| Age | Commit message (Collapse) | Author | |
|---|---|---|---|
| 2011-12-27 | Fixes nasty mpeg2 on ts A/V lag when using ff. | "Torsten Jager" | |
| --HG-- branch : point-release extra : rebase_source : 6e059c732a63d40b65b09f4ef725ec5ca45c4c1c | |||
| 2011-12-16 | flac decoder: int -> size_t | Petri Hintukainen | |
| --HG-- branch : point-release | |||
| 2011-12-16 | Fixed flac_read_callback() signature for flac 1.1.3+ (when sizeof(size_t) != ↵ | Petri Hintukainen | |
| sizeof(unsigned)) --HG-- branch : point-release | |||
| 2011-11-23 | Fix libavcodec version checks for AAC LATM/MPEG parser code | Brad Smith | |
| --HG-- branch : point-release | |||
| 2011-11-18 | Fixed building with internal ffmpeg | Petri Hintukainen | |
| 2011-11-02 | ff_audio_decoder: Use parser for AAC LATM and MPEG. Fixes streams with ↵ | Petri Hintukainen | |
| multiple audio packets in single PES packet and audio packets splitted to multiple PES packets. | |||
| 2011-11-02 | ff_audio_decoder: do not queue any data if opening audio output fails | Petri Hintukainen | |
| 2011-11-02 | ff_audio_decoder: If codec parameters can't be read from first audio packet, ↵ | Petri Hintukainen | |
| try next. Fixes detecting parameters when there are multiple audio packets in single PES packet. | |||
| 2011-11-02 | ff_audio_decoder: open audio out after decoding the data | Petri Hintukainen | |
| - Fixes detecting audio parameters when audio packet is splitted to multiple buffers - Simplifies the code (decode function is called only once for each audio frame) | |||
| 2011-10-27 | ff_audio_decoder: splitted calling avcodec_decode_audio* to separate function | Petri Hintukainen | |
| 2011-09-27 | Fixed mpeg2 decoding with ffmpeg. | Petri Hintukainen | |
| Codec was never opened when using mpeg12 mode. | |||
| 2011-09-27 | ff_audio_open_codec(): initialize codec if it hasn't been initialized. | Petri Hintukainen | |
| This makes HEADERS optional for codecs that don't require extradata. imported patch 10124.diff | |||
| 2011-09-27 | Splitted ff_audio_open_codec() from ff_audio_decode_data() | Petri Hintukainen | |
| imported patch 10123.diff | |||
| 2011-09-27 | Splitted ff_audio_init_codec() from ff_audio_handle_header_buffer() | Petri Hintukainen | |
| imported patch 10121.diff | |||
| 2011-09-27 | Splitted ff_audio_handle_header_buffer() from ff_audio_decode_data() | Petri Hintukainen | |
| imported patch 10120.diff | |||
| 2011-09-27 | ffmpeg audio: make sure decode_buffer is allocated only once | Petri Hintukainen | |
| imported patch 10119.diff | |||
| 2011-09-15 | ffmpeg audio: removed checks that are always true (context is allocated in init) | Petri Hintukainen | |
| imported patch 10116.diff | |||
| 2011-09-15 | ffmpeg audio: make sure context is allocated only once | Petri Hintukainen | |
| imported patch 10115.diff | |||
| 2011-09-16 | Fixed "warning: cast from pointer to integer of different size" | Petri Hintukainen | |
| 2011-09-16 | Simplify: check for BUF_FLAG_SPECIAL only once. Splitted special buffer ↵ | Petri Hintukainen | |
| handling to separate function. | |||
| 2011-09-16 | ffmpeg video: do not require preview buffers for mpeg1/2 | Petri Hintukainen | |
| 2011-09-16 | Splitted ff_init_mpeg12_mode() from ff_handle_preview_buffer() | Petri Hintukainen | |
| 2011-09-10 | Not every audio packet can be used to determine the sample rate and number of | Chris Rankin | |
| audio channels. So we must keep discarding packets that cannot be used to initialise the codec until we receive one that can be. | |||
| 2011-09-10 | Pad end of audio data buffer with zeros, as instructed by the API documentation. | Chris Rankin | |
| 2011-09-10 | Use xine's fast memcpy function instead of standard library one. | Chris Rankin | |
| 2011-09-10 | Optimise flags usage. | Chris Rankin | |
| 2011-09-10 | Tidy up flags usage. | Chris Rankin | |
| 2011-08-29 | Add AAC LATM support from FFmpeg 0.7+ | Chris Rankin | |
