[FFmpeg-cvslog] lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft
Anton Khirnov
git at videolan.org
Tue Mar 12 22:16:03 CET 2013
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Sat Feb 23 08:20:12 2013 +0100| [5d606863c353553488e88dfd89ea5571268d9dca] | committer: Anton Khirnov
lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=5d606863c353553488e88dfd89ea5571268d9dca
---
libavcodec/avcodec.h | 65 --------------------------------------------------
libavcodec/utils.c | 48 -------------------------------------
libavcodec/version.h | 3 ---
3 files changed, 116 deletions(-)
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index b3bdb61..7bc7b61 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -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.
*
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 2591428..ed39c0c 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -1523,54 +1523,6 @@ int attribute_align_arg avcodec_decode_video2(AVCodecContext *avctx, AVFrame *pi
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 = { { 0 } };
- int ret, got_frame = 0;
-
- if (avctx->get_buffer != avcodec_default_get_buffer) {
- av_log(avctx, AV_LOG_ERROR, "Custom get_buffer() for use with"
- "avcodec_decode_audio3() detected. Overriding with avcodec_default_get_buffer\n");
- av_log(avctx, AV_LOG_ERROR, "Please port your application to "
- "avcodec_decode_audio4()\n");
- avctx->get_buffer = avcodec_default_get_buffer;
- }
-
- 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);
- 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;
- }
- return ret;
-}
-
-#endif
-
int attribute_align_arg avcodec_decode_audio4(AVCodecContext *avctx,
AVFrame *frame,
int *got_frame_ptr,
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 76d686d..f1cc749 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -49,9 +49,6 @@
#ifndef FF_API_REQUEST_CHANNELS
#define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 56)
#endif
-#ifndef FF_API_OLD_DECODE_AUDIO
-#define FF_API_OLD_DECODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
-#endif
#ifndef FF_API_OLD_ENCODE_AUDIO
#define FF_API_OLD_ENCODE_AUDIO (LIBAVCODEC_VERSION_MAJOR < 55)
#endif
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