September 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] snd_device_name_hint segfaults
by Forest Bond 03 Oct '13

03 Oct '13

Hi, I'm seeing segfaults with VLC similar to those reported here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-January/059168.ht… This is on Ubuntu 12.04 with the latest alsa-lib package (1.0.25-1ubuntu10.2) and the following patches also applied: * 2cfc8b9b44a8e493c41b3d63d5a00b306a18a5ed snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() * 44c1a623dd1fc9e831616b663bebc54ca98df994 Add snd_lib_error_set_local() to install a thread-local error handler. * 25dbb102810b31c02358904d70d53c960fb0a10e snd_device_name_hint(): do not change the global error handler. * f49b2dc522a2564315c76d075203b15a39941e8a snd_device_name_hint(): do not use global snd_config. * 7f2b2c8c1650a1883b48abfcdb455138943854f9 conf: Fix a memory access violation resulting from improper error propogation So I think there are still issues when calling snd_device_name_hint(-1, ...). Here is the backtrace: #0 0x003f0e78 in __strcpy_chk () from /lib/i386-linux-gnu/libc.so.6 #1 0x0625084b in strcpy (__src=0x0, __dest=0xb6025e80 "pcm.(null)") at /usr/include/i386-li nux-gnu/bits/string3.h:105 #2 try_config (config=0xaf1c4df0, list=0xb52fe804, base=0x323e962 "pcm", name=0x0) at nameh int.c:243 #3 0x06251bc5 in add_software_devices (list=0xb52fe804, config=0xaf1c4df0) at namehint.c:51 2 #4 snd_device_name_hint (card=-1, iface=<optimized out>, hints=0xb52fe90c) at namehint.c:59 1 #5 0x0323c9bb in GetDevices (obj=0xb6271700, item=0x0, prefs_dev=<optimized out>) at alsa.c :721 #6 0x0323d4c0 in Probe (dev=0xb6011df8 "default", obj=0xb6271700) at alsa.c:161 #7 Open (obj=0xb6271700) at alsa.c:539 #8 0x0a6c73d0 in generic_start (func=0x323cd10, ap=0xb52fed98 "\273") at modules/modules.c:422 #9 0x0a6c7a3a in vlc_module_load (p_this=0xb6271700, psz_capability=0xa70b959 "audio output", psz_name=<optimized out>, b_strict=false, probe=0xa6c73c0 <generic_start>) at modules/modules.c:347 #10 0x0a6c8002 in module_need (obj=0xb6271700, cap=0xa70b959 "audio output", name=0xa70d2bd "$aout", strict=false) at modules/modules.c:437 #11 0x0a6b1242 in aout_OutputNew (p_aout=0xb6271700, p_format=0xb52feeac) at audio_output/output.c:59 #12 0x0a6ad77f in aout_DecNew (p_aout=0xb6271700, p_format=0xb52feeac, p_replay_gain=0xaef02a14, p_request_vout=0xb52feed0) at audio_output/dec.c:112 #13 0x0a67066e in aout_new_buffer (p_dec=0xaef02870, i_samples=1024) at input/decoder.c:2286 #14 0x0a672690 in decoder_NewAudioBuffer (p_decoder=0xaef02870, i_size=1024) at input/decoder.c:213 #15 0x04032f8c in DecodeBlock (p_dec=0xaef02870, pp_block=0xb52ff05c) at faad.c:464 #16 0x0a670891 in DecoderDecodeAudio (p_dec=0xaef02870, p_block=0xaef03000) at input/decoder.c:1291 #17 0x0a6716f5 in DecoderProcessAudio (b_flush=false, p_block=0xaef03000, p_dec=0xaef02870) at input/decoder.c:1936 #18 DecoderProcess (p_dec=0xaef02870, p_block=<optimized out>) at input/decoder.c:2059 #19 0x0a671959 in DecoderThread (p_data=0xaef02870) at input/decoder.c:938 #20 0x001fbd4c in start_thread () from /lib/i386-linux-gnu/libpthread.so.0 #21 0x003ddace in clone () from /lib/i386-linux-gnu/libc.so.6 As a work around I will probably patch VLC to not call snd_device_name_hint this way. I see this has been done for other applications: http://code.google.com/p/chromium/issues/detail?id=95797 http://code.mythtv.org/trac/ticket/10809 But I wanted to try diagnosing the issue first. Thoughts? Thanks, Forest -- Forest Bond http://www.alittletooquiet.net http://www.rapidrollout.com

