/**
* SECTION:element-aacparse
+ * @title: aacparse
* @short_description: AAC parser
* @see_also: #GstAmrParse
*
* be determined either. However, ADTS format AAC clips can be seeked, and parser
* can also estimate playback position and clip duration.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
* ]|
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
#define ADIF_MAX_SIZE 40 /* Should be enough */
#define ADTS_MAX_SIZE 10 /* Should be enough */
#define LOAS_MAX_SIZE 3 /* Should be enough */
+#define RAW_MAX_SIZE 1 /* Correct framing is required */
#define ADTS_HEADERS_LENGTH 7UL /* Total byte-length of fixed and variable
headers prepended during raw to ADTS
static gboolean gst_aac_parse_src_event (GstBaseParse * parse,
GstEvent * event);
+static gboolean gst_aac_parse_read_audio_specific_config (GstAacParse *
+ aacparse, GstBitReader * br, gint * object_type, gint * sample_rate,
+ gint * channels, gint * frame_samples);
+
+
#define gst_aac_parse_parent_class parent_class
G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
{
GstStructure *s;
- GstCaps *src_caps = NULL, *allowed;
+ GstCaps *src_caps = NULL, *peercaps;
gboolean res = FALSE;
const gchar *stream_format;
guint8 codec_data[2];
if (stream_format)
gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
- allowed = gst_pad_get_allowed_caps (GST_BASE_PARSE (aacparse)->srcpad);
- if (allowed && !gst_caps_can_intersect (src_caps, allowed)) {
+ peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (aacparse), NULL);
+ if (peercaps && !gst_caps_can_intersect (src_caps, peercaps)) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Caps can not intersect");
if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
"Input is ADTS, trying raw");
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "raw",
NULL);
- if (gst_caps_can_intersect (src_caps, allowed)) {
+ if (gst_caps_can_intersect (src_caps, peercaps)) {
GstBuffer *codec_data_buffer;
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
gst_buffer_fill (codec_data_buffer, 0, codec_data, 2);
gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
codec_data_buffer, NULL);
+ gst_buffer_unref (codec_data_buffer);
}
} else if (aacparse->header_type == DSPAAC_HEADER_NONE) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Input is raw, trying ADTS");
gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "adts",
NULL);
- if (gst_caps_can_intersect (src_caps, allowed)) {
+ if (gst_caps_can_intersect (src_caps, peercaps)) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
"Caps can intersect, we will prepend ADTS headers");
aacparse->output_header_type = DSPAAC_HEADER_ADTS;
}
}
}
- if (allowed)
- gst_caps_unref (allowed);
+ if (peercaps)
+ gst_caps_unref (peercaps);
aacparse->last_parsed_channels = 0;
aacparse->last_parsed_sample_rate = 0;
if (buf && gst_buffer_get_size (buf) >= 2) {
GstMapInfo map;
- guint sr_idx;
+ GstBitReader br;
if (!gst_buffer_map (buf, &map, GST_MAP_READ))
return FALSE;
+ gst_bit_reader_init (&br, map.data, map.size);
+ gst_aac_parse_read_audio_specific_config (aacparse, &br,
+ &aacparse->object_type, &aacparse->sample_rate, &aacparse->channels,
+ &aacparse->frame_samples);
- sr_idx = ((map.data[0] & 0x07) << 1) | ((map.data[1] & 0x80) >> 7);
- aacparse->object_type = (map.data[0] & 0xf8) >> 3;
- aacparse->sample_rate =
- gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
- aacparse->channels = (map.data[1] & 0x78) >> 3;
- if (aacparse->channels == 7)
- aacparse->channels = 8;
- else if (aacparse->channels == 11)
- aacparse->channels = 7;
- else if (aacparse->channels == 12 || aacparse->channels == 14)
- aacparse->channels = 8;
aacparse->header_type = DSPAAC_HEADER_NONE;
aacparse->mpegversion = 4;
- aacparse->frame_samples = (map.data[1] & 4) ? 960 : 1024;
gst_buffer_unmap (buf, &map);
GST_DEBUG ("codec_data: object_type=%d, sample_rate=%d, channels=%d, "
gst_aac_parse_set_src_caps (aacparse, caps);
if (aacparse->header_type == aacparse->output_header_type)
gst_base_parse_set_passthrough (parse, TRUE);
+
+ /* input is already correctly framed */
+ gst_base_parse_set_min_frame_size (parse, RAW_MAX_SIZE);
} else {
return FALSE;
}
/* See table 1.13 in ISO/IEC 14496-3 */
static gboolean
-gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
- GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
+gst_aac_parse_read_audio_specific_config (GstAacParse * aacparse,
+ GstBitReader * br, gint * object_type, gint * sample_rate, gint * channels,
+ gint * frame_samples)
{
guint8 audio_object_type;
guint8 G_GNUC_UNUSED extension_audio_object_type;
if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
return FALSE;
+ if (object_type)
+ *object_type = audio_object_type;
if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
return FALSE;
}
GST_LOG_OBJECT (aacparse,
- "Audio object type 5 or 29, so rereading sampling rate...");
+ "Audio object type 5 or 29, so rereading sampling rate (was %d)...",
+ *sample_rate);
if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
return FALSE;
extension_audio_object_type = 0;
}
- GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
+ GST_INFO_OBJECT (aacparse, "Parsed AudioSpecificConfig: %d Hz, %d channels",
*sample_rate, *channels);
+ if (frame_samples && audio_object_type == 23) {
+ guint8 frame_flag;
+ /* Read the Decoder Configuration (GASpecificConfig) if present */
+ /* We only care about the first bit to know what the number of samples
+ * in a frame is */
+ if (!gst_bit_reader_get_bits_uint8 (br, &frame_flag, 1))
+ return FALSE;
+ *frame_samples = frame_flag ? 960 : 1024;
+ }
+
/* There's LOTS of stuff next, but we ignore it for now as we have
what we want (sample rate and number of channels */
GST_DEBUG_OBJECT (aacparse,
"Need more code to parse humongous LOAS data, currently ignored");
- if (bits)
- *bits = 0;
aacparse->last_parsed_channels = *channels;
return TRUE;
}
}
if (!use_same_config) {
if (v == 0) {
- if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
+ if (!gst_aac_parse_read_audio_specific_config (aacparse, &br, NULL,
sample_rate, channels, NULL))
return FALSE;
} else {
- guint32 bits, asc_len;
+ guint32 asc_len;
if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
return FALSE;
- if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
- sample_rate, channels, &bits))
+ if (!gst_aac_parse_read_audio_specific_config (aacparse, &br, NULL,
+ sample_rate, channels, NULL))
return FALSE;
- asc_len -= bits;
if (!gst_bit_reader_skip (&br, asc_len))
return FALSE;
}
adts_headers[0] = 0xFFU;
adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
- (channel_configuration & 0x4U);
+ ((channel_configuration & 0x4U) >> 2);
adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
(guint8) (frame_size >> 11);
adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
gst_buffer_get_size (frame->out_buffer) - header_size);
}
+ frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
+
return GST_FLOW_OK;
}
aacparse->sent_codec_tag = FALSE;
aacparse->last_parsed_channels = 0;
aacparse->last_parsed_sample_rate = 0;
+ aacparse->object_type = 0;
+ aacparse->bitrate = 0;
+ aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
+ aacparse->output_header_type = DSPAAC_HEADER_NOT_PARSED;
+ aacparse->channels = 0;
+ aacparse->sample_rate = 0;
return TRUE;
}
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