*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
/**
* 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 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
+ * gst-launch-1.0 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
* ]|
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
#include <string.h>
#include <gst/base/gstbitreader.h>
+#include <gst/pbutils/pbutils.h>
#include "gstaacparse.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
+ conversion */
#define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec)
-static const gint loas_sample_rate_table[32] = {
+static const gint loas_sample_rate_table[16] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
-static const gint loas_channels_table[32] = {
+static const gint loas_channels_table[16] = {
0, 1, 2, 3, 4, 5, 6, 8,
- 0, 0, 0, 0, 0, 0, 0, 0
+ 0, 0, 0, 7, 8, 0, 8, 0
};
static gboolean gst_aac_parse_start (GstBaseParse * parse);
static GstCaps *gst_aac_parse_sink_getcaps (GstBaseParse * parse,
GstCaps * filter);
-static gboolean gst_aac_parse_check_valid_frame (GstBaseParse * parse,
- GstBaseParseFrame * frame, guint * size, gint * skipsize);
-
-static GstFlowReturn gst_aac_parse_parse_frame (GstBaseParse * parse,
+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);
-gboolean gst_aac_parse_convert (GstBaseParse * parse,
- GstFormat src_format,
- gint64 src_value, GstFormat dest_format, gint64 * dest_value);
+static gboolean gst_aac_parse_read_audio_specific_config (GstAacParse *
+ aacparse, GstBitReader * br, gint * object_type, gint * sample_rate,
+ gint * channels, gint * frame_samples);
-gint gst_aac_parse_get_frame_overhead (GstBaseParse * parse,
- GstBuffer * buffer);
-
-gboolean gst_aac_parse_event (GstBaseParse * parse, GstEvent * event);
+#define gst_aac_parse_parent_class parent_class
G_DEFINE_TYPE (GstAacParse, gst_aac_parse, GST_TYPE_BASE_PARSE);
-static inline gint
-gst_aac_parse_get_sample_rate_from_index (guint sr_idx)
-{
- static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000, 44100,
- 32000, 24000, 22050, 16000, 12000, 11025, 8000
- };
-
- if (sr_idx < G_N_ELEMENTS (aac_sample_rates))
- return aac_sample_rates[sr_idx];
- GST_WARNING ("Invalid sample rate index %u", sr_idx);
- return 0;
-}
-
/**
* gst_aac_parse_class_init:
* @klass: #GstAacParseClass.
GST_DEBUG_CATEGORY_INIT (aacparse_debug, "aacparse", 0,
"AAC audio stream parser");
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_template));
+ gst_element_class_add_static_pad_template (element_class, &sink_template);
+ gst_element_class_add_static_pad_template (element_class, &src_template);
- gst_element_class_set_details_simple (element_class,
+ gst_element_class_set_static_metadata (element_class,
"AAC audio stream parser", "Codec/Parser/Audio",
"Advanced Audio Coding parser", "Stefan Kost <stefan.kost@nokia.com>");
parse_class->stop = GST_DEBUG_FUNCPTR (gst_aac_parse_stop);
parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_setcaps);
parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_aac_parse_sink_getcaps);
- parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_aac_parse_parse_frame);
- parse_class->check_valid_frame =
- GST_DEBUG_FUNCPTR (gst_aac_parse_check_valid_frame);
+ 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);
}
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;
}
gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
{
GstStructure *s;
- GstCaps *src_caps = NULL;
+ GstCaps *src_caps = NULL, *peercaps;
gboolean res = FALSE;
const gchar *stream_format;
+ guint8 codec_data[2];
+ guint16 codec_data_data;
+ gint sample_rate_idx;
GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
if (sink_caps)
gst_caps_set_simple (src_caps, "framed", G_TYPE_BOOLEAN, TRUE,
"mpegversion", G_TYPE_INT, aacparse->mpegversion, NULL);
+ aacparse->output_header_type = aacparse->header_type;
switch (aacparse->header_type) {
case DSPAAC_HEADER_NONE:
stream_format = "raw";
