May 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl_sai: Allow the SAI driver to turn on SAI_MCLK
by Fabio Estevam 04 May '16

04 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> On mx6ul the General Purpose Register 1 (GPR1) contains the following bits for enabling the output of the SAI MCLKs: SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR Introduce "gpr" and "fsl,sai-enable-mclk" optional properties to allow the enablement of the SAI_MCLK outputs. Tested on a imx6ul-evk board. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- .../devicetree/bindings/sound/fsl-sai.txt | 8 +++++ include/linux/mfd/syscon/imx6q-iomuxc-gpr.h | 3 ++ sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++ 3 files changed, 48 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d7..86755bb 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -48,6 +48,14 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties (for mx6ul): + + - gpr : The phandle to the General Purpose Register (GPR) + node. + + - fsl,sai-enable-mclk : This is a boolean property. If present, indicates + that SAI will output the SAI MCLK clock. + Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h index 238c8db..401f97e 100644 --- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h +++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h @@ -447,5 +447,8 @@ #define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18) #define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17) #define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17) +#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19) +#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20) +#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21) #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..311325e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> #include "fsl_sai.h" #include "imx-pcm.h" @@ -785,11 +787,14 @@ static const struct regmap_config fsl_sai_regmap_config = { static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device_node *gpr_np = of_parse_phandle(np, "gpr", 0); struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + u32 index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -877,6 +882,38 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-enable-mclk", NULL)) { + ret = of_property_read_u32(np, "sai-index", &index); + if (ret) { + dev_err(&pdev->dev, "could not read sai-index\n"); + return ret; + } + + gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "could not find gpr node\n"); + return PTR_ERR(gpr); + } + + switch (index) { + case 1: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI1_MCLK_DIR, + IMX6UL_GPR1_SAI1_MCLK_DIR); + break; + case 2: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI2_MCLK_DIR, + IMX6UL_GPR1_SAI2_MCLK_DIR); + break; + case 3: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI3_MCLK_DIR, + IMX6UL_GPR1_SAI3_MCLK_DIR); + break; + } + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; -- 1.9.1

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[alsa-devel] [PATCH 0/2 v6] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 04 May '16

04 May '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. Changes v5 -> v6 * Use SNDRV_DMA_TYPE_CONTINUOUS Changes v4 -> v5 * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh(a)kernel.org> Cc: Alexey Brodkin <abrodkin(a)synopsys.com> Cc: linux-snps-arc(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Jose Abreu (2): ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../devicetree/bindings/sound/designware-i2s.txt | 9 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 71 +++++++ sound/soc/dwc/designware_i2s.c | 94 ++++++--- sound/soc/dwc/designware_pcm.c | 228 +++++++++++++++++++++ 6 files changed, 383 insertions(+), 29 deletions(-) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables"
by John Keeping 04 May '16

04 May '16

This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38. It turns out that the commit that made these variables unused is wrong so we're about to revert it. Bring back the variables in prepration. Signed-off-by: John Keeping <john(a)metanate.com> --- v2: Add "ASoC: rockchip: " prefix to subject sound/soc/rockchip/rockchip_i2s.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..34743ec61c49 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -77,7 +84,11 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START, I2S_XFER_TXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); @@ -115,7 +126,11 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_RXS_START, I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); -- 2.8.0.rc4.238.g874082a

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[alsa-devel] [PATCH v6 0/4] Propagate errors out from compressed streams
by Charles Keepax 04 May '16

04 May '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v5: - Add a debug print to note which state we transition to in snd_compr_stop_error It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 81 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 +++++- sound/soc/soc-compress.c | 5 ++- 4 files changed, 93 insertions(+), 9 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 04 May '16

04 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

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[alsa-devel] Get dB value using softvol plugin
by Jörg Krause 04 May '16

