ffmpeg: remove unused and broken video framerate computation

ffmpeg already provides us with a framerate if it is known at the stream
level. If it is not known at this level, the framerate may not be fixed
througout the video, and so the best thing to do is to leave the media
kit field_rate unset, and let applications handle frames according to
their presentation timestamps as they come (this is what we had been
doing anyway, the code to output the framerate to the media kit had been
commented out for a long time and the computed value was not used)

Since ffmpeg removed the ticks_per_frame field [1], the code would not
compile anymore with later ffmpeg versions.

[1] https: //patchwork.ffmpeg.org/project/ffmpeg/patch/20230507133255.20881-12-anton@khirnov.net/

Change-Id: Icc0988c507965883a1570e46f904ebf235fb71bc
Reviewed-on: https://review.haiku-os.org/c/haiku/+/7247
Tested-by: Commit checker robot <no-reply+buildbot@haiku-os.org>
Reviewed-by: waddlesplash <waddlesplash@gmail.com>

ffmpeg: av_init_packet is deprecated

https://patchwork.ffmpeg.org/project/ffmpeg/list/?series=3189

Change-Id: I5ac25141ce76b2f7737e2f1b7ce5c8eac4a90082
Reviewed-on: https://review.haiku-os.org/c/haiku/+/7249
Reviewed-by: waddlesplash <waddlesplash@gmail.com>

ffmpeg: avcodec_find_decoder returns a const AVCodec*

Change-Id: I03af13cc74f8bee7b82bb6398aadbfc28ce92df9
Reviewed-on: https://review.haiku-os.org/c/haiku/+/7248
Reviewed-by: waddlesplash <waddlesplash@gmail.com>

Revert the Codec Kit.

All of Barrett's individual reverts have been squashed into this
one commit, save a few actual bugfixes.

Change-Id: Ib0a7d0a841d3ac40b1fca7372c58b7f9229bd1f0

Codec Kit: Introduce BCodecKit namespace

ffmpeg: Update class names

Codec Kit: Rename files

ffmpeg: fix timestamp tracing

Use ffmpeg functions to format the timestamps.

Change-Id: Idd51feb22fc6c5a70e177604eb995ae2d8601cd4
Reviewed-on: https://review.haiku-os.org/663
Reviewed-by: waddlesplash <waddlesplash@gmail.com>

add-ons/media/ffmpeg: Use SWS for color-space conversion universally.

It seems to be as if not faster than the built-in method now as far
as I can tell, and this means one less arch-specific difference.
I haven't ripped all of it out yet, though.

add-ons/media/ffmpeg: Rework usage of AVPicture in AVCodecDecoder.

It has been deprecated since FFmpeg ~3.0, and is internally implemented
using these functions now, so this should largely be a no-op change.

AVCodecEncoder still uses it.

AVCodecDecoder: Rename fContext to fCodecContext

ffmpeg: Remove obsolete code

ffmpeg: implement deinterlace feature for ffmpeg 3.x.

* Use the yadif filter.
* Fixes #12731.

AVCodecDecoder: fix timing problems

A combination of two problems made things go wrong with the timestamp of
decoded audio.

1) The output buffer size is too small to hold the complete input.
swresample handles this by buffering the input for use the next time it
is called, however repeatedly doing this results in lots of buffering,
and our way to compute the output timestamp from the input does not take
it into account so it does weird things. Moreover, we would need to
empty the buffer by calling swr_convert with NULL input in that case.

Fix: make sure to not feed more data to swr_convert than it can output
in our buffer. This way, no buffering occurs, only the matrixing
conversion.

2) When using planar audio, the "frame size" is a bit different. Instead
of adding sample size * channel count to 1 pointer, we need to add
sample size * 1 to each channel buffer.

Fix: add the "fInputFrameSize" which takes this into account, instead of
misusing fOutputFrameSize for the input.

Fixes #12460.

Use swresample to rematrix audio channels, try 2

This code is closer to what we used before for gcc2, and should not
crash there anymore. I could not get exactly the same code to work, but
reduced the difference to a single statement (which makes gcc2 work and
breaks gcc4). This is protected under a guard with appropriate ffmpeg
version test.

This code works for both gcc2 and gcc4 (tested with mp3 and aac files as
well as youtube videos). It still gets the timestamps wrong with ffmpeg
2.4, but I'll try to solve that another time.

Revert "AVCodecDecoder: use swresample to interleave audio channels."

Commit 856cc59e58bb83de4a69daa680748ba9b52592d8 didn't really "fix"
anything; it just broke audio pretty much everywhere but YouTube,
and there videos play at 2x speed so it wasn't really worth it.

Stopgap solution for #12509.

ffmpeg: fix x86_64 build.

