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

[alsa-devel] [PATCH] ALSA: hda - initialize audio InfoFrame to be all zero
by mengdong.lin＠intel.com 23 Jul '14

23 Jul '14

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch initialized the local audio InfoFrame variable 'ai' to be all zero, thus the data bytes will indicate "Refer to Stream Header" by default. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5ef9503..3ab7063 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1062,6 +1062,7 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, { union audio_infoframe ai; + memset(&ai, 0, sizeof(ai)); if (conn_type == 0) { /* HDMI */ struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; -- 1.8.1.2

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[alsa-devel] [PATCH] pcm: pcm_local.h: include <time.h> to enable CLOCK_MONOTONIC
by Shengjiu Wang 23 Jul '14

23 Jul '14

CLOCK_MONITONIC is defined in <bits/time.h>, add <time.h> before <sys/time.h>. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_local.h | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 2206afe..80bbe59 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -24,6 +24,7 @@ #include <stdlib.h> #include <limits.h> #include <sys/uio.h> +#include <time.h> #include <sys/time.h> #define _snd_mask sndrv_mask -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: cs42xx8: Add SND_SOC_DAIFMT_DSP_A support
by Shengjiu Wang 23 Jul '14

23 Jul '14

According to the spec, the definition of TDM and ONELINE_24 for CS42XX8_INTF_DAC and CS42XX8_INTF_ADC is wrong. correct them and enable SND_SOC_DAIFMT_DSP_A support. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- sound/soc/codecs/cs42xx8.c | 3 +++ sound/soc/codecs/cs42xx8.h | 8 ++++---- 2 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ec53ffc..02b1520 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -219,6 +219,9 @@ static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_RIGHT_J: val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; break; + case SND_SOC_DAIFMT_DSP_A: + val = CS42XX8_INTF_DAC_DIF_TDM | CS42XX8_INTF_ADC_DIF_TDM; + break; default: dev_err(codec->dev, "unsupported dai format\n"); return -EINVAL; diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h index da0b94a..b2c10e5 100644 --- a/sound/soc/codecs/cs42xx8.h +++ b/sound/soc/codecs/cs42xx8.h @@ -128,8 +128,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) -#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (5 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (6 << CS42XX8_INTF_DAC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_SHIFT 0 #define CS42XX8_INTF_ADC_DIF_WIDTH 3 #define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) @@ -138,8 +138,8 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap); #define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) #define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) -#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (5 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (6 << CS42XX8_INTF_ADC_DIF_SHIFT) /* ADC Control & DAC De-Emphasis (Address 05h) */ #define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 -- 1.8.0

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[alsa-devel] [PATCH 0/2] ASoC: fsl_sai: Two bug fixes for fsl_sai driver
by Nicolin Chen 23 Jul '14

23 Jul '14

Nicolin Chen (2): ASoC: fsl_sai: Reset FIFOs after disabling TE/RE ASoC: fsl_sai: Fix incorrect register writing in fsl_sai_isr() sound/soc/fsl/fsl_sai.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH v10 0/3] Add ES8328 audio codec
by Sean Cross 23 Jul '14

