audioencoder: Protect samples_in/bytes_out and audio info with object lock

It might cause invalid calculations during the CONVERT query otherwise.
This commit is contained in:
Sebastian Dröge 2016-07-04 11:07:54 +02:00
parent 8d8262a00c
commit 5cbd1a7bca

View file

@ -479,8 +479,10 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
if (full) {
enc->priv->active = FALSE;
GST_OBJECT_LOCK (enc);
enc->priv->samples_in = 0;
enc->priv->bytes_out = 0;
GST_OBJECT_UNLOCK (enc);
g_list_foreach (enc->priv->ctx.headers, (GFunc) gst_buffer_unref, NULL);
g_list_free (enc->priv->ctx.headers);
@ -491,12 +493,14 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
gst_object_unref (enc->priv->ctx.allocator);
enc->priv->ctx.allocator = NULL;
GST_OBJECT_LOCK (enc);
gst_caps_replace (&enc->priv->ctx.input_caps, NULL);
gst_caps_replace (&enc->priv->ctx.caps, NULL);
gst_caps_replace (&enc->priv->ctx.allocation_caps, NULL);
memset (&enc->priv->ctx, 0, sizeof (enc->priv->ctx));
gst_audio_info_init (&enc->priv->ctx.info);
GST_OBJECT_UNLOCK (enc);
if (enc->priv->upstream_tags) {
gst_tag_list_unref (enc->priv->upstream_tags);
@ -911,7 +915,9 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
GST_BUFFER_OFFSET_END (tmpbuf) = priv->bytes_out + size;
}
GST_OBJECT_LOCK (enc);
priv->bytes_out += size;
GST_OBJECT_UNLOCK (enc);
gst_pad_push (enc->srcpad, tmpbuf);
}
@ -973,7 +979,9 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
}
}
GST_OBJECT_LOCK (enc);
priv->bytes_out += size;
GST_OBJECT_UNLOCK (enc);
if (G_UNLIKELY (priv->discont)) {
GST_LOG_OBJECT (enc, "marking discont");
@ -1118,7 +1126,9 @@ gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
/* mark this already as consumed,
* which it should be when subclass gives us data in exchange for samples */
priv->offset += need;
GST_OBJECT_LOCK (enc);
priv->samples_in += need / ctx->info.bpf;
GST_OBJECT_UNLOCK (enc);
/* subclass might not want to be bothered with leftover data,
* so take care of that here if so, otherwise pass along */
@ -1430,8 +1440,10 @@ gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc, GstCaps * caps)
res = klass->set_format (enc, &state);
if (res) {
GST_OBJECT_LOCK (enc);
ctx->info = state;
gst_caps_replace (&enc->priv->ctx.input_caps, caps);
GST_OBJECT_UNLOCK (enc);
} else {
/* invalidate state to ensure no casual carrying on */
GST_DEBUG_OBJECT (enc, "subclass did not accept format");
@ -1717,8 +1729,11 @@ gst_audio_encoder_sink_query_default (GstAudioEncoder * enc, GstQuery * query)
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res = gst_audio_info_convert (&enc->priv->ctx.info,
src_fmt, src_val, dest_fmt, &dest_val)))
GST_OBJECT_LOCK (enc);
res = gst_audio_info_convert (&enc->priv->ctx.info,
src_fmt, src_val, dest_fmt, &dest_val);
GST_OBJECT_UNLOCK (enc);
if (!res)
goto error;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
res = TRUE;
@ -1919,9 +1934,12 @@ gst_audio_encoder_src_query_default (GstAudioEncoder * enc, GstQuery * query)
gint64 src_val, dest_val;
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
if (!(res = __gst_audio_encoded_audio_convert (&enc->priv->ctx.info,
enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
&dest_fmt, &dest_val)))
GST_OBJECT_LOCK (enc);
res = __gst_audio_encoded_audio_convert (&enc->priv->ctx.info,
enc->priv->bytes_out, enc->priv->samples_in, src_fmt, src_val,
&dest_fmt, &dest_val);
GST_OBJECT_UNLOCK (enc);
if (!res)
break;
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
break;