AVCodecDecoder: use swresample to interleave audio channels.

Some codecs will always output audio in planar mode no matter what we
request. This is the case for example with AAC used for youtube. We now
use swresample to convert from planar to packed format.

Note that since swresample does its own buffering, we could probably do
away with some of the code that handled buffering before, making the
audio pipeline simpler and faster.

Fixes audio in youtube, but now the video plays at 2x speed. It seems
something is wrong with the timestamps. Possible things to investigate:
* why do we use the packet dts instead of the pts from the frames anyway?
* the pts and pkt_dts are in "stream time_base units". We seem to assume
microseconds for audio but this is probably not the case. Or did I
miss where the conversion is done?

FFMPEG Plugin: Small refactoring in audio path.

- Lazy initializing the fDecodedData variable fits best in
_ResetRawDecodedAudio() as fRawDecodedAudio is all about managing properties
of fDecodedData. So fDecodedData having some memory allocated is seen as a
property here.

- Updated documentation accordingly

- No functional change intended.

FFMPEG Plugin: Fix performance regression on video path.

- For best performance the color conversion function expects a memory location
aligned to 32 bytes. Without that alignment the color conversion function
falls back to a slower conversion path. This fix was measured via
DO_PROFILING and tested with various 1080p video files.

FFMPEG Plugin: Implement audio input buffer padding.

- Padding is required by FFMPEG for correct operation of all audio decoders.
FFMPEG performs some speed optimizations under the hood that may lead to
reading over the end of the chunk buffer wouldn't there have been padding
applied.

- Resolve TODOs by unifying fVideoChunkBuffer and fChunkBuffer back into
fChunkBuffer because audio path is responsible for freeing fChunkBuffer now.
Resolved TODOs apply to the replacing fVideoChunkBuffer variable by
fChunkBuffer, rename some methods by removing the "Video" part, collapse two
methods into one (_LoadNextChunkIfNeededAndAssignStartTime()).
No functional change intended.

- Enhance "logging stream to file" functionality to write to distinct logging
files for audio and video. Before this commit one could only log video
streams. But with unifying the _LoadNextChunkIfNeededAndAssignStartTime()
audio streams gained the logging functionality for free. But now audio and
video streams would be written in the same log file when watching a media
file containing both audio and video. This is prevented by the distinct
logging mentioned above.

- Update documentation accordingly.

FFMPEG Plugin: Refactor out copying of audio format properties.

- Main reasons for this refactoring of negotiation steps are to increase
readability and to be on par with the video path.

- Rearrange some negotiation steps in video path to be on par with audio path.

- Add documentation for the new method and update existing documentation
accordingly.

- No functional change intended.

FFMPEG Plugin: Refactor out audio frames chunk decoding.

- Main purpose is to prepare auto detection of audio frame properties for
media formats that encode those properties in the frames themself (e.g. MP3)
instead of in the container format (e.g. WMA).
The main difference between akin named methods _DecodeNextAudioFrame() and
_DecodeNextAudioFrameChunk() is that the former method deals with providing
the exact number of audio frames expected by the caller of
BMediaDecoder::Decode() and the latter deals with decoding any number of
audio frames at all.

- New documentation added and existing documentationupdated accordingly.

- No functional change intended.

FFMPEG Plugin: Fill out more media header fields in audio path.

- Some small refactoring when resetting fRawDecodedAudio. Instead of letting
FFMPEG reset fRawDecodedAudio we do it manually to preserve the allocated
memory in fRawDecodedAudio->opaque (otherwise FFMPEG's
avcodec_get_frame_defaults() would NULLify the opaque pointer without
releasing the allocated memory.

- Keep track of the total size of fDecodedData in fRawDecodedAudio->linesize[0]
instead of relying on calculating it every time it is needed. This makes the
code more comprehensible.

FFMPEG Plugin: Refactor out checking of invalid conditions in audio path.

- Main reason for this refactoring is to increase readability and thus make
audio decode path more comprehensible.

- Added documentation for the new method accordingly.

- Small change in calculating the decoded data size to clear when error occurs
during decoding. This way it is more readable and more consistent with
calculations of decoded data size on other locations.

- No functional change intended.

FFMPEG Plugin: Refactor out moving audio frames from buffer.

- Main reason for this refactoring is to increase readability and thus make the
audio decode path more comprehensible.

- Added documentation for the new method accordingly.

- No functional change intended.

FFMPEG Plugin: Refactor out chunk loading in audio path.

- Main reasons are to increase readability of audio path and to demonstrate
that chunk loading in audio and video path is the same code that can be
focused in one method (instead of two at the moment). Added a TODO for
collapsing both methods into one and the conditions that must hold true to
do so (just in case I'll be hitted by a bus and someone else has to proceed).
Collapsing is scheduled for a later commit.

