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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.
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parent
02a7b31f0e
commit
1650272b84
2 changed files with 23 additions and 7 deletions
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@ -757,6 +757,12 @@ theora_enc_chain (GstPad * pad, GstBuffer * buffer)
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running_time =
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gst_segment_to_running_time (&enc->segment, GST_FORMAT_TIME, timestamp);
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if ((gint64) running_time < 0) {
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GST_DEBUG_OBJECT (enc, "Dropping buffer, timestamp: %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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gst_buffer_unref (buffer);
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return GST_FLOW_OK;
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}
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/* see if we need to schedule a keyframe */
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GST_OBJECT_LOCK (enc);
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@ -51,6 +51,7 @@
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#include <gst/gsttagsetter.h>
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#include <gst/tag/tag.h>
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#include <gst/audio/multichannel.h>
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#include <gst/audio/audio.h>
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#include "gstvorbisenc.h"
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GST_DEBUG_CATEGORY_EXTERN (vorbisenc_debug);
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@ -893,7 +894,8 @@ gst_vorbis_enc_buffer_from_packet (GstVorbisEnc * vorbisenc,
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/* update the next timestamp, taking granulepos_offset and subgranule offset
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* into account */
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vorbisenc->next_ts =
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granulepos_to_timestamp_offset (vorbisenc, packet->granulepos);
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granulepos_to_timestamp_offset (vorbisenc, packet->granulepos) +
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vorbisenc->initial_ts;
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GST_BUFFER_DURATION (outbuf) =
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vorbisenc->next_ts - GST_BUFFER_TIMESTAMP (outbuf);
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@ -1097,18 +1099,25 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
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GstBuffer *buf1, *buf2, *buf3;
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gboolean first = FALSE;
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GstClockTime timestamp = GST_CLOCK_TIME_NONE;
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GstClockTime running_time = GST_CLOCK_TIME_NONE;
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vorbisenc = GST_VORBISENC (GST_PAD_PARENT (pad));
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if (!vorbisenc->setup)
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goto not_setup;
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timestamp =
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buffer = gst_audio_buffer_clip (buffer, &vorbisenc->segment,
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vorbisenc->frequency, 4 * vorbisenc->channels);
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if (buffer == NULL) {
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GST_DEBUG_OBJECT (vorbisenc, "Dropping buffer, out of segment");
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return GST_FLOW_OK;
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}
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running_time =
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gst_segment_to_running_time (&vorbisenc->segment, GST_FORMAT_TIME,
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GST_BUFFER_TIMESTAMP (buffer)) + vorbisenc->initial_ts;
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GST_BUFFER_TIMESTAMP (buffer));
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timestamp = running_time + vorbisenc->initial_ts;
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GST_DEBUG_OBJECT (vorbisenc, "Initial ts is %" GST_TIME_FORMAT,
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GST_TIME_ARGS (vorbisenc->initial_ts));
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if (!vorbisenc->header_sent) {
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/* Vorbis streams begin with three headers; the initial header (with
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most of the codec setup parameters) which is mandated by the Ogg
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@ -1168,10 +1177,11 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
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vorbisenc->next_ts = timestamp;
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vorbisenc->expected_ts = timestamp;
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vorbisenc->granulepos_offset = gst_util_uint64_scale
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(timestamp, vorbisenc->frequency, GST_SECOND);
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(running_time, vorbisenc->frequency, GST_SECOND);
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vorbisenc->subgranule_offset = 0;
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vorbisenc->subgranule_offset =
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vorbisenc->next_ts - granulepos_to_timestamp_offset (vorbisenc, 0);
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(vorbisenc->next_ts - vorbisenc->initial_ts) -
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granulepos_to_timestamp_offset (vorbisenc, 0);
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vorbisenc->header_sent = TRUE;
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first = TRUE;
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@ -1221,7 +1231,7 @@ gst_vorbis_enc_chain (GstPad * pad, GstBuffer * buffer)
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/* We need to round to the nearest whole number of samples, not just do
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* a truncating division here */
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vorbisenc->granulepos_offset = gst_util_uint64_scale
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(timestamp + GST_SECOND / vorbisenc->frequency / 2
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(running_time + GST_SECOND / vorbisenc->frequency / 2
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- vorbisenc->subgranule_offset, vorbisenc->frequency, GST_SECOND);
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vorbisenc->header_sent = TRUE;
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