January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] wcd9335: Remove unnecessary null checks in wcd9335_set_channel_map
by Nathan Chancellor 31 Jan '19

31 Jan '19

Clang warns: sound/soc/codecs/wcd9335.c:2004:11: warning: address of array 'wcd->rx_chs' will always evaluate to 'true' [-Wpointer-bool-conversion] if (wcd->rx_chs) { ~~ ~~~~~^~~~~~ sound/soc/codecs/wcd9335.c:2012:11: warning: address of array 'wcd->tx_chs' will always evaluate to 'true' [-Wpointer-bool-conversion] if (wcd->tx_chs) { ~~ ~~~~~^~~~~~ 2 warnings generated. Fixes: 20aedafdf492 ("ASoC: wcd9335: add support to wcd9335 codec") Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 3878187bb512..981f88a5f615 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2001,20 +2001,16 @@ static int wcd9335_set_channel_map(struct snd_soc_dai *dai, return -EINVAL; } - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } + wcd->num_rx_port = rx_num; + for (i = 0; i < rx_num; i++) { + wcd->rx_chs[i].ch_num = rx_slot[i]; + INIT_LIST_HEAD(&wcd->rx_chs[i].list); } - if (wcd->tx_chs) { - wcd->num_tx_port = tx_num; - for (i = 0; i < tx_num; i++) { - wcd->tx_chs[i].ch_num = tx_slot[i]; - INIT_LIST_HEAD(&wcd->tx_chs[i].list); - } + wcd->num_tx_port = tx_num; + for (i = 0; i < tx_num; i++) { + wcd->tx_chs[i].ch_num = tx_slot[i]; + INIT_LIST_HEAD(&wcd->tx_chs[i].list); } return 0; -- 2.20.1

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[alsa-devel] auto_mute for early patch issue
by Kailang 31 Jan '19

31 Jan '19

Hi Takashi, Chrome OS use early patch firmware file to load model and use auto_mute =no hint options. I find an issue for early patch issue about hint option auto_mute = no. This will cause all spec->init_hook can't get hp_pin values. For example: alc225_init() alc225_shutup() alc256_init() alc256_shutup() Early patch file hint auto_mute = no equal spec->suppress_auto_mute = 1 It will return function check_auto_mute_availability() in had_generic.c. hp_pin = spec->gen.autocfg.hp_pins[0]; ==> It will get value 0. I think this issue was for a long time. Please see the attach dmesg. One is kill hint auto_mute option. BR, Kailang

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Re: [alsa-devel] linux-next: Tree for Jan 30 (sound/soc/ti/davinci-mcasp.c)
by Randy Dunlap 30 Jan '19

30 Jan '19

On 1/29/19 11:39 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20190129: > on i386: ld: sound/soc/ti/davinci-mcasp.o: in function `davinci_mcasp_probe': davinci-mcasp.c:(.text+0x1d99): undefined reference to `edma_pcm_platform_register' Full randconfig file is attached. PS: Please fix this line in the source file also: * Steve Chen <schen@.mvista.com> (extra '.') -- ~Randy

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[alsa-devel] [PATCH v3 00/14] Sound Open Firmware (SOF) core
by Pierre-Louis Bossart 30 Jan '19

