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

[alsa-devel] [PATCH 1/1] ASoC: sirf-usp: Fixed a bug for playback and capture work at the same time
by Rongjun Ying 25 Jul '14

25 Jul '14

1. The startup function invoked when the playback and capture. If start playback when capturing, the registers are re-initinitialised. That cause the playback fail. So move the startup code into runtime resume. 2. Modified: If non RUNTIME_PM support, the probe need enable clock and initinitialise registers. 3. Refine code. Signed-off-by: Rongjun Ying <rongjun.ying(a)csr.com> --- sound/soc/sirf/sirf-usp.c | 54 +++++++++++++++++++++++------------------------ 1 file changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 9693bc2..3a73037 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -103,11 +103,8 @@ static int sirf_usp_pcm_set_dai_fmt(struct snd_soc_dai *dai, return 0; } -static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static void sirf_usp_i2s_init(struct sirf_usp *usp) { - struct sirf_usp *usp = snd_soc_dai_get_drvdata(dai); - /* Configure RISC mode */ regmap_update_bits(usp->regmap, USP_RISC_DSP_MODE, USP_RISC_DSP_SEL, ~USP_RISC_DSP_SEL); @@ -119,19 +116,16 @@ static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, regmap_write(usp->regmap, USP_TX_DMA_IO_LEN, 0); regmap_write(usp->regmap, USP_RX_DMA_IO_LEN, 0); - regmap_write(usp->regmap, USP_RX_FRAME_CTRL, USP_SINGLE_SYNC_MODE); - - regmap_write(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_SLAVE_CLK_SAMPLE); - /* Configure Mode2 register */ regmap_write(usp->regmap, USP_MODE2, (1 << USP_RXD_DELAY_LEN_OFFSET) | - (0 << USP_TXD_DELAY_LEN_OFFSET)); + (0 << USP_TXD_DELAY_LEN_OFFSET) | + USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE); /* Configure Mode1 register */ regmap_write(usp->regmap, USP_MODE1, USP_SYNC_MODE | USP_EN | USP_TXD_ACT_EDGE_FALLING | USP_RFS_ACT_LEVEL_LOGIC1 | USP_TFS_ACT_LEVEL_LOGIC1 | - USP_TX_UFLOW_REPEAT_ZERO); + USP_TX_UFLOW_REPEAT_ZERO | USP_CLOCK_MODE_SLAVE); /* Configure RX DMA IO Control register */ regmap_write(usp->regmap, USP_RX_DMA_IO_CTRL, 0); @@ -155,8 +149,6 @@ static int sirf_usp_i2s_startup(struct snd_pcm_substream *substream, /* Congiure TX FIFO Level Check register */ regmap_write(usp->regmap, USP_TX_FIFO_LEVEL_CHK, TX_FIFO_SC(0x1B) | TX_FIFO_LC(0x0E) | TX_FIFO_HC(0x04)); - - return 0; } static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, @@ -204,23 +196,19 @@ static int sirf_usp_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) regmap_update_bits(usp->regmap, USP_TX_FRAME_CTRL, USP_TXC_DATA_LEN_MASK | USP_TXC_FRAME_LEN_MASK - | USP_TXC_SHIFTER_LEN_MASK, + | USP_TXC_SHIFTER_LEN_MASK | USP_TXC_SLAVE_CLK_SAMPLE, ((data_len - 1) << USP_TXC_DATA_LEN_OFFSET) | ((frame_len - 1) << USP_TXC_FRAME_LEN_OFFSET) - | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET)); + | ((shifter_len - 1) << USP_TXC_SHIFTER_LEN_OFFSET) + | USP_TXC_SLAVE_CLK_SAMPLE); else regmap_update_bits(usp->regmap, USP_RX_FRAME_CTRL, USP_RXC_DATA_LEN_MASK | USP_RXC_FRAME_LEN_MASK - | USP_RXC_SHIFTER_LEN_MASK, + | USP_RXC_SHIFTER_LEN_MASK | USP_SINGLE_SYNC_MODE, ((data_len - 1) << USP_RXC_DATA_LEN_OFFSET) | ((frame_len - 1) << USP_RXC_FRAME_LEN_OFFSET) - | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET)); - - regmap_update_bits(usp->regmap, USP_MODE1, - USP_CLOCK_MODE_SLAVE, USP_CLOCK_MODE_SLAVE); - regmap_update_bits(usp->regmap, USP_MODE2, - USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE, - USP_TFS_CLK_SLAVE_MODE | USP_RFS_CLK_SLAVE_MODE); + | ((shifter_len - 1) << USP_RXC_SHIFTER_LEN_OFFSET) + | USP_SINGLE_SYNC_MODE); return 0; } @@ -253,7 +241,6 @@ static int sirf_usp_pcm_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops sirf_usp_pcm_dai_ops = { - .startup = sirf_usp_i2s_startup, .trigger = sirf_usp_pcm_trigger, .set_fmt = sirf_usp_pcm_set_dai_fmt, .hw_params = sirf_usp_pcm_hw_params, @@ -282,7 +269,6 @@ static struct snd_soc_dai_driver sirf_usp_pcm_dai = { .ops = &sirf_usp_pcm_dai_ops, }; -#ifdef CONFIG_PM static int sirf_usp_pcm_runtime_suspend(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); @@ -293,9 +279,15 @@ static int sirf_usp_pcm_runtime_suspend(struct device *dev) static int sirf_usp_pcm_runtime_resume(struct device *dev) { struct sirf_usp *usp = dev_get_drvdata(dev); - return clk_prepare_enable(usp->clk); + int ret; + ret = clk_prepare_enable(usp->clk); + if (ret) { + dev_err(dev, "clk_enable failed: %d\n", ret); + return ret; + } + sirf_usp_i2s_init(usp); + return 0; } -#endif #ifdef CONFIG_PM_SLEEP static int sirf_usp_pcm_suspend(struct device *dev) @@ -369,6 +361,11 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) } pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sirf_usp_pcm_runtime_resume(&pdev->dev); + if (ret) + return ret; + } ret = devm_snd_soc_register_component(&pdev->dev, &sirf_usp_component, &sirf_usp_pcm_dai, 1); @@ -381,7 +378,10 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) static int sirf_usp_pcm_remove(struct platform_device *pdev) { - pm_runtime_disable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) + sirf_usp_pcm_runtime_suspend(&pdev->dev); + else + pm_runtime_disable(&pdev->dev); return 0; } -- 1.9.3 Member of the CSR plc group of companies. 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[alsa-devel] [PATCH] ASoC: tas2552: Return proper error for probe error paths
by Axel Lin 25 Jul '14

