This prevents certain issues I've encountered with devices where they
expect to shut down in a specific thread they started up in, as well as
a number of other issues, such as the configuration dialogs.
The configuration dialogs require that a message loop be present, and
this was not the case previously because everything was in the video
thread, which has no windows-specific code.
Configuration/crossbar/etc dialogs will now execute correctly.
This adds support for dynamic format changes on the fly. Format,
resolution, sample rate, can all now be changed by the current
directshow device on the fly.
On an asynchronous video source, the source resolution is automatically
handled by the core, and set to the resolution of the last video data
that was sent. There is no need to manually specify a resolution.
This is not a com pointer; it should not release/close the handle when
an & operator is used, it should only return the handle value. Clearing
is only used on assignment.
This helps ensure that an asynchronous video source is played as close
to its framerate as possible, reduces the risk of duplication as
much as possible, and helps to ensure that playback is as smooth as
possible.
This prevents multiple needless calls to obs_source_get_frame and other
functions. If the texture has already been processed, then just render
it as-is in any subsequent calls to obs_source_video_render.
This is actually unnecessary now that there's a hard limit on the
maximum offset in which audio can be inserted.
This also assumes too much about the audio; it assumes audio is always
on, where as with some devices (such as the elgato) audio is not on
until the stream starts, and when the video has already incremented the
counter.
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.
When setting up the capture, the plugin will now query pulse for the default
format of the specific source instead of the server.
This is useful if a source has different settings than what the defaults are
for the server, e.g. when the source is an output with 5.1 surround sound
and the microphone input is mono while the server defaults to stereo sound.
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.
the pos_x and pos_y variables were somewhat deceptive, because they were
not actually the poition of the cursor. They represented the position
of the cursor's bitmap on the screen, not the position of the cursor.
Some devices burst their audio (such as when querying audio from
directshow), and the 250 millisecond threshold that sets the audio meter
back to muted status would erroneously cause the meter to appear bounce
back between muted and unmuted. Instead, a one second test should be
sufficient time to prevent that from happening.
This implements audio support, allowing not only the ability to capture
the built-in audio from the video device's audio capture pin, but also
the ability to override the default audio with a custom audio device.
The DShowInput::Update function was split up and refactored a bit, as it
was getting a bit large and messy.
If the audio didn't start at the 0 timestamp, it would misinterpret it
as a timestamp jump because obs_source::next_audio_ts_min is set to 0 on
creation. Timestamp starting values should be allowed to start at any
arbitrary value.
This fixes a bug where the pulseaudio plugin always reported
a speaker layout of stereo to obs, regardless of how many channels
pulseaudio actually recorded.
If the default number of channels was different to 2 this would
cause audio distortion.
jp9000
BtbN
Palana
fryshorts
Zachary Lund