Commit ad67773c authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠
Browse files

ext/flac/gstflacdec.*: Make flac-in-ogg work (#352100).

Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
(gst_flac_dec_write), (gst_flac_dec_loop),
(gst_flac_dec_sink_event), (gst_flac_dec_chain),
(gst_flac_dec_src_query):
* ext/flac/gstflacdec.h:
Make flac-in-ogg work (#352100).
parent 76ff577a
2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
(gst_flac_dec_write), (gst_flac_dec_loop),
(gst_flac_dec_sink_event), (gst_flac_dec_chain),
(gst_flac_dec_src_query):
* ext/flac/gstflacdec.h:
Make flac-in-ogg work (#352100).
2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
......
......@@ -444,9 +444,11 @@ gst_flac_dec_scan_got_frame (GstFlacDec * flacdec, guint8 * data, guint size,
*last_sample_num = val; /* FIXME: + length of last block in samples */
}
GST_DEBUG_OBJECT (flacdec, "last sample %" G_GINT64_FORMAT " = %"
GST_TIME_FORMAT, *last_sample_num,
GST_TIME_ARGS (*last_sample_num * GST_SECOND / flacdec->sample_rate));
if (flacdec->sample_rate > 0) {
GST_DEBUG_OBJECT (flacdec, "last sample %" G_GINT64_FORMAT " = %"
GST_TIME_FORMAT, *last_sample_num,
GST_TIME_ARGS (*last_sample_num * GST_SECOND / flacdec->sample_rate));
}
return TRUE;
}
......@@ -805,6 +807,13 @@ gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
goto done;
}
if (flacdec->cur_granulepos != GST_BUFFER_OFFSET_NONE) {
/* this should be fine since it should be one flac frame per ogg packet */
flacdec->segment.last_stop = flacdec->cur_granulepos - samples;
GST_LOG_OBJECT (flacdec, "granulepos = %" G_GINT64_FORMAT ", samples = %u",
flacdec->cur_granulepos, samples);
}
GST_BUFFER_TIMESTAMP (outbuf) =
gst_util_uint64_scale_int (flacdec->segment.last_stop, GST_SECOND,
frame->header.sample_rate);
......@@ -843,7 +852,7 @@ gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
if (!flacdec->seeking) {
GST_DEBUG ("pushing %d samples at offset %" G_GINT64_FORMAT
"(%" GST_TIME_FORMAT " + %" GST_TIME_FORMAT ")",
" (%" GST_TIME_FORMAT " + %" GST_TIME_FORMAT ")",
samples, GST_BUFFER_OFFSET (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
......@@ -908,6 +917,8 @@ gst_flac_dec_loop (GstPad * sinkpad)
flacdec->init = FALSE;
}
flacdec->cur_granulepos = GST_BUFFER_OFFSET_NONE;
flacdec->last_flow = GST_FLOW_OK;
GST_LOG_OBJECT (flacdec, "processing single");
......@@ -1032,13 +1043,32 @@ gst_flac_dec_sink_event (GstPad * pad, GstEvent * event)
}
case GST_EVENT_NEWSEGMENT:{
GstFormat fmt;
gboolean update;
gdouble rate, applied_rate;
gint64 cur, stop, time;
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
&fmt, &cur, &stop, &time);
gst_event_parse_new_segment (event, NULL, NULL, &fmt, NULL, NULL, NULL);
if (fmt == GST_FORMAT_TIME) {
GstFormat dformat = GST_FORMAT_DEFAULT;
GST_DEBUG_OBJECT (dec, "newsegment event in TIME format => framed");
dec->framed = TRUE;
res = gst_pad_push_event (dec->srcpad, event);
dec->need_newsegment = FALSE;
/* this won't work for the first newsegment event though ... */
if (gst_flac_dec_convert_src (dec->srcpad, GST_FORMAT_TIME, cur,
&dformat, &cur) && cur != -1 &&
gst_flac_dec_convert_src (dec->srcpad, GST_FORMAT_TIME, stop,
&dformat, &stop) && stop != -1) {
