Add avcodec_decode_audio4().

Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.

1 files changed, 20 insertions, 13 deletions

diff --git a/libavcodec/vorbisdec.c b/libavcodec/vorbisdec.c
index b202249e9b..381b61d060 100644
--- a/libavcodec/vorbisdec.c
+++ b/libavcodec/vorbisdec.c
@@ -121,6 +121,7 @@ typedef struct {
 
 typedef struct vorbis_context_s {
     AVCodecContext *avccontext;
+    AVFrame frame;
     GetBitContext gb;
     DSPContext dsp;
     FmtConvertContext fmt_conv;
@@ -1033,6 +1034,9 @@ static av_cold int vorbis_decode_init(AVCodecContext *avccontext)
     avccontext->sample_rate = vc->audio_samplerate;
     avccontext->frame_size  = FFMIN(vc->blocksize[0], vc->blocksize[1]) >> 2;
 
+    avcodec_get_frame_defaults(&vc->frame);
+    avccontext->coded_frame = &vc->frame;
+
     return 0;
 }
 
@@ -1605,16 +1609,15 @@ static int vorbis_parse_audio_packet(vorbis_context *vc)
 
 // Return the decoded audio packet through the standard api
 
-static int vorbis_decode_frame(AVCodecContext *avccontext,
-                               void *data, int *data_size,
-                               AVPacket *avpkt)
+static int vorbis_decode_frame(AVCodecContext *avccontext, void *data,
+                               int *got_frame_ptr, AVPacket *avpkt)
 {
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     vorbis_context *vc = avccontext->priv_data;
     GetBitContext *gb = &(vc->gb);
     const float *channel_ptrs[255];
-    int i, len, out_size;
+    int i, len, ret;
 
     av_dlog(NULL, "packet length %d \n", buf_size);
 
@@ -1625,18 +1628,18 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
 
     if (!vc->first_frame) {
         vc->first_frame = 1;
-        *data_size = 0;
+        *got_frame_ptr = 0;
         return buf_size;
     }
 
     av_dlog(NULL, "parsed %d bytes %d bits, returned %d samples (*ch*bits) \n",
             get_bits_count(gb) / 8, get_bits_count(gb) % 8, len);
 
-    out_size = len * vc->audio_channels *
-               av_get_bytes_per_sample(avccontext->sample_fmt);
-    if (*data_size < out_size) {
-        av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
-        return AVERROR(EINVAL);
+    /* get output buffer */
+    vc->frame.nb_samples = len;
+    if ((ret = avccontext->get_buffer(avccontext, &vc->frame)) < 0) {
+        av_log(avccontext, AV_LOG_ERROR, "get_buffer() failed\n");
+        return ret;
     }
 
     if (vc->audio_channels > 8) {
@@ -1649,12 +1652,15 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
     }
 
     if (avccontext->sample_fmt == AV_SAMPLE_FMT_FLT)
-        vc->fmt_conv.float_interleave(data, channel_ptrs, len, vc->audio_channels);
+        vc->fmt_conv.float_interleave((float *)vc->frame.data[0], channel_ptrs,
+                                      len, vc->audio_channels);
     else
-        vc->fmt_conv.float_to_int16_interleave(data, channel_ptrs, len,
+        vc->fmt_conv.float_to_int16_interleave((int16_t *)vc->frame.data[0],
+                                               channel_ptrs, len,
                                                vc->audio_channels);
 
-    *data_size = out_size;
+    *got_frame_ptr   = 1;
+    *(AVFrame *)data = vc->frame;
 
     return buf_size;
 }
@@ -1678,6 +1684,7 @@ AVCodec ff_vorbis_decoder = {
     .init           = vorbis_decode_init,
     .close          = vorbis_decode_close,
     .decode         = vorbis_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
     .long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
     .channel_layouts = ff_vorbis_channel_layouts,
     .sample_fmts = (const enum AVSampleFormat[]) {