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[alsa-devel] alsa-lib 1.0.27.2 and musl
by Waldemar Brodkorb 02 Oct '13

02 Oct '13

Hi Developers, I am trying to cross-compile alsa-lib with a musl¹ libc based toolchain. Unfortunately src/compat/hsearch_r.c is failing to compile, because of some glibc specific macros (like _P) normally defined in sys/cdefs.h. After grepping through the code I can not find any consumer for any functions implemented in hsearch_r.c. When I just comment out the code, everything compiles fine. Seems that that usage of hsearch functions was removed in the code long time ago. commit 273d115de05574251bdd661747ecb68449a5cf1d Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Jul 15 10:20:32 2000 +0000 Removed old mixer API functions. Added simple control layer (experimental). Can it just be removed? thanks for any advice or comment. best regards Waldemar ¹ http://www.musl-libc.org/

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[alsa-devel] snd-usb-usx2y (Tascam US-122) sequence errors
by Dr Nicholas J Bailey 02 Oct '13

02 Oct '13

I'm on debian jessie, Linux arial 3.10-3-686-pae #1 SMP Debian 3.10.11-1 (2013-09-10) i686 GNU/Linux with a Tascam us-122 that has recently stopped working. When I try to capture with anything, there is a few hundred ms of audio then the capture process stalls and the following dmesg appears: [ 462.596296] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 462.596296] Most probably some urb of usb-frame 1024 is still missing. [ 462.596296] Cause could be too long delays in usb-hcd interrupt handling. [ 477.959230] Sequence Error!(hcd_frame=0 ep=8in;wait=1024,frame=0). [ 477.959230] Most probably some urb of usb-frame 1024 is still missing. [ 477.959230] Cause could be too long delays in usb-hcd interrupt handling. Before this kernel, the box worked OK. The only think that leads me to believe it might be my fault is that for a very long time, I've had to plug the box in: [ 445.529271] usb 2-1.6: new full-speed USB device number 5 using ehci-pci [ 445.622076] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 445.622086] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 then unplug it and plug it in again: [ 445.972743] usb 2-1.6: USB disconnect, device number 5 [ 447.621519] usb 2-1.6: new full-speed USB device number 6 using ehci-pci [ 447.714462] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 447.714471] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 [ 447.834852] usbcore: registered new interface driver snd-usb-usx2y [ 455.194314] usb 2-1.6: USB disconnect, device number 6 [ 457.487433] usb 2-1.6: new full-speed USB device number 7 using ehci-pci [ 457.579960] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8006 [ 457.579969] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 to get the firmware to load [ 457.888884] usb 2-1.6: USB disconnect, device number 7 [ 459.379782] usb 2-1.6: new full-speed USB device number 8 using ehci-pci [ 459.472516] usb 2-1.6: New USB device found, idVendor=1604, idProduct=8007 [ 459.472526] usb 2-1.6: New USB device strings: Mfr=0, Product=0, SerialNumber=0 (green USB light comes on here). I'm not sure about how to start debugging this; there are many things that could be wrong. If it will help, I'll try and file a bug report; if not please tell me to shut up (prefereably after pointing me at the solution :) ) Nick/.

