ALSA: usb: refine delay information with USB frame counter
Existing code only updates the audio delay when URBs were
submitted/retired. This can introduce an uncertainty of 8ms
on the number of samples played out with the default settings,
and a lot more when URBs convey more packets to reduce the
interrupt rate and power consumption.
This patch relies on the USB frame counter to reduce the
uncertainty to less than 2ms worst-case. The delay information
essentially becomes independent of the URB size and number of
packets. This should help applications like PulseAudio which
require accurate audio timing. Clemens Ladisch reported
a decrease of mplayer's A-V difference from nrpacks down to at
most 1ms.
Thanks to Clemens for also pointing out that the implementation
of frame counters varies between different HCDs. Only the
8 lowest-bits are used to estimate the delay.
ALSA: hda: Conexant: Allow different output types to share DAC
Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.
BugLink: http://bugs.launchpad.net/bugs/817943 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Timur Tabi [Tue, 23 Aug 2011 21:48:26 +0000 (16:48 -0500)]
ASoC: MPC5200: replace of_device with platform_device
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma)
accordingly. This fixes a build break.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Takashi Iwai [Wed, 24 Aug 2011 15:57:44 +0000 (17:57 +0200)]
ALSA: hda - Prefer multi-io to speakers for realtek auto-parser
When the multi-io jacks are available, parse them first and assign DACs
before parsing speakers and headphones. This allows a better chance of
surround I/O in some desktops and laptops with limited DACs.
Takashi Iwai [Wed, 24 Aug 2011 15:27:52 +0000 (17:27 +0200)]
ALSA: hda - Fix Center/LFE mixer element creations for Realtek
The commit 23c09b00900c3fa6672148738cad29d6fc6ded7c
ALSA: hda - Support multiple speakers by Realtek auto-parser
changes the return value from alc_get_line_out_pfx(), and it breaks
the center/LFE mixer split check. The caller must test with a string
"CLFE" now.
Takashi Iwai [Wed, 24 Aug 2011 14:22:21 +0000 (16:22 +0200)]
ALSA: hda - Rewrite Lenovo X200 quirk with pincfg-fix using auto-parser
Introduce the pincfg table to patch_conexant.c for fixing up the extra
pin-configuration for auto-parser. As an example, Lenovo X200 model is
replaced with this new mechanism. (This also fixes the wrong mixer
elements for docking-station I/O in the previous model quirk
automagically.)
Stephen Warren [Tue, 23 Aug 2011 17:16:28 +0000 (11:16 -0600)]
ASoC: Tegra: wm8903 machine driver: Drop Ventana support
Board file support for Ventana is not yet mainlined, and probably won't
ever be given the move to Device-Tree. Consequently, the Ventana entry
is being removed from arch/arm/tools/mach-types in the next merge window,
since it was registered over a year ago.
This will also remove function machine_is_ventana(), which is used by
the ASoC Tegra WM8903 machine driver. This will cause compilation
failures. Drop Ventana support to resolve this.
Hopefully, in the not-too-distant future, tegra_wm8903.c will be able to
configure itself from Device-Tree, and hence we'll be able to re-instate
Ventana support just by creating a .dts file for the board.
Also note that Aebl support is in a similar boat. However, that board
isn't scheduled for deprecation for at least another 5 months, and
perhaps we will have completely removed non-Device-Tree support from
tegra_wm8903.c by then and/or adjusted mach-types policy.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Joseph Pentland [Tue, 23 Aug 2011 09:41:50 +0000 (10:41 +0100)]
ASoC: Add Springbank I/O card to Speyside Kconfig
Signed-off-by: Joseph Pentland <jp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are references in the code to 256 sources, so I tested it with 256 aplays,
of which the first and last with real data and the rest playing /dev/zero .
Also increase amount of page tables, so the default aplay size works.
Lu Guanqun [Wed, 24 Aug 2011 06:45:10 +0000 (14:45 +0800)]
ALSA: core: release the constraint check for replace ops
Suppose the ALSA card already has a number of MAX_USER_CONTROLS controls, and
the user wants to replace one, it should not fail at this condition check.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 23 Aug 2011 16:27:14 +0000 (18:27 +0200)]
ALSA: hda - Rewrite ALC662 asus-mode* models with fixups
Re-implement the asus-mode[1-8] quirks with the pin-config tables.
