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

[alsa-devel] [PATCH] ASoC: wm9713: fix regmap free path
by Robert Jarzmik 20 Feb '16

20 Feb '16

In the conversion to regmap, I assumed that the 2 following functions was working symetrically: - snd_soc_codec_init_regmap() - snd_soc_codec_exit_regmap(codec) As a mater of fact with the current code the regmap is freed twice because of the devm_() call: (mutex_lock) from (debugfs_remove_recursive+0x50/0x1d0) (debugfs_remove_recursive) from (regmap_debugfs_exit+0x1c/0xd4) (regmap_debugfs_exit) from (regmap_exit+0x28/0xc8) (regmap_exit) from (release_nodes+0x18c/0x204) (release_nodes) from (device_release+0x18/0x90) (device_release) from (kobject_release+0x90/0x1bc) (kobject_release) from (wm9713_soc_remove+0x1c/0x24) (wm9713_soc_remove) from (soc_remove_component+0x50/0x7c) (soc_remove_component) from (soc_remove_dai_links+0x118/0x228) (soc_remove_dai_links) from (snd_soc_register_card+0x4e4/0xdd4) (snd_soc_register_card) from (devm_snd_soc_register_card+0x34/0x70) Fix this by removing the doubled regmap free. Fixes: 700dadfefc3d ASoC: wm9713: convert to regmap Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/soc/codecs/wm9713.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 79e143625ac3..0242b7dcc1bb 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1230,7 +1230,6 @@ static int wm9713_soc_remove(struct snd_soc_codec *codec) { struct wm9713_priv *wm9713 = snd_soc_codec_get_drvdata(codec); - snd_soc_codec_exit_regmap(codec); snd_soc_free_ac97_codec(wm9713->ac97); return 0; } @@ -1272,3 +1271,2 @@ static int wm9713_remove(struct platform_device *pdev) return 0; } -- 2.1.4

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[alsa-devel] [PATCH 0/11] ASoC: rsnd: cleanup SSI multi mode and sequence
by Kuninori Morimoto 20 Feb '16

20 Feb '16

Hi Mark Current rsnd driver is supporting many features, and now we are confirming these register settings sequence. These patches fixup not good sequence (and bug) Kuninori Morimoto (11): ASoC: rsnd: rename RSND_DVC_CHANNELS to RSND_MAX_CHANNELS ASoC: rsnd: setup SRC_ROUTE_MODE0 before SRC_SRCIR ASoC: rsnd: indicates status failed SSI ASoC: rsnd: SSI function parameter uses "mod" instead of "ssi" ASoC: rsnd: fixup forever loop bug on SSI ASoC: rsnd: add missing .irq callback for DMA ASoC: rsnd: tidyup SSI init/start sequence ASoC: rsnd: Parent SSI attach is not needed if not clock master ASoC: rsnd: move rsnd_ssi_irq() position ASoC: rsnd: judge multi SSI in runtime ASoC: rsnd: judge work SSI in runtime sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/dvc.c | 6 ++--- sound/soc/sh/rcar/rsnd.h | 6 ++--- sound/soc/sh/rcar/src.c | 3 ++- sound/soc/sh/rcar/ssi.c | 166 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------- sound/soc/sh/rcar/ssiu.c | 6 ++--- 6 files changed, 126 insertions(+), 65 deletions(-)

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[alsa-devel] [PATCH v5 00/14] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 20 Feb '16

