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

[alsa-devel] [PATCH] imx-ssi.c: fix I2S slave setup
by Eric Bénard 27 May '10

27 May '10

* without this fix, the waveform on the SSI port are not real I2S signal (at least on i.MX27's SSI4) * original Freescale's driver has if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) && ssi_mode->network_mode) { scr &= ~SSI_SCR_I2S_MODE_MASK; scr |= SSI_SCR_I2S_MODE_SLAVE; } So as now the SSI is set in network mode as a default when running in I2S format, it seems reasonable to also set I2S_MODE_SLAVE Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/imx-ssi.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 80b4fee..c5e1626 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, /* * SSI DAI format configuration. * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S because the I2S mode is only a register preset. */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -98,7 +96,7 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) case SND_SOC_DAIFMT_I2S: /* data on rising edge of bclk, frame low 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; - scr |= SSI_SCR_NET; + scr |= SSI_SCR_NET | SSI_SCR_I2S_MODE_SLAVE; break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ -- 1.6.3.3

3 4

[alsa-devel] [PATCH] alsa-python: add register_poll to alsaseq
by Hector Martin 27 May '10

27 May '10

Attached patch adds register_poll to the sequencer bindings to support nonblocking sequencer access with the select module. I sent this last month but it looks like it was overlooked. Resending. -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

2 1

[alsa-devel] [PATCH: alsa-lib] Fix a string format ambiguity
by Colin Guthrie 27 May '10

27 May '10

Hi, Simply patch to fix a format-string ambiguity. Meant to give this over before 1.0.23 but forgot. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

2 1

[alsa-devel] [PATCH] Add support for Thinkpad Edge conexant chip
by Jerone Young 27 May '10

27 May '10

This quirks in support for the Thinkpad Edge. Signed-off-by: Jerone Young <jerone.young(a)canonical.com> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index e863649..f4a2bd6 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -2975,6 +2975,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),

2 1

[alsa-devel] [PATCH 1/5] Remove unused io map functions.
by linux＠audioscience.com 27 May '10

27 May '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpios.c | 23 ----------------------- sound/pci/asihpi/hpios.h | 9 --------- 2 files changed, 0 insertions(+), 32 deletions(-) diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index de615cf..742ee12 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -89,26 +89,3 @@ u16 hpios_locked_mem_free(struct consistent_dma_area *p_mem_area) void hpios_locked_mem_free_all(void) { } - -void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx, - unsigned int length) -{ - HPI_DEBUG_LOG(DEBUG, "mapping %d %s %08llx-%08llx %04llx len 0x%x\n", - idx, pci_dev->resource[idx].name, - (unsigned long long)pci_resource_start(pci_dev, idx), - (unsigned long long)pci_resource_end(pci_dev, idx), - (unsigned long long)pci_resource_flags(pci_dev, idx), length); - - if (!(pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM)) { - HPI_DEBUG_LOG(ERROR, "not an io memory resource\n"); - return NULL; - } - - if (length > pci_resource_len(pci_dev, idx)) { - HPI_DEBUG_LOG(ERROR, "resource too small for requested %d \n", - length); - return NULL; - } - - return ioremap(pci_resource_start(pci_dev, idx), length); -} diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index a62c3f1..370f39b 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -166,13 +166,4 @@ struct hpi_adapter { void __iomem *ap_remapped_mem_base[HPI_MAX_ADAPTER_MEM_SPACES]; }; -static inline void hpios_unmap_io(void __iomem *addr, - unsigned long size) -{ - iounmap(addr); -} - -void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx, - unsigned int length); - #endif -- 1.7.0.4

2 5

[alsa-devel] [PATCH 1/4] ALSA: usb-audio: parse more format descriptors with structs
by Daniel Mack 27 May '10

