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speexenc: update output segment stop time to match clipped samples

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This will let oggmux generate a granpos on the last page that properly
represents the clipped samples at the end of the stream.
This commit is contained in:
Vincent Penquerc'h 2014-06-10 10:04:07 +01:00
parent b18d8b085a
commit ca9528d0b0
2 changed files with 30 additions and 0 deletions
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29
ext/speex/gstspeexenc.c
View file

@ -253,6 +253,7 @@ gst_speex_enc_start (GstAudioEncoder * benc)
speex_bits_init (&enc->bits); speex_bits_init (&enc->bits);
enc->tags = gst_tag_list_new_empty (); enc->tags = gst_tag_list_new_empty ();
enc->header_sent = FALSE; enc->header_sent = FALSE;
enc->encoded_samples = 0;
return TRUE; return TRUE;
} }
@ -511,6 +512,9 @@ gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
} }
break; break;
} }
case GST_EVENT_SEGMENT:
enc->encoded_samples = 0;
break;
default: default:
break; break;
} }
@ -530,6 +534,8 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
gsize bsize, size; gsize bsize, size;
GstBuffer *outbuf; GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
GstSegment *segment;
GstClockTime duration;
if (G_LIKELY (buf)) { if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ); gst_buffer_map (buf, &map, GST_MAP_READ);
@ -539,6 +545,28 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
if (G_UNLIKELY (bsize % bytes)) { if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples"); GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
/* If encoding part of a frame, and we have no set stop time on
* the output segment, we update the segment stop time to reflect
* the last sample. This will let oggmux set the last page's
* granpos to tell a decoder the dummy samples should be clipped.
*/
segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT,
segment);
if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
int input_samples = bsize / (enc->channels * 2);
GST_DEBUG_OBJECT (enc,
"No stop time and partial frame, updating segment");
duration =
gst_util_uint64_scale (enc->encoded_samples + input_samples,
GST_SECOND, enc->rate);
segment->stop = segment->start + duration;
GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
segment);
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
gst_event_new_segment (segment));
}
size = ((bsize / bytes) + 1) * bytes; size = ((bsize / bytes) + 1) * bytes;
data0 = data = g_malloc0 (size); data0 = data = g_malloc0 (size);
memcpy (data, bdata, bsize); memcpy (data, bdata, bsize);
@ -603,6 +631,7 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc), ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, samples); outbuf, samples);
enc->encoded_samples += frame_size;
done: done:
g_free (data0); g_free (data0);

1
ext/speex/gstspeexenc.h
View file

@ -80,6 +80,7 @@ struct _GstSpeexEnc {
gint rate; gint rate;
gboolean header_sent; gboolean header_sent;
guint64 encoded_samples;
GstTagList *tags; GstTagList *tags;