They are provided in case where BIOS is broken on the device, so it's
not enabled in PCI SSID lookup table. User needs to specify it via model
option explicitly if the driver doesn't work with the BIOS setup as is.
Takashi Iwai [Tue, 23 Aug 2011 16:16:13 +0000 (18:16 +0200)]
ALSA: hda - Add multi-headphone NIDs in multiout struct
For supporting both the multiple headphones and the multiple speakers,
add the new field in struct hda_multi_out, and evaluate in the standard
setup functions.
Takashi Iwai [Tue, 23 Aug 2011 15:28:55 +0000 (17:28 +0200)]
ALSA: hda - Restore VREF50 setup for ALC861-VD dallas/hp models
During the cleanup by commit 6727b12669f255dbf65b3d63c32cce1e3e967398,
the specific setups for dallas and hp models, using VREF50 for mic pins,
were lost. Fixed now.
Takashi Iwai [Tue, 23 Aug 2011 12:57:08 +0000 (14:57 +0200)]
ALSA: hda - Fix double-headphone/speaker paths for Cxt auto-parser
When multiple headphones or speakers are assigned but no individual
DACs are available, the driver should take the first HP/SPK DAC instead
of another primary output. The patch adds a bit-flag to dac field of
struct pin_dac_pair indicating that it's a slave DAC.
Takashi Iwai [Tue, 23 Aug 2011 10:41:09 +0000 (12:41 +0200)]
ALSA: hda - Update jack-sense info even when no automute is set
The internal states, jack_present and line_jack_present should be
updated upon unsolicited events even if no automute is set.
Otherwise the wrong state is referred when the automute behavior is
changed by the mixer control.
Takashi Iwai [Tue, 23 Aug 2011 10:36:28 +0000 (12:36 +0200)]
ALSA: hda - Fix output-path initialization for Realtek auto-parser
When the headphone or speaker output has no own DAC, initialize the path
using the primary DAC. Otherwise the path won't be set properly and
can result in the silence.
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Acked-by: Dong Aisheng <b29396@freescale.com> Reviewed-by: Wolfram Sang <w.sang@pengutronix.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dong Aisheng [Sun, 21 Aug 2011 15:45:40 +0000 (23:45 +0800)]
ASoC: mxs-saif: clear clk gate first before register setting
Saif needs clear clk gate first before writing registers or the write
will not success.
The original xx_get_mclk function clear clk gate after mclk setting
that may cause the former mclk setting unwork, then the real output
mclk maybe inaccurate.
Placing the clear before setting mclk to avoid such an issue.
We also have to clear clk gate in startup instead of in prepare function.
Signed-off-by: Dong Aisheng <b29396@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The first change is to add an of_node_put, since codec_np has previously
been allocated. The rest of the patch reorganizes the error handling code
so the only code executed is that which is needed.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
(
if (...) { ... when != of_node_put(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
of_node_put(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Acked-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
(
if (...) { ... when != of_node_put(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
of_node_put(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Acked-by: Alexander Sverdlin <subaparts@yandex.ru> Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Mon, 22 Aug 2011 14:22:41 +0000 (09:22 -0500)]
sound/soc/fsl/fsl_dma.c: add missing of_node_put
of_parse_phandle increments the reference count of np, so this should be
decremented before trying the next possibility.
Since we don't actually use np, we can decrement the reference count
immediately.
Reported-by: Julia Lawall <julia@diku.dk> Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 17 Aug 2011 01:03:51 +0000 (10:03 +0900)]
ASoC: Initial WM8962 DSP2 support
The WM8962 features a DSP providing a number of signal processing
features including HD Bass and Virtual Surround Sound (VSS). Enable
initial support for this, allowing users to enable and disable the
algorithms using the default coefficient sets. Further patches will
add support for runtime configuration of the DSP coefficients.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Sun, 21 Aug 2011 11:20:00 +0000 (12:20 +0100)]
ASoC: Ensure we only run Speyside WM8962 bias level callbacks once
We get called once per DAPM context but only need to run once. When DAPM
was serialized this was a series of noops but now it can run in parallel
we need to take proper care.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Sat, 20 Aug 2011 18:28:59 +0000 (19:28 +0100)]
ASoC: Optimise WM8996 no interrupt path
This occurs frequently if we are in edge triggered mode as we must poll the
interrupt status register until we get no more interrupts so it's worth
the effort - it means we skip writing null acknowledgements to the chip.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
For non-audio uses like accessory detection we can use a lower quality,
unregulated microphone bias, saving a little power. As the hardware can
manually enable and disable the biases we can select regulating mode
automatically with supply widgets connected to the biases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
(
if (...) { ... when != kfree(x)
when != x = E3
when != E3 = x
* return ...;
}
... when != x = E2
when != I(...,x,...) S
if (...) { ... when != x = E4
kfree(x); ... return ...; }
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sat, 20 Aug 2011 07:14:45 +0000 (09:14 +0200)]
ALSA: hda - Don't spew too many ELD errors
Currently HD-audio driver shows the all error ELD byte as an error
in the kernel message. This is annoying when the video driver doesn't
set the correct ELD from the beginning. e.g. radeon sends a zero-byte
data, but we still check ELD with the fixed 128 byte as a workaround
for some broken devices, it spews 128-times errors.
