In case the application has to deal with fussy servers. User agent
sniffing is so last decade.
Adds a property to set the Flash version on both the sink and the src.
The default stays the same (IIRC, Flash plugin for Linux from 2009).
The former uses a thread-safe way of getting statistics from the
connection without having to protect the fields with a lock.
The latter produces a zeroed statistics structure for use when no
connection exists.
Apply outgoing sizes only after writing the chunk to the peer. This is
important particularly for the set chunk size and allows exposing it
without threading issues.
Move output chunking from gst_rtmp_connection_queue_message into
gst_rtmp_connection_start_write, which effectively moves it from the
streaming thread into the loop thread.
This allows us to handle the outgoing chunk-size message (which is
generated by changing the future chunk-size property) properly, which
could come from any other thread.
Serializes an RTMP message into a series of chunks, all in one buffer.
Similar to what gst_rtmp_connection_queue_message does to serialize
into a GByteArray.
Similar to gst_rtmp_output_stream_write_all_bytes_async, but takes a
GstBuffer instead of a GBytes. It can also return the number of bytes
written, which might be lower in case of an error.
OBJECT_LOCK is used to protect property access only. self->lock is
used to access the RtmpConnection, mostly between the streaming thread
and the loop thread.
To avoid deadlocks involving these two locks, we obey a lock order:
If both self->lock and OBJECT_LOCK are needed, self->lock must be locked
first. Clarify this.
alignment works like in mpegtsmux, joining several MpegTS packets into
one buffer. Default value of 0 joins as many as possible for each
incoming buffer, to optimise CPU usage.
If we have no DTS but a PTS then this means both are the same, and we
should update the last_ts with the PTS. Only if both are unknown then we
don't know the current position and should not update it at all.
Previously we would always update the last_ts to GST_CLOCK_TIME_NONE if
the DTS is unknown, which caused the position to jump around and to
cause spurious gap events to be sent.
Instead of doing it on each packet and doing it based on the distance to
the previous SCR instead of based on the DTS.
Previously we would send gap events for audio all the time if the SCR
distance was 400ms because the threshold for audio is 300ms and by only
ever updating the position when the SCR updates we would always be 100ms
above the threshold and send needless gap events.
This fixes audio glitches on various files caused by gap events.
Some raw h264 encoded files trigger the assignment of wrong PTS to buffers
when some SEI data is provided. This change prevents it to happen.
Also ensure this behavior is being tested.
We might have some old timecodes that are in the future now and have to
drop those to make sure that our queue is correctly ordered and we don't
have multiple timecodes for the same running time.
Directly read them out of the decoder as soon as we passed audio and
then store them in a queue that we handle internally together with their
timestamps. This cleans up memory management and gives us proper control
over the queue instead of guessing how the queue inside the LTC decoder
actually works and when it overflows.
And also introduce 6 instead of 2 frames of latency compared to the LTC
audio input as that seems to be an upper bound for how much the LTC
library is lagging behind.
As the H265/H264 bitstream can support multiple slices,
mastering_display_info_state and content_light_level_state
should be changed only on first slice segment.
Fix#1152
