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flacenc: 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 10:59:13 +01:00
parent 1dcc883261
commit b18d8b085a
2 changed files with 34 additions and 0 deletions
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32
ext/flac/gstflacenc.c
View file

@ -408,6 +408,8 @@ gst_flac_enc_start (GstAudioEncoder * enc)
flacenc->eos = FALSE; flacenc->eos = FALSE;
flacenc->tags = gst_tag_list_new_empty (); flacenc->tags = gst_tag_list_new_empty ();
flacenc->toc = NULL; flacenc->toc = NULL;
flacenc->samples_in = 0;
flacenc->samples_out = 0;
return TRUE; return TRUE;
} }
@ -1124,6 +1126,8 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
GstFlacEnc *flacenc; GstFlacEnc *flacenc;
GstBuffer *outbuf; GstBuffer *outbuf;
GstSegment *segment;
GstClockTime duration;
flacenc = GST_FLAC_ENC (client_data); flacenc = GST_FLAC_ENC (client_data);
@ -1159,6 +1163,28 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (flacenc), outbuf); ret = gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (flacenc), outbuf);
} else { } else {
/* regular frame data, pass to base class */ /* regular frame data, pass to base class */
if (flacenc->eos && flacenc->samples_in == flacenc->samples_out + 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 (flacenc);
if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
GST_DEBUG_OBJECT (flacenc,
"No stop time and partial frame, updating segment");
duration =
gst_util_uint64_scale (flacenc->samples_out + samples,
GST_SECOND,
FLAC__stream_encoder_get_sample_rate (flacenc->encoder));
segment->stop = segment->start + duration;
GST_DEBUG_OBJECT (flacenc, "new output segment %" GST_SEGMENT_FORMAT,
segment);
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (flacenc),
gst_event_new_segment (segment));
}
}
GST_LOG ("Pushing buffer: samples=%u, size=%u, pos=%" G_GUINT64_FORMAT, GST_LOG ("Pushing buffer: samples=%u, size=%u, pos=%" G_GUINT64_FORMAT,
samples, (guint) bytes, flacenc->offset); samples, (guint) bytes, flacenc->offset);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (flacenc), ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (flacenc),
@ -1229,6 +1255,11 @@ gst_flac_enc_sink_event (GstAudioEncoder * enc, GstEvent * event)
} }
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
break; break;
case GST_EVENT_SEGMENT:
flacenc->samples_in = 0;
flacenc->samples_out = 0;
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
break;
default: default:
ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event); ret = GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (enc, event);
break; break;
@ -1328,6 +1359,7 @@ gst_flac_enc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
res = FLAC__stream_encoder_process_interleaved (flacenc->encoder, res = FLAC__stream_encoder_process_interleaved (flacenc->encoder,
(const FLAC__int32 *) data, samples); (const FLAC__int32 *) data, samples);
flacenc->samples_in += samples;
g_free (data); g_free (data);

2
ext/flac/gstflacenc.h
View file

@ -59,6 +59,8 @@ struct _GstFlacEnc {
GstTagList * tags; GstTagList * tags;
GstToc * toc; GstToc * toc;
guint64 samples_in;
guint64 samples_out;
gboolean eos; gboolean eos;
/* queue headers until we have them all so we can add streamheaders to caps */ /* queue headers until we have them all so we can add streamheaders to caps */
gboolean got_headers; gboolean got_headers;