July 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v5 0/6] Add dw_hdmi i2s audio support
by Yakir Yang 25 Aug '15

25 Aug '15

As Russell have send an series patches for dw_hdmi audio, and most important it works well on rockchip platform. Cause some patches have been achieve in Russell's series, so I abondon some patches in this verison. Here are the abondon patches that already be achieved: * drm: bridge/dw_hdmi: adjust n/cts setting order * drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual * drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() * drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks And I also abondon some interrupt patches that is not a common bug in dw_hdmi driver, but is a rockchip platform bug which HDMI registers would reset by CRU module after system suspend/resume (in this case HPD interrupt mask would reset, then driver can't respond hot plug event after system resume). Here are the abondon patches that used to "fix" dw_hdmi interrupt bug: * drm: bridge/dw_hdmi: add irq control to suspend/resume * drm: bridge/dw_hdmi: wrap irq control in fucntions * drm: rockchip/dw_hdmi_rockchip: add resume/suspend support Here are the leave list that I have make in my series: - add n/cts combinations for more display modes - enable audio support when sink device is HDMI monitor and has audio - rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h - add dw_hdmi i2s audio driver - add rockchip_hdmi_audio sound card driver - add dt-bindings for documentation for rockchip_hdmi_audio driver Changes in v5: - make more words in commit message. - Take Russell suggest that create "sink_has_audio" and "sink_is_hdmi" in struct hdmi, and keep vmode's mdvi changeless. - Config fc_invidconf to HDMI mode when sink device is HDMI monitor. - Take Mark Brown suggest that remove jack_status recorded, report jack status directly when hdmi plug happend, and remove devm_kfree and devm_free_irq. - Correct the MODULE_LICENSE to "GPL v2" - Move source from sound/soc/codecs to drivers/gpu/drm/bridge/ - Rename dai driver name to "dw-hdmi-i2s-hifi" - Take Mark Brown suggest that remove the no useful code in probe func, and remove the snd_soc_dai_set_fmt() snd_soc_unregister_card(), remove those noisy dev_info() - Rename codec_dai_name to "dw-hdmi-i2s-hifi" Changes in v4: - Take Dainel suggest that introduce drm_detect_monitor_audio() to judge whether source should config audio. - Replace delaywork with irq thread, and add suspend/resume interfaces, replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Yakir Yang (6): drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio when sink device is HDMI and has audio drm: bridge/dw_hdmi: rename dw_hdmi-ahb-audio.h to dw_hdmi-audio.h drm: bridge/dw_hdmi-i2s-audio: add audio driver ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for rockchip_hdmi_audio driver .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/Kconfig | 9 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 2 +- .../{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} | 0 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c | 398 +++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi.c | 100 +++++- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 138 +++++++ 11 files changed, 658 insertions(+), 16 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt rename drivers/gpu/drm/bridge/{dw_hdmi-ahb-audio.h => dw_hdmi-audio.h} (100%) create mode 100644 drivers/gpu/drm/bridge/dw_hdmi-i2s-audio.c create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

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[alsa-devel] [patch] ALSA: hdsp: silence a sprinft() overflow warning
by Dan Carpenter 22 Aug '15

22 Aug '15

card->shortname is a 32 char string so the sprintf() can theoretically overflow. snd_rawmidi_new() can accept strings up to 64 bytes long. I have made the temporay buf[] array 40 bytes long and changed the sprintf() to snprintf(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index c19e021..c1fe1d3 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1526,7 +1526,7 @@ static struct snd_rawmidi_ops snd_hdsp_midi_input = static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int id) { - char buf[32]; + char buf[40]; hdsp->midi[id].id = id; hdsp->midi[id].rmidi = NULL; @@ -1537,7 +1537,7 @@ static int snd_hdsp_create_midi (struct snd_card *card, struct hdsp *hdsp, int i hdsp->midi[id].pending = 0; spin_lock_init (&hdsp->midi[id].lock); - sprintf (buf, "%s MIDI %d", card->shortname, id+1); + snprintf(buf, sizeof(buf), "%s MIDI %d", card->shortname, id + 1); if (snd_rawmidi_new (card, buf, id, 1, 1, &hdsp->midi[id].rmidi) < 0) return -1;

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[alsa-devel] BugReport: Dice works partially [Broken]
by Stanislav Sergiienko 16 Aug '15

16 Aug '15

Hi, I am a user of TC Electronic DesktopKonnekt6 and latest kernel update 3.19.2 add partial support for my device and thus broke my usual sound workflow. So i want to make good bug report to make device usable on latest kernels. With 3.19.2 kernel device is successfully recognized by system and even sometimes can play sound(!). But always, after some time sound is breaks (usually after 30-90 seconds) and to restore it i must restart pulseaudio(v6.0) and application. So device becomes unusable on latests kernels. Earlier i use my sound card through ffado + jackdbus + pulseaudio. Now it's impossible, so i have to downgrade kernel. Please request any additional info that you need to fix problem and i will post it here soon. Also i can test any "bleeding edge" changes thru dkms as described here: https://github.com/takaswie/snd-firewire-improve Related bug(not mine): https://bugs.archlinux.org/task/44258 Some useful info: cat /proc/asound/cards 0 [DesktopKonnekt6]: DICE - DesktopKonnekt6 TC Electronic DesktopKonnekt6 (serial 12941) at fw1.0, S400 cat /proc/asound/card0/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc1:000100000000 notification: 00000040 nick name: DesktopKonnekt6 clock select: internal 44100 enable: 1 status: locked 44100 ext status: 00000000 sample rate: 44100 version: 1.0.4.0 clock caps: 44100 48000 88200 96000 176400 192000 arx1 arx2 internal clock source names: Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\INTERNAL\\ tx 0: iso channel: 0 audio channels: 6 midi ports: 0 speed: S400 names: ch 1\ch 2\ - \ - \ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 rx 0: iso channel: 1 sequence start: 0 audio channels: 6 midi ports: 0 names: main out 1\main out 2\phones out 1\phones out 2\ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 ext status: clock source: internal locked: 1 rate: 44100 adat user data: 0

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[alsa-devel] [PATCH] ASoC: Intel: fix runtime pm imbalance on error
by Junjie Mao 15 Aug '15

15 Aug '15

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Junjie Mao <junjie.mao(a)enight.me> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d1b9e3..27a9653aa3d9 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -152,6 +152,7 @@ static int sst_power_control(struct device *dev, bool state) dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { + pm_runtime_put_sync(dev); dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } @@ -204,8 +205,10 @@ static int sst_cdev_open(struct device *dev, struct intel_sst_drv *ctx = dev_get_drvdata(dev); retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) + if (retval < 0) { + pm_runtime_put_sync(ctx->dev); return retval; + } str_id = sst_get_stream(ctx, str_params); if (str_id > 0) { @@ -672,8 +675,10 @@ static int sst_send_byte_stream(struct device *dev, if (NULL == bytes) return -EINVAL; ret_val = pm_runtime_get_sync(ctx->dev); - if (ret_val < 0) + if (ret_val < 0) { + pm_runtime_put_sync(ctx->dev); return ret_val; + } ret_val = sst_send_byte_stream_mrfld(ctx, bytes); sst_pm_runtime_put(ctx); -- 1.9.3

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[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9c46c7d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

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[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

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[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2ce9e1d..f3034b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

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[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f3034b9..0b4fcd9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

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[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0b4fcd9..e79dc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

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[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8e5ff5e..37aabe3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

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