20 Feb '16

This patch series adds DP audio and hotplug notification support. In skylake platform the DP is on a different port which is not enabled by default. A vendor widget is programmed to enable playback on all ports. This also enables all the converters. Codec Dais, widgets, graph are now created dynamically based on the nodes enumerated, thus removing the hardcoding. Mux controls are used to establish routing stream to a specific port. For DP audio support, infoframe needs to be different. Based on ELD queried we either pack HDMI or DP infoframe. Also with this series, Jack notification support is added and Jack event is reported to userspace as is done in case of legacy driver. Few fixes are added to check for possible memory leak, reconfiguring register during resume from D3 etc. The 8th patch is adding a small macro for getting connection type in drm header. It is reviewed by drm folks, CCed them. Please merge it through sound trees. Changes in v5: - Added more comments and reworded some commit messages - Jack reporting support to user space added - Fix to check for possible memory leak in hw_params - Fix not to fail in dai startup to make userland APIs happy Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Jeeja KP (2): ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Subhransu S. Prusty (11): ASoC: hdac_hdmi: Add hotplug notification and read eld ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume include/drm/drm_edid.h | 12 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1197 +++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/hdac_hdmi.h | 6 + 4 files changed, 1094 insertions(+), 122 deletions(-) create mode 100644 sound/soc/codecs/hdac_hdmi.h -- 1.9.1

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[alsa-devel] [PATCH v1 00/14] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. Most of these patches are acked by Kenneth. These patches are tested on DB410C with Headset Mic, secondary mic. Thanks, srini Changes since RFC: - made patches bisectable, reported by 0day testing - merged two patches as suggested by Kenneth - used corect device for snd_dma alloc/free apis, spotted by Kenneth - renamed LPAIF_I2SCTL_MICMODE_QUAD02 to LPAIF_I2SCTL_MICMODE_QUAD23, spotted by Kenneth. - Fixed minor issues suggested by Kenneth. Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 +++++++++++++++++------ sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 243 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 390 insertions(+), 177 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 00/14] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. All these patches are acked by Kenneth. These patches are tested on DB410C with Headset Mic, secondary mic. Thanks, srini Changes since v1/v2: - Fixed patch version number. - renamed rdma_port to dma_port in patch 3, as suggested by Kenneth. - fixed correct device for snd_dma_alloc/free as suggested by Kenneth in patch 13. Changes since RFC: - made patches bisectable, reported by 0day testing - merged two patches as suggested by Kenneth - used corect device for snd_dma alloc/free apis, spotted by Kenneth - renamed LPAIF_I2SCTL_MICMODE_QUAD02 to LPAIF_I2SCTL_MICMODE_QUAD23, spotted by Kenneth. - Fixed minor issues suggested by Kenneth. Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 ++++++++++++++++++----- sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 244 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 391 insertions(+), 177 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH RFC 00/15] ASoC: qcom: Add capture support.
by Srinivas Kandagatla 20 Feb '16

20 Feb '16

LPASS IP on QCOM SOC supports both Playback and capture via I2S, but this feature is missing in existing code. This patchset aims at adding capture support to lpass IP. First few patches in this series does cleanup the driver to make easy to add capture support. These patches are tested on DB410C with Headset Mic. Thanks, srini Srinivas Kandagatla (15): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma dma channel start ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: apq8016: add wrdma support ASoC: qcom: add wrdma register definations ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: apq8016-sbc: add mic support ASoC: qcom: add mic support sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +++-- sound/soc/qcom/lpass-cpu.c | 146 +++++++++++++++++------ sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 118 +++++++++++++------ sound/soc/qcom/lpass-platform.c | 243 +++++++++++++++++++++++---------------- sound/soc/qcom/lpass.h | 10 +- 7 files changed, 392 insertions(+), 177 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: cht_bsw_rt5645: Enable jack detection
by Carlo Caione 20 Feb '16

20 Feb '16

From: Carlo Caione <carlo(a)endlessm.com> Add missing DAPM pins and enable jack detection on those pins for Cherrytrail and Braswell. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index e6cf800..c7af271 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -147,6 +147,17 @@ static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Ext Spk"), }; +static struct snd_soc_jack_pin cht_bsw_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + static int cht_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -202,9 +213,9 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) else jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE; - ret = snd_soc_card_jack_new(runtime->card, "Headset Jack", + ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, &ctx->jack, - NULL, 0); + cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); if (ret) { dev_err(runtime->dev, "Headset jack creation failed %d\n", ret); return ret; -- 1.9.1