stream_format = NULL;
}
+ /* Generate codec data to be able to set profile/level on the caps */
+ sample_rate_idx =
+ gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
+ if (sample_rate_idx < 0)
+ goto not_a_known_rate;
+ codec_data_data =
+ (aacparse->object_type << 11) |
+ (sample_rate_idx << 7) | (aacparse->channels << 3);
+ GST_WRITE_UINT16_BE (codec_data, codec_data_data);
+ gst_codec_utils_aac_caps_set_level_and_profile (src_caps, codec_data, 2);
+
s = gst_caps_get_structure (src_caps, 0);
if (aacparse->sample_rate > 0)
gst_structure_set (s, "rate", G_TYPE_INT, aacparse->sample_rate, NULL);
if (stream_format)
gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
+ 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) {
+ GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
+ "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, peercaps)) {
+ GstBuffer *codec_data_buffer;
+
+ GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
+ "Caps can intersect, we will drop the ADTS layer");
+ aacparse->output_header_type = DSPAAC_HEADER_NONE;
+
+ /* The codec_data data is according to AudioSpecificConfig,
+ ISO/IEC 14496-3, 1.6.2.1 */
+ codec_data_buffer = gst_buffer_new_and_alloc (2);
+ 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, 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 (peercaps)
+ gst_caps_unref (peercaps);
+
+ 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);
gst_caps_unref (src_caps);
return res;
+
+not_a_known_rate:
+ GST_ERROR_OBJECT (aacparse, "Not a known sample rate: %d",
+ aacparse->sample_rate);
+ gst_caps_unref (src_caps);
+ return FALSE;
}
if (value) {
GstBuffer *buf = gst_value_get_buffer (value);
- if (buf) {
+ if (buf && gst_buffer_get_size (buf) >= 2) {
GstMapInfo map;
- guint sr_idx;
+ GstBitReader br;
- gst_buffer_map (buf, &map, GST_MAP_READ);
+ 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_aac_parse_get_sample_rate_from_index (sr_idx);
- aacparse->channels = (map.data[1] & 0x78) >> 3;
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, "
/* arrange for metadata and get out of the way */
gst_aac_parse_set_src_caps (aacparse, caps);
- gst_base_parse_set_passthrough (parse, TRUE);
- } else
+ 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;
+ }
/* caps info overrides */
gst_structure_get_int (structure, "rate", &aacparse->sample_rate);
gst_structure_get_int (structure, "channels", &aacparse->channels);
} else {
- gst_base_parse_set_passthrough (parse, FALSE);
- }
+ const gchar *stream_format =
+ gst_structure_get_string (structure, "stream-format");
+ if (g_strcmp0 (stream_format, "raw") == 0) {
+ GST_ERROR_OBJECT (parse, "Need codec_data for raw AAC");
+ return FALSE;
+ } else {
+ aacparse->sample_rate = 0;
+ aacparse->channels = 0;
+ aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
+ gst_base_parse_set_passthrough (parse, FALSE);
+ }
+ }
return TRUE;
}
const guint8 * data, const guint avail, gboolean drain,
guint * framesize, guint * needed_data)
{
+ guint crc_size;
+
*needed_data = 0;
- if (G_UNLIKELY (avail < 2))
+ /* Absolute minimum to perform the ADTS syncword,
+ layer and sampling frequency tests */
+ if (G_UNLIKELY (avail < 3)) {
+ *needed_data = 3;
return FALSE;
+ }
+ /* Syncword and layer tests */
if ((data[0] == 0xff) && ((data[1] & 0xf6) == 0xf0)) {
+
+ /* Sampling frequency test */
+ if (G_UNLIKELY ((data[2] & 0x3C) >> 2 == 15))
+ return FALSE;
+
+ /* This looks like an ADTS frame header but
+ we need at least 6 bytes to proceed */
+ if (G_UNLIKELY (avail < 6)) {
+ *needed_data = 6;
+ return FALSE;
+ }
+
*framesize = gst_aac_parse_adts_get_frame_len (data);
+ /* If frame has CRC, it needs 2 bytes
+ for it at the end of the header */
+ crc_size = (data[1] & 0x01) ? 0 : 2;
+
+ /* CRC size test */
+ if (*framesize < 7 + crc_size) {
+ *needed_data = 7 + crc_size;
+ return FALSE;
+ }
+
/* In EOS mode this is enough. No need to examine the data further.