23 Jul '14

This patchset adds support for the Everest Semi ES8328 audio codec, used in the Novena open source laptop. It also adds support for using the es8328 on IMX boards. We write a machine driver rather than using simple-card because the machine driver needs to support regulators for the speaker amps and as well as supporting headphone jacks, in addition to the i.MX audmux routing. Changes since v9: - Minor formatting corrections - Rework DAPM registers in es8328.h. DAPM widgets use offsets, and the v9 patch turned snd_soc_write() calls into widgets without converting from i2c masks to offsets first. - Rename DAC and ADC PLL to DLL, to reflect latest documentation. Changes since v8: - For the mclk ratios, move from an array to a lookup table, and rename the variable to be clearer. - Move all CHIPPOWER accesses into DAPM supplies. - Fix sloppy compile problems introduced in v8. - Shut down clocks and supplies on suspend, and enable them on resume. - Remove double set of ES8328_MASTERMODE_MSC. Changes since v7: - We now fetch the regulators in the device-level probe rather than in the ASoC-level probe. Changes since v6: - Minor clarification of AUDMUX in imx-audio-es8328.txt - Use u32 instead of int in imx-es8328.c for int-mux and ext-mux - Bounds checking on mux-int-port and mux-ext-port in imx-es8328.c - Remove NULL checks before calling of_node_put() Changes since v5: - Removed empty imx_set_frequency function from imx-es8328.c - Move the clock from imx-audio-es8328 to the es8328 codec itself - Add support for 2x (22.5792 MHz) and 1x (11.2896 MHz) source clocks - Restore registers on resume - Document pins for ES8328, including input pins Changes since v4: - Simplify clock handling and use assigned-clock-parents - Move the codec regulator from the machine driver to the codec - Specify all four regulators, and require their presence - Fix audio recording, such that it enables the PGA Changes since v3: - Add OF bindings for everest,es8328 - Add back in patch to add everest as a vendor prefix - Give clocks more generic names - Document required clocks in DT bindings document Changes since v2: - Change deemph to a bool, and update based on sample rate - Replace value enums with regular enums - Break SPI and I2C drivers into their own modules - Rename everest,es8328 to es8328 as the module wasn't getting detected. Because of this, the patch to vendor-prefixes.txt has been dropped. Changes since v1: - Rename HP/Speaker to OUT1/OUT2 - Use DAPM widgets for startup/shutdown - Add regulator support to machine driver Sean Cross (3): devicetree: bindings: Add Everest Semicodunctor ASoC: add es8328 codec driver ASoC: fsl: add imx-es8328 machine driver Documentation/devicetree/bindings/sound/es8328.txt | 38 ++ .../devicetree/bindings/sound/imx-audio-es8328.txt | 61 ++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/es8328-i2c.c | 60 ++ sound/soc/codecs/es8328-spi.c | 49 ++ sound/soc/codecs/es8328.c | 756 +++++++++++++++++++++ sound/soc/codecs/es8328.h | 314 +++++++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-es8328.c | 233 +++++++ 12 files changed, 1547 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 sound/soc/codecs/es8328-i2c.c create mode 100644 sound/soc/codecs/es8328-spi.c create mode 100644 sound/soc/codecs/es8328.c create mode 100644 sound/soc/codecs/es8328.h create mode 100644 sound/soc/fsl/imx-es8328.c -- 2.0.0

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Re: [alsa-devel] ALSA SOC Driver (mxs-wm8753)
by Kevin Startup 23 Jul '14

23 Jul '14

Hi there I am attempting to configure a Wolfson WM8753 on a Ka-Ro TX28S, I was wondering if you had any success managing this and if you could give me some pointers. Currently, I think my device tree may be wrong. I am getting this message on bootup: mxs-wm8753 sounds.12: ASoC: CODEC DAI wm8753 not registered Any help would be greatly appreciated Many Thanks Kevin Startup A.W. Communication Systems Ltd. email: kevin(a)awcsl.com<mailto:kevin@awcsl.com> Tel: +44(0)1697 748717 Fax: +44(0)1697 748778 A.W. Communication Systems Ltd VAT No. GB 827 4389 00 Registered No. 4891652 Registered Office Crook Barn, The Crook, Roweltown, Carlisle, UK

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[alsa-devel] [PATCH] ASoC: rsnd: add missing src/dst_addr_width for DMAEngine
by Kuninori Morimoto 23 Jul '14

23 Jul '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas new DMAEngine driver requests src/dst_addr_width Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Laurent Could you please check and give your Ack to this patch? sound/soc/sh/rcar/core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 907d480..c48d999 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -338,6 +338,8 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; cfg.src_addr = rsnd_gen_dma_addr(priv, mod_from, is_play, 1); cfg.dst_addr = rsnd_gen_dma_addr(priv, mod_to, is_play, 0); + cfg.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + cfg.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; dev_dbg(dev, "dma : %s %pad -> %pad\n", dma_name, &cfg.src_addr, &cfg.dst_addr); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: arizona: Disable AIF TX/RX before configuring it
by Richard Fitzgerald 23 Jul '14