27 May '10

Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/endpoint.c | 11 +++++++---- sound/usb/format.c | 20 ++++++++++---------- sound/usb/format.h | 7 ++++--- 3 files changed, 21 insertions(+), 17 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index ef07a6d..4887342 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -158,8 +158,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) int i, altno, err, stream; int format = 0, num_channels = 0; struct audioformat *fp = NULL; - unsigned char *fmt, *csep; + unsigned char *csep; int num, protocol; + struct uac_format_type_i_continuous_descriptor *fmt; dev = chip->dev; @@ -256,8 +257,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno); continue; } - if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || - ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { + if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || + ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; @@ -268,7 +269,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * with the previous one, except for a larger packet size, but * is actually a mislabeled two-channel setting; ignore it. */ - if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && + if (fmt->bNrChannels == 1 && + fmt->bSubframeSize == 2 && + altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->formats == SNDRV_PCM_FMTBIT_S16_LE && protocol == UAC_VERSION_1 && diff --git a/sound/usb/format.c b/sound/usb/format.c index b87cf87..caaa3ef 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -278,12 +278,11 @@ err: * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, - struct audioformat *fp, - int format, void *_fmt, + struct audioformat *fp, int format, + struct uac_format_type_i_continuous_descriptor *fmt, struct usb_host_interface *iface) { struct usb_interface_descriptor *altsd = get_iface_desc(iface); - struct uac_format_type_i_discrete_descriptor *fmt = _fmt; int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; @@ -320,7 +319,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, switch (protocol) { case UAC_VERSION_1: fp->channels = fmt->bNrChannels; - ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); break; case UAC_VERSION_2: /* fp->channels is already set in this case */ @@ -392,12 +391,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, } int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface) + int format, struct uac_format_type_i_continuous_descriptor *fmt, + int stream, struct usb_host_interface *iface) { int err; - switch (fmt[3]) { + switch (fmt->bFormatType) { case UAC_FORMAT_TYPE_I: case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); @@ -407,10 +406,11 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, + fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -421,7 +421,7 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == UAC_FORMAT_TYPE_I && + if (fmt->bFormatType == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; diff --git a/sound/usb/format.h b/sound/usb/format.h index 8298c4e..387924f 100644 --- a/sound/usb/format.h +++ b/sound/usb/format.h @@ -1,8 +1,9 @@ #ifndef __USBAUDIO_FORMAT_H #define __USBAUDIO_FORMAT_H -int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream, - struct usb_host_interface *iface); +int snd_usb_parse_audio_format(struct snd_usb_audio *chip, + struct audioformat *fp, int format, + struct uac_format_type_i_continuous_descriptor *fmt, + int stream, struct usb_host_interface *iface); #endif /* __USBAUDIO_FORMAT_H */ -- 1.7.1

2 5

Re: [alsa-devel] [PATCH 17/17] sound: Add missing spin_unlock
by Takashi Iwai 27 May '10

27 May '10

At Wed, 26 May 2010 17:59:27 +0200 (CEST), Julia Lawall wrote: > > From: Julia Lawall <julia(a)diku.dk> > > Add a spin_unlock missing on the error path. > > The semantic match that finds this problem is as follows: > (http://coccinelle.lip6.fr/) > > // <smpl> > @@ > expression E1; > @@ > > * spin_lock(E1,...); > <+... when != E1 > if (...) { > ... when != E1 > * return ...; > } > ...+> > * spin_unlock(E1,...); > // </smpl> > > Signed-off-by: Julia Lawall <julia(a)diku.dk> Thanks, applied now. Takashi > --- > sound/mips/au1x00.c | 1 + > sound/oss/dmasound/dmasound_atari.c | 5 +++-- > 2 files changed, 4 insertions(+), 2 deletions(-) > > diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c > index 3e763d6..446cf97 100644 > --- a/sound/mips/au1x00.c > +++ b/sound/mips/au1x00.c > @@ -516,6 +516,7 @@ get the interrupt driven case to work efficiently */ > break; > if (i == 0x5000) { > printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n"); > + spin_unlock(&au1000->ac97_lock); > return 0; > } > > diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c > index 1f47741..13c2144 100644 > --- a/sound/oss/dmasound/dmasound_atari.c > +++ b/sound/oss/dmasound/dmasound_atari.c > @@ -1277,7 +1277,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > * (almost) like on the TT. > */ > write_sq_ignore_int = 0; > - return IRQ_HANDLED; > + goto out; > } > > if (!write_sq.active) { > @@ -1285,7 +1285,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > * the sq variables, so better don't do anything here. > */ > WAKE_UP(write_sq.sync_queue); > - return IRQ_HANDLED; > + goto out; > } > > /* Probably ;) one frame is finished. Well, in fact it may be that a > @@ -1322,6 +1322,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy) > /* We are not playing after AtaPlay(), so there > is nothing to play any more. Wake up a process > waiting for audio output to drain. */ > +out: > spin_unlock(&dmasound.lock); > return IRQ_HANDLED; > } >