We also relax the check when we have sync, on the assumption that
if we're not looking at random data, we have a much higher chance
*value = 0;
if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2))
return FALSE;
- for (i = 0; i < bytes; ++i) {
+ for (i = 0; i <= bytes; ++i) {
*value <<= 8;
if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8))
return FALSE;
if (!*sample_rate)
return FALSE;
}
+ aacparse->last_parsed_sample_rate = *sample_rate;
return TRUE;
}
/* 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, channel_configuration;
+ guint8 audio_object_type;
+ guint8 G_GNUC_UNUSED extension_audio_object_type;
+ guint8 channel_configuration, extension_channel_configuration;
+ gboolean G_GNUC_UNUSED sbr = FALSE, ps = FALSE;
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;
if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4))
return FALSE;
- GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
*channels = loas_channels_table[channel_configuration];
+ GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration);
if (!*channels)
return FALSE;
- if (audio_object_type == 5) {
+ if (audio_object_type == 5 || audio_object_type == 29) {
+ extension_audio_object_type = 5;
+ sbr = TRUE;
+ if (audio_object_type == 29) {
+ ps = TRUE;
+ /* Parametric stereo. If we have a one-channel configuration, we can
+ * override it to stereo */
+ if (*channels == 1)
+ *channels = 2;
+ }
+
GST_LOG_OBJECT (aacparse,
- "Audio object type 5, 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;
+
+ if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
+ return FALSE;
+
+ if (audio_object_type == 22) {
+ /* extension channel configuration */
+ if (!gst_bit_reader_get_bits_uint8 (br, &extension_channel_configuration,
+ 4))
+ return FALSE;
+ GST_LOG_OBJECT (aacparse, "extension channel_configuration: %d",
+ extension_channel_configuration);
+ *channels = loas_channels_table[extension_channel_configuration];
+ if (!*channels)
+ return FALSE;
+ }
+ } else {
+ 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;
}
gst_bit_reader_init (&br, data, avail);
/* skip sync word (11 bits) and size (13 bits) */
- gst_bit_reader_skip (&br, 11 + 13);
+ if (!gst_bit_reader_skip (&br, 11 + 13))
+ return FALSE;
/* First bit is "use last config" */
if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1))
return FALSE;
if (u8) {
- GST_DEBUG_OBJECT (aacparse, "Frame uses previous config");
- if (!aacparse->sample_rate || !aacparse->channels) {
- GST_WARNING_OBJECT (aacparse, "No previous config to use");
+ GST_LOG_OBJECT (aacparse, "Frame uses previous config");
+ 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;
}
GST_DEBUG_OBJECT (aacparse, "Frame contains new config");
+ /* audioMuxVersion */
if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1))
return FALSE;
if (v) {
+ /* audioMuxVersionA */
if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1))
return FALSE;
} else
guint8 same_time, subframes, num_program, prog;
if (v == 1) {
guint32 value;
+ /* taraBufferFullness */
if (!gst_aac_parse_latm_get_value (aacparse, &br, &value))
return FALSE;
}
}
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;
+ if (!gst_bit_reader_skip (&br, asc_len))
return FALSE;
- asc_len -= bits;
- gst_bit_reader_skip (&br, asc_len);
}
}
}
}
- GST_WARNING_OBJECT (aacparse, "More data ignored");
+ GST_LOG_OBJECT (aacparse, "More data ignored");
} else {
GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"...");
+ return FALSE;
}
return TRUE;
}
*needed_data = 0;
/* 3 byte header */
- if (G_UNLIKELY (avail < 3))
+ if (G_UNLIKELY (avail < 3)) {
+ *needed_data = 3;
return FALSE;
+ }
if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) {
*framesize = gst_aac_parse_loas_get_frame_len (data);
- GST_DEBUG_OBJECT (aacparse, "Found %u byte LOAS frame", *framesize);
+ GST_DEBUG_OBJECT (aacparse, "Found possible %u byte LOAS frame",
+ *framesize);
/* In EOS mode this is enough. No need to examine the data further.
We also relax the check when we have sync, on the assumption that
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
nextlen + LOAS_MAX_SIZE);
return TRUE;
+ } else {
+ GST_DEBUG_OBJECT (aacparse, "That was a false positive");
}
}
return FALSE;
if (rate) {
gint sr_idx = (data[2] & 0x3c) >> 2;
- *rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
+ *rate = gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
}
- if (channels)
+ if (channels) {
*channels = ((data[2] & 0x01) << 2) | ((data[3] & 0xc0) >> 6);
+ if (*channels == 7)
+ *channels = 8;
+ }
if (version)
*version = (data[1] & 0x08) ? 2 : 4;
if (object)
- *object = (data[2] & 0xc0) >> 6;
+ *object = ((data[2] & 0xc0) >> 6) + 1;
}
/**
for (i = 0; i < avail - 4; i++) {
if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) ||
- ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) ||
+ ((data[i] == 0x56) && ((data[i + 1] & 0xe0) == 0xe0)) ||
strncmp ((char *) data + i, "ADIF", 4) == 0) {
- GST_DEBUG_OBJECT (aacparse, "Found ADIF signature at offset %u", i);
+ GST_DEBUG_OBJECT (aacparse, "Found signature at offset %u", i);
found = TRUE;
if (i) {
GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize);
- aacparse->header_type = DSPAAC_HEADER_ADTS;
gst_aac_parse_parse_adts_header (aacparse, data, &rate, &channels,
&aacparse->object_type, &aacparse->mpegversion);
+ if (!channels || !framesize) {
+ GST_DEBUG_OBJECT (aacparse, "impossible ADTS configuration");
+ return FALSE;
+ }
+
+ aacparse->header_type = DSPAAC_HEADER_ADTS;
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
aacparse->frame_samples, 2, 2);
if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
framesize, &need_data_loas)) {
- gint rate, channels;
+ gint rate = 0, channels = 0;
GST_INFO ("LOAS, framesize: %d", *framesize);
if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
&channels, &aacparse->mpegversion)) {
- GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
+ /* This is pretty normal when skipping data at the start of
+ * random stream (MPEG-TS capture for example) */
+ GST_LOG_OBJECT (aacparse, "Error reading LOAS config");
return FALSE;
}
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
aacparse->frame_samples, 2, 2);
+ /* Don't store the sample rate and channels yet -
+ * this is just format detection. */
GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d",
rate, channels, aacparse->object_type, aacparse->mpegversion);
- aacparse->sample_rate = rate;
- aacparse->channels = channels;
}
gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE);
/* FIXME: This gives totally wrong results. Duration calculation cannot
be based on this */
- aacparse->sample_rate = gst_aac_parse_get_sample_rate_from_index (sr_idx);
+ aacparse->sample_rate =
+ gst_codec_utils_aac_get_sample_rate_from_index (sr_idx);
/* baseparse is not given any fps,
* so it will give up on timestamps, seeking, etc */
return FALSE;
}
+/**
+ * gst_aac_parse_get_audio_profile_object_type
+ * @aacparse: #GstAacParse.