- Added documentation for the new method accordingly.

- Make use of full line length in comments of
_LoadNextVideoChunkIfNeededAndAssignStartTime().

- No functional change intended.

FFMPEG Plugin: Refactor out actual audio decoding part.

- Main reason for this refactoring is to increase readability and thus make the
audio decode path more comprehensible.

- Added documentation for the new method accordingly.

- Small refactoring for detecting when to update fRawDecodedAudio's properties.
This is a preparation step for factoring out the flushing of the
fDecodedDataBuffer in a later commit.

- No functional change intended.

FFMPEG Plugin: Refactor start time handling in audio path.

- FFMPEG handles the relationship of start time between encoded and decoded
audio data now by using the fTempPacket->dts and the
fDecodedDataBuffer->pkt_dts fields. We still have to manually keep track of
start times for consecutive audio frames though to support returning a number
of audio frames that may assembled of partial AVFrames.

- The start time of the very first audio frame data packet returned by Decode()
is now correctly calculated based on GetNextChunk() start times instead of
being always zero.

- Introduce fRawDecodedAudio that serves as a container to store properties of
the audio frames stored in fDecodedData. This prepares the population of the
fHeader structure with audio frame properties needed to allow clients of
BMediaDecoder::Decode() detect audio format changes in a later commit.

- Remove fStartTime as it is superflous now.

FFMPEG Plugin: Refactor scope of fTempPacket in audio path.

- There are two main reasons for this refactoring:
1. Prepare using FFMPEGs functionality of audio frame start time assignment
(instead of rolling it ourself) like already done for the video path
(see _LoadNextVideoChunkIfNeededAndAssignStartTime() for reference).
2. Get rid of fChunkBufferOffset (this is a minor reason though).

- Untangle some of the conditional checks to increase readability.

- No functional change intended.

FFMPEG Plugin: Rename some variables in audio path.

- Make the difference between fDecodedData and fDecodedDataBuffer more clear.

- No functional change intended.

FFMPEG Plugin: Refactor audio decoding method into two.

- First method is solely responsible to fill the audio output buffer with
already decoded audio frames.
Second method is solely responsible for decoding the encoded audio data and
put it in the decoded audio output buffer for further processing with the
first method.
This prepares auto detection of audio frame properties for audio formats
where the properties are contained within the encoded audio frame (e.g. MP3),
instead within the audio container format (e.g. WMA). Implementing auto
detection is scheduled for a later commit though.

- Added documentation accordingly.

- No functional change intended.

FFMPEG Plugin: Fix playing video files.

- This should fix the bug where video files that played well before the recent
changes to the FFMPEG Plugin didn't play anymore. Now we apply the essential
video container properties (that were passed by with Setup()) to the
AVCodecContext. Some video formats simply store those properties in the
container only (e.g. AVI, WMV) and not in the video frames itself
(e.g. MPEG2).
Tested with several files from samples.ffmpeg.org and from the FATE suite of
FFMPEG.

FFMPEG Plugin: Fix video start_time handling

- I misinterpreted the semantics of reordered_opaque. I thought it would
establish the correct relationship between the start_time returned by
GetNextChunk() and the next video frame successfully decoded. But for this
to work reordered_opaque expects to be filled with presentation time stamps.
A series of presentation time stamps may be jumping back in time due to the
presence of B-frames. The decoded frame presentation time stamps series would
then be ordered in a monotonically increased way.
But actually GetNextChunk() always returns monotonically increasing start
times. Mapping this behaviour to FFMPEG's expectations means labeling those
start times as decoding time stamps (dts). Though for those start times to be
related to the correct decoded video frames you have to assign the start time
with the AVPacket containing the data to be decoded.

- This commit finally makes DVB video playback working for me with the TV app.
Though no audio yet.

- The documentation was updated accordingly.

FFMPEG Plugin: Fix bug and refactor input buffer padding.

- Fixes a bug using realloc with a memory area that is declared const which
lead to a crash in MediaPlayer playing big_buck_bunny_720p_stereo.ogg.
- The refactoring introduces a strict separation between const memory areas
(chunk data read from GetNextChunk()) and mutable memory areas
(fVideoChunkBuffer) by using a copy operation instead of a casted
assignment operation.
- Updated documentation accordingly.
- Besides fixing the bug, there is no functional change intended.

FFMPEG Plugin: Refactor out loading next video chunk.

- Main reason for refactoring was to increase readability of
_DecodeNextVideoFrame() by simplifying it. Refactoring was tested
successfully for no functional change with mpeg2_decoder_test.
- Reindented the method definition in the header file so that the new method
_LoadNextVideoChunkIfNeededAndUpdateStartTime() fits into 80 chars per line.
Reindentdation is applied to methods only as the member variables have no
space left for reindentation.
- Update documentation accordingly.
- Fix wording of audio part to audio path.
- No functional change intended.

