diff options
| -rw-r--r-- | ChangeLog | 7 | ||||
| -rw-r--r-- | configure.ac | 2 | ||||
| -rw-r--r-- | ext/faad/gstfaad.c | 55 |
3 files changed, 11 insertions, 53 deletions
@@ -1,3 +1,10 @@ +2007-07-24 Sebastian Dröge <slomo@circular-chaos.org> + + * configure.ac: + * ext/faad/gstfaad.c: (gst_faad_chain), (gst_faad_change_state): + Use the new buffer clipping function from gstaudio here and + require gst-plugins-base CVS. + 2007-07-23 Stefan Kost <ensonic@users.sf.net> * configure.ac: diff --git a/configure.ac b/configure.ac index 1311a102..68510f49 100644 --- a/configure.ac +++ b/configure.ac @@ -46,7 +46,7 @@ AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** dnl *** remove rtpmanager/equalizer stuff below when this is updated GST_REQ=0.10.13 -GSTPB_REQ=0.10.13 +GSTPB_REQ=0.10.13.1 dnl *** autotools stuff **** diff --git a/ext/faad/gstfaad.c b/ext/faad/gstfaad.c index 5863e590..3ab62b12 100644 --- a/ext/faad/gstfaad.c +++ b/ext/faad/gstfaad.c @@ -1132,56 +1132,6 @@ looks_like_valid_header (guint8 * input_data, guint input_size) return TRUE; } -/* - clips buffer to currently configured segment. Returns FALSE if the buffer - has to be dropped. -*/ - -static gboolean -clip_outgoing_buffer (GstFaad * faad, GstBuffer * buffer) -{ - gint64 start, stop, cstart, cstop, diff; - gboolean res = TRUE; - - if (faad->segment->format != GST_FORMAT_TIME) - goto beach; - - start = GST_BUFFER_TIMESTAMP (buffer); - stop = start + GST_BUFFER_DURATION (buffer); - - if (gst_segment_clip (faad->segment, GST_FORMAT_TIME, - start, stop, &cstart, &cstop)) { - diff = cstart - start; - if (diff > 0) { - GST_BUFFER_TIMESTAMP (buffer) = cstart; - GST_BUFFER_DURATION (buffer) -= diff; - - /* time->frames->bytes */ - diff = - faad->bps * faad->channels * GST_CLOCK_TIME_TO_FRAMES (diff, - faad->samplerate); - GST_BUFFER_DATA (buffer) += diff; - GST_BUFFER_SIZE (buffer) -= diff; - } - diff = stop - cstop; - if (diff > 0) { - GST_BUFFER_DURATION (buffer) -= diff; - /* time->frames->bytes */ - diff = - faad->bps * faad->channels * GST_CLOCK_TIME_TO_FRAMES (diff, - faad->samplerate); - /* update size */ - GST_BUFFER_SIZE (buffer) -= diff; - } - } else { - GST_DEBUG_OBJECT (faad, "buffer is outside configured segment"); - res = FALSE; - } - -beach: - return res; -} - static GstFlowReturn gst_faad_chain (GstPad * pad, GstBuffer * buffer) { @@ -1389,7 +1339,8 @@ gst_faad_chain (GstPad * pad, GstBuffer * buffer) faad->sum_dur_out += GST_BUFFER_DURATION (outbuf); GST_OBJECT_UNLOCK (faad); - if (clip_outgoing_buffer (faad, outbuf)) { + if ((outbuf = gst_audio_buffer_clip (outbuf, faad->segment, + faad->samplerate, faad->bps * faad->channels))) { GST_LOG_OBJECT (faad, "pushing buffer, off=%" G_GUINT64_FORMAT ", ts=%" GST_TIME_FORMAT, GST_BUFFER_OFFSET (outbuf), @@ -1493,7 +1444,7 @@ gst_faad_change_state (GstElement * element, GstStateChange transition) return GST_STATE_CHANGE_FAILURE; break; case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_segment_init (faad->segment, GST_FORMAT_UNDEFINED); + gst_segment_init (faad->segment, GST_FORMAT_TIME); break; default: break; |