audioencoder: fix tag handling

Merge upstream tags with encoder tags and update whenever
any of those changes.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
This commit is contained in:
Tim-Philipp Müller 2015-08-18 11:45:24 +01:00
parent 8a736f6e98
commit 4c00709e22

View file

@ -258,9 +258,15 @@ struct _GstAudioEncoderPrivate
gboolean hard_min;
gboolean drainable;
/* pending tags */
/* upstream stream tags (global tags are passed through as-is) */
GstTagList *upstream_tags;
/* subclass tags */
GstTagList *tags;
GstTagMergeMode tags_merge_mode;
gboolean tags_changed;
/* pending serialized sink events, will be sent from finish_frame() */
GList *pending_events;
};
@ -490,9 +496,14 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
gst_audio_info_init (&enc->priv->ctx.info);
if (enc->priv->upstream_tags) {
gst_tag_list_unref (enc->priv->upstream_tags);
enc->priv->upstream_tags = NULL;
}
if (enc->priv->tags)
gst_tag_list_unref (enc->priv->tags);
enc->priv->tags = NULL;
enc->priv->tags_merge_mode = GST_TAG_MERGE_APPEND;
enc->priv->tags_changed = FALSE;
g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
@ -611,28 +622,50 @@ gst_audio_encoder_push_pending_events (GstAudioEncoder * enc)
}
}
static inline void
gst_audio_encoder_check_and_push_ending_tags (GstAudioEncoder * enc)
static GstEvent *
gst_audio_encoder_create_merged_tags_event (GstAudioEncoder * enc)
{
if (G_UNLIKELY (enc->priv->tags && enc->priv->tags_changed)) {
GstTagList *merged_tags;
GST_LOG_OBJECT (enc, "upstream : %" GST_PTR_FORMAT, enc->priv->upstream_tags);
GST_LOG_OBJECT (enc, "encoder : %" GST_PTR_FORMAT, enc->priv->tags);
GST_LOG_OBJECT (enc, "mode : %d", enc->priv->tags_merge_mode);
merged_tags =
gst_tag_list_merge (enc->priv->upstream_tags, enc->priv->tags,
enc->priv->tags_merge_mode);
GST_DEBUG_OBJECT (enc, "merged : %" GST_PTR_FORMAT, merged_tags);
if (merged_tags == NULL)
return NULL;
if (gst_tag_list_is_empty (merged_tags)) {
gst_tag_list_unref (merged_tags);
return NULL;
}
/* add codec info to pending tags */
#if 0
GstCaps *caps;
caps = gst_pad_get_current_caps (enc->srcpad);
gst_pb_utils_add_codec_description_to_tag_list (merged_tags,
GST_TAG_AUDIO_CODEC, caps);
#endif
/* add codec info to pending tags */
#if 0
if (!enc->priv->tags)
enc->priv->tags = gst_tag_list_new ();
enc->priv->tags = gst_tag_list_make_writable (enc->priv->tags);
caps = gst_pad_get_current_caps (enc->srcpad);
gst_pb_utils_add_codec_description_to_tag_list (enc->priv->tags,
GST_TAG_CODEC, caps);
gst_pb_utils_add_codec_description_to_tag_list (enc->priv->tags,
GST_TAG_AUDIO_CODEC, caps);
#endif
GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, enc->priv->tags);
gst_audio_encoder_push_event (enc,
gst_event_new_tag (gst_tag_list_ref (enc->priv->tags)));
return gst_event_new_tag (merged_tags);
}
static void
gst_audio_encoder_check_and_push_pending_tags (GstAudioEncoder * enc)
{
if (enc->priv->tags_changed) {
GstEvent *tags_event;
tags_event = gst_audio_encoder_create_merged_tags_event (enc);
if (tags_event != NULL)
gst_audio_encoder_push_event (enc, tags_event);
enc->priv->tags_changed = FALSE;
}
}
@ -760,8 +793,8 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
gst_audio_encoder_push_pending_events (enc);
/* send after pending events, which likely includes newsegment event */
gst_audio_encoder_check_and_push_ending_tags (enc);
/* send after pending events, which likely includes segment event */
gst_audio_encoder_check_and_push_pending_tags (enc);
/* remove corresponding samples from input */
if (samples < 0)
@ -1540,7 +1573,7 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
/* check for pending events and tags */
gst_audio_encoder_push_pending_events (enc);
gst_audio_encoder_check_and_push_ending_tags (enc);
gst_audio_encoder_check_and_push_pending_tags (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
@ -1561,6 +1594,21 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
break;
}
case GST_EVENT_STREAM_START:
{
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
/* Flush upstream tags after a STREAM_START */
GST_DEBUG_OBJECT (enc, "received STREAM_START. Clearing taglist");
if (enc->priv->upstream_tags) {
gst_tag_list_unref (enc->priv->upstream_tags);
enc->priv->upstream_tags = NULL;
enc->priv->tags_changed = TRUE;