FFMPEG Plugin: Implement video input buffer padding.

- Padding is required by FFMPEG for correct operation of all video decoders.
FFMPEG performs some speed optimizations under the hood that may lead to
reading over the end of the chunk buffer wouldn't there have been padding
applied.
- Note: Padding is required for audio decoders, too. I will tackle this in some
later commits. For the time being we have a degradation in code reuse, due to
different memory ownership of chunk buffers in audio and video decoder path.
Audio path must not care about freeing chunk buffers whereas video path must.
- Fix coding style and some typos.
- Update documentation accordingly.

FFMPEG Plugin: Implement flushing of video frames from decoder.

- Video frames still contained in the video decoder are now flushed when there
are no encoded data chunks left (signaled by B_LAST_BUFFER_ERROR).
- New code was successfully tested with mpeg2_decoder_test. The last test image
is now decoded, too, resulting in a bump of the expected number of decoded
images.
- Refactor code to support flushing -and- keep the readability at a sane level.
- Remove some uncommented code for the sake of readability.
- Documentation updated accordingly.

FFMPEG Plugin: Automatic detection of video decoding parameters.

- Kudos to Marcus Overhagen for laying out the general idea of automatic
detection by sharing some of his dvb code examples with me.
- Simplify pixel format search code by removing the loop and let FFMPEG
decide what pixel format to use based on the actual video data.
- Automatically detect the video frame rate based on the actual video data.
- Remove fOutputVideoFormat to avoid synchronizing values in two distinct
places. The member variable fHeader is the main place for important decoder
parameters now.
- Introduce fOutputColorSpace containing the color space that was previously
tracked in the fOutputVideoFormat member variable.
- Update the documentation accordingly.

FFMPEG Plugin: Code styles, typos and code deduplication.

- Kudos to stippi for taking the time pointing those out :)
http://www.freelists.org/post/haiku-commits/haiku-hrev47576-srcaddonsmediapluginsffmpeg,1
- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>

FFMPEG-Plugin: Refactor out update of media_header.

- Main purpose is to make reading the function DecodeNextFrame() easier on the
eyes, by moving out auxiliary code.
Note: The media_header update code for the start_time is still left in
DecodeNextFrame(). This will be addressed in a later commit specially
targetted on handling start_time calculations for incomplete video frames.

- Also updated / added some documentation.

- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>

FFMPEG plugin: Refactor video decoding function.

- Factor out the deinterlacing and color converting part to make the code more
readable. This makes it easier to understand which code belongs to the actual
decoding process and which code to the post processing.

- There seems to be no performance impact involved (I just looked at the spikes
of the process manager) in factoring out this part, but one can always inline
the method if a closer performance assesment (e.g. by enabling the profiling
the existing profiling code) suggests so.

- Document the _DecodeVideo() method a little bit. Maybe someone can document
the info parameter, as I'm a little bit clueless here.

- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>
(cherry picked from commit c5fa095fa73d47e75a46cfc138a56028fcc01819)

FFMPEG plugin: Refactor video decoding function.

- We factor out the code that does the decoding, deinterlacing and color space
converting into its own function. This prepares auto detection of video frame
properties in a later commit. Auto detection means (for example), that you
don't need to know the size of the decoded video frame -before- intitializing
the video codec as it is the case with the current implementation.

- Use the already existent private member variable fHeader to fill out and
return video frame properties.

- Rename some variables to be more self describing.

- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>
(cherry picked from commit 6beab04f48e98022041895750f38c5a35a5f4f9e)

ffmpeg: don't use deprecated API in AVCodecDecoder.

* Renamed fAudioTempPacket to fTempPacket and use it for calling
the new video decoding function. This just avoids a warning
generated from the libavcodec sources. The function used before
did the exact same thing...
* Maintain fStartTime correctly in _DecodeVideo(). Don't overwrite
it with a calculated starttime in Decode(). This will allow drift
to bubble up to the higher layers.
* Do not use the previously required hack to close and reopen the
AVCodec after seeking. avcodec_flush_buffers() seems to work
fine now, and for certain stream types (MPEG1, MPEG2 video for
example) the keyframe is correctly used after seeking.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38806 a95241bf-73f2-0310-859d-f6bbb57e9c96

