July 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] add support for Codec: Conexant ID 5068 (not present in ALSA yet)
by ccamacho＠linuxmail.org 21 Jul '10

21 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

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[alsa-devel] Compiler errors in alsa-lib-1.0.22/test/*.c
by John Simpson 21 Jul '10

21 Jul '10

Howdy, I've been trying to use the alsa-lib API, but I haven't been able to get anything to compile. ================= ~/alsa/alsa-lib-1.0.22/test$ for f in *.c; do echo; cc $f 2>&1 | head -n2; done /tmp/ccNjLWLK.o: In function `dump_event_filter': client_event_filter.c:(.text+0x23): undefined reference to `snd_seq_client_info_event_filter_check' /tmp/ccNLDqdU.o: In function `main': control.c:(.text+0x1c): undefined reference to `snd_ctl_card_info_sizeof' /tmp/ccBehb51.o: In function `setparams_stream': latency.c:(.text+0x24): undefined reference to `snd_pcm_hw_params_any' /usr/lib/gcc/x86_64-linux-gnu/4.4.3/../../../../lib/crt1.o: In function `_start': (.text+0x20): undefined reference to `main' midiloop.c: In function ‘main’: midiloop.c:172: warning: format ‘%i’ expects type ‘int’, but argument 2 has type ‘size_t’ namehint.c: In function ‘main’: namehint.c:12: warning: passing argument 3 of ‘snd_device_name_hint’ from incompatible pointer type oldapi.c: In function ‘main’: oldapi.c:40: warning: assignment from incompatible pointer type /tmp/ccmB528p.o: In function `generate_sine': pcm.c:(.text+0x119): undefined reference to `snd_pcm_format_width' /tmp/ccChgWYC.o: In function `main': pcm_min.c:(.text+0x4d): undefined reference to `snd_pcm_open' /tmp/cc9FiONG.o: In function `write_ev': playmidi1.c:(.text+0x1b65): undefined reference to `snd_seq_event_output' /tmp/ccQkntBX.o: In function `main': queue_timer.c:(.text+0x1d0): undefined reference to `snd_seq_open' /tmp/ccmIMCZ4.o: In function `main': rawmidi.c:(.text+0x4e3): undefined reference to `snd_rawmidi_open' /tmp/cc3g8KGc.o: In function `event_decoder_start_timer': seq.c:(.text+0x4e2): undefined reference to `snd_seq_control_queue' seq-decoder.c:6: error: ‘SND_SEQ_EVENT_SYSTEM’ undeclared here (not in a function) seq-decoder.c:6: error: array index in initializer not of integer type seq-sender.c:68: error: expected ‘)’ before ‘*’ token seq-sender.c:100: error: expected ‘)’ before ‘*’ token /tmp/ccxyZfhg.o: In function `show_status': timer.c:(.text+0x1d): undefined reference to `snd_timer_status_sizeof' ================= Did I miss something? TIA -- John Wesley Simpson SwaJime's Cove℠ www.swajime.com

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[alsa-devel] [PATCH 0/3] Removing dead code
by Christian Dietrich 21 Jul '10

21 Jul '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we are looking at multiple integrity errors in linux' configuration system. I've been running a check on the sound/ sourcetree for config Items not defined in Kconfig and found 3 such cases. Sourcecode blocks depending on these Items are not reachable from a vanilla kernel -- dead code. I've seen such dead blocks made on purpose e.g. while integrating new features into the kernel but generally they're just useless. Each of the patches in this patchset removes on such dead config Item, I'd be glad if you consider applying them. I've been doing deeper analysis of such issues before and can do so again but I'm not so sure they were fastly usefull. I build the patches against a vanilla kernel in order to try if the kernel compiles with this patches Please keep me informed of this patch getting confirmed / merged so we can keep track of it. Regards Christian Dietrich [0] http://vamos1.informatik.uni-erlangen.de/ Christian Dietrich (3): sound/oss: Remove dead CONFIG_SOFTOSS* sound/soc/atmel: Remove dead CONFIG_AT32_ENHANCED_PORTMUX sound/isa/gus: Remove dead CONFIG_SND_DEBUG_ROM sound/isa/gus/interwave.c | 8 -------- sound/oss/vidc.c | 3 --- sound/soc/atmel/playpaq_wm8510.c | 9 --------- 3 files changed, 0 insertions(+), 20 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 21 Jul '10

