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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 09:55:36 +00:00
audioparsers: fix some parsers
This commit is contained in:
parent
bb8cecb424
commit
63a17f4ec9
3 changed files with 66 additions and 79 deletions
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@ -91,12 +91,7 @@ gint gst_aac_parse_get_frame_overhead (GstBaseParse * parse,
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gboolean gst_aac_parse_event (GstBaseParse * parse, GstEvent * event);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0, \
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"AAC audio stream parser");
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GST_BOILERPLATE_FULL (GstAacParse, gst_aac_parse, GstBaseParse,
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GST_TYPE_BASE_PARSE, _do_init);
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G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
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static inline gint
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gst_aac_parse_get_sample_rate_from_index (guint sr_idx)
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@ -112,14 +107,18 @@ gst_aac_parse_get_sample_rate_from_index (guint sr_idx)
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}
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/**
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* gst_aac_parse_base_init:
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* @klass: #GstElementClass.
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* gst_aac_parse_class_init:
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* @klass: #GstAacParseClass.
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*
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*/
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static void
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gst_aac_parse_base_init (gpointer klass)
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gst_aac_parse_class_init (GstAacParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
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"AAC audio stream parser");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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@ -129,18 +128,6 @@ gst_aac_parse_base_init (gpointer klass)
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gst_element_class_set_details_simple (element_class,
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"AAC audio stream parser", "Codec/Parser/Audio",
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"Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
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}
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/**
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* gst_aac_parse_class_init:
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* @klass: #GstAacParseClass.
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*
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*/
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static void
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gst_aac_parse_class_init (GstAacParseClass * klass)
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{
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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parse_class->start = GST_DEBUG_FUNCPTR (gst_aac_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
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@ -158,7 +145,7 @@ gst_aac_parse_class_init (GstAacParseClass * klass)
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*
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*/
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static void
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gst_aac_parse_init (GstAacParse * aacparse, GstAacParseClass * klass)
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gst_aac_parse_init (GstAacParse * aacparse)
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{
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GST_DEBUG ("initialized");
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}
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@ -253,15 +240,19 @@ gst_aac_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
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GstBuffer *buf = gst_value_get_buffer (value);
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if (buf) {
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const guint8 *buffer = GST_BUFFER_DATA (buf);
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guint8 *data;
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gsize size;
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guint sr_idx;
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sr_idx = ((buffer[0] & 0x07) << 1) | ((buffer[1] & 0x80) >> 7);
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aacparse->object_type = (buffer[0] & 0xf8) >> 3;
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data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
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sr_idx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
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aacparse->object_type = (data[0] & 0xf8) >> 3;
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aacparse->sample_rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
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aacparse->channels = (buffer[1] & 0x78) >> 3;
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aacparse->channels = (data[1] & 0x78) >> 3;
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aacparse->header_type = DSPAAC_HEADER_NONE;
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aacparse->mpegversion = 4;
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gst_buffer_unmap (buf, data, size);
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GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d",
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aacparse->object_type, aacparse->sample_rate, aacparse->channels);
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@ -573,7 +564,8 @@ gboolean
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gst_aac_parse_check_valid_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
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{
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const guint8 *data;
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guint8 *data;
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gsize size;
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GstAacParse *aacparse;
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gboolean ret = FALSE;
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gboolean lost_sync;
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@ -581,27 +573,27 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse,
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aacparse = GST_AAC_PARSE (parse);
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buffer = frame->buffer;
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data = GST_BUFFER_DATA (buffer);
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data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
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lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
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if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
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aacparse->header_type == DSPAAC_HEADER_NONE) {
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/* There is nothing to parse */
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*framesize = GST_BUFFER_SIZE (buffer);
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*framesize = size;
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ret = TRUE;
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} else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
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ret = gst_aac_parse_detect_stream (aacparse, data, GST_BUFFER_SIZE (buffer),
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ret = gst_aac_parse_detect_stream (aacparse, data, size,
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GST_BASE_PARSE_DRAINING (parse), framesize, skipsize);
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} else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
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guint needed_data = 1024;
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ret = gst_aac_parse_check_adts_frame (aacparse, data,
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GST_BUFFER_SIZE (buffer), GST_BASE_PARSE_DRAINING (parse),
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framesize, &needed_data);
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ret = gst_aac_parse_check_adts_frame (aacparse, data, size,
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GST_BASE_PARSE_DRAINING (parse), framesize, &needed_data);
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if (!ret) {
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GST_DEBUG ("buffer didn't contain valid frame");
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@ -613,6 +605,7 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse,
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GST_DEBUG ("buffer didn't contain valid frame");
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), 1024);
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}
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gst_buffer_unmap (buffer, data, size);
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return ret;
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}
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@ -648,6 +641,8 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
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GstBuffer *buffer;
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GstFlowReturn ret = GST_FLOW_OK;
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gint rate, channels;
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guint8 *data;
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gsize size;
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aacparse = GST_AAC_PARSE (parse);
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buffer = frame->buffer;
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@ -658,8 +653,11 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
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/* see above */
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frame->overhead = 7;
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gst_aac_parse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer),
