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Author SHA1 Message Date
Justin Ruggles c7f3f1c91e flac parser: set duration instead of frame_size 2012-02-20 15:08:40 -05:00
Justin Ruggles e9cda85351 avcodec: add duration field to AVCodecParserContext 
This will allow parsers to export the duration of the current frame being
output, if known, instead of using AVCodecContext.frame_size.
 2012-02-20 15:08:40 -05:00
Justin Ruggles 0b42a9388c avutil: add av_rescale_q_rnd() to allow different rounding 2012-02-20 15:08:40 -05:00
Justin Ruggles e00959a9b1 libmp3lame: support float and s32 sample formats 2012-02-20 12:32:31 -05:00
Justin Ruggles e232225276 libmp3lame: renaming, rearrangement, alignment, and comments 2012-02-20 12:32:30 -05:00
Justin Ruggles 232e16dd02 libmp3lame: use the LAME default bit rate 
Also, only set bit rate for CBR.
 2012-02-20 12:32:30 -05:00
Justin Ruggles 1f516c0451 libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing 2012-02-20 12:32:30 -05:00
Justin Ruggles e3d2c89e9d libmp3lame: cosmetics: remove some pointless comments 2012-02-20 12:32:30 -05:00
Justin Ruggles bf909fc456 libmp3lame: convert some debugging code to av_dlog() 
also remove unneeded commented-out full frame data debugging
 2012-02-20 12:32:30 -05:00
Justin Ruggles 469d2a8e8e libmp3lame: remove outdated comment. 
We now require at least libmp3lame 3.98.3.
lame_encode_buffer_interleaved() still doesn't work for mono, but it does not
"die"; it just expects a stereo interleaved buffer.
 2012-02-20 12:32:30 -05:00
Justin Ruggles 35cfd7d09c libmp3lame: do not set coded_frame->key_frame. 
it is already set in avcodec_alloc_frame()
 2012-02-20 12:32:30 -05:00
Justin Ruggles 8dad25ebf7 libmp3lame: improve error handling in MP3lame_encode_init() 2012-02-20 12:32:30 -05:00
Justin Ruggles 310c372e12 libmp3lame: remove unneeded 'stereo' field from Mp3AudioContext 2012-02-20 12:32:30 -05:00
Justin Ruggles f2a4559c77 ttadec: use branchless unsigned-to-signed unfolding 2012-02-17 14:35:35 -05:00
Justin Ruggles c9fdf3241a bethsoftvid: synchronize video timestamps with audio sample rate 
According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.
 2012-02-16 10:47:11 -05:00
Justin Ruggles 773ff823da bethsoftvid: add audio stream only after getting the first audio packet 
This avoids initializing a stream with dummy values or when the file does not
contain audio.
Also set duration for audio packets, using the sample rate as the time base.
 2012-02-16 10:47:11 -05:00
Justin Ruggles 9546f331c6 bethsoftvid: Set video packet duration instead of accumulating pts. 2012-02-16 10:47:11 -05:00
Justin Ruggles 05e4ae833c bethsoftvid: set packet key frame flag for audio and I-frame video packets. 
Fixes avconv video stream copy of bethsoft video, which was skipping all
video frames unless the copyinkf option was used.
 2012-02-16 10:47:11 -05:00
Justin Ruggles 17b115591f bethsoftvid: fix read_packet() return codes. 
Use proper AVERROR codes, and return 0 for no error.
 2012-02-16 10:47:11 -05:00
Justin Ruggles f320fb894c bethsoftvid: pass palette in side data instead of in a separate packet. 
Update FATE reference to account for now non-existent palette packet.
This also fixes the FATE test if frame data is not initialized in
get_buffer(), so update comment in avconv accordingly.
 2012-02-16 10:47:11 -05:00
Justin Ruggles b498867d66 FATE: update reference for seek-alac_mp4 
This should have been updated in b590f3a7bf.
 2012-02-11 16:41:01 -05:00
Justin Ruggles 764852d653 alacenc: use AVCodec.encode2() 2012-02-11 12:49:23 -05:00
Justin Ruggles bee80054f7 alacenc: cosmetics: indentation 2012-02-11 12:49:22 -05:00
Justin Ruggles b6e8ff72ea alacenc: consolidate bitstream writing into a single function. 
Simplifies use of verbatim mode.
 2012-02-11 12:49:22 -05:00
Justin Ruggles b590f3a7bf alacenc: only encode frame size in header for a final smaller frame 
Otherwise it is not needed because it matches the frame size as encoded in
the extradata.
 2012-02-11 12:49:22 -05:00
Justin Ruggles ba821b098b alacenc: store current frame size in AlacEncodeContext. 