Enabled any and all decoders and demuxers which are currently compiled into
FFmpeg. It's a bit sad, but this obsoletes pretty much all other decoder
and reader plugins. Some of them were built on external libraries as well
(AC3 (not part of default image anyway, since it's GPL), APE, MusePack),
so it's not really a big difference to using FFmpeg as external library.
The format matching is greatly simplified by using B_MISC_FORMAT_FAMILY
for everything but raw audio, and the actual FFmpeg CodecID as codec tag.
The downside of this is that the AVFormatReader can no longer be used with
other decoder plugins, but it would be easy to add special cases for native
decoders we wish to support. Obviously the out of the box support for file
formats and decoders has greatly increased with this change, so there has
to be a pretty good reason now for writing a "native" decoder or reader.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38786 a95241bf-73f2-0310-859d-f6bbb57e9c96

Finall changed the Seek() method of Decoder plug-ins to
SeekedTo(), since it's only informative to decoders. They
can't modify the seeked frame/time. This also mirrors what
all existing decoders were doing in Seek(). BMediaTrack
is simplified accordingly (resolved two TODOs).


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38705 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Use the new avcodec_decode_audio3() call, avcodec_decode_audio2()
gives the deprecated warning... We need to cache an AVPacket for this.
* Check the allocation of fOutputBuffer.
* When seeking, we need to flush the already decoded stuff
in fOutputBuffer, and throw away the last chunk buffer as well.
* Handle an incomplete input format at least to the point of not
crashing with a divide error (mp3_reader would give us such an
incomplete format for example).
* _DecodeAudio():
- Fixed some edge cases in the audio decoding loop: avcodec_decode_audio3()
can return a 0 length, which means no error, but no decoded frames
either. ffplay throws away the chunk in this case, do the same.
- Convert some invalid situations that were printf()s into debugger()s.
- Add much more comments to explain how everything works.
* Fixed the occasional coding style issue.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38400 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Optionally use libswscale for color space conversion. Have not done any
performance tests, but I wrote this code before David got around to improve
his color space assembler code and didn't want to throw it away. Turned
off for the time being, since David's version works fine.
* Supply a codec sub_id in the media_codec_info. This allows matching
decoders to encoders, which comes in handy for Smart Rendering in Clockwerk.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@34931 a95241bf-73f2-0310-859d-f6bbb57e9c96

Sorry, this was supposed to be part of the last commit...


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@33356 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Automatic whitespace cleanup.
* media_decode_info member was never used.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@32015 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Coding style cleanup, some removal of dead code.
* Refactored NegotiateOutputFormat() and Decode() into two separate private
methods each, one for video and one for audio.
* Keep reading chunks when video decoding, until we have got a picture. This
gets us scrambled video instead of a black picture for h264 in mpegts.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@31457 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Renamed avcodec folder to ffmpeg.
* Cleaned up plugins Jamfile.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@31359 a95241bf-73f2-0310-859d-f6bbb57e9c96

FFMPEG Plugin: Small refactoring in audio path.

- Lazy initializing the fDecodedData variable fits best in
_ResetRawDecodedAudio() as fRawDecodedAudio is all about managing properties
of fDecodedData. So fDecodedData having some memory allocated is seen as a
property here.

- Updated documentation accordingly

- No functional change intended.

FFMPEG Plugin: Fix performance regression on video path.

- For best performance the color conversion function expects a memory location
aligned to 32 bytes. Without that alignment the color conversion function
falls back to a slower conversion path. This fix was measured via
DO_PROFILING and tested with various 1080p video files.

FFMPEG Plugin: Implement audio input buffer padding.

- Padding is required by FFMPEG for correct operation of all audio decoders.
FFMPEG performs some speed optimizations under the hood that may lead to
reading over the end of the chunk buffer wouldn't there have been padding
applied.

- Resolve TODOs by unifying fVideoChunkBuffer and fChunkBuffer back into
fChunkBuffer because audio path is responsible for freeing fChunkBuffer now.
Resolved TODOs apply to the replacing fVideoChunkBuffer variable by
fChunkBuffer, rename some methods by removing the "Video" part, collapse two
methods into one (_LoadNextChunkIfNeededAndAssignStartTime()).
No functional change intended.

- Enhance "logging stream to file" functionality to write to distinct logging
files for audio and video. Before this commit one could only log video
streams. But with unifying the _LoadNextChunkIfNeededAndAssignStartTime()
audio streams gained the logging functionality for free. But now audio and
video streams would be written in the same log file when watching a media
file containing both audio and video. This is prevented by the distinct
logging mentioned above.

- Update documentation accordingly.

FFMPEG Plugin: Refactor out copying of audio format properties.

- Main reasons for this refactoring of negotiation steps are to increase
readability and to be on par with the video path.

- Rearrange some negotiation steps in video path to be on par with audio path.

- Add documentation for the new method and update existing documentation
accordingly.

- No functional change intended.

FFMPEG Plugin: Refactor out audio frames chunk decoding.

