Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (297 commits)
  ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
  ALSA: asihpi - HPI version 4.08
  ALSA: asihpi - Add volume mute controls
  ALSA: asihpi - Control name updates
  ALSA: asihpi - Use size_t for sizeof result
  ALSA: asihpi - Explicitly include mutex.h
  ALSA: asihpi - Add new node and message defines
  ALSA: asihpi - Make local function static
  ALSA: asihpi - Fix minor typos and spelling
  ALSA: asihpi - Remove unused structures, macros and functions
  ALSA: asihpi - Remove spurious adapter index check
  ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
  ALSA: asihpi - DSP code loader API now independent of OS
  ALSA: asihpi - Remove controlex structs and associated special data transfer code
  ALSA: asihpi - Increase request and response buffer sizes
  ALSA: asihpi - Give more meaningful name to hpi request message type
  ALSA: usb-audio - Add quirk for  Roland / BOSS BR-800
  ALSA: hda - Remove a superfluous argument of via_auto_init_output()
  ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
  ALSA: hda - Add documentation for codec-specific mixer controls
  ...

1 files changed, 24 insertions, 22 deletions

diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index e3e8531..28ce17d 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -580,43 +580,45 @@ void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld)
 #endif /* CONFIG_PROC_FS */
 
 /* update PCM info based on ELD */
-void hdmi_eld_update_pcm_info(struct hdmi_eld *eld, struct hda_pcm_stream *pcm,
-			      struct hda_pcm_stream *codec_pars)
+void snd_hdmi_eld_update_pcm_info(struct hdmi_eld *eld,
+			      struct hda_pcm_stream *hinfo)
 {
+	u32 rates;
+	u64 formats;
+	unsigned int maxbps;
+	unsigned int channels_max;
 	int i;
 
 	/* 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;
+	rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
+		SNDRV_PCM_RATE_48000;
+	formats = SNDRV_PCM_FMTBIT_S16_LE;
+	maxbps = 16;
+	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;
+		rates |= a->rates;
+		if (a->channels > channels_max)
+			channels_max = a->channels;
 		if (a->format == AUDIO_CODING_TYPE_LPCM) {
 			if (a->sample_bits & AC_SUPPCM_BITS_20) {
-				pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
-				if (pcm->maxbps < 20)
-					pcm->maxbps = 20;
+				formats |= SNDRV_PCM_FMTBIT_S32_LE;
+				if (maxbps < 20)
+					maxbps = 20;
 			}
 			if (a->sample_bits & AC_SUPPCM_BITS_24) {
-				pcm->formats |= SNDRV_PCM_FMTBIT_S32_LE;
-				if (pcm->maxbps < 24)
-					pcm->maxbps = 24;
+				formats |= SNDRV_PCM_FMTBIT_S32_LE;
+				if (maxbps < 24)
+					maxbps = 24;
 			}
 		}
 	}
 
-	if (!codec_pars)
-		return;
-
 	/* restrict the parameters by the values the codec provides */
-	pcm->rates &= codec_pars->rates;
-	pcm->formats &= codec_pars->formats;
-	pcm->channels_max = min(pcm->channels_max, codec_pars->channels_max);
-	pcm->maxbps = min(pcm->maxbps, codec_pars->maxbps);
+	hinfo->rates &= rates;
+	hinfo->formats &= formats;
+	hinfo->maxbps = min(hinfo->maxbps, maxbps);
+	hinfo->channels_max = min(hinfo->channels_max, channels_max);
 }