21 Jul '10

The following changes since commit 66b47fdb851924956b6e4696fb43a3496ae2c462: Mark Brown (1): ASoC: Implement WM8994 OPCLK support are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Jorge Eduardo Candelaria (1): ASoC: TWL6040: Correct widget handling for drivers Peter Ujfalusi (3): ASoC: TWL4030: Fix for digital loopback gain range ASoC: TWL4030: DAC power optimization ASoC: TWL4030: Add configurable delay after digimic enable sound/soc/codecs/twl4030.c | 47 ++++++++++++++++++++++++++---------- sound/soc/codecs/twl4030.h | 1 + sound/soc/codecs/twl6040.c | 56 ++++++++++++++++++++++---------------------- 3 files changed, 63 insertions(+), 41 deletions(-)

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[alsa-devel] mmotm 2010-07-19 - more pm_qos woes - audio this time
by Valdis.Kletnieks＠vt.edu 21 Jul '10

21 Jul '10

On Mon, 19 Jul 2010 16:38:09 PDT, akpm(a)linux-foundation.org said: > The mm-of-the-moment snapshot 2010-07-19-16-37 has been uploaded to > > http://userweb.kernel.org/~akpm/mmotm/ I've accumulated about 1,000 of these overnight. Seems to be consistently landing in pairs based on the timestamps. [42875.544006] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42875.544006] Hardware name: Latitude E6500 [42875.544006] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42875.544006] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42875.544006] Call Trace: [42875.544006] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42875.544006] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42875.544006] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42875.544006] [<ffffffff812052c7>] plist_add+0x1b/0xb1 [42875.544006] [<ffffffff81057b2e>] update_target+0x94/0xd0 [42875.544006] [<ffffffff81057d54>] pm_qos_add_request+0x79/0x7b [42875.544006] [<ffffffff8141d199>] snd_pcm_hw_params+0x25a/0x28a [42875.544006] [<ffffffff8141da29>] snd_pcm_common_ioctl1+0x23b/0x980 [42875.544006] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42875.544006] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42875.544006] [<ffffffff810d9178>] ? fd_install+0x36/0xd5 [42875.544006] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42875.544006] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42875.544006] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42875.544006] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42875.544006] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42875.544006] ---[ end trace 23d666d936caf587 ]--- [42875.544006] ------------[ cut here ]------------ [42875.544006] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42875.544006] Hardware name: Latitude E6500 [42875.544006] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42875.544006] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42875.544006] Call Trace: [42875.544006] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42875.544006] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42875.544006] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42875.544006] [<ffffffff81205354>] plist_add+0xa8/0xb1 [42875.544006] [<ffffffff81057b2e>] update_target+0x94/0xd0 [42875.544006] [<ffffffff81057d54>] pm_qos_add_request+0x79/0x7b [42875.544006] [<ffffffff8141d199>] snd_pcm_hw_params+0x25a/0x28a [42875.544006] [<ffffffff8141da29>] snd_pcm_common_ioctl1+0x23b/0x980 [42875.544006] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42875.544006] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42875.544006] [<ffffffff810d9178>] ? fd_install+0x36/0xd5 [42875.544006] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42875.544006] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42875.544006] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42875.544006] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42875.544006] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42875.544006] ---[ end trace 23d666d936caf588 ]--- [42882.427310] ------------[ cut here ]------------ [42882.428016] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42882.428016] Hardware name: Latitude E6500 [42882.428016] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42882.428016] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42882.428016] Call