June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/7] asoc: Add audio for sti platforms
by Arnaud Pouliquen 10 Jul '15

10 Jul '15

version 1: - First patches submission Set of patches concerns the audio part of the STi platform Hardware is split in two main blocks: Unipheriperal IPs and codec IPs Uniperipheral IPs: Ips in charge of the I2S buses Uniperipheral IPs are in charge the PCM rendering/capturing on/from I2S. - 4 uniperipheral player IPs receive the PCM audio data or audio encoded bit stream data from the memory (via FDMA)and performs optional IEC-60958 or IEC-61937 formatting on the audio data and sends them out on the I2S output. - 2 uniperipheral reader IPS receive the PCM audio or encoded bit stream on the I2S input interface and writes the processed data in memory (via FDMA) Codec IPs: - HDMI_TX: format and generate HDMI signal. - SPDIF: format and generate biphase signal for SPDIF output - DAC: format and generate stereo analogic signal - HDMI_RX: receive HDMI signal from external source. In addition 2 I2S/TDM interfaces are available to connect external ICs - PCM_IN to external capture codec (ADC, BT...) - PCM_OUT for external player codec (DAC, BT..) On Sti platform audio driver is splitted in 2 part to support Hw architecture - sti_audio_platform for uniperipheral IPs mmannagement - sti-sas codec for codec IP management +-----------------------+ +-----------------+ | +-----------------+ | | | | | +---------+ | | | +---------+ | | | |Uniperif | | | | | HDMI | | +---+ | | |player 0 +---------------> TX +-------> | | | +---------+ | | | +---------+ | +---+ | | +---------+ | | | | | | |Uniperif | | | | | +---+ | | |player 1 +---------------------------------> | | | +---------+ | | | | +---+ | | +---------+ | | | +---------+ | +---+ | | |Uniperif | | | | | DAC | | | | | | |player 2 +---------------> +------^----+ | | +---------+ | | | +---------+ | | | +---------+ | | | +---------+ | | | |Uniperif | | | | | SPDIF | | +---+ | | |player 3 +---------------> +-------> | | | +---------+ | | | +---------+ | +---+ | | +---------+ | | | | | | |Uniperif | | | | | +---+ | | |reader 0 <---------------------------------+ | | | +---------+ | | | | +---+ | | +---------+ | | | +---------+ | | | |Uniperif | | | | | HDMIF | | +---+ | | |reader 1 <---------------+ RX <-------+ | | | +---------+ | | | +---------+ | +---+ | | | | | | | | CPU DAI | | | | | +-----------------+ | | | | STI platform | | SAS CODEC | +-----------------------+ +-----------------+ In the first version of the driver only SPDIF and DAC are supported. in future HDMI TX, HDMI_RX and support for TDM mode should be added. Arnaud Pouliquen (7): ASoC: sti: add binding for ASoc driver Asoc: sti: add uniperipheral header file Asoc: sti: add CPU DAI driver for playback Asoc: sti: add CPU DAI driver for capture Asoc: sti: Add platform driver ASoc: Add ability to build sti drivers ASoc: Codec: add sti platform codec .../devicetree/bindings/sound/st,sti-asoc-card.txt | 176 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 663 ++++++++++ sound/soc/sti/Kconfig | 17 + sound/soc/sti/Makefile | 5 + sound/soc/sti/sti_platform.c | 643 +++++++++ sound/soc/sti/uniperif.h | 1236 ++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1367 ++++++++++++++++++++ sound/soc/sti/uniperif_reader.c | 493 +++++++ 12 files changed, 4608 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_platform.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

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[alsa-devel] [PATCH 0/6] ASoC: Intel: fix kconfig, symbols and sparse
by Vinod Koul 09 Jul '15

09 Jul '15

So this series has two parts part 1, first three patches which fix the issue of BYT driver not getting compiled in -next and then cleans up symbol usage in various driver Second part fixes (next three) spare warnings on drivers Vinod Koul (6): ASoC: Intel: use CONFIG_SND_SOC for intel boards ASoC: Intel: kconfig - Move DW_DMAC_CORE dependency to machines ASoC: intel - kconfig: remove SND_SOC_INTEL_SST prompt ASOC: intel - hsw: fix sparse unused varible warn ASoC: intel - atom: statify sst_slot_enum_info ASoC: Intel -atom: fix sst_platform_compr_ops declaration sound/soc/intel/Kconfig | 24 +++++++++++++----------- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 - sound/soc/intel/atom/sst-mfld-platform.h | 1 + sound/soc/intel/haswell/sst-haswell-pcm.c | 1 - 6 files changed, 16 insertions(+), 15 deletions(-) -- 1.9.1

