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lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft

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This commit is contained in:
Anton Khirnov
2013-02-23 08:20:12 +01:00
parent 6327c10702
commit 5d606863c3
3 changed files with 0 additions and 116 deletions
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65
libavcodec/avcodec.h
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@@ -487,11 +487,6 @@ typedef struct AVCodecDescriptor {
*/
#define AV_CODEC_PROP_LOSSLESS (1 << 2)
#if FF_API_OLD_DECODE_AUDIO
/* in bytes */
#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
#endif
/**
* @ingroup lavc_decoding
* Required number of additionally allocated bytes at the end of the input bitstream for decoding.
@@ -3360,66 +3355,6 @@ void avcodec_align_dimensions(AVCodecContext *s, int *width, int *height);
void avcodec_align_dimensions2(AVCodecContext *s, int *width, int *height,
int linesize_align[AV_NUM_DATA_POINTERS]);
#if FF_API_OLD_DECODE_AUDIO
/**
* Wrapper function which calls avcodec_decode_audio4.
*
* @deprecated Use avcodec_decode_audio4 instead.
*
* Decode the audio frame of size avpkt->size from avpkt->data into samples.
* Some decoders may support multiple frames in a single AVPacket, such
* decoders would then just decode the first frame. In this case,
* avcodec_decode_audio3 has to be called again with an AVPacket that contains
* the remaining data in order to decode the second frame etc.
* If no frame
* could be outputted, frame_size_ptr is zero. Otherwise, it is the
* decompressed frame size in bytes.
*
* @warning You must set frame_size_ptr to the allocated size of the
* output buffer before calling avcodec_decode_audio3().
*
* @warning The input buffer must be 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 avpkt->data should be set to 0 to ensure that
* no overreading happens for damaged MPEG streams.
*
* @warning You must not provide a custom get_buffer() when using
* avcodec_decode_audio3(). Doing so will override it with
* avcodec_default_get_buffer. Use avcodec_decode_audio4() instead,
* which does allow the application to provide a custom get_buffer().
*
* @note You might have to align the input buffer avpkt->data and output buffer
* samples. The alignment requirements depend on the CPU: On some CPUs it isn't
* 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, avpkt->data should have 4 byte alignment at minimum and
* samples should be 16 byte aligned unless the CPU doesn't need it
* (AltiVec and SSE do).
*
* @note Codecs which have the CODEC_CAP_DELAY capability set have a delay
* between input and output, these need to be fed with avpkt->data=NULL,
* avpkt->size=0 at the end to return the remaining frames.
*
* @param avctx the codec context
* @param[out] samples the output buffer, sample type in avctx->sample_fmt
* If the sample format is planar, each channel plane will
* be the same size, with no padding between channels.
* @param[in,out] frame_size_ptr the output buffer size in bytes
* @param[in] avpkt The input AVPacket containing the input buffer.
* You can create such packet with av_init_packet() and by then setting
* data and size, some decoders might in addition need other fields.
* All decoders are designed to use the least fields possible though.
* @return On error a negative value is returned, otherwise the number of bytes
* used or zero if no frame data was decompressed (used) from the input AVPacket.
*/
attribute_deprecated int avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
int *frame_size_ptr,
AVPacket *avpkt);
#endif
/**
* Decode the audio frame of size avpkt->size from avpkt->data into frame.
*