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  1. 18 4月, 2013 4 次提交
    • D
      ALSA: add DSD formats · ef7a4f97
      由 Daniel Mack 提交于 
      This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital

DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.

The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).

DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:

                                                  configured hardware
        176.4KHz   352.8kHz   705.6KHz     <----       sample rate

8-bit                2.8MHz     5.6MHz
16-bit    2.8Mhz     5.6MHz    11.2MHz

         `-----------------------------'
             actual DSD sample rates
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      ef7a4f97
    • T
      ALSA: hda - Disable the sanity check in snd_hda_add_pincfg() · d5657ec9
      由 Takashi Iwai 提交于 
      When pin default configs are overridden via patch option, these are
evaluated before fixups are applied.  Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.

We may reorder the execution, but an easier fix is simply to disable
this sanity check.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      d5657ec9
    • W
      ALSA: hda - fix error return code in patch_alc662() · 6134b1a2
      由 Wei Yongjun 提交于 
      Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.
Signed-off-by: NWei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      6134b1a2
    • T
      ALSA: hda - Don't call vmaster hook when bus->shutdown is set · 594813ff
      由 Takashi Iwai 提交于 
      The flag bus->shutdown implies that the control elements might have
been already destroyed.  When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.

This patch just adds a check of the flag in the caller side for
avoiding such a crash.

Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily.  So let it be for a while.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      594813ff
  3. 17 4月, 2013 1 次提交
  5. 16 4月, 2013 3 次提交
    • T
      ALSA: hda - Use the primary DAC for all aamix outputs · 5ead56f2
      由 Takashi Iwai 提交于 
      When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default.  Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      5ead56f2
    • T
      ALSA: hda - Fix aamix activation with loopback control on VIA codecs · 65033cc8
      由 Takashi Iwai 提交于 
      When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not.  But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.

Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing.  This leaves the aamix path even though the
loopback control is turned off.

This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
  true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback
Reported-by: NLydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      65033cc8
    • D
      ALSA: hda - Add codec delay to the capture time stamp. · ae03bbb8
      由 Dylan Reid 提交于 
      For capture, the delay through the codec contributes to the time stamp
of the sample recorded at the A to D.  Rename the codec time stamp
function appropriately.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      ae03bbb8
  7. 15 4月, 2013 3 次提交
  9. 13 4月, 2013 1 次提交
    • C
      ALSA: usb: Add quirk for 192KHz recording on E-Mu devices · 1539d4f8
      由 Calvin Owens 提交于 
      When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.

Userspace expected:  L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1

Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.

Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.
Signed-off-by: NCalvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      1539d4f8
  11. 12 4月, 2013 6 次提交
  13. 11 4月, 2013 4 次提交
  15. 10 4月, 2013 3 次提交
  17. 09 4月, 2013 2 次提交
  19. 08 4月, 2013 2 次提交
  21. 07 4月, 2013 1 次提交
  23. 05 4月, 2013 8 次提交
  25. 04 4月, 2013 2 次提交