gst_segment_set_newsegment_full (&dec->segment, update, rate,
applied_rate, dformat, cur, stop, time);
GST_DEBUG_OBJECT (dec, "segment %" GST_SEGMENT_FORMAT, &dec->segment);
} else {
GST_WARNING_OBJECT (dec, "couldn't convert time => samples");
}
} else if (fmt == GST_FORMAT_BYTES || TRUE) {
GST_DEBUG_OBJECT (dec, "newsegment event in %s format => not framed",
gst_format_get_name (fmt));
......@@ -1075,6 +1105,7 @@ gst_flac_dec_chain (GstPad * pad, GstBuffer * buf)
{
FLAC__SeekableStreamDecoderState s;
GstFlacDec *dec;
gboolean got_audio_frame;
dec = GST_FLAC_DEC (GST_PAD_PARENT (pad));
......@@ -1099,6 +1130,22 @@ gst_flac_dec_chain (GstPad * pad, GstBuffer * buf)
FLAC__stream_decoder_flush (dec->stream_decoder);
}
if (dec->framed) {
gint64 unused;
/* check if this is a flac audio frame (rather than a header or junk) */
got_audio_frame = gst_flac_dec_scan_got_frame (dec, GST_BUFFER_DATA (buf),
GST_BUFFER_SIZE (buf), &unused);
/* oggdemux will set granulepos in OFFSET_END instead of timestamp */
if (got_audio_frame && !GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
dec->cur_granulepos = GST_BUFFER_OFFSET_END (buf);
}
} else {
dec->cur_granulepos = GST_BUFFER_OFFSET_NONE;
got_audio_frame = TRUE;
}
gst_adapter_push (dec->adapter, buf);
buf = NULL;
......@@ -1109,8 +1156,10 @@ gst_flac_dec_chain (GstPad * pad, GstBuffer * buf)
* interface is a bit dumb it seems (if we don't have as much data as
* it wants it will call our read callback repeatedly and the only
* way to stop that is to error out or EOS, which will affect the
* decoder state). Requiring MAX_BLOCK_SIZE should make sure it
* always gets enough data to decode at least one block */
* decoder state). And the decoder seems to always ask for MAX_BLOCK_SIZE
* bytes rather than the max. block size from the header). Requiring
* MAX_BLOCK_SIZE bytes here should make sure it always gets enough data
* to decode at least one block */
while (gst_adapter_available (dec->adapter) >= FLAC__MAX_BLOCK_SIZE &&
dec->last_flow == GST_FLOW_OK) {
GST_LOG_OBJECT (dec, "%u bytes available",
......@@ -1120,13 +1169,15 @@ gst_flac_dec_chain (GstPad * pad, GstBuffer * buf)
break;
}
}
} else {
} else if (dec->framed && got_audio_frame) {
/* framed - there should always be enough data to decode something */
GST_LOG_OBJECT (dec, "%u bytes available",
gst_adapter_available (dec->adapter));
if (!FLAC__stream_decoder_process_single (dec->stream_decoder)) {
GST_DEBUG_OBJECT (dec, "process_single failed");
}
} else {
GST_DEBUG_OBJECT (dec, "don't have all headers yet");
}
return dec->last_flow;
......@@ -1367,7 +1418,7 @@ gst_flac_dec_src_query (GstPad * pad, GstQuery * query)
if (fmt == GST_FORMAT_TIME && peer && gst_pad_query (peer, query)) {
gst_query_parse_duration (query, NULL, &len);
GST_DEBUG_OBJECT (flacdec, "peer returned duration %" GST_TIME_FORMAT,
len);
GST_TIME_ARGS (len));
res = TRUE;
goto done;
}
......
......@@ -73,6 +73,8 @@ struct _GstFlacDec {
/* from the stream info, needed for scanning */
guint16 min_blocksize;
guint16 max_blocksize;
gint64 cur_granulepos; /* only used in framed mode (flac-in-ogg) */
};
struct _GstFlacDecClass {
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment