pins. Shengjiu Wang (2): ASoC: fsl_esai: refine esai for TDM support Revert "ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support." sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/fsl_esai.c | 16 +++++++++++----- sound/soc/fsl/fsl_esai.h | 8 ++++---- 3 files changed, 15 insertions(+), 10 deletions(-) -- 1.7.9.5

1 2

Re: [alsa-devel] ASUS Phoebus (CMI8888) HDA Support
by Takashi Iwai 07 Aug '14

07 Aug '14

At Mon, 4 Aug 2014 12:44:00 -0700, Vincent Lejeune wrote: > > Here it is. OK, blow is an untested fix patch, based on the latest sound git tree. For applying to the older kernels, you may need some manual adjustment. Takashi --- diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5db1948699d8..aa302fb03fc5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -265,6 +265,7 @@ enum { AZX_DRIVER_TERA, AZX_DRIVER_CTX, AZX_DRIVER_CTHDA, + AZX_DRIVER_CMEDIA, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -330,6 +331,7 @@ static char *driver_short_names[] = { [AZX_DRIVER_TERA] = "HDA Teradici", [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", + [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -1373,6 +1375,7 @@ static void azx_check_snoop_available(struct azx *chip) snoop = false; break; case AZX_DRIVER_CTHDA: + case AZX_DRIVER_CMEDIA: snoop = false; break; } @@ -2154,6 +2157,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_CTX | AZX_DCAPS_CTX_WORKAROUND | AZX_DCAPS_RIRB_PRE_DELAY | AZX_DCAPS_POSFIX_LPIB }, #endif + /* CM8888 */ + { PCI_DEVICE(0x13f6, 0x5011), + .driver_data = AZX_DRIVER_CMEDIA | + AZX_DCAPS_NO_MSI | AZX_DCAPS_POSFIX_LPIB }, /* Vortex86MX */ { PCI_DEVICE(0x17f3, 0x3010), .driver_data = AZX_DRIVER_GENERIC }, /* VMware HDAudio */ diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index ed3d133ffbb6..3ed5d5613bc1 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -75,15 +75,62 @@ static int patch_cmi9880(struct hda_codec *codec) return err; } +static int patch_cmi8888(struct hda_codec *codec) +{ + struct cmi_spec *spec; + struct auto_pin_cfg *cfg; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + + codec->spec = spec; + cfg = &spec->gen.autocfg; + snd_hda_gen_spec_init(&spec->gen); + + /* mask NID 0x10 from the playback volume selection; + * it's a headphone boost volume handled manually below + */ + spec->gen.out_vol_mask = (1ULL << 0x10); + + err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); + if (err < 0) + goto error; + err = snd_hda_gen_parse_auto_config(codec, cfg); + if (err < 0) + goto error; + + if (get_defcfg_device(snd_hda_codec_get_pincfg(codec, 0x10)) == + AC_JACK_HP_OUT) { + static const struct snd_kcontrol_new amp_kctl = + HDA_CODEC_VOLUME("Headphone Amp Playback Volume", + 0x10, 0, HDA_OUTPUT); + if (!snd_hda_gen_add_kctl(&spec->gen, NULL, &amp_kctl)) { + err = -ENOMEM; + goto error; + } + } + + codec->patch_ops = cmi_auto_patch_ops; + return 0; + + error: + snd_hda_gen_free(codec); + return err; +} + /* * patch entries */ static const struct hda_codec_preset snd_hda_preset_cmedia[] = { + { .id = 0x13f68888, .name = "CMI8888", .patch = patch_cmi8888 }, { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 }, { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 }, {} /* terminator */ }; +MODULE_ALIAS("snd-hda-codec-id:13f68888"); MODULE_ALIAS("snd-hda-codec-id:13f69880"); MODULE_ALIAS("snd-hda-codec-id:434d4980");

2 8

[alsa-devel] [PATCH v2] ASoC: rsnd: enable SRC setting via DT
by Kuninori Morimoto 07 Aug '14

07 Aug '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Some special board wants to exchange sampling rate for special codec (or for very simple DA/AD converter) which can accept only 1 sampling rate. This patch enables its settings. 48kHz ---> 48kHz 44.1kHz ---> [SRC] ---> [codec] 96kHz ---> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - add detail git log .../devicetree/bindings/sound/renesas,rsnd.txt | 2 +- sound/soc/sh/rcar/src.c | 14 ++++++++++++-- 2 files changed, 13 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index aa697ab..652573e 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -27,7 +27,7 @@ SSI subnode properties: - no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case SRC subnode properties: -no properties at this point +- convert-rate : SRC convert rate if needed DAI subnode properties: - playback : list of playback modules diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 4d39505..7f3c7d7 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -612,11 +612,11 @@ static void rsnd_of_parse_src(struct platform_device *pdev, const struct rsnd_of_data *of_data, struct rsnd_priv *priv) { - struct device_node *src_node; + struct device_node *src_node, *np; struct rcar_snd_info *info = rsnd_priv_to_info(priv); struct rsnd_src_platform_info *src_info; struct device *dev = &pdev->dev; - int nr; + int nr, i; if (!of_data) return; @@ -640,6 +640,16 @@ static void rsnd_of_parse_src(struct platform_device *pdev, info->src_info = src_info; info->src_info_nr = nr; + i = 0; + for_each_child_of_node(src_node, np) { + src_info = info->src_info + i; + + of_property_read_u32(np, "convert-rate", + &src_info->convert_rate); + + i++; + } + rsnd_of_parse_src_end: of_node_put(src_node); } -- 1.7.9.5