For avoiding this, the driver aborts reading when the first byte is
invalid. In such a case, the whole data is certainly invalid.
Takashi Iwai [Fri, 19 Aug 2011 07:10:29 +0000 (09:10 +0200)]
ALSA: hda - Allow different assoc numbers for multiple speakers
In snd_hda_parse_pin_def_config(), we checked the associated number
of speaker pins and accepts only one number exclusively. But many BIOS
seem to give different assoc number for surround speakers, thus we'd
better to accept all speaker pins no matter which assoc number, and sort
like done for the headphone pins.
Takashi Iwai [Fri, 19 Aug 2011 07:05:35 +0000 (09:05 +0200)]
ALSA: hda - Support multiple speakers by Realtek auto-parser
Add the support of multiple speakers by Realtek auto-parser.
When all speaker pins have individual DACs, create each speaker volume
control. Otherwise, create a bind-volume control for all speaker outs.
Takashi Iwai [Fri, 19 Aug 2011 06:30:53 +0000 (08:30 +0200)]
ALSA: usb-audio - Check the dB-range validity in the later read, too
When the initial check of dB-range failed due to the read error, try to
check again at the later read, too. When an invalid dB range is found,
remove TLV flags and notify the mixer info change.
Takashi Iwai [Fri, 19 Aug 2011 05:55:10 +0000 (07:55 +0200)]
ALSA: usb-audio - Fix missing mixer dB information
The recent fix for testing dB range at the mixer creation time seems
to cause regressions in some devices. In such devices, reading the dB
info at probing time gives an error, thus both dBmin and dBmax are still
zero, and TLV flag isn't set although the later read of dB info succeeds.
This patch adds a workaround for such a case by assuming that the later
read will succeed. In future, a similar test should be performed in a
case where a wrong dB range is seen even in the later read.
Axel Lin [Thu, 18 Aug 2011 07:31:04 +0000 (15:31 +0800)]
ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
This patch includes below small fixes:
1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove().
2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(),
as we will call regulator_bulk_free() in sta32x_remove().
3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove.
The i2c core will set the clientdata to NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Johannes Stezenbach <js@sig21.net> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Thu, 18 Aug 2011 12:02:47 +0000 (15:02 +0300)]
ASoC: omap: Fix build errors in ams-delta
Fix "error: too few arguments to function 'ams_delta_set_bias_level'"
build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC:
Add context parameter to card DAPM callbacks") by adding dapm context
to ams_delta_set_bias_level calls.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sascha Hauer [Wed, 17 Aug 2011 07:20:01 +0000 (09:20 +0200)]
ASoC: Fix check for symmetric rate enforcement
The ASoC core tries to not enforce symmetric rates when
two streams open simultaneously. It does so by checking
rtd->rate being zero. This works exactly once after booting
because it is not set to zero again when the streams close.
Fix this by setting rtd->rate when no active stream is left.
[This leads to lots of warnings about not enforcing the symmetry in some
situations as there's a race in the userspace API where we know we've
got two applications but don't know what rates they want to set.
-- broonie ]
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Timur Tabi [Tue, 16 Aug 2011 22:47:45 +0000 (18:47 -0400)]
ASoC: claim the IRQ when the fsl_ssi device is probed, not opened
The PowerPC Freescale SSI driver is claiming the IRQ when the IRQ when
the device is opened, which means that the /proc/interrupts entry for
the SSI exists only during playback or capture. This also meant that
the user won't know that the IRQ number is wrong until he tries to use
the device. Instead, we should claim the IRQ when the device is probed.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>