Merge remote-tracking branch 'qatar/master'
* qatar/master: (36 commits) adpcmenc: Use correct frame_size for Yamaha ADPCM. avcodec: add ff_samples_to_time_base() convenience function to internal.h adx parser: set duration mlp parser: set duration instead of frame_size gsm parser: set duration mpegaudio parser: set duration instead of frame_size (e)ac3 parser: set duration instead of frame_size flac parser: set duration instead of frame_size avcodec: add duration field to AVCodecParserContext avutil: add av_rescale_q_rnd() to allow different rounding pnmdec: remove useless .pix_fmts libmp3lame: support float and s32 sample formats libmp3lame: renaming, rearrangement, alignment, and comments libmp3lame: use the LAME default bit rate libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing libmp3lame: cosmetics: remove some pointless comments libmp3lame: convert some debugging code to av_dlog() libmp3lame: remove outdated comment. libmp3lame: do not set coded_frame->key_frame. libmp3lame: improve error handling in MP3lame_encode_init() ... Conflicts: doc/APIchanges libavcodec/libmp3lame.c libavcodec/pcxenc.c libavcodec/pnmdec.c libavcodec/pnmenc.c libavcodec/sgienc.c libavcodec/utils.c libavformat/hls.c libavutil/avutil.h libswscale/x86/swscale_mmx.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
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@@ -29,8 +29,8 @@
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "bytestream.h"
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#include "internal.h"
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#include "vorbis.h"
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#include "libavutil/mathematics.h"
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#undef NDEBUG
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#include <assert.h>
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@@ -268,7 +268,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
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op2->packet = context->buffer + sizeof(ogg_packet);
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l = op2->bytes;
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avccontext->coded_frame->pts = av_rescale_q(op2->granulepos, (AVRational) { 1, avccontext->sample_rate }, avccontext->time_base);
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avccontext->coded_frame->pts = ff_samples_to_time_base(avccontext,
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op2->granulepos);
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//FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
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if (l > buf_size) {
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