Trace: [42882.428016] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42882.428016] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42882.428016] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42882.428016] [<ffffffff8120537c>] plist_del+0x1f/0x8b [42882.428016] [<ffffffff81057b21>] update_target+0x87/0xd0 [42882.428016] [<ffffffff81057bb6>] pm_qos_remove_request+0x4c/0x5c [42882.428016] [<ffffffff8141dad9>] snd_pcm_common_ioctl1+0x2eb/0x980 [42882.428016] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42882.428016] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42882.428016] [<ffffffff8109589d>] ? trace_preempt_on+0x15/0x28 [42882.428016] [<ffffffff810c727e>] ? unmap_region+0xbd/0xcc [42882.428016] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42882.428016] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42882.428016] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42882.428016] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42882.428016] [<ffffffff810c746a>] ? unlink_file_vma+0x2d/0x4d [42882.428016] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42882.428016] ---[ end trace 23d666d936caf589 ]--- [42882.428016] ------------[ cut here ]------------ [42882.428016] WARNING: at lib/plist.c:57 plist_check_head+0x47/0x114() [42882.428016] Hardware name: Latitude E6500 [42882.428016] Modules linked in: ppp_async sunrpc microcode usb_storage [last unloaded: scsi_wait_scan] [42882.428016] Pid: 58070, comm: pulseaudio Tainted: G W 2.6.35-rc5-mmotm0719 #2 [42882.428016] Call Trace: [42882.428016] [<ffffffff81037335>] warn_slowpath_common+0x80/0x98 [42882.428016] [<ffffffff81037362>] warn_slowpath_null+0x15/0x17 [42882.428016] [<ffffffff812051df>] plist_check_head+0x47/0x114 [42882.428016] [<ffffffff812053df>] plist_del+0x82/0x8b [42882.428016] [<ffffffff81057b21>] update_target+0x87/0xd0 [42882.428016] [<ffffffff81057bb6>] pm_qos_remove_request+0x4c/0x5c [42882.428016] [<ffffffff8141dad9>] snd_pcm_common_ioctl1+0x2eb/0x980 [42882.428016] [<ffffffff811c63db>] ? inode_has_perm.clone.18+0x92/0xa2 [42882.428016] [<ffffffff8141e613>] snd_pcm_playback_ioctl1+0x231/0x246 [42882.428016] [<ffffffff8109589d>] ? trace_preempt_on+0x15/0x28 [42882.428016] [<ffffffff810c727e>] ? unmap_region+0xbd/0xcc [42882.428016] [<ffffffff8141e652>] snd_pcm_playback_ioctl+0x2a/0x2e [42882.428016] [<ffffffff810e91a0>] vfs_ioctl+0x31/0xa2 [42882.428016] [<ffffffff810e9b1f>] do_vfs_ioctl+0x496/0x4c9 [42882.428016] [<ffffffff810e9ba9>] sys_ioctl+0x57/0x96 [42882.428016] [<ffffffff810c746a>] ? unlink_file_vma+0x2d/0x4d [42882.428016] [<ffffffff8100272b>] system_call_fastpath+0x16/0x1b [42882.428016] ---[ end trace 23d666d936caf58a ]---

2 1

[alsa-devel] Missing config option for SFFSDR_FPGA
by Christian Dietrich 21 Jul '10

21 Jul '10

Hi all! As part of the VAMOS[0] research project at the University of Erlangen we are looking at multiple integrity errors in linux' configuration system. I've been running a check on the sound/ sourcetree for config Items not defined in Kconfig and found such a case. Sourcecode blocks depending on these Items are not reachable from a vanilla kernel -- dead code. I've seen such dead blocks made on purpose e.g. while integrating new features into the kernel but generally they're just useless. There are two ifdef blocks, which do a check on CONFIG_SFFSDR_FPGA, but this flag isn't defined anywhere. There is an select in sound/soc/davinci[1], but Kconfig doesn't define the symbol, if there isn't a config option for it. So this blocks can never be selected. I don't think this is intended, so i didn't wrote a patch. Please just notice it. Regards Christian Dietrich [0] http://vamos1.informatik.uni-erlangen.de/ [1] sound/soc/davinci/Kconfig: select SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifdef CONFIG_SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifndef CONFIG_SFFSDR_FPGA sound/soc/davinci/davinci-sffsdr.c:#ifndef CONFIG_SFFSDR_FPGA -- (λ x . x x) (λ x . x x) -- See how beatiful the lambda is No documentation is better than bad documentation -- Das Ausdrucken dieser Mail wird urheberrechtlich verfolgt.

