March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: ssm2602: Replace reg_default_raw with reg_default
by James Kelly 20 Mar '18

20 Mar '18

SSM2602 driver is broken on recent kernels (at least since 4.9). User space applications such as amixer or alsamixer get EIO when attempting to access codec controls via the relevant IOCTLs. Root cause of these failures is the regcache_hw_init function in drivers/base/regmap/regcache.c, which prevents regmap cache initalization from the reg_defaults_raw element of the regmap_config structure when registers are write only. It also disables the regmap cache entirely when all registers are write only or volatile as is the case for the SSM2602 driver. Using the reg_defaults element of the regmap_config structure rather than the reg_defaults_raw element to initalize the regmap cache avoids the logic in the regcache_hw_init function entirely. It also makes this driver consistent with other ASoC codec drivers, as this driver was the ONLY codec driver that used the reg_defaults_raw element to initalize the cache. Tested on Digilent Zybo Z7 development board which has a SSM2603 codec chip connected to a Xilinx Zynq SoC. --- sound/soc/codecs/ssm2602.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 031e0eabcc1f..501a4e73b185 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -54,10 +54,17 @@ struct ssm2602_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 +static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { + { .reg = 0x00, .def = 0x0097 }, + { .reg = 0x01, .def = 0x0097 }, + { .reg = 0x02, .def = 0x0079 }, + { .reg = 0x03, .def = 0x0079 }, + { .reg = 0x04, .def = 0x000a }, + { .reg = 0x05, .def = 0x0008 }, + { .reg = 0x06, .def = 0x009f }, + { .reg = 0x07, .def = 0x000a }, + { .reg = 0x08, .def = 0x0000 }, + { .reg = 0x09, .def = 0x0000 } }; @@ -621,8 +628,8 @@ const struct regmap_config ssm2602_regmap_config = { .volatile_reg = ssm2602_register_volatile, .cache_type = REGCACHE_RBTREE, - .reg_defaults_raw = ssm2602_reg, - .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), + .reg_defaults = ssm2602_reg, + .num_reg_defaults = ARRAY_SIZE(ssm2602_reg), }; EXPORT_SYMBOL_GPL(ssm2602_regmap_config); -- 2.14.3 (Apple Git-98)

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[alsa-devel] [PATCH] ASoC: rt286: don't turn off HV and VREF if headset is detected
by Bard Liao 20 Mar '18

20 Mar '18

"HV" and "VREF" will be powered up when jack type detection process is started and will be powered off when jack type detection process is finished. It will generate an unexpected interrupt signal when they are powered up during the capture process. Codec driver will do the jack type detection process and we can't capture properly before the jack detection process is finished. This patch will not power off the "HV" and "VREF" widgets if headset is detected and it will solve the unexpected interrupt issue. As a result, it will also solve the silence data captured at the beginning in headset mic recording issue. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e164357..0b0f748 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -275,9 +275,10 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - - snd_soc_dapm_disable_pin(dapm, "HV"); - snd_soc_dapm_disable_pin(dapm, "VREF"); + if (!*mic) { + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); + } if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); snd_soc_dapm_sync(dapm); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt298: don't turn off HV and VREF if headset is detected
by Bard Liao 20 Mar '18

20 Mar '18

"HV" and "VREF" will be powered up when jack type detection process is started and will be powered off when jack type detection process is finished. It will generate an unexpected interrupt signal when they are powered up during the capture process. Codec driver will do the jack type detection process and we can't capture properly before the jack detection process is finished. This patch will not power off the "HV" and "VREF" widgets if headset is detected and it will solve the unexpected interrupt issue. As a result, it will also solve the silence data captured at the beginning in headset mic recording issue. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7bf4b31..06cdba4 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -290,9 +290,10 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) regmap_read(rt298->regmap, RT298_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - - snd_soc_dapm_disable_pin(dapm, "HV"); - snd_soc_dapm_disable_pin(dapm, "VREF"); + if (!*mic) { + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); + } if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); snd_soc_dapm_sync(dapm); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: twl6040: remove duplicated remove callback
by Kuninori Morimoto 20 Mar '18

