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aacparse: improve channel/rate handling

Browse source 
Keep track of the last parsed channels/rate fields so they can be
used even if the element was not yet configured.

https://bugzilla.gnome.org/show_bug.cgi?id=769278
This commit is contained in:
Vincent Penquerc'h 2016-09-05 09:37:02 +01:00
parent 740749ac55
commit 559546dd3a
2 changed files with 33 additions and 4 deletions
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34
gst/audioparsers/gstaacparse.c
View file

@ -97,7 +97,10 @@ static GstFlowReturn gst_aac_parse_handle_frame (GstBaseParse * parse,
GstBaseParseFrame * frame, gint * skipsize);
static GstFlowReturn gst_aac_parse_pre_push_frame (GstBaseParse * parse,
GstBaseParseFrame * frame);
static gboolean gst_aac_parse_src_event (GstBaseParse * parse,
GstEvent * event);
#define gst_aac_parse_parent_class parent_class
G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
/**
@ -128,6 +131,7 @@ gst_aac_parse_class_init (GstAacParseClass * klass)
parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_handle_frame);
parse_class->pre_push_frame =
GST_DEBUG_FUNCPTR (gst_aac_parse_pre_push_frame);
parse_class->src_event = GST_DEBUG_FUNCPTR (gst_aac_parse_src_event);
}
@ -143,6 +147,9 @@ gst_aac_parse_init (GstAacParse * aacparse)
GST_DEBUG ("initialized");
GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (aacparse));
GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (aacparse));
aacparse->last_parsed_sample_rate = 0;
aacparse->last_parsed_channels = 0;
}
@ -251,6 +258,9 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
if (allowed)
gst_caps_unref (allowed);
aacparse->last_parsed_channels = 0;
aacparse->last_parsed_sample_rate = 0;
GST_DEBUG_OBJECT (aacparse, "setting src caps: %" GST_PTR_FORMAT, src_caps);
res = gst_pad_set_caps (GST_BASE_PARSE (aacparse)->srcpad, src_caps);
@ -347,7 +357,6 @@ gst_aac_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
gst_base_parse_set_passthrough (parse, FALSE);
}
}
return TRUE;
}
@ -524,6 +533,7 @@ gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
if (!*sample_rate)
return FALSE;
}
aacparse->last_parsed_sample_rate = *sample_rate;
return TRUE;
}
@ -563,6 +573,7 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
"Need more code to parse humongous LOAS data, currently ignored");
if (bits)
*bits = 0;
aacparse->last_parsed_channels = *channels;
return TRUE;
}
@ -589,13 +600,13 @@ gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
return FALSE;
if (u8) {
GST_LOG_OBJECT (aacparse, "Frame uses previous config");
if (!aacparse->sample_rate || !aacparse->channels) {
if (!aacparse->last_parsed_sample_rate || !aacparse->last_parsed_channels) {
GST_DEBUG_OBJECT (aacparse,
"No previous config to use. We'll look for more data.");
return FALSE;
}
*sample_rate = aacparse->sample_rate;
*channels = aacparse->channels;
*sample_rate = aacparse->last_parsed_sample_rate;
*channels = aacparse->last_parsed_channels;
return TRUE;
}
@ -1439,6 +1450,8 @@ gst_aac_parse_start (GstBaseParse * parse)
aacparse->frame_samples = 1024;
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), ADTS_MAX_SIZE);
aacparse->sent_codec_tag = FALSE;
aacparse->last_parsed_channels = 0;
aacparse->last_parsed_sample_rate = 0;
return TRUE;
}
@ -1586,3 +1599,16 @@ gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
return res;
}
static gboolean
gst_aac_parse_src_event (GstBaseParse * parse, GstEvent * event)
{
GstAacParse *aacparse = GST_AAC_PARSE (parse);
if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
aacparse->last_parsed_channels = 0;
aacparse->last_parsed_sample_rate = 0;
}
return GST_BASE_PARSE_CLASS (parent_class)->src_event (parse, event);
}

3
gst/audioparsers/gstaacparse.h
View file

@ -83,6 +83,9 @@ struct _GstAacParse {
GstAacHeaderType output_header_type;
gboolean sent_codec_tag;
gint last_parsed_sample_rate;
gint last_parsed_channels;
};
/**