3 13

[alsa-devel] [PATCH 0/2] ASoC: Intel: skylake - match braces for if-else
by Vinod Koul 09 Jul '15

09 Jul '15

Coding style mandates that if-else should match braces even if one of them is a single line statement. Fix it in two places missed earlier Mark, you can also fold these two patches with SKL driver series (1st and 2nd patch can fold in 1st and 2nd patch respectively) or apply independetly, either way okay Vinod Koul (2): ASoC: Intel: skl-pcm.c match braces for if-else ASoC: Intel: skl.c match braces for if-else sound/soc/intel/skylake/skl-pcm.c | 3 ++- sound/soc/intel/skylake/skl.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 1/4] ASoC: max98090: Check for clk_prepare_enable() error
by Fabio Estevam 07 Jul '15

07 Jul '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/max98090.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 78268f05..1697340 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1801,10 +1801,13 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, if (IS_ERR(max98090->mclk)) break; - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { clk_disable_unprepare(max98090->mclk); - else - clk_prepare_enable(max98090->mclk); + } else { + ret = clk_prepare_enable(max98090->mclk); + if (ret) + return ret; + } break; case SND_SOC_BIAS_STANDBY: -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: topology: Fix TLV size calculation.
by Liam Girdwood 07 Jul '15

07 Jul '15

TLV size calculation was incorrectly calculated. Fix this according to include/sound/tlv.h Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index d096068..7a19df3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -580,27 +580,26 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, } static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, u32 tlv_size) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) { - struct snd_soc_tplg_ctl_tlv *tplg_tlv; struct snd_ctl_tlv *tlv; + int size; - if (tlv_size == 0) + if (tplg_tlv->count == 0) return 0; - tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos; - tplg->pos += tlv_size; - - tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL); + size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & + ~(sizeof(unsigned int) - 1)); + tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); if (tlv == NULL) return -ENOMEM; dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, tplg_tlv->size); + tplg_tlv->numid, size); tlv->numid = tplg_tlv->numid; - tlv->length = tplg_tlv->size; - memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->size); + tlv->length = size; + memcpy(&tlv->tlv[0], tplg_tlv->data, size); kc->tlv.p = (void *)tlv; return 0; @@ -773,7 +772,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size); + soc_tplg_create_tlv(tplg, &kc, &mc->tlv); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- 2.1.4

2 7

[alsa-devel] [PATCH 0/2] ASoc: simple card: extend for MCLK support
by Arnaud Pouliquen 07 Jul '15

07 Jul '15

Patches propose to extend the management of the mclk-fs ratio. This extension allows to support following card constraints: - mclk is generated by cpu_dai, depending on frame rate. cpu_dai needs to be informed on the ratio supported by codec. - Each codec needs a specific ratio. In this case mclk-fs needs to be defined for the dai_link. Arnaud Pouliquen (2): ASoC: simple card: Add mclk-fs property in dai-link ASoC: simple card: set cpu-dai sysclk with mclk-fs .../devicetree/bindings/sound/simple-card.txt | 6 ++++- sound/soc/generic/simple-card.c | 27 +++++++++++++++++++--- 2 files changed, 29 insertions(+), 4 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: wm8731: initialize the hardware when loading the codec driver
by Sergey Kiselev 07 Jul '15