+ *
+ * Gets the MPEG-2 profile or the MPEG-4 object type value corresponding to the
+ * mpegversion and profile of @aacparse's src pad caps, according to the
+ * values defined by table 1.A.11 in ISO/IEC 14496-3.
+ *
+ * Returns: the profile or object type value corresponding to @aacparse's src
+ * pad caps, if such a value exists; otherwise G_MAXUINT8.
+ */
+static guint8
+gst_aac_parse_get_audio_profile_object_type (GstAacParse * aacparse)
+{
+ GstCaps *srccaps;
+ GstStructure *srcstruct;
+ const gchar *profile;
+ guint8 ret;
+
+ srccaps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse));
+ if (G_UNLIKELY (srccaps == NULL)) {
+ return G_MAXUINT8;
+ }
+
+ srcstruct = gst_caps_get_structure (srccaps, 0);
+ profile = gst_structure_get_string (srcstruct, "profile");
+ if (G_UNLIKELY (profile == NULL)) {
+ gst_caps_unref (srccaps);
+ return G_MAXUINT8;
+ }
+
+ if (g_strcmp0 (profile, "main") == 0) {
+ ret = (guint8) 0U;
+ } else if (g_strcmp0 (profile, "lc") == 0) {
+ ret = (guint8) 1U;
+ } else if (g_strcmp0 (profile, "ssr") == 0) {
+ ret = (guint8) 2U;
+ } else if (g_strcmp0 (profile, "ltp") == 0) {
+ if (G_LIKELY (aacparse->mpegversion == 4))
+ ret = (guint8) 3U;
+ else
+ ret = G_MAXUINT8; /* LTP Object Type allowed only for MPEG-4 */
+ } else {
+ ret = G_MAXUINT8;
+ }
+
+ gst_caps_unref (srccaps);
+ return ret;
+}
+
+/**
+ * gst_aac_parse_get_audio_channel_configuration
+ * @num_channels: number of audio channels.
+ *
+ * Gets the Channel Configuration value, as defined by table 1.19 in ISO/IEC
+ * 14496-3, for a given number of audio channels.
+ *
+ * Returns: the Channel Configuration value corresponding to @num_channels, if
+ * such a value exists; otherwise G_MAXUINT8.
+ */
+static guint8
+gst_aac_parse_get_audio_channel_configuration (gint num_channels)
+{
+ if (num_channels >= 1 && num_channels <= 6) /* Mono up to & including 5.1 */
+ return (guint8) num_channels;
+ else if (num_channels == 8) /* 7.1 */
+ return (guint8) 7U;
+ else
+ return G_MAXUINT8;
+
+ /* FIXME: Add support for configurations 11, 12 and 14 from
+ * ISO/IEC 14496-3:2009/PDAM 4 based on the actual channel layout
+ */
+}
+
+/**
+ * gst_aac_parse_get_audio_sampling_frequency_index:
+ * @sample_rate: audio sampling rate.
+ *
+ * Gets the Sampling Frequency Index value, as defined by table 1.18 in ISO/IEC
+ * 14496-3, for a given sampling rate.
+ *
+ * Returns: the Sampling Frequency Index value corresponding to @sample_rate,
+ * if such a value exists; otherwise G_MAXUINT8.