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data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
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gst_aac_parse_parse_adts_header (aacparse, data,
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&rate, &channels, NULL, NULL);
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gst_buffer_unmap (buffer, data, size);
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GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
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if (G_UNLIKELY (rate != aacparse->sample_rate
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@ -164,12 +164,20 @@ static gboolean gst_ac3_parse_check_valid_frame (GstBaseParse * parse,
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static GstFlowReturn gst_ac3_parse_parse_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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GST_BOILERPLATE (GstAc3Parse, gst_ac3_parse, GstBaseParse, GST_TYPE_BASE_PARSE);
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#define gst_ac3_parse_parent_class parent_class
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G_DEFINE_TYPE (GstAc3Parse, gst_ac3_parse, GST_TYPE_BASE_PARSE);
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static void
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gst_ac3_parse_base_init (gpointer klass)
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gst_ac3_parse_class_init (GstAc3ParseClass * klass)
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{
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0,
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"AC3 audio stream parser");
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object_class->finalize = gst_ac3_parse_finalize;
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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@ -179,18 +187,6 @@ gst_ac3_parse_base_init (gpointer klass)
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gst_element_class_set_details_simple (element_class,
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"AC3 audio stream parser", "Codec/Parser/Audio",
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"AC3 parser", "Tim-Philipp Müller <tim centricular net>");
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}
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static void
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gst_ac3_parse_class_init (GstAc3ParseClass * klass)
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{
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GObjectClass *object_class = G_OBJECT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (ac3_parse_debug, "ac3parse", 0,
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"AC3 audio stream parser");
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object_class->finalize = gst_ac3_parse_finalize;
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parse_class->start = GST_DEBUG_FUNCPTR (gst_ac3_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_ac3_parse_stop);
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@ -208,7 +204,7 @@ gst_ac3_parse_reset (GstAc3Parse * ac3parse)
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}
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static void
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gst_ac3_parse_init (GstAc3Parse * ac3parse, GstAc3ParseClass * klass)
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gst_ac3_parse_init (GstAc3Parse * ac3parse)
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{
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (ac3parse), 64 * 2);
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gst_ac3_parse_reset (ac3parse);
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@ -83,23 +83,21 @@ gboolean gst_amr_parse_check_valid_frame (GstBaseParse * parse,
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GstFlowReturn gst_amr_parse_parse_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame);
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#define _do_init(bla) \
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GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0, \
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"AMR-NB audio stream parser");
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GST_BOILERPLATE_FULL (GstAmrParse, gst_amr_parse, GstBaseParse,
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GST_TYPE_BASE_PARSE, _do_init);
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G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);
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/**
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* gst_amr_parse_base_init:
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* @klass: #GstElementClass.
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* gst_amr_parse_class_init:
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* @klass: GstAmrParseClass.
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*
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*/
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static void
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gst_amr_parse_base_init (gpointer klass)
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gst_amr_parse_class_init (GstAmrParseClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0,
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"AMR-NB audio stream parser");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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@ -110,18 +108,6 @@ gst_amr_parse_base_init (gpointer klass)
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"AMR audio stream parser", "Codec/Parser/Audio",
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"Adaptive Multi-Rate audio parser",
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"Ronald Bultje <rbultje@ronald.bitfreak.net>");
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}
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/**
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* gst_amr_parse_class_init:
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* @klass: GstAmrParseClass.
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*
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*/
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static void
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gst_amr_parse_class_init (GstAmrParseClass * klass)
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{
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GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
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parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);
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parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);
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@ -139,7 +125,7 @@ gst_amr_parse_class_init (GstAmrParseClass * klass)
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*
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*/
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static void
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gst_amr_parse_init (GstAmrParse * amrparse, GstAmrParseClass * klass)
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gst_amr_parse_init (GstAmrParse * amrparse)
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{
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/* init rest */
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gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);
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@ -270,14 +256,17 @@ gst_amr_parse_check_valid_frame (GstBaseParse * parse,
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GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
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{
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GstBuffer *buffer;
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const guint8 *data;
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guint8 *data;
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gsize size;
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gint fsize, mode, dsize;
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GstAmrParse *amrparse;
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gboolean ret = FALSE;
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amrparse = GST_AMR_PARSE (parse);
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buffer = frame->buffer;
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data = GST_BUFFER_DATA (buffer);
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dsize = GST_BUFFER_SIZE (buffer);
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data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
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dsize = size;
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GST_LOG ("buffer: %d bytes", dsize);
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}
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/* We return FALSE, so this frame won't get pushed forward. Instead,
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the "skip" value is set, so next time we will receive a valid frame. */
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return FALSE;
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goto done;
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}
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/* Does this look like a possible frame header candidate? */
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@ -311,12 +300,16 @@ gst_amr_parse_check_valid_frame (GstBaseParse * parse,
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(!GST_BASE_PARSE_LOST_SYNC (parse) || GST_BASE_PARSE_DRAINING (parse)
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|| (dsize > fsize && (data[fsize] & 0x83) == 0))) {
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*framesize = fsize;
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return TRUE;
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ret = TRUE;
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goto done;
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}
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}
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GST_LOG ("sync lost");
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return FALSE;
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done:
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gst_buffer_unmap (buffer, data, size);
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return ret;
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}
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