- Main purpose is to prepare auto detection of audio frame properties for
media formats that encode those properties in the frames themself (e.g. MP3)
instead of in the container format (e.g. WMA).
The main difference between akin named methods _DecodeNextAudioFrame() and
_DecodeNextAudioFrameChunk() is that the former method deals with providing
the exact number of audio frames expected by the caller of
BMediaDecoder::Decode() and the latter deals with decoding any number of
audio frames at all.

- New documentation added and existing documentationupdated accordingly.

- No functional change intended.

FFMPEG Plugin: Fill out more media header fields in audio path.

- Some small refactoring when resetting fRawDecodedAudio. Instead of letting
FFMPEG reset fRawDecodedAudio we do it manually to preserve the allocated
memory in fRawDecodedAudio->opaque (otherwise FFMPEG's
avcodec_get_frame_defaults() would NULLify the opaque pointer without
releasing the allocated memory.

- Keep track of the total size of fDecodedData in fRawDecodedAudio->linesize[0]
instead of relying on calculating it every time it is needed. This makes the
code more comprehensible.

FFMPEG Plugin: Refactor out checking of invalid conditions in audio path.

- Main reason for this refactoring is to increase readability and thus make
audio decode path more comprehensible.

- Added documentation for the new method accordingly.

- Small change in calculating the decoded data size to clear when error occurs
during decoding. This way it is more readable and more consistent with
calculations of decoded data size on other locations.

- No functional change intended.

FFMPEG Plugin: Refactor out moving audio frames from buffer.

- Main reason for this refactoring is to increase readability and thus make the
audio decode path more comprehensible.

- Added documentation for the new method accordingly.

- No functional change intended.

FFMPEG Plugin: Refactor out chunk loading in audio path.

- Main reasons are to increase readability of audio path and to demonstrate
that chunk loading in audio and video path is the same code that can be
focused in one method (instead of two at the moment). Added a TODO for
collapsing both methods into one and the conditions that must hold true to
do so (just in case I'll be hitted by a bus and someone else has to proceed).
Collapsing is scheduled for a later commit.

- Added documentation for the new method accordingly.

- Make use of full line length in comments of
_LoadNextVideoChunkIfNeededAndAssignStartTime().

- No functional change intended.

FFMPEG Plugin: Refactor out actual audio decoding part.

- Main reason for this refactoring is to increase readability and thus make the
audio decode path more comprehensible.

- Added documentation for the new method accordingly.

- Small refactoring for detecting when to update fRawDecodedAudio's properties.
This is a preparation step for factoring out the flushing of the
fDecodedDataBuffer in a later commit.

- No functional change intended.

FFMPEG Plugin: Refactor start time handling in audio path.

- FFMPEG handles the relationship of start time between encoded and decoded
audio data now by using the fTempPacket->dts and the
fDecodedDataBuffer->pkt_dts fields. We still have to manually keep track of
start times for consecutive audio frames though to support returning a number
of audio frames that may assembled of partial AVFrames.

- The start time of the very first audio frame data packet returned by Decode()
is now correctly calculated based on GetNextChunk() start times instead of
being always zero.

- Introduce fRawDecodedAudio that serves as a container to store properties of
the audio frames stored in fDecodedData. This prepares the population of the
fHeader structure with audio frame properties needed to allow clients of
BMediaDecoder::Decode() detect audio format changes in a later commit.

- Remove fStartTime as it is superflous now.

FFMPEG Plugin: Refactor scope of fTempPacket in audio path.

- There are two main reasons for this refactoring:
1. Prepare using FFMPEGs functionality of audio frame start time assignment
(instead of rolling it ourself) like already done for the video path
(see _LoadNextVideoChunkIfNeededAndAssignStartTime() for reference).
2. Get rid of fChunkBufferOffset (this is a minor reason though).

- Untangle some of the conditional checks to increase readability.

- No functional change intended.

FFMPEG Plugin: Rename some variables in audio path.

- Make the difference between fDecodedData and fDecodedDataBuffer more clear.

- No functional change intended.

FFMPEG Plugin: Refactor audio decoding method into two.

- First method is solely responsible to fill the audio output buffer with
already decoded audio frames.
Second method is solely responsible for decoding the encoded audio data and
put it in the decoded audio output buffer for further processing with the
first method.
This prepares auto detection of audio frame properties for audio formats
where the properties are contained within the encoded audio frame (e.g. MP3),
instead within the audio container format (e.g. WMA). Implementing auto
detection is scheduled for a later commit though.

- Added documentation accordingly.

- No functional change intended.

FFMPEG Plugin: Fix playing video files.

- This should fix the bug where video files that played well before the recent
changes to the FFMPEG Plugin didn't play anymore. Now we apply the essential
video container properties (that were passed by with Setup()) to the
AVCodecContext. Some video formats simply store those properties in the
container only (e.g. AVI, WMV) and not in the video frames itself
(e.g. MPEG2).
Tested with several files from samples.ffmpeg.org and from the FATE suite of
FFMPEG.

