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ffmpeg: Add codec mapping for AAC LATM/LOAS
Also add the stream-format fields to the CODEC_ID_AAC caps. Fixes bug #650695.
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parent
5ad61c55a6
commit
df40381a83
2 changed files with 36 additions and 4 deletions
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@ -440,6 +440,7 @@ gst_ff_aud_caps_new (AVCodecContext * context, enum CodecID codec_id,
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case CODEC_ID_AC3:
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case CODEC_ID_EAC3:
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case CODEC_ID_AAC:
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case CODEC_ID_AAC_LATM:
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case CODEC_ID_DTS:
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maxchannels = 6;
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break;
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@ -957,8 +958,36 @@ gst_ffmpeg_codecid_to_caps (enum CodecID codec_id,
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break;
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case CODEC_ID_AAC:
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{
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/mpeg",
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"mpegversion", G_TYPE_INT, 4, NULL);
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if (!encode) {
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GValue arr = { 0, };
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GValue item = { 0, };
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g_value_init (&arr, GST_TYPE_LIST);
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g_value_init (&item, G_TYPE_STRING);
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g_value_set_string (&item, "raw");
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gst_value_list_append_value (&arr, &item);
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g_value_set_string (&item, "adts");
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gst_value_list_append_value (&arr, &item);
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g_value_set_string (&item, "adif");
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gst_value_list_append_value (&arr, &item);
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g_value_unset (&item);
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gst_caps_set_value (caps, "stream-format", &arr);
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g_value_unset (&arr);
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} else {
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gst_caps_set_simple (caps, "stream-format", G_TYPE_STRING, "raw", NULL);
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}
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break;
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}
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case CODEC_ID_AAC_LATM: /* LATM/LOAS AAC syntax */
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caps = gst_ff_aud_caps_new (context, codec_id, "audio/mpeg",
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"mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "loas",
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NULL);
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break;
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case CODEC_ID_ASV1:
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@ -827,11 +827,13 @@ gst_ffmpegdec_setcaps (GstPad * pad, GstCaps * caps)
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ffmpegdec->context->flags |= CODEC_FLAG_EMU_EDGE;
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}
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/* for AAC we only use av_parse if not on raw caps */
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if (oclass->in_plugin->id == CODEC_ID_AAC) {
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/* for AAC we only use av_parse if not on stream-format==raw or ==loas */
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if (oclass->in_plugin->id == CODEC_ID_AAC
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|| oclass->in_plugin->id == CODEC_ID_AAC_LATM) {
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const gchar *format = gst_structure_get_string (structure, "stream-format");
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if (format == NULL || strcmp (format, "raw") == 0) {
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if (format == NULL || strcmp (format, "raw") == 0 ||
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oclass->in_plugin->id == CODEC_ID_AAC_LATM) {
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ffmpegdec->turnoff_parser = TRUE;
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}
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}
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@ -2134,7 +2136,8 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
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/* If we don't error out after the first failed read with the AAC decoder,
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* we must *not* carry on pushing data, else we'll cause segfaults... */
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if (len == -1 && in_plugin->id == CODEC_ID_AAC) {
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if (len == -1 && (in_plugin->id == CODEC_ID_AAC
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|| in_plugin->id == CODEC_ID_AAC_LATM)) {
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GST_ELEMENT_ERROR (ffmpegdec, STREAM, DECODE, (NULL),
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("Decoding of AAC stream by FFMPEG failed."));
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*ret = GST_FLOW_ERROR;
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