+ */
+static guint8
+gst_aac_parse_get_audio_sampling_frequency_index (gint sample_rate)
+{
+ switch (sample_rate) {
+ case 96000:
+ return 0x0U;
+ case 88200:
+ return 0x1U;
+ case 64000:
+ return 0x2U;
+ case 48000:
+ return 0x3U;
+ case 44100:
+ return 0x4U;
+ case 32000:
+ return 0x5U;
+ case 24000:
+ return 0x6U;
+ case 22050:
+ return 0x7U;
+ case 16000:
+ return 0x8U;
+ case 12000:
+ return 0x9U;
+ case 11025:
+ return 0xAU;
+ case 8000:
+ return 0xBU;
+ case 7350:
+ return 0xCU;
+ default:
+ return G_MAXUINT8;
+ }
+}
+
+/**
+ * gst_aac_parse_prepend_adts_headers:
+ * @aacparse: #GstAacParse.
+ * @frame: raw AAC frame to which ADTS headers shall be prepended.
+ *
+ * Prepends ADTS headers to a raw AAC audio frame.
+ *
+ * Returns: TRUE if ADTS headers were successfully prepended; FALSE otherwise.
+ */
+static gboolean
+gst_aac_parse_prepend_adts_headers (GstAacParse * aacparse,
+ GstBaseParseFrame * frame)
+{
+ GstMemory *mem;
+ guint8 *adts_headers;
+ gsize buf_size;
+ gsize frame_size;
+ guint8 id, profile, channel_configuration, sampling_frequency_index;
+
+ id = (aacparse->mpegversion == 4) ? 0x0U : 0x1U;
+ profile = gst_aac_parse_get_audio_profile_object_type (aacparse);
+ if (profile == G_MAXUINT8) {
+ GST_ERROR_OBJECT (aacparse, "Unsupported audio profile or object type");
+ return FALSE;
+ }
+ channel_configuration =
+ gst_aac_parse_get_audio_channel_configuration (aacparse->channels);
+ if (channel_configuration == G_MAXUINT8) {
+ GST_ERROR_OBJECT (aacparse, "Unsupported number of channels");
+ return FALSE;
+ }
+ sampling_frequency_index =
+ gst_aac_parse_get_audio_sampling_frequency_index (aacparse->sample_rate);
+ if (sampling_frequency_index == G_MAXUINT8) {
+ GST_ERROR_OBJECT (aacparse, "Unsupported sampling frequency");
+ return FALSE;
+ }
+
+ frame->out_buffer = gst_buffer_copy (frame->buffer);
+ buf_size = gst_buffer_get_size (frame->out_buffer);
+ frame_size = buf_size + ADTS_HEADERS_LENGTH;
+
+ if (G_UNLIKELY (frame_size >= 0x4000)) {
+ GST_ERROR_OBJECT (aacparse, "Frame size is too big for ADTS");
+ return FALSE;
+ }
+
+ adts_headers = (guint8 *) g_malloc0 (ADTS_HEADERS_LENGTH);
+
+ /* Note: no error correction bits are added to the resulting ADTS frames */
+ adts_headers[0] = 0xFFU;
+ adts_headers[1] = 0xF0U | (id << 3) | 0x1U;
+ adts_headers[2] = (profile << 6) | (sampling_frequency_index << 2) | 0x2U |
+ ((channel_configuration & 0x4U) >> 2);
+ adts_headers[3] = ((channel_configuration & 0x3U) << 6) | 0x30U |
+ (guint8) (frame_size >> 11);
+ adts_headers[4] = (guint8) ((frame_size >> 3) & 0x00FF);
+ adts_headers[5] = (guint8) (((frame_size & 0x0007) << 5) + 0x1FU);
+ adts_headers[6] = 0xFCU;
+
+ mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
+ ADTS_HEADERS_LENGTH, adts_headers, g_free);
+ gst_buffer_prepend_memory (frame->out_buffer, mem);
+
+ return TRUE;
+}
/**
* gst_aac_parse_check_valid_frame:
* @parse: #GstBaseParse.
- * @buffer: #GstBuffer.
- * @framesize: If the buffer contains a valid frame, its size will be put here
+ * @frame: #GstBaseParseFrame.
* @skipsize: How much data parent class should skip in order to find the
* frame header.
*
- * Implementation of "check_valid_frame" vmethod in #GstBaseParse class.
+ * Implementation of "handle_frame" vmethod in #GstBaseParse class.
+ *
+ * Also determines frame overhead.
+ * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
+ * a per-frame header. LOAS has 3 bytes.
+ *
+ * We're making a couple of simplifying assumptions:
+ *
+ * 1. We count Program Configuration Elements rather than searching for them
+ * in the streams to discount them - the overhead is negligible.
+ *
+ * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
+ * bits, which should still not be significant enough to warrant the
+ * additional parsing through the headers
*
- * Returns: TRUE if buffer contains a valid frame.