07 Jul '15

From: Sergey Kiselev <sergey.kiselev(a)intel.com> This patch moves the requesting supplies, hardware reset and initialization from wm8731_probe to wm8731_i2c_probe and wm8731_spi_probe. So that the codec hardware is initialized when loading the codec driver, and not when loading the machine driver. This avoids unnecesary hardware resets and re-initializations when re-loading the machine driver. Signed-off-by: Sergey Kiselev <sergey.kiselev(a)intel.com> --- sound/soc/codecs/wm8731.c | 74 +++++++++++++++++++++++++++-------------------- 1 file changed, 42 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 915ea11..f22935a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -84,7 +84,7 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg) return reg <= WM8731_RESET; } -#define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) +#define wm8731_reset(m) regmap_write(m, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -571,69 +571,63 @@ static struct snd_soc_dai_driver wm8731_dai = { .symmetric_rates = 1, }; -static int wm8731_probe(struct snd_soc_codec *codec) +static int wm8731_request_supplies(struct device *dev, + struct wm8731_priv *wm8731) { - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + dev_err(dev, "Failed to request supplies: %d\n", ret); return ret; } ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + dev_err(dev, "Failed to enable supplies: %d\n", ret); return ret; } - ret = wm8731_reset(codec); + return 0; +} + +static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) +{ + int ret = 0; + + ret = wm8731_reset(wm8731->regmap); if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + dev_err(dev, "Failed to issue reset: %d\n", ret); goto err_regulator_enable; } - snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Clear POWEROFF, keep everything else disabled */ + regmap_write(wm8731->regmap, WM8731_PWR, 0x7f); /* Latch the update bits */ - snd_soc_update_bits(codec, WM8731_LOUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_ROUT1V, 0x100, 0); - snd_soc_update_bits(codec, WM8731_LINVOL, 0x100, 0); - snd_soc_update_bits(codec, WM8731_RINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LOUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_ROUT1V, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_LINVOL, 0x100, 0); + regmap_update_bits(wm8731->regmap, WM8731_RINVOL, 0x100, 0); /* Disable bypass path by default */ - snd_soc_update_bits(codec, WM8731_APANA, 0x8, 0); + regmap_update_bits(wm8731->regmap, WM8731_APANA, 0x8, 0); - /* Regulators will have been enabled by bias management */ - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - - return 0; + regcache_mark_dirty(wm8731->regmap); err_regulator_enable: + /* Regulators will be enabled by bias management */ regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); return ret; } -/* power down chip */ -static int wm8731_remove(struct snd_soc_codec *codec) -{ - struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); - - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { - .probe = wm8731_probe, - .remove = wm8731_remove, .set_bias_level = wm8731_set_bias_level, .suspend_bias_off = true, @@ -690,6 +684,12 @@ static int wm8731_spi_probe(struct spi_device *spi) mutex_init(&wm8731->lock); + spi_set_drvdata(spi, wm8731); + + ret = wm8731_request_supplies(&spi->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -698,7 +698,9 @@ static int wm8731_spi_probe(struct spi_device *spi) return ret; } - spi_set_drvdata(spi, wm8731); + ret = wm8731_hw_init(&spi->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); @@ -754,6 +756,12 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, mutex_init(&wm8731->lock); + i2c_set_clientdata(i2c, wm8731); + + ret = wm8731_request_supplies(&i2c->dev, wm8731); + if (ret != 0) + return ret; + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); @@ -762,7 +770,9 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, return ret; } - i2c_set_clientdata(i2c, wm8731); + ret = wm8731_hw_init(&i2c->dev, wm8731); + if (ret != 0) + return ret; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: move RT5645 muxes to rt5645_specific_dapm_widgets
by Oder Chiou 07 Jul '15

07 Jul '15

From: Bard Liao <bardliao(a)realtek.com> This is a similar patch to "move RT5650 muxes to rt5650_specific_ dapm_widgets" patch. The purpose is to silence the "has no paths" warnings. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index d0750be..8c8a2b6 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1697,15 +1697,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), /* IF1 2 Mux */ - SND_SOC_DAPM_MUX("RT5645 IF1 ADC1 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc1_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC2 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc2_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC3 Swap Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc3_in_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, - 0, 0, &rt5645_if1_adc_in_mux), - SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, 0, 0, &rt5645_if2_adc_in_mux), @@ -1716,14 +1707,6 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 DAC3", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac0_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac1_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac2_tdm_sel_mux), - SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, - &rt5645_if1_dac3_tdm_sel_mux), SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1856,6 +1839,25 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5645_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac0_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC1 R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac1_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac2_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 DAC2 R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_if1_dac3_tdm_sel_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC1 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc1_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC2 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc2_in_mux), + SND_SOC_DAPM_MUX("RT5645 IF1 ADC3 Swap Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc3_in_mux), +}; + static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5650_a_dac1_l_mux), @@ -3050,6 +3052,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: + snd_soc_dapm_new_controls(&codec->dapm, + rt5645_specific_dapm_widgets, + ARRAY_SIZE(rt5645_specific_dapm_widgets)); snd_soc_dapm_add_routes(&codec->dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: codecs: max98088: fix typo in constant
by Nik Nyby 07 Jul '15