This avoids an indirection and will simplify implementation of encode2()
 2012-02-11 12:49:22 -05:00
Justin Ruggles 65d15aec77 alacenc: return AVERROR codes in alac_encode_frame() 2012-02-11 12:49:22 -05:00
Justin Ruggles 302daf5800 alacenc: calculate a new max frame size for the final small frame 
Gives a better estimate of buffer requirements and a better decision of
whether or not to use verbatim mode.
 2012-02-11 12:49:22 -05:00
Justin Ruggles fc9cf0b2a6 alacenc: pretty-printing and other cosmetics 2012-02-11 12:49:22 -05:00
Justin Ruggles 51c2483862 alacenc: fix error handling and potential memleaks in alac_encode_init() 2012-02-11 12:49:22 -05:00
Justin Ruggles 6e63228323 alacenc: do not set coded_frame->key_frame 
It is already set in avcodec_alloc_frame()
 2012-02-11 12:49:21 -05:00
Justin Ruggles 64fe3eaeb3 alacenc: do not set bits_per_coded_sample 
encoded ALAC does not have a fixed number of bits per sample
 2012-02-11 12:49:21 -05:00
Justin Ruggles 43a4cb070b alacenc: remove unneeded frame_size check in alac_encode_frame() 2012-02-11 12:49:21 -05:00
Justin Ruggles 6ab681a4c1 ttadec: fix invalid free when an error occurs while decoding 24-bit tta 2012-02-10 20:35:10 -05:00
Justin Ruggles 9d7cee50aa wavpack: add needed braces for 2 statements inside an if block 2012-02-10 20:27:55 -05:00
Justin Ruggles b758cf7343 avcodec: set avpkt->size to 0 if encode2() did not output a packet 2012-02-09 21:51:41 -05:00
Justin Ruggles a75bc764ec avcodec: for audio encoding, set packet dts to packet pts. 
There are no audio encoders which do frame reordering.
 2012-02-09 21:51:29 -05:00
Justin Ruggles d483bb58c3 ac3dsp: do not use pshufb in ac3_extract_exponents_ssse3() 
We need to do unsigned saturation in order to cover the corner case when the
absolute coefficient value is 16777215 (the maximum value).

Fixes Bug #216
 2012-02-09 21:04:44 -05:00
Justin Ruggles 37390d5cca apedec: allow the user to set the maximum number of output samples per call 
It makes sense in some cases to split up the output packet to save on memory
usage (ape frames can be very large), but the current/default size is
arbitrary. Allowing the user to configure this gives more flexibility and
requires minimal additional code.
 2012-02-07 10:01:15 -05:00
Justin Ruggles 39575eead2 apedec: do not unnecessarily zero output samples for mono frames 2012-02-07 10:01:15 -05:00
Justin Ruggles 1d3c672d27 apedec: allocate a single flat buffer for decoded samples 
This will allow the decoder to return samples for the full packet, and it also
makes the decoded buffer pointers aligned.
 2012-02-07 10:01:14 -05:00
Justin Ruggles 32c61400c0 apedec: use sizeof(field) instead of sizeof(type) 2012-02-07 10:01:14 -05:00
Justin Ruggles ec1f3cab20 avplay: reset decoder flush state when seeking 
Fixes seeking after decoder has already been flushed for codecs using
CODEC_CAP_DELAY.
 2012-02-06 13:22:01 -05:00
Justin Ruggles f1c3d4a68a ape: skip packets with invalid size 2012-02-06 13:22:01 -05:00
Justin Ruggles ac3f8d317c ape: calculate final packet size instead of guessing 
Calculates based on total file size and wavetaillength from the header.
Falls back to multiplying finalframeblocks by 8 instead of 4 so that it will
at least be overestimating for 24-bit. Currently it can underestimate the
final packet size, leading to decoding errors.
 2012-02-06 13:22:01 -05:00
Justin Ruggles c2c316158f ape: stop reading after the last frame has been read 
This avoids buffer overread when the last packet size estimate is too small.
 2012-02-06 13:09:11 -05:00
Justin Ruggles 66f7be3603 ape: return AVERROR_EOF instead of AVERROR(EIO) when demuxing is finished 2012-02-06 13:09:11 -05:00
Justin Ruggles 1bc035bc03 ape: return error if seeking to the current packet fails in ape_read_packet() 2012-02-06 13:09:11 -05:00
Justin Ruggles 6f1a5e8d6b dsputil: remove debug message in dsputil_init(). 
AVCodecContext.bits_per_raw_sample is used for audio too, and values other
than 8, 9, and 10 are valid.
 2012-02-03 16:00:30 -05:00
Justin Ruggles 22c0babbb2 apedec: remove unneeded #include of get_bits.h and associated macro 2012-02-02 21:07:01 -05:00
Justin Ruggles e4169612a1 apedec: av_fast_malloc() instead of av_realloc() 
av_realloc() does not guarantee alignment, which is required for
DSPContext.bswap_buf().