+ * Returns: a #GstFlowReturn.
*/
-static gboolean
-gst_aac_parse_check_valid_frame (GstBaseParse * parse,
- GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
+static GstFlowReturn
+gst_aac_parse_handle_frame (GstBaseParse * parse,
+ GstBaseParseFrame * frame, gint * skipsize)
{
GstMapInfo map;
GstAacParse *aacparse;
gboolean ret = FALSE;
gboolean lost_sync;
GstBuffer *buffer;
+ guint framesize;
+ gint rate = 0, channels = 0;
aacparse = GST_AAC_PARSE (parse);
buffer = frame->buffer;
gst_buffer_map (buffer, &map, GST_MAP_READ);
+ *skipsize = -1;
lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
if (aacparse->header_type == DSPAAC_HEADER_ADIF ||
aacparse->header_type == DSPAAC_HEADER_NONE) {
/* There is nothing to parse */
- *framesize = map.size;
+ framesize = map.size;
ret = TRUE;
} else if (aacparse->header_type == DSPAAC_HEADER_NOT_PARSED || lost_sync) {
ret = gst_aac_parse_detect_stream (aacparse, map.data, map.size,
- GST_BASE_PARSE_DRAINING (parse), framesize, skipsize);
+ GST_BASE_PARSE_DRAINING (parse), &framesize, skipsize);
} else if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
guint needed_data = 1024;
ret = gst_aac_parse_check_adts_frame (aacparse, map.data, map.size,
- GST_BASE_PARSE_DRAINING (parse), framesize, &needed_data);
+ GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
- if (!ret) {
+ if (!ret && needed_data) {
GST_DEBUG ("buffer didn't contain valid frame");
+ *skipsize = 0;
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
needed_data);
}
guint needed_data = 1024;
ret = gst_aac_parse_check_loas_frame (aacparse, map.data,
- map.size, GST_BASE_PARSE_DRAINING (parse), framesize, &needed_data);
+ map.size, GST_BASE_PARSE_DRAINING (parse), &framesize, &needed_data);
- if (!ret) {
+ if (!ret && needed_data) {
GST_DEBUG ("buffer didn't contain valid frame");
+ *skipsize = 0;
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
needed_data);
}
gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse),
ADTS_MAX_SIZE);
}
- gst_buffer_unmap (buffer, &map);
-
- return ret;
-}
-
-/**
- * gst_aac_parse_parse_frame:
- * @parse: #GstBaseParse.
- * @buffer: #GstBuffer.
- *
- * Implementation of "parse_frame" vmethod in #GstBaseParse class.
- *
- * Also determines frame overhead.
- * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have
- * a per-frame header. LOAS has 3 bytes.
- *
- * We're making a couple of simplifying assumptions:
- *
- * 1. We count Program Configuration Elements rather than searching for them
- * in the streams to discount them - the overhead is negligible.
- *
- * 2. We ignore CRC. This has a worst-case impact of (num_raw_blocks + 1)*16
- * bits, which should still not be significant enough to warrant the
- * additional parsing through the headers
- *
- * Returns: GST_FLOW_OK if frame was successfully parsed and can be pushed
- * forward. Otherwise appropriate error is returned.
- */
-static GstFlowReturn
-gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
-{
- GstAacParse *aacparse;
- GstBuffer *buffer;
- GstFlowReturn ret = GST_FLOW_OK;
- gint rate, channels;
- GstMapInfo map;
-
- aacparse = GST_AAC_PARSE (parse);
- buffer = frame->buffer;
+ if (G_UNLIKELY (!ret))
+ goto exit;
if (aacparse->header_type == DSPAAC_HEADER_ADTS) {
/* see above */
frame->overhead = 7;
- gst_buffer_map (buffer, &map, GST_MAP_READ);
gst_aac_parse_parse_adts_header (aacparse, map.data,
&rate, &channels, NULL, NULL);
- gst_buffer_unmap (buffer, &map);
GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels);
if (G_UNLIKELY (rate != aacparse->sample_rate
|| channels != aacparse->channels)) {
- GstCaps *sinkcaps;
-
aacparse->sample_rate = rate;
aacparse->channels = channels;
- if ((sinkcaps =
- gst_pad_get_current_caps (GST_BASE_PARSE (aacparse)->sinkpad))) {
- if (!gst_aac_parse_set_src_caps (aacparse, sinkcaps)) {
- /* If linking fails, we need to return appropriate error */
- ret = GST_FLOW_NOT_LINKED;
- }
- gst_caps_unref (sinkcaps);
+ if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
+ /* If linking fails, we need to return appropriate error */
+ ret = GST_FLOW_NOT_LINKED;
}
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
/* see above */
frame->overhead = 3;
- gst_buffer_map (buffer, &map, GST_MAP_READ);
if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
- &channels, NULL)) {
- GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
- } else if (G_UNLIKELY (rate != aacparse->sample_rate
+ &channels, NULL) || !rate || !channels) {
+ /* This is pretty normal when skipping data at the start of
+ * random stream (MPEG-TS capture for example) */
+ GST_DEBUG_OBJECT (aacparse, "Error reading LOAS config. Skipping.");
+ /* Since we don't fully parse the LOAS config, we don't know for sure
+ * how much to skip. Just skip 1 to end up to the next marker and
+ * resume parsing from there */
+ *skipsize = 1;
+ goto exit;
+ }
+
+ if (G_UNLIKELY (rate != aacparse->sample_rate
|| channels != aacparse->channels)) {
aacparse->sample_rate = rate;
aacparse->channels = channels;
GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
channels);
}
- gst_buffer_unmap (buffer, &map);
/* We want to set caps both at start, and when rate/channels change.
