avcodec: remove deprecated old audio decode API

3 files changed, 0 insertions, 108 deletions

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 22aeca361f..bc9db05823 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -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.
  *
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index cfc561346f..b6312b21dd 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -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,
diff --git a/libavcodec/version.h b/libavcodec/version.h
index b7b502a939..454ddcd30b 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -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