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aacparse: put codec data on caps for loas format

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gst-libav audio decoder also needs codec data for LOAS format, otherwise
it will complain about not having a decoder config and skip all packets

https://bugzilla.gnome.org/show_bug.cgi?id=596772
This commit is contained in:
Thiago Santos 2014-02-25 01:12:05 -03:00
parent f3163fb45f
commit e459cf3e01
1 changed files with 56 additions and 32 deletions
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88
gst/audioparsers/gstaacparse.c
View file

@ -146,6 +146,30 @@ gst_aac_parse_init (GstAacParse * aacparse)
GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (aacparse));
}
static GstBuffer *
gst_aac_parse_create_codec_data (GstAacParse * aacparse)
{
GstMapInfo map;
int idx;
GstBuffer *codec_data;
guint16 codec_data_data;
idx = gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
if (idx < 0)
return NULL;
/* 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);
return codec_data;
}
/**
* gst_aac_parse_set_src_caps:
@ -165,7 +189,7 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
gboolean res = FALSE;
const gchar *stream_format;
GstBuffer *codec_data;
guint16 codec_data_data;
gboolean need_codec_data = FALSE;
GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
if (sink_caps)
@ -189,6 +213,7 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
break;
case DSPAAC_HEADER_LOAS:
stream_format = "loas";
need_codec_data = TRUE;
break;
default:
stream_format = NULL;
@ -202,6 +227,12 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
if (stream_format)
gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
if (need_codec_data) {
codec_data = gst_aac_parse_create_codec_data (aacparse);
if (codec_data != NULL)
gst_structure_set (s, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
}
allowed = gst_pad_get_allowed_caps (GST_BASE_PARSE (aacparse)->srcpad);
if (!gst_caps_can_intersect (src_caps, allowed)) {
GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
@ -212,28 +243,14 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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;
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 = 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 = gst_aac_parse_create_codec_data (aacparse);
if (codec_data == NULL)
goto codec_data_failed;
gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
codec_data, NULL);
}
@ -257,7 +274,7 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
gst_caps_unref (src_caps);
return res;
not_a_known_rate:
codec_data_failed:
gst_caps_unref (allowed);
gst_caps_unref (src_caps);
return FALSE;
@ -510,11 +527,12 @@ gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
/* 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)
GstBitReader * br, gint * sample_rate, gint * channels,
guint8 * object_type, guint32 * bits)
{
guint8 audio_object_type, channel_configuration;
guint8 channel_configuration;
if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
if (!gst_aac_parse_get_audio_object_type (aacparse, br, object_type))
return FALSE;
if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
@ -527,15 +545,15 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
if (!*channels)
return FALSE;
if (audio_object_type == 5) {
if (*object_type == 5) {
GST_LOG_OBJECT (aacparse,
"Audio object type 5, so rereading sampling rate...");
if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
return FALSE;
}
GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
*sample_rate, *channels);
GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels,"
" audio object type: %u", *sample_rate, *channels, *object_type);
/* There's LOTS of stuff next, but we ignore it for now as we have
what we want (sample rate and number of channels */
@ -549,7 +567,8 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
static gboolean
gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
guint avail, gint * sample_rate, gint * channels, gint * version)
guint avail, gint * sample_rate, gint * channels,
guint8 * object_type, gint * version)
{
GstBitReader br;
guint8 u8, v, vA;
@ -621,14 +640,14 @@ gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
if (!use_same_config) {
if (v == 0) {
if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
sample_rate, channels, NULL))
sample_rate, channels, object_type, NULL))
return FALSE;
} else {
guint32 bits, 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))
sample_rate, channels, object_type, &bits))
return FALSE;
asc_len -= bits;
if (!gst_bit_reader_skip (&br, asc_len))
@ -851,17 +870,20 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
framesize, &need_data_loas)) {
gint rate, channels;
guint8 object_type;
GST_INFO ("LOAS, framesize: %d", *framesize);
aacparse->header_type = DSPAAC_HEADER_LOAS;
if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
&channels, &aacparse->mpegversion)) {
&channels, &object_type, &aacparse->mpegversion)) {
GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
return FALSE;
}
aacparse->object_type = object_type;
if (rate && channels) {
gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
aacparse->frame_samples, 2, 2);
@ -1187,6 +1209,7 @@ gst_aac_parse_handle_frame (GstBaseParse * parse,
GstBuffer *buffer;
guint framesize;
gint rate, channels;
guint8 object_type;
aacparse = GST_AAC_PARSE (parse);
buffer = frame->buffer;
@ -1269,15 +1292,16 @@ gst_aac_parse_handle_frame (GstBaseParse * parse,
frame->overhead = 3;
if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
&channels, NULL)) {
&channels, &object_type, NULL)) {
GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
} else if (G_UNLIKELY (rate != aacparse->sample_rate
|| channels != aacparse->channels)) {
aacparse->sample_rate = rate;
aacparse->channels = channels;
aacparse->object_type = object_type;
setcaps = TRUE;
GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
channels);
GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels, "
"audio object type: %u", rate, channels, object_type);
}
/* We want to set caps both at start, and when rate/channels change.