ALSA: hda - Always allow basic audio irrespective of ELD info

commit 3dc86429032910bdf762adeb2969112bb303924c upstream.

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>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index 1c849902d7af825c5de50d28befe7cdcc3fd0ca9..b6c74642d9e2d5438f76a33545d2ac906eb1f03b 100644 (file)
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -604,21 +604,19 @@ 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)