3 3

[alsa-devel] [PATCH] ASoC: TWL4030: Add configurable delay after digimic enable
by Peter Ujfalusi 21 Jul '10

21 Jul '10

When digital microphones are connected to twl, delay is needed after enabling the digimic interface of the codec. Add new parameter for the setup data, which can be used to pass the apropriate delay in ms after the digimic interface has been enabled. Without certain delay (in certain HW configuration) the beggining of the recorded sample contains a glitch, which is generated by the digital microphones. Delaying the micbias1, 2 (which is the bias for the digimic0 or 1) does not help, since the glitch is coming after switching the digimic interface. Reversing the micbias and digimic enable order does not work either (in that case the wait need to be added after the micbias enabled). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, I have been looking after this problem for a year now (with low priority), and finally I did figured out what is caused the glitch at the beginning of the recording via the digimic interface. Adding platform dependent delay after enabling the digimic interface fixes the issue I am tracking, in other HW setups with different components the delay might not be needed. Peter sound/soc/codecs/twl4030.c | 25 +++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 1 + 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6fd6d0b..bd557c2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -143,6 +143,9 @@ struct twl4030_priv { u8 earpiece_enabled; u8 predrivel_enabled, predriver_enabled; u8 carkitl_enabled, carkitr_enabled; + + /* Delay needed after enabling the digimic interface */ + unsigned int digimic_delay; }; /* @@ -312,6 +315,8 @@ static void twl4030_init_chip(struct platform_device *pdev) if (!setup) return; + twl4030->digimic_delay = setup->digimic_delay; + /* Configuration for headset ramp delay from setup data */ if (setup->sysclk != twl4030->sysclk) dev_warn(codec->dev, @@ -855,6 +860,16 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int digimic_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(w->codec); + + if (twl4030->digimic_delay) + mdelay(twl4030->digimic_delay); + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1439,10 +1454,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ADC Physical Right", TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic0 Enable", - TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("Digimic1 Enable", - TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Digimic0 Enable", + TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Digimic1 Enable", + TWL4030_REG_ADCMICSEL, 3, 0, NULL, 0, + digimic_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0), SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0), diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 788e3d1..6c57430 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -41,6 +41,7 @@ extern struct snd_soc_codec_device soc_codec_dev_twl4030; struct twl4030_setup_data { unsigned int ramp_delay_value; + unsigned int digimic_delay; /* in ms */ unsigned int sysclk; unsigned int offset_cncl_path; unsigned int check_defaults:1; -- 1.7.1.1

3 2

[alsa-devel] [PATCHv2] ASoC: multi-component - Add Aquila sound driver
by Chanwoo Choi 21 Jul '10