}
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = gst_audio_encoder_push_event (enc, event);
break;
}
case GST_EVENT_TAG:
{
GstTagList *tags;
@ -1568,31 +1616,34 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
gst_event_parse_tag (event, &tags);
if (gst_tag_list_get_scope (tags) == GST_TAG_SCOPE_STREAM) {
tags = gst_tag_list_copy (tags);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (enc->priv->upstream_tags != tags) {
tags = gst_tag_list_copy (tags);
/* FIXME: make generic based on GST_TAG_FLAG_ENCODED */
gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_VIDEO_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_SUBTITLE_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
gst_tag_list_remove_tag (tags, GST_TAG_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_NOMINAL_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_MAXIMUM_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_MINIMUM_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_ENCODER);
gst_tag_list_remove_tag (tags, GST_TAG_ENCODER_VERSION);
/* FIXME: make generic based on GST_TAG_FLAG_ENCODED */
gst_tag_list_remove_tag (tags, GST_TAG_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_VIDEO_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_SUBTITLE_CODEC);
gst_tag_list_remove_tag (tags, GST_TAG_CONTAINER_FORMAT);
gst_tag_list_remove_tag (tags, GST_TAG_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_NOMINAL_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_MAXIMUM_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_MINIMUM_BITRATE);
gst_tag_list_remove_tag (tags, GST_TAG_ENCODER);
gst_tag_list_remove_tag (tags, GST_TAG_ENCODER_VERSION);
gst_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
gst_tag_list_unref (tags);
if (enc->priv->upstream_tags)
gst_tag_list_unref (enc->priv->upstream_tags);
enc->priv->upstream_tags = tags;
GST_INFO_OBJECT (enc, "upstream stream tags: %" GST_PTR_FORMAT, tags);
}
gst_event_unref (event);
event = NULL;
res = TRUE;
break;
event = gst_audio_encoder_create_merged_tags_event (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
/* fall through */
}
default:
/* Forward non-serialized events immediately. */
if (!GST_EVENT_IS_SERIALIZED (event)) {
@ -2591,16 +2642,16 @@ gst_audio_encoder_get_drainable (GstAudioEncoder * enc)
/**
* gst_audio_encoder_merge_tags:
* @enc: a #GstAudioEncoder
* @tags: a #GstTagList to merge
* @mode: the #GstTagMergeMode to use
* @tags: (allow-none): a #GstTagList to merge, or NULL to unset
* previously-set tags
* @mode: the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
*
* Adds tags to so-called pending tags, which will be processed
* before pushing out data downstream.
* Sets the audio encoder tags and how they should be merged with any
* upstream stream tags. This will override any tags previously-set
* with gst_audio_encoder_merge_tags().
*
* Note that this is provided for convenience, and the subclass is
* not required to use this and can still do tag handling on its own,
* although it should be aware that baseclass already takes care
* of the usual CODEC/AUDIO_CODEC tags.
* not required to use this and can still do tag handling on its own.
*
* MT safe.
*/
@ -2608,19 +2659,25 @@ void
gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
const GstTagList * tags, GstTagMergeMode mode)
{
GstTagList *otags;
g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
g_return_if_fail (tags == NULL || GST_IS_TAG_LIST (tags));
g_return_if_fail (tags == NULL || mode != GST_TAG_MERGE_UNDEFINED);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (tags)
GST_DEBUG_OBJECT (enc, "merging tags %" GST_PTR_FORMAT, tags);
otags = enc->priv->tags;
enc->priv->tags = gst_tag_list_merge (enc->priv->tags, tags, mode);
if (otags)
gst_tag_list_unref (otags);
enc->priv->tags_changed = TRUE;
if (enc->priv->tags != tags) {
if (enc->priv->tags) {
gst_tag_list_unref (enc->priv->tags);
enc->priv->tags = NULL;
enc->priv->tags_merge_mode = GST_TAG_MERGE_APPEND;
}
if (tags) {
enc->priv->tags = gst_tag_list_ref ((GstTagList *) tags);
enc->priv->tags_merge_mode = mode;
}
GST_DEBUG_OBJECT (enc, "setting encoder tags to %" GST_PTR_FORMAT, tags);
enc->priv->tags_changed = TRUE;
}
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}