There's a "Render -> Render Audio" operator that pretty much calls into Audaspace functionality directly. This PR adds two new options in there, that boThere's a "Render -> Render Audio" operator that pretty much calls
into Audaspace functionality directly. This PR adds two new options in
there, that both feel like an oversight:
- There's an option for AAC container, when someone wants to render
into .aac file. Previously you could render AAC audio, but only in
MKV (Matroska) container. This is a user request #131980.
- When using MKV (Matroska) or Ogg container, it did not list Opus as
an audio codec that can be used. This felt like an oversight; both
the rest of Blender and Audaspace can handle that, jus the option
was not spelled out in the RNA enums.
Pull Request: https://projects.blender.org/blender/blender/pulls/132877th feel like an oversight:
- There's an option for AAC container, when someone wants to render into `.aac` file. Previously you _could_ render AAC audio, but only in MKV (Matroska) container. This is a user request #131980.
- When using MKV (Matroska) or Ogg container, it did not list Opus as an audio codec that can be used. This felt like an oversight; both the rest of Blender and Audaspace can handle that, jus the option was not spelled out in the RNA enums. Note however that without fix in #132872 the Matroska container continues to be useless in that it produces wrong result.
Upstream Audaspace PR for the 2nd point: https://github.com/neXyon/audaspace/pull/47 (was just merged!)
Pull Request: https://projects.blender.org/blender/blender/pulls/132877
Matroska container really really wants to internally keep on using
milliseconds as a time base. Even if at initialization time both the
m_stream and m_codecCtx are set to use {1, sample_rate} timebase,
later during the avformat_write_header call the audio stream gets set
to {1, 1000}.
Doing av_packet_rescale_ts for each packet after receiving it but
before writing it seems to do all the correct work. This is what is
being done inside Blender movie_write_audio.cc, and in official ffmpeg
mux.c example.
Pull Request: https://projects.blender.org/blender/blender/pulls/132872
The issue was caused by an integer overflow in the `FFMPEGReader::seek`, which
lead to different results depending on an exact compiler version and platform.
The overflow happened in the equation which calculates `m_position`, in its
`packet.pts - st_time` part, which is effectively `int64_t - uint64_t` which is
expected to result in `uint64_t` due to the way how upcasting works in C++.
With the values `packet.pts=0` and `st_time=353600` from the test file the
calculation resulted in a negative value, which then wrapped around due to the
unsigned nature of the type, resulting in very big value on arm64. The value
was wrong, but it did not manifest itself as an error, since the seek code
believed it found the place.
However, when built on x86 platform, the result value was very large negative
value. It is possible that the type upcasting went different, due to the
multiplication with a double value later in the formula. And this large negative
value lead to the seeking code to read the entire file, trying to make it so
the `m_position` is not less than the requested position.
Fortunately, the `AVStream::start_time` is `int64_t`, which leads to a simple
fix which avoids signed/unsigned cast, making the seeking code to properly
calculate `m_position` on both arm64 and x64 platforms, regardless of the
compiler.
Pull Request: https://projects.blender.org/blender/blender/pulls/122149
In ffmpeg 5.0, several variables were made const to try to prevent bad API usage.
Removed some dead code that wasn't used anymore as well.
Reviewed By: Richard Antalik
Differential Revision: http://developer.blender.org/D14063
On the blender side this commit fixes importing video files with audio
and video streams that do not share the same start time and duration.
Differential Revision: https://developer.blender.org/D12353
The duration and start time for audio strips were not correctly read in
audaspace.
Some video files have a "lead in" section of audio that plays before the
video starts playing back. Before this patch, we would play this lead in
audio at the same time as the video started and thus the audio would not
be in sync anymore.
Now the lead in audio is cut off and the duration should be correctly
calculated with this in mind.
If the audio starts after the video, the audio strip is shifted to
account for this, but it will also lead to cut off audio which might not
be wanted. However we don't have a simple way to solve this at this
point.
Differential Revision: http://developer.blender.org/D11917
Due to some floating point errors the last frame of a VSE audio strip can
cause integer overflow and crash Blender. This overflow was caused by a
cast from `int64_t` to `int` without prior check. The crash is fixed by
keeping the variable as `int64_t` for as long as possible.
