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Revert "aacparse: put codec data on caps for loas format"
This reverts commit e459cf3e01.
This was pushed by accident, the bug should likely be fixed in
libav https://bugzilla.libav.org/show_bug.cgi?id=644
This commit is contained in:
Thiago Santos
parent
04bd422432
commit
0443c2593a
1 changed files with 32 additions and 56 deletions
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@ -146,30 +146,6 @@ gst_aac_parse_init (GstAacParse * aacparse)
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GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (aacparse));
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}
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static GstBuffer *
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gst_aac_parse_create_codec_data (GstAacParse * aacparse)
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{
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GstMapInfo map;
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int idx;
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GstBuffer *codec_data;
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guint16 codec_data_data;
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idx = gst_codec_utils_aac_get_index_from_sample_rate (aacparse->sample_rate);
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if (idx < 0)
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return NULL;
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/* The codec_data data is according to AudioSpecificConfig,
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ISO/IEC 14496-3, 1.6.2.1 */
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codec_data = gst_buffer_new_and_alloc (2);
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gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
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codec_data_data =
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(aacparse->object_type << 11) | (idx << 7) | (aacparse->channels << 3);
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GST_WRITE_UINT16_BE (map.data, codec_data_data);
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gst_buffer_unmap (codec_data, &map);
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return codec_data;
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}
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/**
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* gst_aac_parse_set_src_caps:
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@ -189,7 +165,7 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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gboolean res = FALSE;
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const gchar *stream_format;
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GstBuffer *codec_data;
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gboolean need_codec_data = FALSE;
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guint16 codec_data_data;
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GST_DEBUG_OBJECT (aacparse, "sink caps: %" GST_PTR_FORMAT, sink_caps);
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if (sink_caps)
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@ -213,7 +189,6 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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break;
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case DSPAAC_HEADER_LOAS:
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stream_format = "loas";
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need_codec_data = TRUE;
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break;
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default:
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stream_format = NULL;
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@ -227,12 +202,6 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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if (stream_format)
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gst_structure_set (s, "stream-format", G_TYPE_STRING, stream_format, NULL);
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if (need_codec_data) {
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codec_data = gst_aac_parse_create_codec_data (aacparse);
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if (codec_data != NULL)
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gst_structure_set (s, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
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}
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allowed = gst_pad_get_allowed_caps (GST_BASE_PARSE (aacparse)->srcpad);
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if (!gst_caps_can_intersect (src_caps, allowed)) {
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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@ -243,14 +212,28 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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gst_caps_set_simple (src_caps, "stream-format", G_TYPE_STRING, "raw",
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NULL);
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if (gst_caps_can_intersect (src_caps, allowed)) {
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GstMapInfo map;
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int idx;
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idx =
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gst_codec_utils_aac_get_index_from_sample_rate
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(aacparse->sample_rate);
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if (idx < 0)
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goto not_a_known_rate;
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GST_DEBUG_OBJECT (GST_BASE_PARSE (aacparse)->srcpad,
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"Caps can intersect, we will drop the ADTS layer");
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aacparse->output_header_type = DSPAAC_HEADER_NONE;
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codec_data = gst_aac_parse_create_codec_data (aacparse);
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if (codec_data == NULL)
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goto codec_data_failed;
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/* The codec_data data is according to AudioSpecificConfig,
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ISO/IEC 14496-3, 1.6.2.1 */
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codec_data = gst_buffer_new_and_alloc (2);
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gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
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codec_data_data =
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(aacparse->object_type << 11) |
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(idx << 7) | (aacparse->channels << 3);
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GST_WRITE_UINT16_BE (map.data, codec_data_data);
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gst_buffer_unmap (codec_data, &map);
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gst_caps_set_simple (src_caps, "codec_data", GST_TYPE_BUFFER,
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codec_data, NULL);
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}
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@ -274,7 +257,7 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps)
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gst_caps_unref (src_caps);
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return res;
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codec_data_failed:
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not_a_known_rate:
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gst_caps_unref (allowed);
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gst_caps_unref (src_caps);
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return FALSE;
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@ -527,12 +510,11 @@ gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br,
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/* See table 1.13 in ISO/IEC 14496-3 */
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static gboolean
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gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
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GstBitReader * br, gint * sample_rate, gint * channels,
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guint8 * object_type, guint32 * bits)
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GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits)
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{
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guint8 channel_configuration;
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guint8 audio_object_type, channel_configuration;
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if (!gst_aac_parse_get_audio_object_type (aacparse, br, object_type))
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if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type))
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return FALSE;
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if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
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@ -545,15 +527,15 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
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if (!*channels)
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return FALSE;
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if (*object_type == 5) {
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if (audio_object_type == 5) {
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GST_LOG_OBJECT (aacparse,
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"Audio object type 5, so rereading sampling rate...");
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if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate))
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return FALSE;
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}
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GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels,"
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" audio object type: %u", *sample_rate, *channels, *object_type);
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GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels",
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*sample_rate, *channels);
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/* There's LOTS of stuff next, but we ignore it for now as we have
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what we want (sample rate and number of channels */
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@ -567,8 +549,7 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse,
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static gboolean
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gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
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guint avail, gint * sample_rate, gint * channels,
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guint8 * object_type, gint * version)
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guint avail, gint * sample_rate, gint * channels, gint * version)
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{
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GstBitReader br;
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guint8 u8, v, vA;
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@ -640,14 +621,14 @@ gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data,
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if (!use_same_config) {
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if (v == 0) {
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if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
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sample_rate, channels, object_type, NULL))
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sample_rate, channels, NULL))
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return FALSE;
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} else {
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guint32 bits, asc_len;
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if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len))
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return FALSE;
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if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br,
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sample_rate, channels, object_type, &bits))
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sample_rate, channels, &bits))
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return FALSE;
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asc_len -= bits;
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if (!gst_bit_reader_skip (&br, asc_len))
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@ -870,20 +851,17 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse,
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if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain,
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framesize, &need_data_loas)) {
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gint rate, channels;
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guint8 object_type;
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GST_INFO ("LOAS, framesize: %d", *framesize);
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aacparse->header_type = DSPAAC_HEADER_LOAS;
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if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate,
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&channels, &object_type, &aacparse->mpegversion)) {
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&channels, &aacparse->mpegversion)) {
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GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
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return FALSE;
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}
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aacparse->object_type = object_type;
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if (rate && channels) {
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gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate,
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aacparse->frame_samples, 2, 2);
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@ -1209,7 +1187,6 @@ gst_aac_parse_handle_frame (GstBaseParse * parse,
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GstBuffer *buffer;
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guint framesize;
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gint rate, channels;
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guint8 object_type;
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aacparse = GST_AAC_PARSE (parse);
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buffer = frame->buffer;
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@ -1292,16 +1269,15 @@ gst_aac_parse_handle_frame (GstBaseParse * parse,
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frame->overhead = 3;
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if (!gst_aac_parse_read_loas_config (aacparse, map.data, map.size, &rate,
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&channels, &object_type, NULL)) {
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&channels, NULL)) {
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GST_WARNING_OBJECT (aacparse, "Error reading LOAS config");
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} else if (G_UNLIKELY (rate != aacparse->sample_rate
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|| channels != aacparse->channels)) {
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aacparse->sample_rate = rate;
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aacparse->channels = channels;
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aacparse->object_type = object_type;
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setcaps = TRUE;
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GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels, "
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"audio object type: %u", rate, channels, object_type);
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GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate,
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channels);
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}
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/* We want to set caps both at start, and when rate/channels change.
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