 2012-02-02 21:07:01 -05:00
Justin Ruggles 0759c8eb10 apedec: fix handling of packet sizes that are not a multiple of 4 bytes 2012-02-02 21:07:01 -05:00
Justin Ruggles c3a06615bd bethsoftvideo: fix palette reading. 
Return the correct number of consumed bytes and set *data_size = 0.
Returned size is 1 too small, leading to that 1 byte being read as the next
frame, which results in an extra blank frame at the beginning of the stream.
 2012-02-01 19:41:39 -05:00
Justin Ruggles 83ce51cc7d mpc7: use av_fast_padded_malloc() 
Avoids doing malloc/free for each frame.
Also fixes valgrind errors due to use of uninitialized padding bytes.

Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>
 2012-02-01 19:35:59 -05:00
Justin Ruggles b5b825c381 mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes 2012-02-01 19:35:59 -05:00
Justin Ruggles 236a550c3f Fix a typo in the x86 asm version of ff_vector_clip_int32() 
Specifies the correct number of xmm registers used so that they can be saved
and restored on Win64 if necessary.
 2012-02-01 19:02:32 -05:00
Justin Ruggles 3c432631e9 pcmenc: Do not set avpkt->size. 
It is already the correct size as set by ff_alloc_packet().
 2012-02-01 16:33:27 -05:00
Justin Ruggles 1a670973a7 ff_alloc_packet: modify the size of the packet to match the requested size 
This will simplify encoders which use this function to request the exact
packet size rather than the maximum size.
 2012-02-01 16:33:27 -05:00
Justin Ruggles a3a0691bde adpcmenc: remove some unneeded casts 2012-01-30 19:12:55 -05:00
Justin Ruggles 149f2058a4 adpcmenc: use int16_t and uint8_t instead of short and unsigned char. 2012-01-30 19:12:55 -05:00
Justin Ruggles dd88ae831a adpcmenc: fix adpcm_ms extradata allocation 
Add FF_INPUT_BUFFER_PADDING_SIZE.
If allocation fails, also free memory which was allocated previously in
adpcm_encode_init().
 2012-01-30 19:12:55 -05:00
Justin Ruggles 877a1d409c adpcmenc: return proper AVERROR codes instead of -1 2012-01-30 19:12:55 -05:00
Justin Ruggles cb023d9afe adpcmenc: check for coded_frame allocation failure 2012-01-30 19:12:55 -05:00
Justin Ruggles ddf70db6d7 adpcmenc: Do not set coded_frame->key_frame. 
It is already set in avcodec_alloc_frame().
 2012-01-30 19:12:54 -05:00
Justin Ruggles 3c4add27f7 mpc7: check for allocation failure 2012-01-30 19:11:29 -05:00
Justin Ruggles eac31dd163 mpc7: align local temp buffer 
DSPContext.bswap_buf() requires aligned output
 2012-01-30 19:11:29 -05:00
Justin Ruggles 89eea6df28 aacenc: make sure to encode enough frames to cover all input samples. 
Currently, any samples in the final frame are not decoded because they are
only represented by one frame instead of two. So we encode two final frames to
cover both the analysis delay and the MDCT delay.
 2012-01-30 14:20:24 -05:00
Justin Ruggles f44005b610 aacenc: only use the number of input samples provided by the user. 
Fixes handling of CODEC_CAP_SMALL_LAST_FRAME.
 2012-01-30 14:20:24 -05:00
Justin Ruggles 2c577155a7 libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat 
Fixes build with --enable-libspeex
 2012-01-27 11:28:21 -05:00
Justin Ruggles b8dd555c63 aud: remove unneeded field, audio_stream_index from context 
There is only one stream, so it will always be index 0.
This also allows for removal of the private context.
 2012-01-24 14:13:47 -05:00
Justin Ruggles be14a37066 aud: fix time stamp calculation for ADPCM IMA WS 
also allows for removing some unused context fields
 2012-01-24 14:13:47 -05:00
Justin Ruggles 10fef6bd6b aud: simplify header parsing 
also allows for removing some unused context fields
 2012-01-24 14:13:47 -05:00
Justin Ruggles 0e6a8b5cf7 aud: set pts_wrap_bits to 64. 
Time stamps are not read from the bitstream, so no wrapping is needed.
 2012-01-24 14:13:47 -05:00
Justin Ruggles 98c290cc08 cosmetics: indentation 2012-01-24 14:13:46 -05:00
Justin Ruggles d481227c54 aud: support Westwood SND1 audio in AUD files. 
Fixes decoding of the handful of "death scream" samples from
"Command and Conquer" that use SND1. e.g.
http://samples.libav.org/game-formats/westwood-aud/nuyell1.aud
http://samples.libav.org/game-formats/westwood-aud/nuyell7.aud
 2012-01-24 14:13:46 -05:00
Justin Ruggles 02e7dbf5ad adpcm_ima_ws: fix stereo decoding 
Stereo ADPCM IMA WS is planar for VQA version 3 and 2-sample interleaved for
VQA version 2.