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[alsa-devel] [PATCH v4 0/2] ASoC: Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 20 Feb '16

20 Feb '16

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver. For now just the SPDIF transmitter has been tested on a Mele A2000(A10) and a Itead Ibox(A20). In order for this patch set to be functional we require an audio clock patch which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK --- Changes since v3: - removed code required for spdif capture - added disable/enable of apb clk to suspend/resume. - removed unnecessary dt checks in probe - fixed error conditions in probe/resume - modified pm_runtime functionality(confirmed driver worked with PM disabled). - removed .owner from platform driver - renamed bindings file Changes since v2: - removed sunxi-machine-spdif and replaced with simple audio card - removed untested compatibilities from device tree documentation - added pm_runtime and remove shutdown - removed pll2 as it's the parent of the spdif clock - rename clk to spdif_clk - removed enabling of mclk output - removed interrupts status being cleared as it's not being used - use default fifo settings for now - fixed alignments for wrapped lines Changes since v1: - Moved sunxi-machine-spdif.c to separate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into separate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. Marcus Cooper (2): dt-bindings: sunxi :add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block .../bindings/sound/sunxi,sun4i-spdif.txt | 39 ++ sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 561 +++++++++++++++++++++ 4 files changed, 609 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c -- 2.7.0

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[alsa-devel] [PATCH] SoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 20 Feb '16

20 Feb '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index e011311..6f88d1c 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -30,6 +30,7 @@ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_DEEP_BUFFER, MERR_DPCM_COMPR, + MERR_DPCM_BE, }; /* define a bit for each mixer input */ diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index e6cf800..89b4c032 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -282,7 +282,7 @@ static struct snd_soc_dai_link cht_dailink[] = { }, /* CODEC<->CODEC link */ /* back ends */ - { + [MERR_DPCM_BE] = { .name = "SSP2-Codec", .be_id = 1, .cpu_dai_name = "ssp2-port", @@ -357,7 +357,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + card->dai_link[MERR_DPCM_BE].codec_name = kstrdup(codec_name, GFP_KERNEL); snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 1.9.1

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[alsa-devel] [PATCH] fireworks: serialize transactions to update connections at bus reset
by Takashi Sakamoto 20 Feb '16

20 Feb '16

In IEC 61883-1, at bus-reset, applications can continue isochronous streaming by updating connections. In ALSA fireworks driver, the operation is executed in 'update' handler for bus driver. The connection resources are also changed in process contexts of PCM/MIDI applications. Therefore, bus-reset handling has race condition against connection. Current ALSA fireworks driver has a bug for the condition. This commit fixes the bug, by expand critical section with mutex. As a result, connection updating operation in bus-reset handler and connection changing operation in process context are serialized. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks.c | 3 +++ sound/firewire/fireworks/fireworks_stream.c | 6 ++---- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index d5b19bc..8f27b67 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -301,7 +301,10 @@ static void efw_update(struct fw_unit *unit) struct snd_efw *efw = dev_get_drvdata(&unit->device); snd_efw_transaction_bus_reset(efw->unit); + + mutex_lock(&efw->mutex); snd_efw_stream_update_duplex(efw); + mutex_unlock(&efw->mutex); } static void efw_remove(struct fw_unit *unit) diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 968a40a..425db8d 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -313,12 +313,10 @@ void snd_efw_stream_stop_duplex(struct snd_efw *efw) void snd_efw_stream_update_duplex(struct snd_efw *efw) { - if ((cmp_connection_update(&efw->out_conn) < 0) || - (cmp_connection_update(&efw->in_conn) < 0)) { - mutex_lock(&efw->mutex); + if (cmp_connection_update(&efw->out_conn) < 0 || + cmp_connection_update(&efw->in_conn) < 0) { stop_stream(efw, &efw->rx_stream); stop_stream(efw, &efw->tx_stream); - mutex_unlock(&efw->mutex); } else { amdtp_stream_update(&efw->rx_stream); amdtp_stream_update(&efw->tx_stream); -- 2.5.0

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