Commit 1650272b authored by Zaheer Abbas Merali's avatar Zaheer Abbas Merali

vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment

With vorbisenc, compute the granulepos with running time and clip incoming
buffers to segment.
With theoraenc, drop out of segment buffers.
parent 02a7b31f
......@@ -757,6 +757,12 @@ theora_enc_chain (GstPad * pad, GstBuffer * buffer)
running_time =
gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME, timestamp);
if ((gint64) running_time < 0) {
GST_DEBUG_OBJECT (enc, "Dropping buffer, timestamp: %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
/* see if we need to schedule a keyframe */
GST_OBJECT_LOCK (enc);
......
......@@ -51,6 +51,7 @@
#include <gst/gsttagsetter.h>
#include <gst/tag/tag.h>
#include <gst/audio/multichannel.h>
#include <gst/audio/audio.h>
#include "gstvorbisenc.h"
GST_DEBUG_CATEGORY_EXTERN (vorbisenc_debug);
......@@ -893,7 +894,8 @@ gst_vorbis_enc_buffer_from_packet (GstVorbisEnc * vorbisenc,
/* update the next timestamp, taking granulepos_offset and subgranule offset
* into account */
vorbisenc->next_ts =
granulepos_to_timestamp_offset (vorbisenc, packet->granulepos);
granulepos_to_timestamp_offset (vorbisenc, packet->granulepos) +
vorbisenc->initial_ts;
GST_BUFFER_DURATION (outbuf) =
vorbisenc->next_ts - GST_BUFFER_TIMESTAMP (outbuf);
......@@ -1097,18 +1099,25 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
GstBuffer *buf1, *buf2, *buf3;
gboolean first = FALSE;
GstClockTime timestamp = GST_CLOCK_TIME_NONE;
GstClockTime running_time = GST_CLOCK_TIME_NONE;
vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
if (!vorbisenc->setup)
goto not_setup;
timestamp =
buffer = gst_audio_buffer_clip (buffer, &vorbisenc->segment,
vorbisenc->frequency, 4 * vorbisenc->channels);
if (buffer == NULL) {
GST_DEBUG_OBJECT (vorbisenc, "Dropping buffer, out of segment");
return GST_FLOW_OK;
}
running_time =
gst_segment_to_running_time (&vorbisenc->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buffer)) + vorbisenc->initial_ts;
GST_BUFFER_TIMESTAMP (buffer));
timestamp = running_time + vorbisenc->initial_ts;
GST_DEBUG_OBJECT (vorbisenc, "Initial ts is %" GST_TIME_FORMAT,
GST_TIME_ARGS (vorbisenc->initial_ts));
if (!vorbisenc->header_sent) {
/* Vorbis streams begin with three headers; the initial header (with
most of the codec setup parameters) which is mandated by the Ogg
......@@ -1168,10 +1177,11 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
vorbisenc->next_ts = timestamp;
vorbisenc->expected_ts = timestamp;
vorbisenc->granulepos_offset = gst_util_uint64_scale
(timestamp, vorbisenc->frequency, GST_SECOND);
(running_time, vorbisenc->frequency, GST_SECOND);
vorbisenc->subgranule_offset = 0;
vorbisenc->subgranule_offset =
vorbisenc->next_ts - granulepos_to_timestamp_offset (vorbisenc, 0);
(vorbisenc->next_ts - vorbisenc->initial_ts) -
granulepos_to_timestamp_offset (vorbisenc, 0);
vorbisenc->header_sent = TRUE;
first = TRUE;
......@@ -1221,7 +1231,7 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
/* We need to round to the nearest whole number of samples, not just do
* a truncating division here */
vorbisenc->granulepos_offset = gst_util_uint64_scale
(timestamp + GST_SECOND / vorbisenc->frequency / 2
(running_time + GST_SECOND / vorbisenc->frequency / 2
- vorbisenc->subgranule_offset, vorbisenc->frequency, GST_SECOND);
vorbisenc->header_sent = TRUE;
......
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