FFMPEG Plugin: Fix video start_time handling

- I misinterpreted the semantics of reordered_opaque. I thought it would
establish the correct relationship between the start_time returned by
GetNextChunk() and the next video frame successfully decoded. But for this
to work reordered_opaque expects to be filled with presentation time stamps.
A series of presentation time stamps may be jumping back in time due to the
presence of B-frames. The decoded frame presentation time stamps series would
then be ordered in a monotonically increased way.
But actually GetNextChunk() always returns monotonically increasing start
times. Mapping this behaviour to FFMPEG's expectations means labeling those
start times as decoding time stamps (dts). Though for those start times to be
related to the correct decoded video frames you have to assign the start time
with the AVPacket containing the data to be decoded.

- This commit finally makes DVB video playback working for me with the TV app.
Though no audio yet.

- The documentation was updated accordingly.

FFMPEG Plugin: Fix bug and refactor input buffer padding.

- Fixes a bug using realloc with a memory area that is declared const which
lead to a crash in MediaPlayer playing big_buck_bunny_720p_stereo.ogg.
- The refactoring introduces a strict separation between const memory areas
(chunk data read from GetNextChunk()) and mutable memory areas
(fVideoChunkBuffer) by using a copy operation instead of a casted
assignment operation.
- Updated documentation accordingly.
- Besides fixing the bug, there is no functional change intended.

FFMPEG Plugin: Refactor out loading next video chunk.

- Main reason for refactoring was to increase readability of
_DecodeNextVideoFrame() by simplifying it. Refactoring was tested
successfully for no functional change with mpeg2_decoder_test.
- Reindented the method definition in the header file so that the new method
_LoadNextVideoChunkIfNeededAndUpdateStartTime() fits into 80 chars per line.
Reindentdation is applied to methods only as the member variables have no
space left for reindentation.
- Update documentation accordingly.
- Fix wording of audio part to audio path.
- No functional change intended.

FFMPEG Plugin: Implement video input buffer padding.

- Padding is required by FFMPEG for correct operation of all video decoders.
FFMPEG performs some speed optimizations under the hood that may lead to
reading over the end of the chunk buffer wouldn't there have been padding
applied.
- Note: Padding is required for audio decoders, too. I will tackle this in some
later commits. For the time being we have a degradation in code reuse, due to
different memory ownership of chunk buffers in audio and video decoder path.
Audio path must not care about freeing chunk buffers whereas video path must.
- Fix coding style and some typos.
- Update documentation accordingly.

FFMPEG Plugin: Implement flushing of video frames from decoder.

- Video frames still contained in the video decoder are now flushed when there
are no encoded data chunks left (signaled by B_LAST_BUFFER_ERROR).
- New code was successfully tested with mpeg2_decoder_test. The last test image
is now decoded, too, resulting in a bump of the expected number of decoded
images.
- Refactor code to support flushing -and- keep the readability at a sane level.
- Remove some uncommented code for the sake of readability.
- Documentation updated accordingly.

FFMPEG Plugin: Automatic detection of video decoding parameters.

- Kudos to Marcus Overhagen for laying out the general idea of automatic
detection by sharing some of his dvb code examples with me.
- Simplify pixel format search code by removing the loop and let FFMPEG
decide what pixel format to use based on the actual video data.
- Automatically detect the video frame rate based on the actual video data.
- Remove fOutputVideoFormat to avoid synchronizing values in two distinct
places. The member variable fHeader is the main place for important decoder
parameters now.
- Introduce fOutputColorSpace containing the color space that was previously
tracked in the fOutputVideoFormat member variable.
- Update the documentation accordingly.

FFMPEG Plugin: Code styles, typos and code deduplication.

- Kudos to stippi for taking the time pointing those out :)
http://www.freelists.org/post/haiku-commits/haiku-hrev47576-srcaddonsmediapluginsffmpeg,1
- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>

FFMPEG-Plugin: Refactor out update of media_header.

- Main purpose is to make reading the function DecodeNextFrame() easier on the
eyes, by moving out auxiliary code.
Note: The media_header update code for the start_time is still left in
DecodeNextFrame(). This will be addressed in a later commit specially
targetted on handling start_time calculations for incomplete video frames.

- Also updated / added some documentation.

- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>

FFMPEG plugin: Refactor video decoding function.

- Factor out the deinterlacing and color converting part to make the code more
readable. This makes it easier to understand which code belongs to the actual
decoding process and which code to the post processing.

- There seems to be no performance impact involved (I just looked at the spikes
of the process manager) in factoring out this part, but one can always inline
the method if a closer performance assesment (e.g. by enabling the profiling
the existing profiling code) suggests so.

