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| author | Anssi Hannula <anssi.hannula@iki.fi> | 2010-12-07 21:19:23 +0200 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2010-12-08 08:36:20 +0100 |
| commit | 0bbaee3a58c379c4f7bab9635c71d7bad9c422a2 (patch) | |
| tree | 143b422842c08ce1deb32fc78529924a24b26823 /sound | |
| parent | 3dc86429032910bdf762adeb2969112bb303924c (diff) | |
| download | lwn-0bbaee3a58c379c4f7bab9635c71d7bad9c422a2.tar.gz lwn-0bbaee3a58c379c4f7bab9635c71d7bad9c422a2.zip | |
ALSA: hda - Reset sample sizes and max bitrates when reading ELD
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.
The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.
The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.
Fix that by always clearing sample_bits and max_bitrate when reading
SADs.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'sound')
| -rw-r--r-- | sound/pci/hda/hda_eld.c | 4 |
1 files changed, 3 insertions, 1 deletions
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 009031fae2ba..4a663471dadc 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -189,6 +189,9 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, a->channels = GRAB_BITS(buf, 0, 0, 3); a->channels++; + a->sample_bits = 0; + a->max_bitrate = 0; + a->format = GRAB_BITS(buf, 0, 3, 4); switch (a->format) { case AUDIO_CODING_TYPE_REF_STREAM_HEADER: @@ -198,7 +201,6 @@ static void hdmi_update_short_audio_desc(struct cea_sad *a, case AUDIO_CODING_TYPE_LPCM: val = GRAB_BITS(buf, 2, 0, 3); - a->sample_bits = 0; for (i = 0; i < 3; i++) if (val & (1 << i)) a->sample_bits |= cea_sample_sizes[i + 1]; |