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[alsa-devel] [PATCH] ASoC: kirkwood: fix compilation warning in kirkwood_dma_open
by Vladimir Murzin 01 Oct '13

01 Oct '13

writel() supposes the first argument of type unsigned int. This fix the warning: sound/soc/kirkwood/kirkwood-dma.c: In function 'kirkwood_dma_open': sound/soc/kirkwood/kirkwood-dma.c:164:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] Signed-off-by: Vladimir Murzin <murzin.v(a)gmail.com> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434..1135c30 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -161,7 +161,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); -- 1.8.1.5

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[alsa-devel] [PATCH] snd_tlv_convert_from_dB: fix decreasing gain across entries
by Stephen Warren 01 Oct '13

01 Oct '13

From: Stephen Warren <swarren(a)nvidia.com> Currently, for a TLV consisting of TLV_DB_SCALE_ITEMs, if e.g. alsamixer calls snd_mixer_selem_set_playback_dB() with a value that is in-between two TLV_DB_SCALE_ITEMs, and xdir is negative, the selected raw hardware value is the minimum in the first range above that value, rather than the maximum in the last range below that value. The user-visible symptom is that in alsamixer, pressing the down key to reduce the value sticks at certain points, and cannot be incrementally reduced any further, although directly selecting a much lower value (e.g. by pressing 0..9) works as expected. This is triggered e.g. by sound/soc/codec/max98090.c's max98090_hp_tlv[]. Fix this by checking whether xdir is positive or not, rather than checking whether it has a non-zero value. The code to select the previous range's max value is already present. This matches how xdir is used in other parts of the code. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- src/control/tlv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/control/tlv.c b/src/control/tlv.c index 6b0b9f4..b08d887 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -312,7 +312,7 @@ int snd_tlv_convert_from_dB(unsigned int *tlv, long rangemin, long rangemax, submin, submax, db_gain, value, xdir); else if (db_gain < dbmin) { - *value = xdir || pos == 2 ? submin : prev_submax; + *value = xdir > 0 || pos == 2 ? submin : prev_submax; return 0; } prev_submax = submax; -- 1.8.1.5

2 1

[alsa-devel] [PATCH] amixer: actually print all TLVs in a container TLV
by Anssi Hannula 01 Oct '13

01 Oct '13

Due to a wrong indexing of tlv[] decode_tlv() always skips to the end of the container TLV since the used tlv[1] contains the container TLV size instead of the intended element size. This causes, for example, only the first HDMI channel map TLV to be shown. Fix the indexing to actually use the element size. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> --- amixer/amixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 7727943..b83b0c3 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -468,7 +468,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ return; } decode_tlv(spaces + 2, tlv + idx, tlv[idx+1] + 8); - idx += 2 + (tlv[1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); + idx += 2 + (tlv[idx+1] + sizeof(unsigned int) - 1) / sizeof(unsigned int); } break; case SND_CTL_TLVT_DB_SCALE: -- 1.8.1.5

2 1

[alsa-devel] [PATCH v2] ASoC: imx-sgtl5000: do not use devres on a foreign device
by Philipp Zabel 01 Oct '13