21 Jul '10

This patch add sound support for the Aquila board based on S5PC110. The Aquila board is based on Samsung SoC(S5PC110) and include WM8994 codec over I2S to support sound. This uses the I2Sv4 driver compatible with I2Sv5 to run sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Aquila board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/s3c24xx/Kconfig | 9 + sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/aquila_wm8994.c | 301 +++++++++++++++++++++++++++++++++++++ 3 files changed, 312 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/aquila_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 213963a..c09b402 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -131,3 +131,12 @@ config SND_S3C64XX_SOC_SMARTQ depends on SND_S3C24XX_SOC && MACH_SMARTQ select SND_S3C64XX_SOC_I2S select SND_SOC_WM8750 + +config SND_S5PC110_SOC_AQUILA_WM8994 + tristate "SoC I2S Audio support for AQUILA - WM8994" + depends on SND_S3C24XX_SOC && MACH_AQUILA + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on aquila + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 50172c3..02dd12c 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -30,6 +30,7 @@ snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o +snd-soc-aquila-wm8994-objs := aquila_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv32 obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o +obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c new file mode 100644 index 0000000..375f4c0 --- /dev/null +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -0,0 +1,301 @@ +/* + * aquila_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define WM8994_DAI_HIFI 0 +#define WM8994_DAI_VOICE 1 + +static struct snd_soc_card aquila; +static struct platform_device *aquila_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget aquila_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route aquila_dapm_routes[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int aquila_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add aquila specific widgets */ + snd_soc_dapm_new_controls(codec, aquila_dapm_widgets, + ARRAY_SIZE(aquila_dapm_widgets)); + + /* set up aquila specific audio routes */ + snd_soc_dapm_add_routes(codec, aquila_dapm_routes, + ARRAY_SIZE(aquila_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRN"); + snd_soc_dapm_nc_pin(codec, "SPKOUTRP"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&aquila, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int aquila_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops aquila_hifi_ops = { + .hw_params = aquila_hifi_hw_params, +}; + +static int aquila_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "Voice", + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops aquila_voice_ops = { + .hw_params = aquila_voice_hw_params, +}; + +static struct snd_soc_dai_link aquila_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_drv = &s3c64xx_i2s_v4_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_HIFI], + .codec_dai_id = WM8994_DAI_HIFI, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .init = aquila_wm8994_init, + .ops = &aquila_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_drv = &voice_dai, + .codec_dai_drv = &wm8994_dai[WM8994_DAI_VOICE], + .codec_dai_id = WM8994_DAI_VOICE, + .platform_drv = &s3c24xx_soc_platform, + .codec_drv = &soc_codec_dev_wm8994, + .ops = &aquila_voice_ops, +}, +}; + +static struct snd_soc_card aquila = { + .name = "aquila", + .dai_link = aquila_dai, + .num_links = ARRAY_SIZE(aquila_dai), +}; + +static int __init aquila_init(void) +{ + int ret; + + if (!machine_is_aquila()) + return -ENODEV; + + aquila_snd_device = platform_device_alloc("soc-audio", -1); + if (!aquila_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&aquila_snd_device->dev, + 0, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(aquila_snd_device, &aquila); + ret = platform_device_add(aquila_snd_device); + + if (ret) + platform_device_put(aquila_snd_device); + + return ret; +} + +static void __exit aquila_exit(void) +{ + platform_device_unregister(aquila_snd_device); +} + +module_init(aquila_init); +module_exit(aquila_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 Aquila(S5PC110)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] [PATCH] Memory leak in namehint.c
by John Lindgren 21 Jul '10

21 Jul '10

get_card_name() can be called more than once on the same list, so it must free the previous list->cardname before replacing it. Signed-off-by: John Lindgren <john.lindgren(a)tds.net> --- src/control/namehint.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/src/control/namehint.c b/src/control/namehint.c index 78572d8..fc1cb27 100644 --- a/src/control/namehint.c +++ b/src/control/namehint.c @@ -471,6 +471,8 @@ static int get_card_name(struct hint_list *list, int card) char scard[16], *s; int err; + if (list->cardname) + free(list->cardname); err = snd_card_get_name(card, &list->cardname); if (err <= 0) return 0;

2 2

[alsa-devel] Question for ALSA I/O Plugin
by Raul Xiong 21 Jul '10

21 Jul '10

Hi all, I know that there are two kinds of ALSA external Plugin: filter-type and I/O type. Now I have an I/O Plugin named phone. Can it be a slave of a internal Plugin? E.g. can I use this way: aplay -Dplug:phone wav. Thanks!

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