In certain cases the video thread could end up trying to lock itself
recursively.  This just allows the mutexes to safely be locked
recursively.

This is useful for keeping track of what the current colorspace/range is
if a YUV format is being used.

In video-io.c, video frames could skip, but what would happen is the
frame's timestamp would repeat for the next frame, giving the next frame
a non-monotonic timestamp, and then jump.  This could mess up syncing
slightly when the frame is finally given to an outputs.

70 milliseconds is a bit too high for the default audio timestamp
smoothing threshold.  The full range of error thus becomes 140
milliseconds, which is a bit more than necessary to worry about.  For
the time being, I feel it may be worth it to try 50 milliseconds.

This Fixes a minor flaw with the API where data had to always be mutable
to be usable by the API.

Functions that do not modify the fundamental underlying data of a
structure should be marked as constant, both for safety and to signify
that the parameter is input only and will not be modified by the
function using it.

Typedef pointers are unsafe.  If you do:
typedef struct bla *bla_t;
then you cannot use it as a constant, such as: const bla_t, because
that constant will be to the pointer itself rather than to the
underlying data.  I admit this was a fundamental mistake that must
be corrected.

All typedefs that were pointer types will now have their pointers
removed from the type itself, and the pointers will be used when they
are actually used as variables/parameters/returns instead.

This does not break ABI though, which is pretty nice.

Audio that goes below the minimum expecting timing (current time -
buffering time) is automatically removed.  However, delayed audio is not
removed regardless of its delay.  This puts a hard cap of 6 seconds from
current time that the maximum delay audio can have.  This will also
prevent the circular buffer from dynamically growing too large.

Doing timestamp smoothing in obs-source.c is good because timestamps can
typically operate on a different timebase, however, obs-source.c can
also change that time base dynamically (such as with async video and
unexpected timestamp jumps), so in order to ensure that audio is
seamless in the output as well, perform timestamp smoothing in
audio-io.c as well just as an extra precautionary measure.

This is sort of hard to explain:  the scale_video_output function was
overwriting the current frame.  If scaling was disabled, it would do
nothing, and return success, and all would be well.  If it was enabled,
it would then call the scaler, and then replace the contents of the
'data' function parameter with the scaled frame data.  The problem with
this is that I was passing video_output::cur_frame directly, which
overwrites its previous value with the scaled frame data.  Then if
cur_frame was not updated on time, it would end up trying to scale the
previously scaled image, if that makes sense.  it would call the video
scaler with the same from for both the source and destination.

So the simple fix was to simply use a local variable and pass that in as
a parameter to prevent this bug from occurring.

Renamed:                        To:
-----------------------------------------------------------
audio_output_blocksize          audio_output_get_block_size
audio_output_planes             audio_output_get_planes
audio_output_channels           audio_output_get_channels
audio_output_samplerate         audio_output_get_sample_rate
audio_output_getinfo            audio_output_get_info
audio_output_createline         audio_output_create_line
video_output_getinfo            video_output_get_info
video_gettime                   video_output_get_time
video_getframetime              video_output_get_frame_time
video_output_width              video_output_get_width
video_output_height             video_output_get_height
video_output_framerate          video_output_get_frame_rate
video_output_num_skipped_frames video_output_get_skipped_frames
video_output_total_frames       video_output_get_total_frames

The bug here is that when conversion is active, the source video frame
is initialized with the destination height/width/format instead of the
source height/width/format.

This implements the 'frame skipping' mechanism to forcibly cause frames
to be duplicated in order to reduce encoder complexity so the encoder
can catch up to the video, otherwise it will continue to be
progressively behind and will cause a desync of the video.

Typically, if a user gets this issue, they should turn down their
settings.  For the love of god do not tell them that 'frames are
skipping', just tell them that CPU usage is high, and that they should
consider turning down their settings.

Also, Remove the volume level processing from audio-io.c, it was moved
to obs_source instead.

This fixes a crash on Linux that supposedly occurs when pthread_join
is called multiple times.

Also remove an unused variable from obs-encoder.c (via clang static
analysis)

 - Implement the RTMP output module.  This time around, we just use a
   simple FLV muxer, then just write to the stream with RTMP_Write.
   Easy and effective.

 - Fix the FLV muxer, the muxer now outputs proper FLV packets.

