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[Gnash-dev] Sortie's weekly GSoC activity log

From: Jonas 'Sortie' Termansen
Subject: [Gnash-dev] Sortie's weekly GSoC activity log
Date: Fri, 22 Jul 2011 21:24:38 +0200
User-agent: Mozilla/5.0 (X11; U; Linux x86_64; en-US; rv: Gecko/20110617 Thunderbird/3.1.11

Hi, I'm Jonas 'Sortie' Termansen, and I'm the Google Summer of Code student working on Gnash's media playback.

I decided to post weekly logs of my activities here on the mailing list, so you fine folks can see what I am doing and comment if you have anything to say.

This is just a bullet-point list of my activities, my thoughts, and considerations.

* Decided to make testcases for libmedia's various handlers, codecs and considered various methods of doing so. * Studied a bit how digital sound works and wrote a small program 'mksound' that generates wav/raw PCM files with a specific tone. <> * Investigated whether any existing testcases could be easily adapted. Doesn't look like it.
* Submitted bug <>
* Studying in what manner FFmpeg and gst post-processes its output and how it would affect any testcases.
* Decided to test performance as well when writing libmedia testcases.
* Read a lot of documentation on how libmedia integrates with the main player. * Wondered whether it is possible/useful to compile both GST and FFmpeg into gnash, and then search for a decoder in both backends.
* That would make it easier for make check to check all possible backends.
* Figuring out how GnashFactory works.
* Fixed bug where gnash/ wasn't in EXTRA_DIST. (commit 3bf4171)
* Learned what libtool is, what it is good for, and how it actually works.
* Began writing the libmedia.all/test_decoding.cpp testcase for audio decoding. * Made makeFileChannel also accept a filepath string and not just a C FILE* stream. <>
* Created a mini media player for libmedia.all/test_decoding.cpp.
* My testcase only tests sound files at this point. Video testing will be added soon. * Had some nasty MediaParser race conditions when implementing the testcase that made it difficult to see when MediaParser::getAudioInfo() is available, and how should the code wait for the next frame? * It could possibly be beneficial for AudioDecoder to contain a string about which codec it is using, making it easier to print debug output in my testcase. It could also be added to AudioInfo. * FFmpeg fails to decode .wav, .ogg, .mkv, .webm, .flac sound files (could not initialize parser for ffmpeg codec). * The parameter 'parse' in "boost::uint8_t* AudioDecoder::decode(const boost::uint8_t* input, boost::uint32_t inputSize, boost::uint32_t& outputSize, boost::uint32_t& decodedData, bool parse);" seems to be redundant as it always is true, except in very rare cases (decoding RAW sound), but then the audiodecoder is able to figure it out anyways. * The FFmpeg parse system is causing trouble when decoding many of my non-flv testfiles. I'll have to refactor it. Getting rid of the needless 'parse' parameter would simplify matters. * Investigating whether (and how) the parse parameter can be safely removed when dealing with other backends (gst, haiku, simple, speex). * AudioDecoder{Simple,Speex,Gst,Haiku} doesn't use the 'parse' parameter at all.
* Haiku media backend is a stub. No need to worry much about it.
* Removed the 'parse' param from all backends.
* Changed parsing logic in AudioDecoderFfmpeg such that the presence of a parser for a given format is detected automatically. * The above change now fixes vorbis/.ogg and .wav playback in my testcase for FFmpeg. * The .webm, .mkv, and .flac containers are still broken for FFmpeg, they get a premature EOF and halt. * Does Gnash have a built-in way to scan a directory for files? I'll just write a quick-and-dirty method myself that is portable (tested with Wine).
* My testcase now automatically scans a sample directory for files to test.
* I wrote a script using mksound(see above), ffmpeg(1), and lame(1) to generate test files for my test case. * Gstreamer appears to have some EOF halting trouble on .mp4 and .ogg files during my testcase (.flv files works just fine). * A similar bug was described on the IRC; it is probably the same bug. I'll investigate.
* Gstreamer appears to output a kilobytes more data than FFmpeg.
* Gstreamer appears to freeze on my test .wav file (created by mksound) (probably a bug in my code). * AudioDecoder::decode() allocates 4096 bytes on every call that is soon freed. While malloc shouldn't take long to handle that, it would probably be better to switch to memory-pooled buffers. (Oh hey, wasn't that officially my GSoC project?)

Alright, thanks. That was this week's achievements. I'll be back next week.


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