01 Oct '13

Calling devm_clk_get with any device pointer other than our own confuses devres. Use clk_get instead. This avoids hitting the following warning in the imx-sgtl5000 error path: imx-sgtl5000 sound.12: snd_soc_register_card failed (-517) platform sound.12: Driver imx-sgtl5000 requests probe deferral ------------[ cut here ]------------ WARNING: CPU: 0 PID: 75 at drivers/base/dd.c:272 driver_probe_device+0x194/0x218() Modules linked in: snd_soc_sgtl5000(+) snd_soc_imx_sgtl5000 coda snd_soc_imx_audmux imx_sdma snd_soc_fsl_spdif snd_soc_fsl_ssi CPU: 0 PID: 75 Comm: udevd Not tainted 3.11.0-rc6+ #4682 Backtrace: [<80010bc4>] (dump_backtrace+0x0/0x10c) from [<80010d60>] (show_stack+0x18/0x1c) r6:00000110 r5:00000009 r4:00000000 r3:00000000 [<80010d48>] (show_stack+0x0/0x1c) from [<804f0764>] (dump_stack+0x20/0x28) [<804f0744>] (dump_stack+0x0/0x28) from [<8001a4a4>] (warn_slowpath_common+0x6c/0x8c) [<8001a438>] (warn_slowpath_common+0x0/0x8c) from [<8001a4e8>] (warn_slowpath_null+0x24/0x2c) r8:7f032000 r7:7f02f93c r6:cf8eaa54 r5:cf8eaa20 r4:80728a0c [<8001a4c4>] (warn_slowpath_null+0x0/0x2c) from [<80286bdc>] (driver_probe_device+0x194/0x218) [<80286a48>] (driver_probe_device+0x0/0x218) from [<80286cf4>] (__driver_attach+0x94/0x98) r7:00000000 r6:cf8eaa54 r5:7f02f93c r4:cf8eaa20 [<80286c60>] (__driver_attach+0x0/0x98) from [<802851c8>] (bus_for_each_dev+0x5c/0x90) r6:80286c60 r5:7f02f93c r4:00000000 r3:cf8ef03c [<8028516c>] (bus_for_each_dev+0x0/0x90) from [<80286654>] (driver_attach+0x24/0x28) r6:806d0424 r5:cf16a580 r4:7f02f93c [<80286630>] (driver_attach+0x0/0x28) from [<802861e4>] (bus_add_driver+0xdc/0x234) [<80286108>] (bus_add_driver+0x0/0x234) from [<802871d4>] (driver_register+0x80/0x154) r8:7f032000 r7:00000001 r6:7f02fa68 r5:7f02fa74 r4:7f02f93c [<80287154>] (driver_register+0x0/0x154) from [<8033c278>] (i2c_register_driver+0x34/0xbc) [<8033c244>] (i2c_register_driver+0x0/0xbc) from [<7f032018>] (sgtl5000_i2c_driver_init+0x18/0x24 [snd_soc_sgtl5000]) r5:7f02fa74 r4:cfb7ff48 [<7f032000>] (sgtl5000_i2c_driver_init+0x0/0x24 [snd_soc_sgtl5000]) from [<80008738>] (do_one_initcall+0xf4/0x150) [<80008644>] (do_one_initcall+0x0/0x150) from [<80053f64>] (load_module+0x174c/0x1db4) [<80052818>] (load_module+0x0/0x1db4) from [<800546ac>] (SyS_init_module+0xe0/0xf4) [<800545cc>] (SyS_init_module+0x0/0xf4) from [<8000e540>] (ret_fast_syscall+0x0/0x30) r6:00005b22 r5:00afed68 r4:00000000 ---[ end trace b24c5c3bb145dbdd ]--- Signed-off-by: Philipp Zabel <p.zabel(a)pengutronix.de> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> Reviewed-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Rebased onto asoc/for-next, adding back imx_sgtl5000_remove. --- sound/soc/fsl/imx-sgtl5000.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 78f86d8..c49e4cb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (!IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -180,6 +182,15 @@ fail: return ret; } +static int imx_sgtl5000_remove(struct platform_device *pdev) +{ + struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + + clk_put(data->codec_clk); + + return 0; +} + static const struct of_device_id imx_sgtl5000_dt_ids[] = { { .compatible = "fsl,imx-audio-sgtl5000", }, { /* sentinel */ } -- 1.8.4.rc3

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[alsa-devel] [PATCH 2/2] [RESEND] snd-ac97-codec: Use complete ID when the codec is not known
by Ondrej Zary 30 Sep '13

30 Sep '13

Currently, if the codec ID is not known, the last byte of the ID is used as name. It's not helpful at all, use full codec ID instead. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index bf578ba2..30a40a8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1830,7 +1830,7 @@ void snd_ac97_get_name(struct snd_ac97 *ac97, unsigned int id, char *name, int m pid->patch(ac97); } } else - sprintf(name + strlen(name), " id %x", id & 0xff); + sprintf(name + strlen(name), " id %x", id); } /** -- Ondrej Zary

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[alsa-devel] [PATCH] ASoC: generic: Add generic DT based simple codec
by Jean-Francois Moine 30 Sep '13