 - Output API:
   * When using encoders, automatically interleave encoded packets
     before sending it to the output.

   * Pair encoders and have them automatically wait for the other to
     start to ensure sync.

   * Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop'
     because it was a bit confusing, and doing this makes a lot more
     sense for outputs that need to stop suddenly (disconnections/etc).

 - Encoder API:
   * Remove some unnecessary encoder functions from the actual API and
     make them internal.  Most of the encoder functions are handled
     automatically by outputs anyway, so there's no real need to expose
     them and end up inadvertently confusing plugin writers.

   * Have audio encoders wait for the video encoder to get a frame, then
     start at the exact data point that the first video frame starts to
     ensure the most accrate sync of video/audio possible.

   * Add a required 'frame_size' callback for audio encoders that
     returns the expected number of frames desired to encode with.  This
     way, the libobs encoder API can handle the circular buffering
     internally automatically for the encoder modules, so encoder
     writers don't have to do it themselves.

 - Fix a few bugs in the serializer interface.  It was passing the wrong
   variable for the data in a few cases.

 - If a source has video, make obs_source_update defer the actual update
   callback until the tick function is called to prevent threading
   issues.

 - obs-outputs module:  Add preliminary code to send out data, and add
   an FLV muxer.  This time we don't really need to build the packets
   ourselves, we can just use the FLV muxer and send it directly to
   RTMP_Write and it should automatically parse the entire stream for us
   without us having to do much manual code at all.  We'll see how it
   goes.

 - libobs:  Add AVC NAL packet parsing code

 - libobs/media-io:  Add quick helper functions for audio/video to get
   the width/height/fps/samplerate/etc rather than having to query the
   info structures each time.

 - libobs (obs-output.c):  Change 'connect' signal to 'start' and 'stop'
   signals.  'start' now specifies an error code rather than whether it
   simply failed, that way the client can actually know *why* a failure
   occurred.  Added those error codes to obs-defs.h.

 - libobs:  Add a few functions to duplicate/free encoder packets

 - Implement OBS encoder interface.  It was previously incomplete, but
   now is reaching some level of completion, though probably should
   still be considered preliminary.

   I had originally implemented it so that encoders only have a 'reset'
   function to reset their parameters, but I felt that having both a
   'start' and 'stop' function would be useful.

   Encoders are now assigned to a specific video/audio media output each
   rather than implicitely assigned to the main obs video/audio
   contexts.  This allows separate encoder contexts that aren't
   necessarily assigned to the main video/audio context (which is useful
   for things such as recording specific sources).  Will probably have
   to do this for regular obs outputs as well.

   When creating an encoder, you must now explicitely state whether that
   encoder is an audio or video encoder.

   Audio and video can optionally be automatically converted depending
   on what the encoder specifies.

   When something 'attaches' to an encoder, the first attachment starts
   the encoder, and the encoder automatically attaches to the media
   output context associated with it.  Subsequent attachments won't have
   the same effect, they will just start receiving the same encoder data
   when the next keyframe plays (along with SEI if any).  When detaching
   from the encoder, the last detachment will fully stop the encoder and
   detach the encoder from the media output context associated with the
   encoder.

   SEI must actually be exported separately; because new encoder
   attachments may not always be at the beginning of the stream, the
   first keyframe they get must have that SEI data in it.  If the
   encoder has SEI data, it needs only add one small function to simply
   query that SEI data, and then that data will be handled automatically
   by libobs for all subsequent encoder attachments.

 - Implement x264 encoder plugin, move x264 files to separate plugin to
   separate necessary dependencies.

 - Change video/audio frame output structures to not use const
   qualifiers to prevent issues with non-const function usage elsewhere.
   This was an issue when writing the x264 encoder, as the x264 encoder
   expects non-const frame data.

   Change stagesurf_map to return a non-const data type to prevent this
   as well.

 - Change full range parameter of video scaler to be an enum rather than
   boolean

...The reason why audio didn't work was because I overwrote the bitrate
values.

As for semaphores, mac doesn't support unnamed semaphores without using
mach semaphores.  So, I just implemented a semaphore wrapper for each
OS.

Had the audio restart slot connected to things that didn't require a
restart

The strings didn't have ending double quotes.  No clue why this didn't
fail in GCC and VC.  Well, VC is horrible but I expected better out of
GCC.