/**
* 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
#include <string.h>
#include <gst/base/gstbitreader.h>
-#include <gst/pbutils/codec-utils.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
#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);
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);
+
+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_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_static_metadata (element_class,
"AAC audio stream parser", "Codec/Parser/Audio",
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, *allowed;
+ GstCaps *src_caps = NULL, *peercaps;
gboolean res = FALSE;
const gchar *stream_format;
- GstBuffer *codec_data;
+ 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)
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);
- allowed = gst_pad_get_allowed_caps (GST_BASE_PARSE (aacparse)->srcpad);
- if (!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)) {
- GstMapInfo map;
- int idx;
-
- idx =
- gst_codec_utils_aac_get_index_from_sample_rate
- (aacparse->sample_rate);
- if (idx < 0)
- goto not_a_known_rate;
+ 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");
/* The codec_data data is according to AudioSpecificConfig,
ISO/IEC 14496-3, 1.6.2.1 */
- codec_data = gst_buffer_new_and_alloc (2);
- gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
- codec_data_data =
- (aacparse->object_type << 11) |
- (idx << 7) | (aacparse->channels << 3);
- GST_WRITE_UINT16_BE (map.data, codec_data_data);
- gst_buffer_unmap (codec_data, &map);
+ 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, NULL);
+ 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;
}
}
}
- gst_caps_unref (allowed);
+ 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);
return res;
not_a_known_rate:
- gst_caps_unref (allowed);
+ 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_codec_utils_aac_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, "
gst_aac_parse_set_src_caps (aacparse, caps);
if (aacparse->header_type == aacparse->output_header_type)
gst_base_parse_set_passthrough (parse, TRUE);
- } else
+
+ /* 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 {
- aacparse->sample_rate = 0;
- aacparse->channels = 0;
- aacparse->header_type = DSPAAC_HEADER_NOT_PARSED;
- 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;
}
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;
- asc_len -= bits;
if (!gst_bit_reader_skip (&br, asc_len))
return FALSE;
}
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;
*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;
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 signature at offset %u", i);
found = TRUE;
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);
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)) {
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
+ */
}
/**
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);
adts_headers[6] = 0xFCU;
mem = gst_memory_new_wrapped (0, adts_headers, ADTS_HEADERS_LENGTH, 0,
- ADTS_HEADERS_LENGTH, NULL, NULL);
+ ADTS_HEADERS_LENGTH, adts_headers, g_free);
gst_buffer_prepend_memory (frame->out_buffer, mem);
return TRUE;
gboolean lost_sync;
GstBuffer *buffer;
guint framesize;
- gint rate, channels;
+ gint rate = 0, channels = 0;
aacparse = GST_AAC_PARSE (parse);
buffer = frame->buffer;
ret = gst_aac_parse_check_adts_frame (aacparse, map.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);
}
ret = gst_aac_parse_check_loas_frame (aacparse, map.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);
}
frame->overhead = 3;
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;
{
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;
- gst_buffer_map (frame->buffer, &map, GST_MAP_READ);
+ 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->buffer, &map);
- gst_buffer_resize (frame->buffer, header_size,
- gst_buffer_get_size (frame->buffer) - header_size);
+ 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;
}
}
}
+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 *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_caps_make_writable (peercaps);
/* Remove the fields we convert */
remove_fields (peercaps);
+ add_conversion_fields (peercaps);
res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
- res = gst_caps_make_writable (res);
-
- /* Append the template caps because we still want to accept
- * caps without any fields in the case upstream does not
- * know anything.
- */
- gst_caps_append (res, templ);
+ gst_caps_unref (templ);
} else {
res = templ;
}
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