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 27 May '10

27 May '10

The following changes since commit ad8332c1302bcb4f80d593fd3eb477be9d7f5604: ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies (2010-05-20 10:28:39 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 This fixes up the build of the i.MX SSI driver, a minor warning fix for SH and fix the dB scales for a bunch of Wolfson CODECs. Guennadi Liakhovetski (1): ASoC: fix uninitialised variable in siu_dai.c Mark Brown (3): ASoC: Fix dB scales for WM835x ASoC: Fix dB scales for WM8400 ASoC: Fix dB scales for WM8990 Stuart Longland (1): ASoC: Update Freescale i.MX SSI driver DMA parameter handling sound/soc/codecs/wm8350.c | 4 ++-- sound/soc/codecs/wm8400.c | 18 +++++++++--------- sound/soc/codecs/wm8990.c | 18 +++++++++--------- sound/soc/imx/imx-pcm-dma-mx2.c | 7 ++++--- sound/soc/sh/siu_dai.c | 2 ++ 5 files changed, 26 insertions(+), 23 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASOC: S5PC100: Enable AC97 support
by Jassi Brar 27 May '10

27 May '10

From: Jassi Brar <jassi.brar(a)samsung.com> The S5PC100 has the AC97 controller same as S3C6410. Simply enable the options to build the drivers for S5PC100 also. Signed-off-by: Jassi Brar <jassi.brar(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 4 ++-- sound/soc/s3c24xx/smdk_wm9713.c | 1 + 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 2a7cc22..aa112d3 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,6 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX + depends on ARCH_S3C2410 || ARCH_S3C64XX || ARCH_S5PC100 select S3C64XX_DMA if ARCH_S3C64XX help Say Y or M if you want to add support for codecs attached to @@ -120,7 +120,7 @@ config SND_S3C24XX_SOC_SIMTEC_HERMES config SND_SOC_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 + depends on SND_S3C24XX_SOC && (MACH_SMDK6410 || MACH_SMDKC100) select SND_SOC_WM9713 select SND_S3C_SOC_AC97 help diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c index 24fd39f..7dd85e5 100644 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ b/sound/soc/s3c24xx/smdk_wm9713.c @@ -25,6 +25,7 @@ static struct snd_soc_card smdk; * Default CFG switch settings to use this driver: * * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + * SMDKC100: Set CFG6 1-3 On, CFG7 1 On */ /* -- 1.6.2.5

3 2

[alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by VDR User 27 May '10

27 May '10

Hi. I had been mostly successfully using the snd_hda_intel driver from the stable kernel 2.6.34 for HDMI audio with my Nvidia GT220 video card. In an attempt to see if any recent changes to snd_hda_intel would fix a problem I've been having with "pcm_hw.c: snd_pcm_hw_delay() SNDRV_PCM_IOCTL_DELAY failed" I decided to download the latest snapshot. Unfortunately with these drivers I have no audio. All my settings, and everything have remained the same. The only noticable difference I've found is /sys/module/snd_hda_intel/parameters/probe_only is 0,0,0,0,0,0,0,0 in the snapshot drivers, and N,N,N,N,N,N,N,N in the 2.6.34 kernel drivers (which are compiled into the kernel). Any help? Thanks, Derek

2 2