gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.

Makes this work again:

gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
endianness=(int)4321, signed=(boolean)true, width=(int)16,
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
audioresample ! alsasink

diff --git a/ChangeLog b/ChangeLog
index 2b1bf4cea838852cc736b50b1ba147310bc5bb22..724a81f6e0667b0bb7fb1ff38becb59ce1e1787b 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,17 @@
+2006-01-17  Jan Schmidt  <thaytan@mad.scientist.com>
+
+       * gst-libs/gst/audio/gstbaseaudiosink.c:
+       (gst_base_audio_sink_render):
+         Fix playback of non-synchronised streams by assuming a rate
+         of 1.0 instead of a random one.
+
+         Makes this work again:
+
+         gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
+         endianness=(int)4321, signed=(boolean)true, width=(int)16,
+         depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
+         audioresample ! alsasink
+
 === release 0.10.2 ===
 
 2006-01-16  Thomas Vander Stichele <thomas at apestaart dot org>

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 6c697f653f0b8d61436ec8bf10bb405113fd2b17..ea0850006c1d46e9afb7214a10703f4cfd3f352b 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -405,7 +405,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
   guint size;
   guint samples;
   gint bps;
-  gdouble crate;
+  gdouble crate = 1.0;
   GstClockTime crate_num;
   GstClockTime crate_denom;
   GstClockTime cinternal, cexternal;
@@ -438,6 +438,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
    * sample ASAP */
   if (!GST_CLOCK_TIME_IS_VALID (time) || !bsink->sync) {
     render_offset = gst_base_audio_sink_get_offset (sink);
+    GST_DEBUG ("Buffer of size %u has no time. Using render_offset=%"
+        G_GUINT64_FORMAT, GST_BUFFER_SIZE (buf), render_offset);
     goto no_sync;
   }
 
@@ -488,12 +490,11 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
     GST_DEBUG ("resync");
   }
 
-no_sync:
   crate = ((gdouble) crate_num) / crate_denom;
   GST_DEBUG_OBJECT (sink,
       "internal %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", rate %g",
       cinternal, cexternal, crate);
-
+no_sync:
   /* clip length based on rate */
   samples = MIN (samples, samples / (crate * ABS (bsink->segment.rate)));