04 May '16

Hi, I configured my softvol plugin with a min_dB value of -6.0. However, if I store the state with alsactl store, the comment section shows a dbmin value of 0. I figured out that the softvol plugin adds the tlv info when it adds the user ctl and stores the db values in snd_pcm_softvol_t. Why differ the values of the tlv and the values of snd_pcm_softvol_t? My goal is to get information of the current dB value used by the softvol plugin, e.g. -6.0dB for a raw volume value of 255 and -51.0dB for 0. However, I only get values from +0 to +51dB using snd_ctl_convert_to_dB(). Do I miss something? Note, that I started from a clean asound.state. --- The state produced by alsactl: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } --- The output of aplay -D softvol -v: Playing WAVE '/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Soft volume PCM Control: Master min_dB: -57 max_dB: -6 resolution: 256 --- My asound.conf: pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0" } control { name "Master" card 0 } min_dB -57.0 max_dB -6.0 } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } --- Best regards Jörg Krause

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Re: [alsa-devel] [Alsa-user] How do I run arecord on raw digital mic buffer?
by Clemens Ladisch 04 May '16

04 May '16

Rob Nertney wrote: >>> is the best method to build a sound card and a capture PCM? >> >> As compared to what? Any other method will not allow your hardware to >> be used for audio recording. > > As compared to my driver configuring the DMA to capture the data into > a known buffer, and then directly accessing the buffer to wrap it in > a WAV format. I should have rephrased my initial question as, "my goal > is to simply wrap multichannel digital mic input in WAV format; is > ALSA/arecord the most efficient way to do this?" If you bypass ALSA, you have to write all the software yourself, and you will not be able to use your hardware for anything else. > I should make my buffer at least 2x the size of a period. Yes. (You don't need periods if you don't use interrupts to wait for data being available.) > I have a buffer ready, under which function does it get passed back to > the ALSA framework? Normally, ALSA knows about the buffer, and the driver just reports the current position. > * dma completes/fires period-based IRQ > * <<question>> buffer is transferred in function X (?) to Y (?) When a period is finished, you must call snd_pcm_period_elapsed(). ALSA will then call your .pointer callback to know where in the buffer the hardware has finished writing. (Please not that .pointer can also be called at other times.) Regards, Clemens

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[alsa-devel] [patch] ASoC: hdac_hdmi: Potential NULL deref in hdac_hdmi_get_spk_alloc()
by Dan Carpenter 03 May '16

03 May '16

We intended || here instead of &&. The original code potentially leads to a NULL dereference. Fixes: 2889099eb8cd ('ASoC: hdac_hdmi: Register chmap controls and ops') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3043120..7e7ff1b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1609,7 +1609,7 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_pin *pin = pcm->pin; - if (!pin && !pin->eld.eld_valid) + if (!pin || !pin->eld.eld_valid) return 0; return pin->eld.info.spk_alloc;

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Re: [alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by sugar 03 May '16