3 8

[alsa-devel] [PATCH] ASoC: cs35l32: fix compile warning for i2c_probe
by Brian Austin 07 Aug '14

07 Aug '14

Forgot to add a return for err_disable goto statement. Causes compile warning of control reaching end of non-void Signed-off-by: Brian Austin <briann.austin(a)cirrus.com> --- sound/soc/codecs/cs35l32.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 90565d5..9c6b272 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -549,6 +549,7 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, err_disable: regulator_bulk_disable(ARRAY_SIZE(cs35l32->supplies), cs35l32->supplies); + return ret; } static int cs35l32_i2c_remove(struct i2c_client *i2c_client) -- 1.7.9.5

1 0

[alsa-devel] [PATCH V2 0/2] refine esai for tdm support
by Shengjiu Wang 07 Aug '14

07 Aug '14

These patchs is to refine esai for tdm support. Changes for V2 - update the comments according the reviewer's suggestion - add init value for slots and change pin to pins. Shengjiu Wang (2): ASoC: fsl_esai: refine esai for TDM support Revert "ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support." sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/fsl_esai.c | 16 +++++++++++----- sound/soc/fsl/fsl_esai.h | 8 ++++---- 3 files changed, 15 insertions(+), 10 deletions(-) -- 1.7.9.5

2 3

[alsa-devel] [PATCH 0/3] ASoC: Intel: mfld-pcm: Fix to remove ioctls
by Subhransu S. Prusty 07 Aug '14

07 Aug '14

This patch series removes the ioctl type interface and uses dedicated ops. Subhransu S. Prusty (2): ASoC: Intel: mfld-pcm: Use function instead of ioctl ASoC: Intel: mfld-pcm: Change sst_ops prototypes to take dev parameter Vinod Koul (1): ASoC: Intel: Fix to use byte control interface sound/soc/intel/sst-mfld-platform-pcm.c | 26 +++++++++++--------------- sound/soc/intel/sst-mfld-platform.h | 27 +++++++++------------------ 2 files changed, 20 insertions(+), 33 deletions(-) -- 1.9.0

2 4

[alsa-devel] [PATCH] ASoC: fsl_sai: Add asynchronous mode support
by Nicolin Chen 07 Aug '14

07 Aug '14

From: Nicolin Chen <Guangyu.Chen(a)freescale.com> SAI supports these operation modes: 1) asynchronous mode Both Tx and Rx are set to be asynchronous. 2) synchronous mode (Rx sync with Tx) Tx is set to be asynchronous, Rx is set to be synchronous. 3) synchronous mode (Tx sync with Rx) Rx is set to be asynchronous, Tx is set to be synchronous. 4) synchronous mode (Tx/Rx sync with another SAI's Tx) 5) synchronous mode (Tx/Rx sync with another SAI's Rx) * 4) and 5) are beyond this patch because they are related with another SAI. As the initial version of this SAI driver, it supported 2) as default while the others were totally missing. So this patch just adds supports for 1) and 3). Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- .../devicetree/bindings/sound/fsl-sai.txt | 16 ++++++++++++ sound/soc/fsl/fsl_sai.c | 30 +++++++++++++++++++--- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 0f4e238..77864f4 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -24,6 +24,22 @@ Required properties: - big-endian-data: If this property is absent, the little endian mode will be in use as default, or the big endian mode will be in use for all the fifo data. +- fsl,sai-synchronous-rx: This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transimitter and receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. +- fsl,sai-asynchronous: This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which means both transimitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + +Note: +- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transimitter and receiver will send and receive data by following clocks + of transimitter. +- fsl,sai-asynchronous will be ignored if fsl,sai-synchronous-rx property is + already present. Example: sai2: sai@40031000 { diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9f10575..dc84f98 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -330,12 +330,14 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, u32 xcsr, count = 100; /* - * The transmitter bit clock and frame sync are to be - * used by both the transmitter and receiver. + * Asynchronous mode: Clear SYNC for both Tx and Rx. + * Rx sync with Tx clocks: Clear SYNC for Tx, set it for Rx. + * Tx sync with Rx clocks: Clear SYNC for Rx, set it for Tx. */ - regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, 0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + sai->synchronous[TX] ? FSL_SAI_CR2_SYNC : 0); regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, - FSL_SAI_CR2_SYNC); + sai->synchronous[RX] ? FSL_SAI_CR2_SYNC : 0); /* * It is recommended that the transmitter is the last enabled @@ -620,6 +622,26 @@ static int fsl_sai_probe(struct platform_device *pdev) return ret; } + /* Sync Tx with Rx as default by following old DT binding */ + sai->synchronous[RX] = true; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 1; + fsl_sai_dai.symmetric_channels = 1; + fsl_sai_dai.symmetric_samplebits = 1; + + if (of_find_property(np, "fsl,sai-synchronous-rx", NULL)) { + /* Sync Rx with Tx */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = true; + } else if (of_find_property(np, "fsl,sai-asynchronous", NULL)) { + /* Discard all settings for asynchronous mode */ + sai->synchronous[RX] = false; + sai->synchronous[TX] = false; + fsl_sai_dai.symmetric_rates = 0; + fsl_sai_dai.symmetric_channels = 0; + fsl_sai_dai.symmetric_samplebits = 0; + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0e6c9f5..c24d3fd 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -135,9 +135,13 @@ struct fsl_sai { bool big_endian_data; bool is_dsp_mode; bool sai_on_imx; + bool synchronous[2]; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; }; +#define TX 1 +#define RX 0 + #endif /* __FSL_SAI_H */ -- 1.8.4

4 9