25 Jul '14

Return error if devm_regulator_bulk_get() or snd_soc_register_codec() fails. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index f0760af..05f3737 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -480,8 +480,10 @@ static int tas2552_probe(struct i2c_client *client, ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), data->supplies); - if (ret != 0) + if (ret != 0) { dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } pm_runtime_set_active(&client->dev); pm_runtime_set_autosuspend_delay(&client->dev, 1000); @@ -498,7 +500,7 @@ static int tas2552_probe(struct i2c_client *client, if (ret < 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); - return 0; + return ret; } static int tas2552_i2c_remove(struct i2c_client *client) -- 1.9.1

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[alsa-devel] [PATCH 0/3] ASoC: fsl_sai: Fix some issues in fsl_sai_trigger()
by Nicolin Chen 25 Jul '14

25 Jul '14

The series of patches focus on issue fix inside fsl_sai_trigger(). Nicolin Chen (3): ASoC: fsl_sai: Reduce race condition during TE/RE enabling ASoC: fsl_sai: Don't reset FIFO until TE/RE bit is unset ASoC: fsl_sai: Improve enable flow in fsl_sai_trigger() sound/soc/fsl/fsl_sai.c | 40 +++++++++++++++++++++++----------------- 1 file changed, 23 insertions(+), 17 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH] ASoC: rsnd: fixup dai remove callback operation
by Kuninori Morimoto 25 Jul '14

25 Jul '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd driver is using SSI/SRC/DVC which are using "mod" base operation. These "mod" are supporting "probe" and "remove" callbacks. Current rsnd_probe should call "remove" if "probe" was failed, since "probe" might be having DMAEngine handle. Some mod's "remove" callback might be called without calling "probe", but it is no problem. because "remove" do nothing in such case. So, all mod's "remove" should be called when error case of rsnd_probe() and rsnd_remove(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 907d480..1b4d8700 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1041,11 +1041,11 @@ static int rsnd_probe(struct platform_device *pdev) for_each_rsnd_dai(rdai, priv, i) { ret = rsnd_dai_call(probe, &rdai->playback, rdai); if (ret) - return ret; + goto exit_snd_probe; ret = rsnd_dai_call(probe, &rdai->capture, rdai); if (ret) - return ret; + goto exit_snd_probe; } /* @@ -1073,6 +1073,11 @@ static int rsnd_probe(struct platform_device *pdev) exit_snd_soc: snd_soc_unregister_platform(dev); +exit_snd_probe: + for_each_rsnd_dai(rdai, priv, i) { + rsnd_dai_call(remove, &rdai->playback, rdai); + rsnd_dai_call(remove, &rdai->capture, rdai); + } return ret; } @@ -1081,21 +1086,16 @@ static int rsnd_remove(struct platform_device *pdev) { struct rsnd_priv *priv = dev_get_drvdata(&pdev->dev); struct rsnd_dai *rdai; - int ret, i; + int ret = 0, i; pm_runtime_disable(&pdev->dev); for_each_rsnd_dai(rdai, priv, i) { - ret = rsnd_dai_call(remove, &rdai->playback, rdai); - if (ret) - return ret; - - ret = rsnd_dai_call(remove, &rdai->capture, rdai); - if (ret) - return ret; + ret |= rsnd_dai_call(remove, &rdai->playback, rdai); + ret |= rsnd_dai_call(remove, &rdai->capture, rdai); } - return 0; + return ret; } static struct platform_driver rsnd_driver = { -- 1.7.9.5