20 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We don't need 2 .remove callback Fixes: 7480389fb0d8 ("ASoC: twl6040: replace codec to component") Reported-by: Julia Lawall <julia.lawall(a)lip6.fr> Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/twl6040.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b56a647..bfd1abd 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1158,7 +1158,6 @@ static void twl6040_remove(struct snd_soc_component *component) .remove = twl6040_remove, .read = twl6040_read, .write = twl6040_write, - .remove = twl6040_remove, .set_bias_level = twl6040_set_bias_level, .controls = twl6040_snd_controls, .num_controls = ARRAY_SIZE(twl6040_snd_controls), -- 1.9.1

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[alsa-devel] [PATCH] ASoC: topology: Fix kcontrol name string handling.
by Liam Girdwood 20 Mar '18

20 Mar '18

Fix the topology kcontrol string handling so that string pointer references are strdup()ed instead of being copied. This fixes issues with kcontrol templates on the stack or ones that are freed. Remember and free the strings too when topology is unloaded. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 78197e82fc4f..dd2b118a35b3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -530,6 +530,7 @@ static void remove_widget(struct snd_soc_component *comp, kfree(se->dobj.control.dtexts[j]); kfree(se); + kfree(w->kcontrol_news[i].name); } } else { /* volume mixer or bytes controls */ @@ -546,6 +547,7 @@ static void remove_widget(struct snd_soc_component *comp, */ kfree((void *)kcontrol->private_value); snd_ctl_remove(card, kcontrol); + kfree(w->kcontrol_news[i].name); } } @@ -1256,7 +1258,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); - kc[i].name = mc->hdr.name; + kc[i].name = kstrdup(mc->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err_str; kc[i].private_value = (long)sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = mc->hdr.access; @@ -1304,8 +1308,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( err_str: kfree(sm); err: - for (--i; i >= 0; i--) + for (--i; i >= 0; i--) { kfree((void *)kc[i].private_value); + kfree(kc[i].name); + } kfree(kc); return NULL; } @@ -1336,7 +1342,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); - kc[i].name = ec->hdr.name; + kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err_se; kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1412,6 +1420,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( kfree(se->dobj.control.dtexts[j]); kfree(se); + kfree(kc[i].name); } err: kfree(kc); @@ -1450,7 +1459,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - kc[i].name = be->hdr.name; + kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access; @@ -1480,8 +1491,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( return kc; err: - for (--i; i >= 0; i--) + for (--i; i >= 0; i--) { kfree((void *)kc[i].private_value); + kfree(kc[i].name); + } kfree(kc); return NULL; -- 2.14.1

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[alsa-devel] [PATCH v2 0/2] USB Audio Device Class 3.0 support
by Ruslan Bilovol 20 Mar '18

20 Mar '18

Hi Takashi, This patch adds initial USB Audio Device Class 3.0 [1] support to the ALSA that we discussed at ELCE. The patch was tested with UAC3 gadget [2] that I posted to USB mailing list before. It is good for working with BADD device which implements such topologies like BAIF (Basic Audio Input Functions), BAOF (Basic Audio Output Functions) or compbination of both. UAC3 spec has changed descriptors laout and many other codes comparing to UAC2 spec, thus making reuse of existing sources impossible or quite complex. There are still many areas of improvement, as this patch doesn't have UAC3 Mixer Unit support nor some new features like Power Management I tested this with BeagleBone Black as UAC3 gadget device. [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] http://www.spinics.net/lists/linux-usb/msg162482.html v2 changes: - moved audioformat quirks to quirks.c - addressed many comments from Pierre-Louis Bossart - reorganized audio-v3.h - added more UAC3 channel relationship definitions Ruslan Bilovol (2): ALSA: usb-audio: move audioformat quirks to quirks.c ALSA: usb: initial USB Audio Device Class 3.0 support include/linux/usb/audio-v2.h | 4 +- include/linux/usb/audio-v3.h | 395 +++++++++++++++++++++++++++++++++++++++++ include/uapi/linux/usb/audio.h | 1 + sound/usb/card.c | 7 +- sound/usb/card.h | 2 +- sound/usb/clock.c | 228 +++++++++++++++++++++--- sound/usb/clock.h | 4 +- sound/usb/format.c | 91 ++++++++-- sound/usb/format.h | 6 +- sound/usb/mixer.c | 337 +++++++++++++++++++++++------------ sound/usb/quirks.c | 34 ++++ sound/usb/quirks.h | 4 + sound/usb/stream.c | 393 +++++++++++++++++++++++++++++++++------- 13 files changed, 1284 insertions(+), 222 deletions(-) create mode 100644 include/linux/usb/audio-v3.h -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix parsing descriptor of UAC2 processing unit
by Kirill Marinushkin 20 Mar '18