Since only some LOAS frames have that info, we may receive frames
before knowing about rate/channels. */
if (setcaps
|| !gst_pad_has_current_caps (GST_BASE_PARSE_SRC_PAD (aacparse))) {
- GstCaps *sinkcaps;
-
- if ((sinkcaps =
- gst_pad_get_current_caps (GST_BASE_PARSE (aacparse)->sinkpad))) {
- if (!gst_aac_parse_set_src_caps (aacparse, sinkcaps)) {
- /* If linking fails, we need to return appropriate error */
- ret = GST_FLOW_NOT_LINKED;
- }
- gst_caps_unref (sinkcaps);
+ if (!gst_aac_parse_set_src_caps (aacparse, NULL)) {
+ /* If linking fails, we need to return appropriate error */
+ ret = GST_FLOW_NOT_LINKED;
}
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse),
}
}
- return ret;
+ if (aacparse->header_type == DSPAAC_HEADER_NONE
+ && aacparse->output_header_type == DSPAAC_HEADER_ADTS) {
+ if (!gst_aac_parse_prepend_adts_headers (aacparse, frame)) {
+ GST_ERROR_OBJECT (aacparse, "Failed to prepend ADTS headers to frame");
+ ret = GST_FLOW_ERROR;
+ }
+ }
+
+exit:
+ gst_buffer_unmap (buffer, &map);
+
+ if (ret) {
+ /* found, skip if needed */
+ if (*skipsize > 0)
+ return GST_FLOW_OK;
+ *skipsize = 0;
+ } else {
+ if (*skipsize < 0)
+ *skipsize = 1;
+ }
+
+ if (ret && framesize <= map.size) {
+ return gst_base_parse_finish_frame (parse, frame, framesize);
+ }
+
+ return GST_FLOW_OK;
+}
+
+static GstFlowReturn
+gst_aac_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)
+{
+ GstAacParse *aacparse = GST_AAC_PARSE (parse);
+
+ if (!aacparse->sent_codec_tag) {
+ GstTagList *taglist;
+ GstCaps *caps;
+
+ /* codec tag */
+ caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));
+ if (caps == NULL) {
+ if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {
+ GST_INFO_OBJECT (parse, "Src pad is flushing");
+ return GST_FLOW_FLUSHING;
+ } else {
+ GST_INFO_OBJECT (parse, "Src pad is not negotiated!");
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+ }
+
+ taglist = gst_tag_list_new_empty ();
+ gst_pb_utils_add_codec_description_to_tag_list (taglist,
+ GST_TAG_AUDIO_CODEC, caps);
+ gst_caps_unref (caps);
+
+ gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);
+ gst_tag_list_unref (taglist);
+
+ /* also signals the end of first-frame processing */
+ aacparse->sent_codec_tag = TRUE;
+ }
+
+ /* As a special case, we can remove the ADTS framing and output raw AAC. */
+ if (aacparse->header_type == DSPAAC_HEADER_ADTS
+ && aacparse->output_header_type == DSPAAC_HEADER_NONE) {
+ guint header_size;
+ GstMapInfo map;
+ frame->out_buffer = gst_buffer_make_writable (frame->buffer);
+ frame->buffer = NULL;
+ gst_buffer_map (frame->out_buffer, &map, GST_MAP_READ);
+ header_size = (map.data[1] & 1) ? 7 : 9; /* optional CRC */
+ gst_buffer_unmap (frame->out_buffer, &map);
+ gst_buffer_resize (frame->out_buffer, header_size,
+ gst_buffer_get_size (frame->out_buffer) - header_size);
+ }
+
+ frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
+
+ return GST_FLOW_OK;
}
GST_DEBUG ("start");
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;
+ 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;
}
return TRUE;
}
+static void
+remove_fields (GstCaps * caps)
+{
+ guint i, n;
+
+ n = gst_caps_get_size (caps);
+ for (i = 0; i < n; i++) {
+ GstStructure *s = gst_caps_get_structure (caps, i);
+
+ gst_structure_remove_field (s, "framed");
+ }
+}
+
+static void
+add_conversion_fields (GstCaps * caps)
+{
+ guint i, n;
+
+ n = gst_caps_get_size (caps);