03 May '16

Hi John, On 4/30/2016 15:00, John Keeping Wrote: > Hi Enric, > > On Fri, Apr 29, 2016 at 04:59:27PM +0200, Enric Balletbo Serra wrote: >> 2015-12-09 11:32 GMT+01:00 John Keeping <john(a)metanate.com>: >>> If we only clear the tx/rx state when both are disabled it is not >>> possible to start/stop one multiple times while the other is running. >>> Since the two are independently controlled, treat them as such and >>> remove the false dependency between capture and playback. >>> >>> Signed-off-by: John Keeping <john(a)metanate.com> >>> --- >>> sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- >>> 1 file changed, 32 insertions(+), 40 deletions(-) >>> >>> diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c >>> index 83b1b9c..acc6225 100644 >>> --- a/sound/soc/rockchip/rockchip_i2s.c >>> +++ b/sound/soc/rockchip/rockchip_i2s.c >>> @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >>> I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); >>> >>> regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >>> + I2S_XFER_TXS_START, >>> + I2S_XFER_TXS_START); >>> >>> i2s->tx_start = true; >>> } else { >>> @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >>> regmap_update_bits(i2s->regmap, I2S_DMACR, >>> I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); >>> >>> - if (!i2s->rx_start) { >>> - regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | >>> - I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_STOP | >>> - I2S_XFER_RXS_STOP); >>> + regmap_update_bits(i2s->regmap, I2S_XFER, >>> + I2S_XFER_TXS_START, >>> + I2S_XFER_TXS_STOP); >>> >>> - regmap_update_bits(i2s->regmap, I2S_CLR, >>> - I2S_CLR_TXC | I2S_CLR_RXC, >>> - I2S_CLR_TXC | I2S_CLR_RXC); >>> + regmap_update_bits(i2s->regmap, I2S_CLR, >>> + I2S_CLR_TXC, >>> + I2S_CLR_TXC); >>> >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> >>> - /* Should wait for clear operation to finish */ >>> - while (val) { >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> - retry--; >>> - if (!retry) { >>> - dev_warn(i2s->dev, "fail to clear\n"); >>> - break; >>> - } >>> + /* Should wait for clear operation to finish */ >>> + while (val & I2S_CLR_TXC) { >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> + retry--; >>> + if (!retry) { >>> + dev_warn(i2s->dev, "fail to clear\n"); >>> + break; >>> } >>> } >>> } >>> @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >>> I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); >>> >>> regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >>> + I2S_XFER_RXS_START, >>> + I2S_XFER_RXS_START); >>> >>> i2s->rx_start = true; >>> } else { >>> @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >>> regmap_update_bits(i2s->regmap, I2S_DMACR, >>> I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); >>> >>> - if (!i2s->tx_start) { >>> - regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | >>> - I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_STOP | >>> - I2S_XFER_RXS_STOP); >>> + regmap_update_bits(i2s->regmap, I2S_XFER, >>> + I2S_XFER_RXS_START, >>> + I2S_XFER_RXS_STOP); >>> >>> - regmap_update_bits(i2s->regmap, I2S_CLR, >>> - I2S_CLR_TXC | I2S_CLR_RXC, >>> - I2S_CLR_TXC | I2S_CLR_RXC); >>> + regmap_update_bits(i2s->regmap, I2S_CLR, >>> + I2S_CLR_RXC, >>> + I2S_CLR_RXC); >>> >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> >>> - /* Should wait for clear operation to finish */ >>> - while (val) { >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> - retry--; >>> - if (!retry) { >>> - dev_warn(i2s->dev, "fail to clear\n"); >>> - break; >>> - } >>> + /* Should wait for clear operation to finish */ >>> + while (val & I2S_CLR_RXC) { >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> + retry--; >>> + if (!retry) { >>> + dev_warn(i2s->dev, "fail to clear\n"); >>> + break; >>> } >>> } >>> } >>> -- >>> 2.6.3.462.gbe2c914 >> >> Using my Veyron Jerry Chromebook wiht latest kernel I found that the >> speakers doesn't work. Bisect points to this patch as the offending >> commit. However your changes looks reasonable the fact is that >> reverting the patch makes the audio work again on my device. I need to >> dig a bit more into the issue, but meanwhile, any idea on what is >> happening ? Can I ask which device did you test? > > I tested this on a Radxa Rock2 square. This change fixed an issue I > found when running separate processes for playback and recording (I was > testing with arecord and speaker-test IIRC) but I don't think that > should be relevant. I just read through the patch again and I can't see > anything obviously wrong so I'm afraid I don't have any idea what's > causing the problem you're seeing. > Do you have test playback/capture at the same time? I think you may reproduce this issue if you do as follows: step 1: tinyplay test.wav -D 0 -d 0 -p 1024 -n 4 & step 2: tinycap r.wav -D 0 -d 0 -p 1024 -n 4 & you will find that there is no music now. because of this problem, we enable tx/rx at the same time to WA this issue. May I know what's the issue that you met before? > > John > > _______________________________________________ > Linux-rockchip mailing list > Linux-rockchip(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-rockchip > > >

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[alsa-devel] [PATCH v3 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 03 May '16

03 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 81 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 70 insertions(+), 11 deletions(-) -- 1.9.1

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