20 Mar '18

Currently, the offsets in the UAC2 processing unit descriptor are calculated incorrectly. It causes an issue when connecting the device which provides such a feature: ~~~~ [84126.724420] usb 1-1.3.1: invalid Processing Unit descriptor (id 18) ~~~~ After this patch is applied, the UAC2 processing unit inits w/o this error. Fixes: 5e1ddb481776 ("UAPI: (Scripted) Disintegrate include/linux/usb") Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Felipe Balbi <felipe.balbi(a)linux.intel.com> Cc: Philippe Ombredanne <pombredanne(a)nexb.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- include/uapi/linux/usb/audio.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index 17a022c5b414..da3315ed1bcd 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -370,7 +370,7 @@ static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_d { return (protocol == UAC_VERSION_1) ? desc->baSourceID[desc->bNrInPins + 4] : - desc->baSourceID[desc->bNrInPins + 6]; + 2; /* in UAC2, this value is constant */ } static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc, @@ -378,7 +378,7 @@ static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_de { return (protocol == UAC_VERSION_1) ? &desc->baSourceID[desc->bNrInPins + 5] : - &desc->baSourceID[desc->bNrInPins + 7]; + &desc->baSourceID[desc->bNrInPins + 6]; } static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc, -- 2.13.6

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[alsa-devel] [PATCH V2] ASoC: amd: dma driver changes for BT I2S controller instance
by Vijendar Mukunda 19 Mar '18

