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ffmpeg:avcodec_decode_audio

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ffmpeg:avcodec_decode_audio [2007/12/30 14:50] – created devaffmpeg:avcodec_decode_audio [2007/12/30 14:51] (current) deva
Line 1: Line 1:
 +<code c>
 /** /**
  * @deprecated Use avcodec_decode_audio2() instead.  * @deprecated Use avcodec_decode_audio2() instead.
Line 5: Line 6:
                          int *frame_size_ptr,                          int *frame_size_ptr,
                          uint8_t *buf, int buf_size);                          uint8_t *buf, int buf_size);
- +</code>
-/** +
- * Decodes an audio frame from \p buf into \p samples. +
- * The avcodec_decode_audio2() function decodes an audio frame from the input +
- * buffer \p buf of size \p buf_size. To decode it, it makes use of the +
- * audio codec which was coupled with \p avctx using avcodec_open(). The +
- * resulting decoded frame is stored in output buffer \p samples.  If no frame +
- * could be decompressed, \p frame_size_ptr is zero. Otherwise, it is the +
- * decompressed frame size in \e bytes. +
- * +
- * @warning You \e must set \p frame_size_ptr to the allocated size of the +
- * output buffer before calling avcodec_decode_audio2(). +
- * +
- * @warning The input buffer must be \c FF_INPUT_BUFFER_PADDING_SIZE larger than +
- * the actual read bytes because some optimized bitstream readers read 32 or 64 +
- * bits at once and could read over the end. +
- * +
- * @warning The end of the input buffer \p buf should be set to 0 to ensure that +
- * no overreading happens for damaged MPEG streams. +
- * +
- * @note You might have to align the input buffer \p buf and output buffer \p +
- * samples. The alignment requirements depend on the CPU: On some CPUs it isn'+
- * necessary at all, on others it won't work at all if not aligned and on others +
- * it will work but it will have an impact on performance. In practice, the +
- * bitstream should have 4 byte alignment at minimum and all sample data should +
- * be 16 byte aligned unless the CPU doesn't need it (AltiVec and SSE do). If +
- * the linesize is not a multiple of 16 then there's no sense in aligning the +
- * start of the buffer to 16. +
- * +
- * @param avctx the codec context +
- * @param[out] samples the output buffer +
- * @param[in,out] frame_size_ptr the output buffer size in bytes +
- * @param[in] buf the input buffer +
- * @param[in] buf_size the input buffer size in bytes +
- * @return On error a negative value is returned, otherwise the number of bytes +
- * used or zero if no frame could be decompressed. +
- */ +
-int avcodec_decode_audio2(AVCodecContext *avctx, int16_t *samples, +
-                         int *frame_size_ptr, +
-                         uint8_t *buf, int buf_size);+
  
ffmpeg/avcodec_decode_audio.1199022624.txt.gz · Last modified: 2007/12/30 14:50 by deva