30 Jan '19

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core and utilities. Support for Intel devices is provided as a follow-up series. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Main changes since v2 (end of August): There are honestly too many changes to be listed... The biggest changes are the introduction of an ABI check and runtime_pm, as well as countless improvements coming from product validation. We will provide a revision history starting with the next update. Precisions: Since I've been maintaining the SOF kernel parts for most of 2018, Liam and I mutually agreed that it would be more efficient for me to send the patches upstream to avoid any disconnects. The code in this patchset is directly squashed from the SOF development branch [1], which tracks Mark Brown's for-next branch on a weekly basis and the configurations used for testing are based on the defconfigs at [2] Full disclosure on known limitation and issues (full list at [3]) a) the interaction with ASoC is not always perfect, there are a couple of points where a better solution is desired (use of private data, support for link DMA, etc). The known points are explicitly documented in the code and will be updated. To be clearer, it's our belief that SOF does not cripple ASoC in any way, and that merging this SOF core does not harm others. It's just complicated to get things right, especially on a couple of cases where the topology and DPCM frameworks generate complex flows that very few people understand. b) The get/put methods for controls generate an IPC and possibly a wake-up. This is not optimal and is being fixed. c) The debugfs parts will be updated for runtime_pm parts d) The topology free, firmware release and Hdaudio exit sequences will be updated. It is our intention to fully support module load/unload and free all resources. e) runtime_pm is being hardened and the use of SMART_SUSPEND was suggested. Thank you for reviews and comments, we appreciate the time spent commenting on this large patchset. Thanks in particular to Alan Cox and Andy Shevchenko for their comments on an earlier version. Pierre [1] https://github.com/thesofproject/linux [2] https://github.com/thesofproject/kconfig [3] https://github.com/thesofproject/linux/issues Liam Girdwood (12): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware logger support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add PM support ASoC: SOF: Add Nocodec machine driver support Pierre-Louis Bossart (2): ASoC: SOF: Add xtensa support ASoC: SOF: Add utils include/sound/soc.h | 3 + include/sound/sof.h | 91 + include/sound/sof/control.h | 125 ++ include/sound/sof/dai-intel.h | 175 ++ include/sound/sof/dai.h | 75 + include/sound/sof/header.h | 158 ++ include/sound/sof/info.h | 118 ++ include/sound/sof/pm.h | 48 + include/sound/sof/stream.h | 149 ++ include/sound/sof/topology.h | 256 +++ include/sound/sof/trace.h | 66 + include/sound/sof/xtensa.h | 44 + include/uapi/sound/sof/abi.h | 63 + include/uapi/sound/sof/eq.h | 164 ++ include/uapi/sound/sof/fw.h | 67 + include/uapi/sound/sof/header.h | 27 + include/uapi/sound/sof/manifest.h | 188 ++ include/uapi/sound/sof/tokens.h | 95 + include/uapi/sound/sof/tone.h | 21 + include/uapi/sound/sof/trace.h | 93 + sound/soc/sof/control.c | 421 +++++ sound/soc/sof/core.c | 398 +++++ sound/soc/sof/debug.c | 176 ++ sound/soc/sof/ipc.c | 813 +++++++++ sound/soc/sof/loader.c | 322 ++++ sound/soc/sof/nocodec.c | 81 + sound/soc/sof/ops.c | 200 +++ sound/soc/sof/ops.h | 379 ++++ sound/soc/sof/pcm.c | 755 ++++++++ sound/soc/sof/pm.c | 403 +++++ sound/soc/sof/sof-priv.h | 574 ++++++ sound/soc/sof/topology.c | 2775 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 295 +++ sound/soc/sof/utils.c | 173 ++ sound/soc/sof/xtensa/Kconfig | 3 + sound/soc/sof/xtensa/Makefile | 5 + sound/soc/sof/xtensa/core.c | 152 ++ 37 files changed, 9951 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/sound/sof/control.h create mode 100644 include/sound/sof/dai-intel.h create mode 100644 include/sound/sof/dai.h create mode 100644 include/sound/sof/header.h create mode 100644 include/sound/sof/info.h create mode 100644 include/sound/sof/pm.h create mode 100644 include/sound/sof/stream.h create mode 100644 include/sound/sof/topology.h create mode 100644 include/sound/sof/trace.h create mode 100644 include/sound/sof/xtensa.h create mode 100644 include/uapi/sound/sof/abi.h create mode 100644 include/uapi/sound/sof/eq.h create mode 100644 include/uapi/sound/sof/fw.h create mode 100644 include/uapi/sound/sof/header.h create mode 100644 include/uapi/sound/sof/manifest.h create mode 100644 include/uapi/sound/sof/tokens.h create mode 100644 include/uapi/sound/sof/tone.h create mode 100644 include/uapi/sound/sof/trace.h create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/nocodec.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/pm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c create mode 100644 sound/soc/sof/utils.c create mode 100644 sound/soc/sof/xtensa/Kconfig create mode 100644 sound/soc/sof/xtensa/Makefile create mode 100644 sound/soc/sof/xtensa/core.c -- 2.17.1

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[alsa-devel] [PATCH 1/2] ASoC: pcm512x: Implement the set_bclk_ratio interface
by Dimitris Papavasiliou 30 Jan '19