23 Jul '14

If we don't disable the AIF TX/RX then we may fall into a situation where the new AIF settings are ignored by the device. For example, this problem manifests when switching between different sample rates. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 20 ++++++++++++++++++-- 1 files changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index fb30c82..270e7ad 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1249,6 +1249,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int tdm_width = arizona->tdm_width[dai->id - 1]; int tdm_slots = arizona->tdm_slots[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; + unsigned int aif_tx_state, aif_rx_state; if (params_rate(params) % 8000) rates = &arizona_44k1_bclk_rates[0]; @@ -1299,9 +1300,18 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + /* Save AIF TX/RX state */ + aif_tx_state = snd_soc_read(codec, base + ARIZONA_AIF_TX_ENABLES); + aif_rx_state = snd_soc_read(codec, base + ARIZONA_AIF_RX_ENABLES); + /* Disable AIF TX/RX before configuring it */ + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_ENABLES, + 0xff, 0x0); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_ENABLES, + 0xff, 0x0); + ret = arizona_hw_params_rate(substream, params, dai); if (ret != 0) - return ret; + goto restore_aif; regmap_update_bits_async(arizona->regmap, base + ARIZONA_AIF_BCLK_CTRL, @@ -1320,7 +1330,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, ARIZONA_AIF1RX_WL_MASK | ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); - return 0; +restore_aif: + /* Restore AIF TX/RX state */ + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_ENABLES, + 0xff, aif_tx_state); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_ENABLES, + 0xff, aif_rx_state); + return ret; } static const char *arizona_dai_clk_str(int clk_id) -- 1.7.2.5

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[alsa-devel] [PATCH 1/3] ASoC: wm5110: Set symmetric samplebits on AIFs
by Richard Fitzgerald 23 Jul '14

23 Jul '14

Different playback and capture bits-per-sample are not supported on the AIFs Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 62ef544..2f2ec26 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1485,6 +1485,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif2", @@ -1506,6 +1507,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-aif3", @@ -1527,6 +1529,7 @@ static struct snd_soc_dai_driver wm5110_dai[] = { }, .ops = &arizona_dai_ops, .symmetric_rates = 1, + .symmetric_samplebits = 1, }, { .name = "wm5110-slim1", -- 1.7.2.5

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[alsa-devel] [PATCH 0/4] More componentization preparation
by Lars-Peter Clausen 23 Jul '14

23 Jul '14

Now that the multi-CODEC patches have been merged we can continue with the componentization. Since it is already rather late in the development cycle again this series only has a small set of cleanup and preparation patches so we have that out of the way. The first patch removes the unused snd_soc_card platform_list, the other three patches move the card field from the snd_soc_platform and snd_soc_codec struct and consolidates the registration of controls on top of that. Series depends on topic/multi - Lars Lars-Peter Clausen (4): ASoC: Remove per card platform list ASoC: tegra: Replace instances of rtd->codec->card with rtd->card ASoC: Move card field form platform/codec to component ASoC: Add function to register component controls include/sound/soc.h | 12 ++++------ sound/soc/codecs/ac97.c | 4 ++-- sound/soc/codecs/cx20442.c | 6 ++--- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm_adsp.c | 4 ++-- sound/soc/omap/ams-delta.c | 2 +- sound/soc/soc-core.c | 52 +++++++++++++++++++++++++--------------- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-jack.c | 4 ++-- sound/soc/tegra/tegra_alc5632.c | 5 ++-- sound/soc/tegra/tegra_max98090.c | 5 ++-- sound/soc/tegra/tegra_rt5640.c | 5 ++-- sound/soc/tegra/tegra_wm8753.c | 3 +-- sound/soc/tegra/tegra_wm8903.c | 5 ++-- sound/soc/tegra/trimslice.c | 3 +-- 16 files changed, 61 insertions(+), 55 deletions(-) -- 1.8.0

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