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic31xx: Add dependency for I2C and select REGMAP_I2C in Kconfig
by Peter Ujfalusi 25 Jul '14

25 Jul '14

The codec can be configured via I2C and using regmap_i2c. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1090135de0ba..9fb6fb210251 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -565,6 +565,8 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC31XX tristate + depends on I2C + select REGMAP_I2C config SND_SOC_TLV320AIC32X4 tristate -- 2.0.2

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[alsa-devel] [PATCH 0/3 v4] arizona: Improvements to codec DVFS control
by Charles Keepax 25 Jul '14

25 Jul '14

Hi, Respin of Richard's DVFS series, from me since he is on holiday. I have based these on Mark's ASoC for-next branch as there a few changes in there to the same parts of code. But let me know if you would rather have a series based on Lee's tree instead. This set of changes cleans up the way the drivers handle the control of internal subsystem clocks for codecs with DVFS. This previously relied on the clock control happening as a side-effect of changing the regulator voltage, rather than explicitly, which assumed that you were using the built-in regulator. Also DVFS control was being done from the ADSP driver despite that driver supposedly being codec- agnostic. There are other cases on some codecs where DVFS control is needed so this sequence of patches cleans up the way DVFS is implemented and adds the extra case of DVFS boost for higher sample rates on WM8997 and WM5102 Changes since v3: - Changed error messages from raising/lowering DCVDD Changes since v2: - Correct some small issues (wrong printf specifier, clarity on PRE_PMD/POST_PMD event) - Added warning for calling arizona_dvfs_down with flags that are not enabled - Squashed patches 2 and 4, so the SUBSYS bit is not set from two seperate places Changes since v1: - Changed mask variable to flags to be more consistent with its usage - Fixed minor tabbing issue - Adding missing SND_SOC_DAPM_PRE_PMD event flag in WM_ADSP2_E Richard Fitzgerald (3): mfd: arizona: Export function to control subsystem DVFS ASoC: wm_adsp: Move DVFS control into codec driver ASoC: arizona: Add DVFS handling for sample rate control drivers/mfd/arizona-core.c | 89 ++++++++++++++++++++++++++++++++++++++ drivers/regulator/arizona-ldo1.c | 5 -- include/linux/mfd/arizona/core.h | 12 +++++ sound/soc/codecs/arizona.c | 22 +++++++++- sound/soc/codecs/wm5102.c | 47 +++++++++++++++++++- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm_adsp.c | 73 +------------------------------ sound/soc/codecs/wm_adsp.h | 13 +++--- 8 files changed, 176 insertions(+), 87 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH 1/2] ALSA: hda - add PCI IDs for Intel Braswell
by libin.yang＠intel.com 25 Jul '14

25 Jul '14

From: Libin Yang <libin.yang(a)intel.com> Add HD Audio Device PCI ID for the Intel Braswell platform. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 387fc73..5db1948 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2017,6 +2017,9 @@ static const struct pci_device_id azx_ids[] = { /* BayTrail */ { PCI_DEVICE(0x8086, 0x0f04), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, + /* Braswell */ + { PCI_DEVICE(0x8086, 0x2284), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.9.1

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[alsa-devel] [PATCH v5 0/2] Add Freescale ASRC driver
by Nicolin Chen 25 Jul '14

