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authorAnssi Hannula <anssi.hannula@iki.fi>2010-12-07 20:19:23 +0100
committerTakashi Iwai <tiwai@suse.de>2010-12-08 08:36:20 +0100
commit0bbaee3a58c379c4f7bab9635c71d7bad9c422a2 (patch)
tree143b422842c08ce1deb32fc78529924a24b26823 /sound/pci/hda/hda_eld.c
parentALSA: hda - Always allow basic audio irrespective of ELD info (diff)
downloadlinux-0bbaee3a58c379c4f7bab9635c71d7bad9c422a2.tar.xz
linux-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/pci/hda/hda_eld.c')
-rw-r--r--sound/pci/hda/hda_eld.c4
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];