+ for (i = 0; i < n; i++) {
+ GstStructure *s = gst_caps_get_structure (caps, i);
+
+ if (gst_structure_has_field (s, "stream-format")) {
+ const GValue *v = gst_structure_get_value (s, "stream-format");
+
+ if (G_VALUE_HOLDS_STRING (v)) {
+ const gchar *str = g_value_get_string (v);
+
+ if (strcmp (str, "adts") == 0 || strcmp (str, "raw") == 0) {
+ GValue va = G_VALUE_INIT;
+ GValue vs = G_VALUE_INIT;
+
+ g_value_init (&va, GST_TYPE_LIST);
+ g_value_init (&vs, G_TYPE_STRING);
+ g_value_set_string (&vs, "adts");
+ gst_value_list_append_value (&va, &vs);
+ g_value_set_string (&vs, "raw");
+ gst_value_list_append_value (&va, &vs);
+ gst_structure_set_value (s, "stream-format", &va);
+ g_value_unset (&va);
+ g_value_unset (&vs);
+ }
+ } else if (GST_VALUE_HOLDS_LIST (v)) {
+ gboolean contains_raw = FALSE;
+ gboolean contains_adts = FALSE;
+ guint m = gst_value_list_get_size (v), j;
+
+ for (j = 0; j < m; j++) {
+ const GValue *ve = gst_value_list_get_value (v, j);
+ const gchar *str;
+
+ if (G_VALUE_HOLDS_STRING (ve) && (str = g_value_get_string (ve))) {
+ if (strcmp (str, "adts") == 0)
+ contains_adts = TRUE;
+ else if (strcmp (str, "raw") == 0)
+ contains_raw = TRUE;
+ }
+ }
+
+ if (contains_adts || contains_raw) {
+ GValue va = G_VALUE_INIT;
+ GValue vs = G_VALUE_INIT;
+
+ g_value_init (&va, GST_TYPE_LIST);
+ g_value_init (&vs, G_TYPE_STRING);
+ g_value_copy (v, &va);
+
+ if (!contains_raw) {
+ g_value_set_string (&vs, "raw");
+ gst_value_list_append_value (&va, &vs);
+ }
+ if (!contains_adts) {
+ g_value_set_string (&vs, "adts");
+ gst_value_list_append_value (&va, &vs);
+ }
+
+ gst_structure_set_value (s, "stream-format", &va);
+
+ g_value_unset (&vs);
+ g_value_unset (&va);
+ }
+ }
+ }
+ }
+}
+
static GstCaps *
gst_aac_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)
{
- GstCaps *peercaps;
+ GstCaps *peercaps, *templ;
GstCaps *res;
- /* FIXME: handle filter caps */
+ templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));
- peercaps = gst_pad_get_allowed_caps (GST_BASE_PARSE_SRC_PAD (parse));
- if (peercaps) {
- guint i, n;
+ if (filter) {
+ GstCaps *fcopy = gst_caps_copy (filter);
+ /* Remove the fields we convert */
+ remove_fields (fcopy);
+ add_conversion_fields (fcopy);
+ peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), fcopy);
+ gst_caps_unref (fcopy);
+ } else
+ peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), NULL);
- /* Remove the framed field */
+ if (peercaps) {
peercaps = gst_caps_make_writable (peercaps);
- n = gst_caps_get_size (peercaps);
- for (i = 0; i < n; i++) {
- GstStructure *s = gst_caps_get_structure (peercaps, i);
-
- gst_structure_remove_field (s, "framed");
- }
+ /* Remove the fields we convert */
+ remove_fields (peercaps);
+ add_conversion_fields (peercaps);
- res =
- gst_caps_intersect_full (peercaps,
- gst_pad_get_pad_template_caps (GST_BASE_PARSE_SRC_PAD (parse)),
- GST_CAPS_INTERSECT_FIRST);
+ res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
+ gst_caps_unref (templ);
} else {
- res =
- gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD
- (parse)));
+ res = templ;
+ }
+
+ if (filter) {
+ GstCaps *intersection;
+
+ intersection =
+ gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
+ gst_caps_unref (res);
+ res = intersection;
}
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);
+}
projects / platform / upstream / gstreamer.git / blobdiff