25 Jul '14

This series of patches add Freescale ASRC module driver support along with the extra request in imx-sdma structure required by SDMA Device to Device script. The previous version has been in the maillist for nearly six months without any comment and reply. So I decide to drop the SDMA part's change since this series focus on ASRC support (Later, we can send SDMA part separately.) so that ASRC driver may get upstream first. Changelog v5: (All follow Varka's suggestions) * PATCH-2: Added missing '\n' to pair_err() calls. * PATCH-2: Return the final regmap_write() for fsl_asrc_init(). * PATCH-2: Dropped fsl_asrc_devtype and use simple ids behind fsl_asrc_pm. * PATCH-2: Use of_device_is_compatible() to differ compatible directly. * PATCH-2: Fixed indentation for devm_regmap_init_mmio_clk() call. v4: (All follow Varka's suggestions) * PATCH-2: Added '\n' to dev_dbg() inside fsl_asrc_isr(). * PATCH-2: Added indentations to DT binding document. * PATCH-2: Redefined fsl_asrc_set_watermarks() as void type. * PATCH-2: Put fsl_asrc_isr() and fsl_asrc_ids before probe() v3: * PATCH-1: Added an Acked-by from Shawn Guo * PATCH-2: Dropped THIS_MODULE as Tobias suggests * PATCH-2: Use dma_coerce_mask_and_coherent() in fsl_asrc_dma_pcm_new() * PATCH-2: Added substream check in fsl_asrc_dma_pcm_new() v2: * PATCH-1: Dropped the change to SDMA driver. * PATCH-2: Dropped useless member in private data structures. * PATCH-2: Refined some comments in fsl_asrc_dma.c driver. * PATCH-2: Refined commit comments. Nicolin Chen (2): ARM: imx: Add the secondary request into the structure for imx-sdma ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers .../devicetree/bindings/sound/fsl,asrc.txt | 60 ++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 992 +++++++++++++++++++++ sound/soc/fsl/fsl_asrc.h | 461 ++++++++++ sound/soc/fsl/fsl_asrc_dma.c | 386 ++++++++ 7 files changed, 1911 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 sound/soc/fsl/fsl_asrc.c create mode 100644 sound/soc/fsl/fsl_asrc.h create mode 100644 sound/soc/fsl/fsl_asrc_dma.c -- 1.8.4

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[alsa-devel] snd_seq_drain_output returns ENOENT when event filter is used
by Wladimir 25 Jul '14

25 Jul '14

Hello, I noticed that writing ALSA sequencer events sometimes fails. When I debugged further, the cause seems to be that the receiving side (Yoshi softsynth) has set an event filter using snd_seq_client_info_event_filter_add, and the sending application sends events that are filtered out. >From ptrace: write(3, "\10\0\0\375\0\0\0\0\0\0\0\0\0\0\376\375\t) \0\0\0\0\0\0\0\0\0", 28) = -1 ENOENT (No such file or directory) I think it is wrong that the sender gets errors in this case. It can send events all day succesfuly as long as the port is not connected (meaning they all get dropped), but when you connect it to a filtering client (and some get dropped), it fails. Details: OS: Ubuntu 14.04 Kernel: 3.13.0-30-generic libasound2 version: 1.0.27.2-3ubuntu7 Wladimir

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[alsa-devel] [PATCH v4 0/2] Add Freescale ASRC driver
by Nicolin Chen 25 Jul '14

25 Jul '14

This series of patches add Freescale ASRC module driver support along with the extra request in imx-sdma structure required by SDMA Device to Device script. The previous version has been in the maillist for nearly six months without any comment and reply. So I decide to drop the SDMA part's change since this series focus on ASRC support (Later, we can send SDMA part separately.) so that ASRC driver may get upstream first. Changelog v4: (All follow Varka's suggestions) * PATCH-2: Added '\n' to dev_dbg() inside fsl_asrc_isr(). * PATCH-2: Added indentations to DT binding document. * PATCH-2: Redefined fsl_asrc_set_watermarks() as void type. * PATCH-2: Put fsl_asrc_isr() and fsl_asrc_ids before probe() v3: * PATCH-1: Added an Acked-by from Shawn Guo * PATCH-2: Dropped THIS_MODULE as Tobias suggests * PATCH-2: Use dma_coerce_mask_and_coherent() in fsl_asrc_dma_pcm_new() * PATCH-2: Added substream check in fsl_asrc_dma_pcm_new() v2: * PATCH-1: Dropped the change to SDMA driver. * PATCH-2: Dropped useless member in private data structures. * PATCH-2: Refined some comments in fsl_asrc_dma.c driver. * PATCH-2: Refined commit comments. Nicolin Chen (2): ARM: imx: Add the secondary request into the structure for imx-sdma ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers .../devicetree/bindings/sound/fsl,asrc.txt | 60 ++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 1032 ++++++++++++++++++++ sound/soc/fsl/fsl_asrc.h | 461 +++++++++ sound/soc/fsl/fsl_asrc_dma.c | 386 ++++++++ 7 files changed, 1951 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 sound/soc/fsl/fsl_asrc.c create mode 100644 sound/soc/fsl/fsl_asrc.h create mode 100644 sound/soc/fsl/fsl_asrc_dma.c -- 1.8.4

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