* vm_manager: Handle multiple areas in ChangeMemoryState
It is possible that a few areas have the same permisson and state, but with different backing pointers. Currently, this function assumes that only one continous area is found, but this is not always the case.
* service/ldr_ro: Handle multiple areas in VerifyBufferState
It is possible that the buffer passed from the game is made up of multiple areas with the same permisson and state but different backing pointers. Change the check to allow that.
The most important one being adding a mutex to protect the format_context. Apparently it wasn't thread safe (as one'd expect) but I didn't think about that.
Should fix some of the strange issues happening with MP4 muxers, etc.
The file's size is stored in FileSessionSlot and retrieved when the game calls GetSize. However, it is not updated when the file is written to, which can possibly change the file size. Therefore, this can cause GetSize to return incorrect results.
According to HW tests, this vsync event is signaled for activated cameras at about the same frequency as the frame rate. The last 5 vsync timings are recorded (in microseconds) and can be retrieved with the service function.
Also, corrected the default frame_rate to 15, according to HW test.
This should fix the missing camera images in certain games.
You can now directly place ExeFS overrides/patches inside the mod folder (instead of the exefs subfolder). This allows us to have drop-in compatibility with Luma3DS mods.
These two functions allow the frontend to get a list of encoders/formats and their specific options.
Retrieving the options is harder than it sounds due to FFmpeg's strange AVClass and AVOption system. For example, for integer and flags options, 'named constants' can be set. They are of type `AV_OPT_TYPE_CONST` and are categoried according to the `unit` field. An option can recognize all constants of the same `unit`.
Previously, we just used the native sample rate for encoding. However, some encoders like libmp3lame doesn't support it. Therefore, we now use a supported sample rate (preferring the native one if possible).
FFmpeg requires audio data to be sent in a sequence of frames, each containing the same specific number of samples. Previously, we buffered input samples in FFmpegBackend. However, as the source and destination sample rates can now be different, we should buffer resampled data instead. swresample have an internal input buffer, so we now just forward all data to it and 'gradually' receive resampled data, at most one frame_size at a time. When there is not enough resampled data to form a frame, we will record the current offset and request for less data on the next call.
Additionally, this commit also fixes a flaw. When an encoder supports variable frame sizes, its frame size is reported to be 0, which breaks our buffering system. Now we treat variable frame size encoders as having a frame size of 160 (the size of a HLE audio frame).
We previously assumed that the first preferred sample format is planar, but that may not be true for all codecs. Instead we should find a supported sample format that is planar.
While YUV420P is widely used, not all encoders accept it (e.g. Intel QSV only accepts NV12). We should use the codec's preferred pixel format instead as we need to rescale the frame anyway.
