At Tue, 7 Dec 2010 20:56:19 +0200, Anssi Hannula wrote:
Commit bbbe33900d1f3c added functionality to restrict PCM parameters based on ELD info (derived from EDID data) of the audio sink.
However, according to CEA-861-D no SAD is needed for basic audio (32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a basic audio flag in the CEA EDID Extension.
The flag is not present in ELD. However, as all audio capable sinks are required to support basic audio, we can assume it to be always available.
Fix allowed audio formats with sinks that have SADs (Short Audio Descriptors) which do not completely overlap with the basic audio formats (there are no reports of affected devices so far) by always assuming that basic audio is supported.
Reported-by: Stephen Warren swarren@nvidia.com Signed-off-by: Anssi Hannula anssi.hannula@iki.fi
Tested on 2.6.36 and I saw no obvious regressions.
The issue affects stable 2.6.36 as well, but as there are no reports of affected devices, I'll leave the decision on whether to send this to stable@ for the ALSA subsystem maintainers.
I'm inclined for adding stable in this case, because this is more "safer" move for users.
Will the patch "ALSA: hda - Do not wrongly restrict min_channels based on ELD" be resent or shall I apply it as is now together with this?
thanks,
Takashi
sound/pci/hda/hda_eld.c | 16 +++++++--------- 1 files changed, 7 insertions(+), 9 deletions(-)
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 47ef8aa..009031f 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -598,21 +598,19 @@ void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm, { int i;
- pcm->rates = 0;
- pcm->formats = 0;
- pcm->maxbps = 0;
- pcm->channels_max = 0;
- /* assume basic audio support (the basic audio flag is not in ELD;
* however, all audio capable sinks are required to support basic* audio) */- pcm->rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000;
- pcm->formats = SNDRV_PCM_FMTBIT_S16_LE;
- pcm->maxbps = 16;
- pcm->channels_max = 2; for (i = 0; i < eld->sad_count; i++) { struct cea_sad *a = &eld->sad[i]; pcm->rates |= a->rates; if (a->channels > pcm->channels_max) pcm->channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) {
if (a->sample_bits & AC_SUPPCM_BITS_16) {pcm->formats |= SNDRV_PCM_FMTBIT_S16_LE;if (pcm->maxbps < 16)pcm->maxbps = 16;} if (a->sample_bits & AC_SUPPCM_BITS_20) { pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE; if (pcm->maxbps < 20)-- 1.7.3