30 Sep '13

This patch adds a simple sound codec which is described by the DT. This codec may be used when no specific codec action is needed. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/simple-codec.txt | 103 +++++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/simple-codec.c | 197 +++++++++++++++++++++++++ 4 files changed, 3088 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-codec.txt b/Documentation/devicetree/bindings/sound/simple-codec.txt new file mode 100644 index 0000000..75be747 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-codec.txt @@ -0,0 +1,103 @@ +Device-Tree bindings for the simple codec + +Required properties: +- compatible: should be "linux,simple-codec". +- dai-name: name of the codec + +Optional properties: +- capture: information about capture +- playback: information about playback +At least one of the 'capture' or 'playback' nodes must be present. + +Child 'capture' and 'playback' required properties: +- stream-name: name of the stream +- formats: list of the supported formats (see below) +- rates: list of the supported rates (see below) +- #channels: minimum and maximum numbers of channels + +Formats: + "s8" + "u8" + "s16_le" + "s16_be" + "u16_le" + "u16_be" + "s24_le" + "s24_be" + "u24_le" + "u24_be" + "s32_le" + "s32_be" + "u32_le" + "u32_be" + "float_le" + "float_be" + "float64_le" + "float64_be" + "iec958_subframe_le" + "iec958_subframe_be" + "mu_law" + "a_law" + "ima_adpcm" + "mpeg" + "gsm" + "special" + "s24_3l" + "s24_3be" + "u24_3le" + "u24_3be" + "s20_3le" + "s20_3be" + "u20_3le" + "u20_3be" + "s18_3le" + "s18_3b" + "u18_3le" + "u18_3be" + "g723_24" + "g723_24_1b" + "g723_40" + "g723_40_1b" + "dsd_u8" + "dsd_u16_le" + +Rates: + 5512 + 8000 + 11025 + 16000 + 22050 + 32000 + 44100 + 48000 + 64000 + 88200 + 96000 + 176400 + 192000 + +Examples: + + spdifo: spdif-transmitter { + compatible = "linux,simple-codec"; + dai-name = "dit-hifi"; + playback { + stream-name = "S/PDIF Playback"; + formats = "s16_le"; + rates = <8000 11025 16000 22050 32000 44100 + 48000 64000 88200 96000>; + #channels = <1 384>; + }; + }; + + hdmio: hdmi-transmitter { + compatible = "linux,simple-codec"; + dai-name = "hdmi-hifi"; + playback { + stream-name = "HDMI Playback"; + formats = "s16_le", "s24_le"; + rates = <32000 44100 48000 88200 96000 + 176400 192000>; + #channels = <2 8>; + }; + }; diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 610f612..d9221539 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -2,3 +2,9 @@ config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" help This option enables generic simple sound card support + +config SND_SIMPLE_CODEC + tristate "ASoC Simple sound codec support" + depends on OF + help + This option enables generic simple sound codec support diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..f94620e 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,5 @@ snd-soc-simple-card-objs := simple-card.o +snd-soc-simple-codec-objs := simple-codec.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CODEC) += snd-soc-simple-codec.o diff --git a/sound/soc/generic/simple-codec.c b/sound/soc/generic/simple-codec.c new file mode 100644 index 0000000..217cd0e --- /dev/null +++ b/sound/soc/generic/simple-codec.c @@ -0,0 +1,197 @@ +/* + * ALSA SoC simple codec driver + * + * This driver is used by controllers which do not need any codec + * (s/pdif, hdmi..). + * Its parameters are built from a DT description. + * + * Copyright: (C) 2013 Jean-François Moine <moinejf(a)free.fr> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <linux/of.h> + +#define DRV_NAME "simple-codec" + +static struct snd_soc_codec_driver soc_codec_simple_codec; + +static char *format_tb[] = { + "s8", "u8", "s16_le", "s16_be", + "u16_le", "u16_be", "s24_le", "s24_be", + "u24_le", "u24_be", "s32_le", "s32_be", + "u32_le", "u32_be", "float_le", "float_be", + + "float64_le", "float64_be", "iec958_subframe_le", "iec958_subframe_be", + "mu_law", "a_law", "ima_adpcm", "mpeg", + "gsm", "", "", "", + "", "", "", "special", + + "s24_3l", "s24_3be", "u24_3le", "u24_3be", + "s20_3le", "s20_3be", "u20_3le", "u20_3be", + "s18_3le", "s18_3b", "u18_3le", "u18_3be", + "g723_24", "g723_24_1b", "g723_40", "g723_40_1b", + + "dsd_u8", "dsd_u16_le", +}; + +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#error "Change this table" +#endif +static u32 rate_tb[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000 }; + +static int get_format(u64 *p_formats, char *p) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(format_tb); i++) { + if (strcmp(format_tb[i], p) == 0) { + *p_formats |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int get_rate(u32 *p_rates, u32 val) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rate_tb); i++) { + if (rate_tb[i] == val) { + *p_rates |= 1 << i; + return 0; + } + } + return -EINVAL; +} + +static int stream_populate(struct device *dev, + struct snd_soc_pcm_stream *stream, + struct device_node *np) +{ + u64 formats; + u32 rates, val; + char *p; + int i, ret; + + ret = of_property_read_string(np, "stream-name", &stream->stream_name); + if (ret) { + dev_err(dev, "Failed to parse stream-name string\n"); + return ret; + } + + i = 0; + formats = 0; + for (;;) { + ret = of_property_read_string_index(np, "formats", i, + (const char **) &p); + if (ret) + break; + ret = get_format(&formats, p); + if (ret) { + dev_err(dev, "Bad codec format\n"); + return ret; + } + i++; + } + stream->formats = formats; + + i = 0; + rates = 0; + for (;;) { + ret = of_property_read_u32_index(np, "rates", i, &val); + if (ret) + break; + ret = get_rate(&rates, val); + if (ret) { + dev_err(dev, "Bad rate\n"); + return ret; + } + i++; + } + stream->rates = rates; + + ret = of_property_read_u32_index(np, "#channels", 0, &val); + if (ret) { + dev_err(dev, "Bad min channel\n"); + return ret; + } + stream->channels_min = val; + ret = of_property_read_u32_index(np, "#channels", 1, &val); + if (ret) { + dev_err(dev, "Bad max channel\n"); + return ret; + } + stream->channels_max = val; + + return 0; +} + +static int simple_codec_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct device_node *np = dev->of_node; + struct device_node *np2; + struct snd_soc_dai_driver *dai; + int ret; + + dai = devm_kzalloc(dev, sizeof *dai, GFP_KERNEL); + if (!dai) + return -ENOMEM; + + ret = of_property_read_string(np, "dai-name", &dai->name); + if (ret) { + dev_err(dev, "Failed to parse dai-name\n"); + return ret; + } + + np2 = of_find_node_by_name(np, "capture"); + if (np2) { + ret = stream_populate(dev, &dai->capture, np2); + if (ret) + return ret; + } + np2 = of_find_node_by_name(np, "playback"); + if (np2) { + ret = stream_populate(dev, &dai->playback, np2); + if (ret) + return ret; + } + + return snd_soc_register_codec(dev, &soc_codec_simple_codec, + dai, 1); +} + +static int simple_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id simple_codec_of_match[] = { + { .compatible = "linux,simple-codec", }, + { } +}; +MODULE_DEVICE_TABLE(of, simple_codec_of_match); + +static struct platform_driver simple_codec_driver = { + .probe = simple_codec_probe, + .remove = simple_codec_remove, + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = simple_codec_of_match, + }, +}; + +module_platform_driver(simple_codec_driver); + +MODULE_AUTHOR("Jean-Francois Moine <moinejf(a)free.fr>"); +MODULE_DESCRIPTION("simple codec driver"); +MODULE_LICENSE("GPL"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: pcm: Remove extra spaces from dev_ prints
by Jarkko Nikula 30 Sep '13

30 Sep '13

dev_ prints are already prefixed by ": " before format string so there is no need for extra spaces. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6..d449872 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -721,7 +721,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); @@ -749,7 +749,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, if (dpcm->fe == fe) continue; - dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + dev_dbg(fe->dev, "reparent %s path %s %s %s\n", stream ? "capture" : "playback", dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -773,7 +773,7 @@ static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; - dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -2116,7 +2116,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- 1.8.4.rc3

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