19 Mar '18

With in ACP, There are three I2S controllers can be configured/enabled ( I2S SP, I2S MICSP, I2S BT). Default enabled I2S controller instance is I2S SP. This patch provides required changes to support I2S BT controller Instance. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 106 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp.h | 10 +++++ 2 files changed, 94 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 540088d..a61c4e0 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -697,6 +697,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *prtd = substream->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(prtd->cpu_dai); struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); if (adata == NULL) @@ -710,7 +711,21 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } + adata->i2s_play_instance = dev->i2s_instance; + if (adata->i2s_play_instance == I2S_SP_INSTANCE) + adata->i2ssp_renderbytescount = 0; + else if (adata->i2s_play_instance == I2S_BT_INSTANCE) + adata->i2sbt_renderbytescount = 0; + else + return -EINVAL; } else { + adata->i2s_capture_instance = dev->i2s_instance; + if (adata->i2s_capture_instance == I2S_SP_INSTANCE) + adata->i2ssp_capturebytescount = 0; + else if (adata->i2s_capture_instance == I2S_BT_INSTANCE) + adata->i2sbt_capturebytescount = 0; + else + return -EINVAL; switch (intr_data->asic_type) { case CHIP_STONEY: runtime->hw = acp_st_pcm_hardware_capture; @@ -736,11 +751,20 @@ static int acp_dma_open(struct snd_pcm_substream *substream) * This enablement is not required for another stream, if current * stream is not closed */ - if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) + if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && + !intr_data->play_i2sbt_stream && + !intr_data->capture_i2sbt_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; + switch (adata->i2s_play_instance) { + case I2S_BT_INSTANCE: + intr_data->play_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->play_i2ssp_stream = substream; + } /* For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. @@ -751,7 +775,14 @@ static int acp_dma_open(struct snd_pcm_substream *substream) bank, true); } } else { - intr_data->capture_i2ssp_stream = substream; + switch (adata->i2s_capture_instance) { + case I2S_BT_INSTANCE: + intr_data->capture_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->capture_i2ssp_stream = substream; + } if (intr_data->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(intr_data->acp_mmio, @@ -1010,34 +1041,49 @@ static int acp_dma_close(struct snd_pcm_substream *substream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_i2ssp_stream = NULL; - /* For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - * added condition checks for Carrizo platform only - */ - if (adata->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->play_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->play_i2ssp_stream = NULL; + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks + * is ON.Setting SRAM bank state code skipped for STONEY + * platform.Added condition checks for Carrizo platform + * only. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } else { - adata->capture_i2ssp_stream = NULL; - if (adata->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->capture_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->capture_i2ssp_stream = NULL; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. - */ - if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) + */ + if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } @@ -1089,6 +1135,8 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_i2ssp_stream = NULL; audio_drv_data->capture_i2ssp_stream = NULL; + audio_drv_data->play_i2sbt_stream = NULL; + audio_drv_data->capture_i2sbt_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1177,6 +1225,20 @@ static int acp_pcm_resume(struct device *dev) adata->capture_i2ssp_stream->runtime->private_data, adata->asic_type); } + if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2sbt_stream && + adata->play_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->play_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + if (adata->capture_i2sbt_stream && + adata->capture_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->capture_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index ba01510..8bc78cc 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -4,6 +4,8 @@ #include "include/acp_2_2_d.h" #include "include/acp_2_2_sh_mask.h" +#include <sound/designware_i2s.h> +#include "../dwc/local.h" #define ACP_PAGE_SIZE_4K_ENABLE 0x02 @@ -72,6 +74,8 @@ #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 +#define I2S_SP_INSTANCE 1 +#define I2S_BT_INSTANCE 2 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, @@ -88,12 +92,18 @@ struct audio_substream_data { uint64_t size; u64 i2ssp_renderbytescount; u64 i2ssp_capturebytescount; + u64 i2sbt_renderbytescount; + u64 i2sbt_capturebytescount; void __iomem *acp_mmio; + u16 i2s_play_instance; + u16 i2s_capture_instance; }; struct audio_drv_data { struct snd_pcm_substream *play_i2ssp_stream; struct snd_pcm_substream *capture_i2ssp_stream; + struct snd_pcm_substream *play_i2sbt_stream; + struct snd_pcm_substream *capture_i2sbt_stream; void __iomem *acp_mmio; u32 asic_type; }; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: dwc: I2S Controller instance param added
by Vijendar Mukunda 19 Mar '18

19 Mar '18

When multiple I2S controller instances created, i2s_instance parameter refers to i2s controller instance value. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- include/sound/designware_i2s.h | 4 ++++ sound/soc/dwc/dwc-i2s.c | 1 + sound/soc/dwc/local.h | 1 + 3 files changed, 6 insertions(+) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 830f5ca..2a94e7c 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -44,6 +44,10 @@ struct i2s_platform_data { int channel; u32 snd_fmts; u32 snd_rates; + /* i2s_instance parameter returns I2S controller instance value + * when multiple I2S controllers instantiated + */ + u32 i2s_instance; #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e27e21f..20145c0 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -549,6 +549,7 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) idx = 1; + dev->i2s_instance = pdata->i2s_instance; /* Set DMA slaves info */ dev->play_dma_data.pd.data = pdata->play_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data; diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 91dc70a..e89e464 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -98,6 +98,7 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 i2s_instance; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; -- 2.7.4

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[alsa-devel] [PATCH v2] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 19 Mar '18

19 Mar '18

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 11 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 232 +++++++++++++++++++++++++++++++++++++ 3 files changed, 245 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..822b9143dcb1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,17 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on MFD_INTEL_LPSS && ACPI && I2C + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..2b49cb6aa84a 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -21,6 +21,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -44,3 +45,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..04986c015f0f --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,232 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2016-18 Intel Corporation. + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack cnl_headset; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_component *component = codec_dai->component; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_component_set_jack(component, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = KBUILD_MODNAME, + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = cnl_rt274_probe, +}; + +module_platform_driver(cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.16.2

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