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aacparse: streamline src caps setting

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In particular, also set src caps whenever changes in stream warrant doing so.
This commit is contained in:
Mark Nauwelaerts 2010-10-12 18:07:49 +02:00 committed by Tim-Philipp Müller
parent 46ea4eaa1f
commit 5369b97246
2 changed files with 46 additions and 18 deletions
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62
gst/audioparsers/gstaacparse.c
View file

@ -234,11 +234,11 @@ gst_aacparse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
} }
s = gst_caps_get_structure (src_caps, 0); s = gst_caps_get_structure (src_caps, 0);
if (!gst_structure_has_field (s, "rate") && aacparse->sample_rate > 0) if (aacparse->sample_rate > 0)
gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL); gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
if (!gst_structure_has_field (s, "channels") && aacparse->channels > 0) if (aacparse->channels > 0)
gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL); gst_structure_set (s, "channels", G_TYPE_INT, aacparse->channels, NULL);
if (!gst_structure_has_field (s, "stream-format") && stream_format) if (stream_format)
gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL); gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps); GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
@ -300,6 +300,10 @@ gst_aacparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
gst_base_parse_set_passthrough (parse, TRUE); gst_base_parse_set_passthrough (parse, TRUE);
} else } else
return FALSE; return FALSE;
/* caps info overrides */
gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
gst_structure_get_int (structure, "channels", &aacparse->channels);
} }
return TRUE; return TRUE;
@ -391,6 +395,25 @@ gst_aacparse_check_adts_frame (GstAacParse * aacparse,
return FALSE; return FALSE;
} }
/* caller ensure sufficient data */
static inline void
gst_aacparse_parse_adts_header (GstAacParse * aacparse, const guint8 * data,
gint * rate, gint * channels, gint * object, gint * version)
{
if (rate) {
gint sr_idx = (data[2] & 0x3c) >> 2;
*rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
}
if (channels)
*channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
if (version)
*version = (data[1] & 0x08) ? 2 : 4;
if (object)
*object = (data[2] & 0xc0) >> 6;
}
/** /**
* gst_aacparse_detect_stream: * gst_aacparse_detect_stream:
@ -451,22 +474,19 @@ gst_aacparse_detect_stream (GstAacParse * aacparse,
if (gst_aacparse_check_adts_frame (aacparse, data, avail, if (gst_aacparse_check_adts_frame (aacparse, data, avail,
framesize, &need_data)) { framesize, &need_data)) {
gint sr_idx; gint rate, channels;
GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize); GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
aacparse->header_type = DSPAAC_HEADER_ADTS; aacparse->header_type = DSPAAC_HEADER_ADTS;
sr_idx = (data[2] & 0x3c) >> 2; gst_aacparse_parse_adts_header (aacparse, data, &rate, &channels,
&aacparse->object_type, &aacparse->mpegversion);
aacparse->sample_rate = gst_aacparse_get_sample_rate_from_index (sr_idx);
aacparse->mpegversion = (data[1] & 0x08) ? 2 : 4;
aacparse->object_type = (data[2] & 0xc0) >> 6;
aacparse->channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse), gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse),
aacparse->sample_rate, 1024, 2, 2); rate, 1024, 2, 2);
GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d", GST_DEBUG ("ADTS: samplerate %d, channels %d, objtype %d, version %d",
aacparse->sample_rate, aacparse->channels, aacparse->object_type); rate, channels, aacparse->object_type, aacparse->mpegversion);
return TRUE; return TRUE;
} else if (need_data) { } else if (need_data) {
@ -630,16 +650,27 @@ gst_aacparse_parse_frame (GstBaseParse * parse, GstBuffer * buffer)
{ {
GstAacParse *aacparse; GstAacParse *aacparse;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
gint rate, channels;
aacparse = GST_AACPARSE (parse); aacparse = GST_AACPARSE (parse);
if (!aacparse->src_caps_set) { gst_aacparse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer),
&rate, &channels, NULL, NULL);
GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
if (G_UNLIKELY (rate != aacparse->sample_rate
|| channels != aacparse->channels)) {
aacparse->sample_rate = rate;
aacparse->channels = channels;
if (!gst_aacparse_set_src_caps (aacparse, if (!gst_aacparse_set_src_caps (aacparse,
GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) { GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) {
/* If linking fails, we need to return appropriate error */ /* If linking fails, we need to return appropriate error */
ret = GST_FLOW_NOT_LINKED; ret = GST_FLOW_NOT_LINKED;
} }
aacparse->src_caps_set = TRUE;
gst_base_parse_set_frame_props (GST_BASE_PARSE (aacparse),
aacparse->sample_rate, 1024, 2, 2);
} }
gst_buffer_set_caps (buffer, GST_PAD_CAPS (parse->srcpad)); gst_buffer_set_caps (buffer, GST_PAD_CAPS (parse->srcpad));
@ -662,7 +693,6 @@ gst_aacparse_start (GstBaseParse * parse)
aacparse = GST_AACPARSE (parse); aacparse = GST_AACPARSE (parse);
GST_DEBUG ("start"); GST_DEBUG ("start");
aacparse->src_caps_set = FALSE;
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024); gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
gst_base_parse_set_passthrough (parse, FALSE); gst_base_parse_set_passthrough (parse, FALSE);
return TRUE; return TRUE;

2
gst/audioparsers/gstaacparse.h
View file

@ -90,8 +90,6 @@ struct _GstAacParse {
gint mpegversion; gint mpegversion;
GstAacHeaderType header_type; GstAacHeaderType header_type;
gboolean src_caps_set;
}; };
/** /**