30 Jan '19

Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external oscillators, to generate 44.1 or 48kHz multiples and are forced to resort to hacks [1] in order to support 24-bit data without ending up with fractional dividers. This patch allows the machine driver to use 32-bit frames for 24-bit data to avoid such issues. Although the datasheet (p. 15) seems to suggest that only a handful of ratios are supported, it's not very explicit about it, so we allow the full range of values supported by the underlying register in the callback, to avoid needlessly rejecting potentially usable configurations. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.h… Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: This is essentially the same patch submitted previously as an RFC[1]. It has been split in two, in order to separate the set_bclk_ratio implementation from the clocking calculation fix and some conservative checking has been added to the argument of the callback. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144213.ht… sound/soc/codecs/pcm512x.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 6cb1653be804..b63d9392bae3 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -55,6 +55,7 @@ struct pcm512x_priv { unsigned long overclock_dsp; int mute; struct mutex mutex; + unsigned int bclk_ratio; }; /* @@ -915,10 +916,15 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, int fssp; int gpio; - lrclk_div = snd_soc_params_to_frame_size(params); - if (lrclk_div == 0) { - dev_err(dev, "No LRCLK?\n"); - return -EINVAL; + if (pcm512x->bclk_ratio > 0) { + lrclk_div = pcm512x->bclk_ratio; + } else { + lrclk_div = snd_soc_params_to_frame_size(params); + + if (lrclk_div == 0) { + dev_err(dev, "No LRCLK?\n"); + return -EINVAL; + } } if (!pcm512x->pll_out) { @@ -1383,6 +1389,19 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + if (ratio > 256) + return -EINVAL; + + pcm512x->bclk_ratio = ratio; + + return 0; +} + static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; @@ -1438,6 +1457,7 @@ static const struct snd_soc_dai_ops pcm512x_dai_ops = { .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, .digital_mute = pcm512x_digital_mute, + .set_bclk_ratio = pcm512x_set_bclk_ratio, }; static struct snd_soc_dai_driver pcm512x_dai = { -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rockchip: fix small typo
by lsh 30 Jan '19

30 Jan '19

codec_dai -> cpu_dai Signed-off-by: lsh <lshua312(a)163.com> --- sound/soc/rockchip/rockchip_rt5645.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 881c324..f88adf0 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -95,14 +95,16 @@ static int rk_aif1_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT); if (ret < 0) { - dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + dev_err(cpu_dai->dev, + "Can't set clock for cpu_dai %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(codec_dai->dev, "Can't set codec clock %d\n", ret); + dev_err(codec_dai->dev, + "Can't set clock for codec_dai %d\n", ret); return ret; } -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda/tegra: enable clock during probe
by Sameer Pujar 30 Jan '19

30 Jan '19

If CONFIG_PM is disabled or runtime PM calls are forbidden, the clocks will not be ON. This could cause issue during probe, where hda init setup is done. This patch checks whether runtime PM is enabled or not. If disabled, clocks are enabled in probe() and disabled in remove() This patch does following minor changes as cleanup, * return code check for pm_runtime_get_sync() to take care of failure and exit gracefully. * In remove path runtime PM is disabled before calling snd_card_free(). * hda_tegra_disable_clocks() is moved out of CONFIG_PM_SLEEP check. * runtime PM callbacks moved out of CONFIG_PM check Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> Reviewed-by: Ravindra Lokhande <rlokhande(a)nvidia.com> Reviewed-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 31 ++++++++++++++++++++++--------- 1 file changed, 22 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c8d18dc..155c2f5 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -219,7 +219,6 @@ static int hda_tegra_enable_clocks(struct hda_tegra *data) return rc; } -#ifdef CONFIG_PM_SLEEP static void hda_tegra_disable_clocks(struct hda_tegra *data) { clk_disable_unprepare(data->hda2hdmi_clk); @@ -227,6 +226,7 @@ static void hda_tegra_disable_clocks(struct hda_tegra *data) clk_disable_unprepare(data->hda_clk); } +#ifdef CONFIG_PM_SLEEP /* * power management */ @@ -257,7 +257,6 @@ static int hda_tegra_resume(struct device *dev) } #endif /* CONFIG_PM_SLEEP */ -#ifdef CONFIG_PM static int hda_tegra_runtime_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); @@ -283,7 +282,7 @@ static int hda_tegra_runtime_resume(struct device *dev) int rc; rc = hda_tegra_enable_clocks(hda); - if (rc != 0) + if (rc) return rc; if (chip && chip->running) { hda_tegra_init(hda); @@ -292,7 +291,6 @@ static int hda_tegra_runtime_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops hda_tegra_pm = { SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) @@ -555,6 +553,13 @@ static int hda_tegra_probe(struct platform_device *pdev) if (!azx_has_pm_runtime(chip)) pm_runtime_forbid(hda->dev); + /* explicit resume if runtime PM is disabled */ + if (!pm_runtime_enabled(hda->dev)) { + err = hda_tegra_runtime_resume(hda->dev); + if (err) + goto out_free; + } + schedule_work(&hda->probe_work); return 0; @@ -571,7 +576,14 @@ static void hda_tegra_probe_work(struct work_struct *work) struct platform_device *pdev = to_platform_device(hda->dev); int err; - pm_runtime_get_sync(hda->dev); + err = pm_runtime_get_sync(hda->dev); + if (err < 0) { + dev_err(hda->dev, + "failed in pm_runtime_get_syc with err = %d\n", + err); + return; + } + err = hda_tegra_first_init(chip, pdev); if (err < 0) goto out_free; @@ -599,12 +611,13 @@ static void hda_tegra_probe_work(struct work_struct *work) static int hda_tegra_remove(struct platform_device *pdev) { - int ret; - - ret = snd_card_free(dev_get_drvdata(&pdev->dev)); pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) { + hda_tegra_runtime_suspend(&pdev->dev); + pm_runtime_set_suspended(&pdev->dev); + } - return ret; + return snd_card_free(dev_get_drvdata(&pdev->dev)); } static void hda_tegra_shutdown(struct platform_device *pdev) -- 2.7.4

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[alsa-devel] [PATCH 0/2] ALSA: pcm: implement the anonymous dup v2
by Jaroslav Kysela 30 Jan '19

30 Jan '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) Cc: Phil Burk <philburk(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the ioctl/mmap filter for the anonymous dup include/sound/pcm.h | 9 +-- include/uapi/sound/asound.h | 12 +++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 ++-- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 153 ++++++++++++++++++++++++++++++++++++++++++-- 6 files changed, 171 insertions(+), 19 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH] ALSA: pcm: Drop unused snd_pcm_substream.file field
by Takashi Iwai 30 Jan '19

30 Jan '19

It's assigned but nowhere used. Let's remove it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 1 - sound/core/oss/pcm_oss.c | 1 - sound/core/pcm_native.c | 4 +--- 3 files changed, 1 insertion(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2c30c1ad1b0d..a20d3a48df00 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -470,7 +470,6 @@ struct snd_pcm_substream { struct snd_pcm_group self_group; /* fake group for non linked substream (with substream lock inside) */ struct snd_pcm_group *group; /* pointer to current group */ /* -- assigned files -- */ - void *file; int ref_count; atomic_t mmap_count; unsigned int f_flags; diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 467039b342b5..d5b0d7ba83c4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2427,7 +2427,6 @@ static int snd_pcm_oss_open_file(struct file *file, } pcm_oss_file->streams[idx] = substream; - substream->file = pcm_oss_file; snd_pcm_oss_init_substream(substream, &setup[idx], minor); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 26afb6b0889a..63640d3af9db 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2452,10 +2452,8 @@ static int snd_pcm_open_file(struct file *file, return -ENOMEM; } pcm_file->substream = substream; - if (substream->ref_count == 1) { - substream->file = pcm_file; + if (substream->ref_count == 1) substream->pcm_release = pcm_release_private; - } file->private_data = pcm_file; return 0; -- 2.16.4

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[alsa-devel] [PATCH 0/3] ALSA: pcm: implement the anonymous dup
by Jaroslav Kysela 30 Jan '19

30 Jan '19

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup The filtering might be refined. It depends on the real requirements. Perhaps, we may create more ioctl groups. Any comments are more than welcome. Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (3): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: remove the file member from struct pcm ALSA: pcm: implement the ioctl/mmap filter for the anonymous dup include/sound/pcm.h | 10 +-- include/uapi/sound/asound.h | 12 +++- sound/core/oss/pcm_oss.c | 3 +- sound/core/pcm.c | 48 +++++++++----- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 154 +++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 195 insertions(+), 33 deletions(-) -- 2.13.6

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