07 Jul '15

This fixes a typo in the M98088_REG_02_JACK_STATUS constant. Signed-off-by: Nik Nyby <nikolas(a)gnu.org> --- sound/soc/codecs/max98088.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index be89a4f..efa39bf 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -16,7 +16,7 @@ */ #define M98088_REG_00_IRQ_STATUS 0x00 #define M98088_REG_01_MIC_STATUS 0x01 -#define M98088_REG_02_JACK_STAUS 0x02 +#define M98088_REG_02_JACK_STATUS 0x02 #define M98088_REG_03_BATTERY_VOLTAGE 0x03 #define M98088_REG_0F_IRQ_ENABLE 0x0F #define M98088_REG_10_SYS_CLK 0x10 -- 2.1.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: arizona: Implement stability check for EQ coefficients
by Charles Keepax 07 Jul '15

07 Jul '15

Specifying unstable coefficients for the EQ can have a severe impact on the audio. This patchs adds a stability check on the coefficients written to the EQ, for this it is necessary to merge the mode control and the coefficients as some coefficients may only be unstable with a certain mode setting so it is ideal if these are always updated in sync. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 59 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 10 +++++++ sound/soc/codecs/wm5102.c | 12 +++------ sound/soc/codecs/wm5110.c | 12 +++------ sound/soc/codecs/wm8997.c | 12 +++------ 5 files changed, 81 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 802e05e..3996786 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2313,6 +2313,65 @@ const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { }; EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); +static bool arizona_eq_filter_unstable(bool mode, __be16 _a, __be16 _b) +{ + s16 a = be16_to_cpu(_a); + s16 b = be16_to_cpu(_b); + + if (!mode) { + return abs(a) >= 4096; + } else { + if (abs(b) >= 4096) + return true; + + return (abs((a << 16) / (4096 - b)) >= 4096 << 4); + } +} + +int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct soc_bytes *params = (void *)kcontrol->private_value; + unsigned int val; + __be16 *data; + int len; + int ret; + + len = params->num_regs * regmap_get_val_bytes(arizona->regmap); + + data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + data[0] &= cpu_to_be16(ARIZONA_EQ1_B1_MODE); + + if (arizona_eq_filter_unstable(!!data[0], data[1], data[2]) || + arizona_eq_filter_unstable(true, data[4], data[5]) || + arizona_eq_filter_unstable(true, data[8], data[9]) || + arizona_eq_filter_unstable(true, data[12], data[13]) || + arizona_eq_filter_unstable(false, data[16], data[17])) { + dev_err(arizona->dev, "Rejecting unstable EQ coefficients\n"); + ret = -EINVAL; + goto out; + } + + ret = regmap_read(arizona->regmap, params->base, &val); + if (ret != 0) + goto out; + + val &= ~ARIZONA_EQ1_B1_MODE; + data[0] |= cpu_to_be16(val); + + ret = regmap_raw_write(arizona->regmap, params->base, data, len); + +out: + kfree(data); + return ret; +} +EXPORT_SYMBOL_GPL(arizona_eq_coeff_put); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 43deb04..ebe50cf 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -194,6 +194,13 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name " Preloader", name "L"), \ ARIZONA_MIXER_ROUTES(name " Preloader", name "R") +#define ARIZONA_EQ_CONTROL(xname, xbase) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \ + .put = arizona_eq_coeff_put, .private_value = \ + ((unsigned long)&(struct soc_bytes) { .base = xbase, \ + .num_regs = 20, .mask = ~ARIZONA_EQ1_B1_MODE }) } + #define ARIZONA_RATE_ENUM_SIZE 4 extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; @@ -229,6 +236,9 @@ extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int arizona_eq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index d097f09e5..556bd98 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -788,8 +788,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -801,8 +800,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -814,8 +812,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -827,8 +824,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 709fcc6..dfa4d4b 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,8 +247,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -260,8 +259,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -273,8 +271,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -286,8 +283,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4134dc7..53e0a46 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -174,8 +174,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), -SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ1 Coefficients", ARIZONA_EQ1_2), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -187,8 +186,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), -SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ2 Coefficients", ARIZONA_EQ2_2), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -200,8 +198,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), -SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ3 Coefficients", ARIZONA_EQ3_2), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -213,8 +210,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), -SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), +ARIZONA_EQ_CONTROL("EQ4 Coefficients", ARIZONA_EQ4_2), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- 1.7.2.5

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