- Document the _DecodeVideo() method a little bit. Maybe someone can document
the info parameter, as I'm a little bit clueless here.

- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>
(cherry picked from commit c5fa095fa73d47e75a46cfc138a56028fcc01819)

FFMPEG plugin: Refactor video decoding function.

- We factor out the code that does the decoding, deinterlacing and color space
converting into its own function. This prepares auto detection of video frame
properties in a later commit. Auto detection means (for example), that you
don't need to know the size of the decoded video frame -before- intitializing
the video codec as it is the case with the current implementation.

- Use the already existent private member variable fHeader to fill out and
return video frame properties.

- Rename some variables to be more self describing.

- No functional change intended.

Signed-off-by: Colin Günther <coling@gmx.de>
(cherry picked from commit 6beab04f48e98022041895750f38c5a35a5f4f9e)

ffmpeg: don't use deprecated API in AVCodecDecoder.

* Renamed fAudioTempPacket to fTempPacket and use it for calling
the new video decoding function. This just avoids a warning
generated from the libavcodec sources. The function used before
did the exact same thing...
* Maintain fStartTime correctly in _DecodeVideo(). Don't overwrite
it with a calculated starttime in Decode(). This will allow drift
to bubble up to the higher layers.
* Do not use the previously required hack to close and reopen the
AVCodec after seeking. avcodec_flush_buffers() seems to work
fine now, and for certain stream types (MPEG1, MPEG2 video for
example) the keyframe is correctly used after seeking.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38806 a95241bf-73f2-0310-859d-f6bbb57e9c96

Enabled any and all decoders and demuxers which are currently compiled into
FFmpeg. It's a bit sad, but this obsoletes pretty much all other decoder
and reader plugins. Some of them were built on external libraries as well
(AC3 (not part of default image anyway, since it's GPL), APE, MusePack),
so it's not really a big difference to using FFmpeg as external library.
The format matching is greatly simplified by using B_MISC_FORMAT_FAMILY
for everything but raw audio, and the actual FFmpeg CodecID as codec tag.
The downside of this is that the AVFormatReader can no longer be used with
other decoder plugins, but it would be easy to add special cases for native
decoders we wish to support. Obviously the out of the box support for file
formats and decoders has greatly increased with this change, so there has
to be a pretty good reason now for writing a "native" decoder or reader.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38786 a95241bf-73f2-0310-859d-f6bbb57e9c96

Finall changed the Seek() method of Decoder plug-ins to
SeekedTo(), since it's only informative to decoders. They
can't modify the seeked frame/time. This also mirrors what
all existing decoders were doing in Seek(). BMediaTrack
is simplified accordingly (resolved two TODOs).


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38705 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Use the new avcodec_decode_audio3() call, avcodec_decode_audio2()
gives the deprecated warning... We need to cache an AVPacket for this.
* Check the allocation of fOutputBuffer.
* When seeking, we need to flush the already decoded stuff
in fOutputBuffer, and throw away the last chunk buffer as well.
* Handle an incomplete input format at least to the point of not
crashing with a divide error (mp3_reader would give us such an
incomplete format for example).
* _DecodeAudio():
- Fixed some edge cases in the audio decoding loop: avcodec_decode_audio3()
can return a 0 length, which means no error, but no decoded frames
either. ffplay throws away the chunk in this case, do the same.
- Convert some invalid situations that were printf()s into debugger()s.
- Add much more comments to explain how everything works.
* Fixed the occasional coding style issue.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@38400 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Optionally use libswscale for color space conversion. Have not done any
performance tests, but I wrote this code before David got around to improve
his color space assembler code and didn't want to throw it away. Turned
off for the time being, since David's version works fine.
* Supply a codec sub_id in the media_codec_info. This allows matching
decoders to encoders, which comes in handy for Smart Rendering in Clockwerk.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@34931 a95241bf-73f2-0310-859d-f6bbb57e9c96

Sorry, this was supposed to be part of the last commit...


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@33356 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Automatic whitespace cleanup.
* media_decode_info member was never used.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@32015 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Coding style cleanup, some removal of dead code.
* Refactored NegotiateOutputFormat() and Decode() into two separate private
methods each, one for video and one for audio.
* Keep reading chunks when video decoding, until we have got a picture. This
gets us scrambled video instead of a black picture for h264 in mpegts.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@31457 a95241bf-73f2-0310-859d-f6bbb57e9c96

* Renamed avcodec folder to ffmpeg.
* Cleaned up plugins Jamfile.


git-svn-id: file:///srv/svn/repos/haiku/haiku/trunk@31359 a95241bf-73f2-0310-859d-f6bbb57e9c96