 2012-01-24 14:13:46 -05:00
Justin Ruggles 220506d23f avcodec: add a new codec_id for CRYO APC IMA ADPCM. 
The stereo layout and extradata is significantly different from that in
Westwood IMA ADPCM, so a separate codec_id is warranted.
 2012-01-24 14:13:41 -05:00
Justin Ruggles 4340a6363e vqa: remove unused context fields, audio_samplerate and audio_bits 2012-01-24 14:08:29 -05:00
Justin Ruggles 7e813d5933 vqa: clean up audio header parsing 
Fix setting of bit_rate and bits_per_coded_sample.
Ask for sample for unsupported VQA versions.
 2012-01-24 14:08:29 -05:00
Justin Ruggles 5b4c562897 vqa: set time base to frame rate as coded in the header. 
According to Multimedia Wiki documentation, the frame rate is not always 15,
and the actual value can be found in the header.
 2012-01-24 14:08:29 -05:00
Justin Ruggles 9ae846971f vqa: set packet duration. 
Also remove unused context field, audio_frame_counter.
 2012-01-24 14:08:29 -05:00
Justin Ruggles 53ed79a260 vqa: use 1/sample_rate as the audio stream time base 2012-01-24 14:08:29 -05:00
Justin Ruggles e1ac69fa27 vqa: set stream start_time to 0. 
The format has no coded timestamps and must use packet durations to generate
them.
 2012-01-24 14:08:29 -05:00
Justin Ruggles e924a4ba71 westwood: split the AUD and VQA demuxers into separate files. 
They have no code in common.
 2012-01-23 10:34:48 -05:00
Justin Ruggles 01ed1c390d tta: cast output data pointer to the correct type 
fixes "warning: assignment from incompatible pointer type"
 2012-01-17 09:41:13 -05:00
Justin Ruggles ee458cb1fa avconv: use avcodec_encode_audio2() 2012-01-15 21:24:18 -05:00
Justin Ruggles 05f95443ca pcmenc: use AVCodec.encode2() 2012-01-15 21:24:18 -05:00
Justin Ruggles a6ccae3f4c avcodec: bump minor version and add APIChanges for the new audio encoding API 2012-01-15 21:24:17 -05:00
Justin Ruggles b2c75b6e63 avcodec: Add avcodec_encode_audio2() as replacement for avcodec_encode_audio() 
This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.

This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.
 2012-01-15 21:24:17 -05:00
Justin Ruggles 5ee5fa021f avcodec: add a public function, avcodec_fill_audio_frame(). 
This is a convenience function for the user to fill audio AVFrame information.
 2012-01-15 21:24:17 -05:00
Justin Ruggles 4c82c6d993 FATE: convert output to rgba for the targa tests which currently output pal8 
fixes the tests on big-endian systems
 2012-01-13 18:40:12 -05:00
Justin Ruggles 9c2f9b0e2d FATE: enable the 2 remaining targa conformance suite tests 2012-01-13 17:13:12 -05:00
Justin Ruggles faaebcdf6b targa: add support for rgb555 palette 2012-01-13 17:11:41 -05:00
Justin Ruggles e1a7af6f29 FATE: fix targa tests on big-endian systems 2012-01-13 16:14:48 -05:00
Justin Ruggles 68d6012c72 FATE: add tests for targa 
Based on a patch by Oana Stratulat <oanaandreeastratulat@gmail.com>
 2012-01-13 15:45:30 -05:00
Justin Ruggles 56f22b7e16 pcmenc: set frame_size to 0. 
This indicates that the actual frame size is based on the buf_size passed to
avcodec_encode_audio().
 2012-01-11 09:09:56 -05:00
Justin Ruggles f01f6a8474 gsm demuxer: use generic seeking instead of a gsm-specific function. 
Since packets all contain only a single block, the generic seek function can
be used while still maintaining block-accuracy.
 2012-01-11 08:56:24 -05:00
Justin Ruggles 467879eb54 gsm demuxer: return packets with only 1 gsm block at a time. 
fixes stream copy of raw gsm to mov.
tested with QuickTime.
 2012-01-11 08:56:24 -05:00
Justin Ruggles 82390f57d1 avcodec: add GSM parser 
The WAVE demuxer returns packets with many blocks per frame, which needs to be
parsed into single blocks. This has a side-effect of fixing the timestamps.
 2012-01-11 08:56:24 -05:00
Justin Ruggles 4556ebfb7d ipmovie: fix pts for CODEC_ID_INTERPLAY_DPCM 
frame sample count calculation was incorrect
 2012-01-10 14:25:07 -05:00