virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2025-01-13 21:28:01 +02:00
Code Issues Releases Activity

avcodec: remove deprecated old audio decode API

Browse Source
This commit is contained in:
Hendrik Leppkes 2015-09-05 21:07:25 +02:00
parent c956cb2c02
commit 2c8ee2547e
3 changed files with 0 additions and 108 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

60
libavcodec/avcodec.h
View File

@ -4145,66 +4145,6 @@ int avcodec_enum_to_chroma_pos(int *xpos, int *ypos, enum AVChromaLocation pos);
*/
enum AVChromaLocation avcodec_chroma_pos_to_enum(int xpos, int ypos);
#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.
*

45
libavcodec/utils.c
View File

@ -2280,51 +2280,6 @@ fail:
return ret;
}
#if FF_API_OLD_DECODE_AUDIO
int attribute_align_arg avcodec_decode_audio3(AVCodecContext *avctx, int16_t *samples,
int *frame_size_ptr,
AVPacket *avpkt)
{
AVFrame *frame = av_frame_alloc();
int ret, got_frame = 0;
if (!frame)
return AVERROR(ENOMEM);
ret = avcodec_decode_audio4(avctx, frame, &got_frame, avpkt);
if (ret >= 0 && got_frame) {
int ch, plane_size;
int planar = av_sample_fmt_is_planar(avctx->sample_fmt);
int data_size = av_samples_get_buffer_size(&plane_size, avctx->channels,
frame->nb_samples,
avctx->sample_fmt, 1);
if (*frame_size_ptr < data_size) {
av_log(avctx, AV_LOG_ERROR, "output buffer size is too small for "
"the current frame (%d < %d)\n", *frame_size_ptr, data_size);
av_frame_free(&frame);
return AVERROR(EINVAL);
}
memcpy(samples, frame->extended_data[0], plane_size);
if (planar && avctx->channels > 1) {
uint8_t *out = ((uint8_t *)samples) + plane_size;
for (ch = 1; ch < avctx->channels; ch++) {
memcpy(out, frame->extended_data[ch], plane_size);
out += plane_size;
}
}
*frame_size_ptr = data_size;
} else {
*frame_size_ptr = 0;
}
av_frame_free(&frame);
return ret;
}
#endif
int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx,
AVFrame *frame,
int *got_frame_ptr,

3
libavcodec/version.h
View File

@ -55,9 +55,6 @@
#ifndef FF_API_VIMA_DECODER
#define FF_API_VIMA_DECODER (LIBAVCODEC_VERSION_MAJOR < 57)
#endif
#ifndef FF_API_OLD_DECODE_AUDIO
#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 57)
#endif
#ifndef FF_API_OLD_ENCODE_AUDIO
#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 57)
#endif