| I've now tested this patch on Fedora 15 (FFmpeg 0.7) and Fedora 14 (FFmpeg 0.6), and am happy to report that it works fine on F15 and doesn't break xine-lib on F14. On F14, it also has the happy side effect of no longer trying to decode an LATM AAC stream with the xineplug_decode_faad.so plugin. (Which was something which never ended well anyway.) | |||
| 2011-08-09 | Cosmetics: reordered functions | Petri Hintukainen | |
| 2011-08-09 | ffmpeg VC-1: scan for extradata (sequence header) from preview buffers | Petri Hintukainen | |
| 2011-08-14 | Fixed multithreaded decoding with lavc >= 52.112.0. | Petri Hintukainen | |
| avcodec_thread_init() was deprecated in lavc 52.112.0 (2011-02-09) | |||
| 2011-08-13 | rv30 & rv40 support | Torsten Jager | |
| 2011-08-13 | VP8 support | Torsten Jager | |
| 2011-08-13 | ffmpeg audio crash fix (sse2 alignment) | Torsten Jager | |
| Certain ffmpeg audio decoders use 32 bit float samples internally (wma, eac3, ...). They are then exported to the calling application as 16 bit integer. That conversion is done by faster sse2 code if your processor supports it. However, sse2 instructions require data buffers to be 16 byte aligned, or hit a segfault otherwise. Plain malloc() / realloc() ensures only 8 byte alignment, giving a 50% chance of a crash. FFmpeg internally uses aligned buffers a lot. It seems to be a good idea to do likewise for input buffers as well, even if current version does not strictly need it yet. Libavutil/av_realloc() has a bug that can break the alignment when enlarging an existing buffer. Thus I included a fixed version of it within ff_audio_decoder.c. | |||
| 2011-05-17 | Fix build with very recent copies of FFmpeg | Brad Smith | |
| This is a backport of the 1.2 code that was commited to utilize the new API provided by FFmpeg for awhile now but this is especially important because the old API has been eliminated all together from said copies of FFmpeg. | |||
| 2010-03-23 | Added new audio buffer type (EAC3) | Petri Hintukainen | |
| 2010-03-23 | Fixed using uninitialized decode_buffer_size | Petri Hintukainen | |
| 2010-03-10 | Handle odd widths properly (for ffmpeg-decoded video). | Darren Salt | |
| 2010-02-25 | Fix build with the old, outdated and deprecated internal ffmpeg. | Darren Salt | |
| 2010-02-21 | WMAPro support | Christopher Martin | |
| Rename "wmav3" to "wmapro" in xine-lib's internals to line up xine-lib's nomenclature with what everyone else calls it and knows it as. [Tweaked by ds to avoid API change.] Tell xine-lib that when it finds wmapro, look to ffmpeg. ffmpeg's wmapro decoder is unique in that it puts out samples that are floats, not 16-bit ints. These need to be converted. This requires external ffmpeg. | |||
| 2010-02-04 | Work around an ffmpeg SVQ3 bug; check for avcodec_thread_init failure. | Darren Salt | |
| 2010-02-03 | Kill a "missing return" warning. | Darren Salt | |
| 2010-01-21 | Undo libavutil workaround (fixed upstream). | Darren Salt | |
| 2010-01-17 | Cope with libavutil no longer defining some endian-specific macros. | Darren Salt | |
| 2010-01-17 | "Fix" playback of 24-bit FLAC. | Darren Salt | |
| We pretend that it's 16-bit to avoid "audio device unavailable" (ALSA). Also, the clock is a bit fast. | |||
| 2009-12-05 | Build fix (undefined symbol) for when using older ffmpeg. | Darren Salt | |
| 2009-12-04 | Bump the FLAC decoder's priority above ffmpegaudio, and build it by default. | Darren Salt | |
| 2009-11-30 | Trim trailing space & reduce space+tab. | Darren Salt | |
| 2009-11-17 | VC1 support fixes | Petri Hintukainen | |
| There are two tricks to make VC1 decoding work: 1) VC1 sequence and entry point headers must be present in context->extradata. 2) video width and height must be known when opening decoder. Some container formats store required extra data, but mpeg-ts does not. 1) is fixed by scanning the stream for headers and discarding all data until proper headers are found. 2) is fixed by re-opening decoder with width and height information from first open. | |||
| 2009-10-15 | Quick hack to prevent segfaulting at end-of-stream when this->context == NULL. | Darren Salt | |
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