rtmp: Factorize the code by adding handle_invoke_status

[ffmpeg] / libavcodec / g726.c

diff --git a/libavcodec/g726.c b/libavcodec/g726.c
index 85711f854c9d2b302cd2400cf995bf2cc48dcd9a..34aa9c84683685a8d6734b3cbcd5ae686bfb55a8 100644 (file)
--- a/libavcodec/g726.c
+++ b/libavcodec/g726.c
@@ -330,10 +330,12 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
 
     g726_reset(c);
 
+#if FF_API_OLD_ENCODE_AUDIO
     avctx->coded_frame = avcodec_alloc_frame();
     if (!avctx->coded_frame)
         return AVERROR(ENOMEM);
     avctx->coded_frame->key_frame = 1;
+#endif
 
     /* select a frame size that will end on a byte boundary and have a size of
        approximately 1024 bytes */
@@ -342,28 +344,37 @@ static av_cold int g726_encode_init(AVCodecContext *avctx)
     return 0;
 }
 
+#if FF_API_OLD_ENCODE_AUDIO
 static av_cold int g726_encode_close(AVCodecContext *avctx)
 {
     av_freep(&avctx->coded_frame);
     return 0;
 }
+#endif
 
-static int g726_encode_frame(AVCodecContext *avctx,
-                            uint8_t *dst, int buf_size, void *data)
+static int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                             const AVFrame *frame, int *got_packet_ptr)
 {
     G726Context *c = avctx->priv_data;
-    const int16_t *samples = data;
+    const int16_t *samples = (const int16_t *)frame->data[0];
     PutBitContext pb;
-    int i;
+    int i, ret, out_size;
 
-    init_put_bits(&pb, dst, 1024*1024);
+    out_size = (frame->nb_samples * c->code_size + 7) / 8;
+    if ((ret = ff_alloc_packet(avpkt, out_size))) {
+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
+        return ret;
+    }
+    init_put_bits(&pb, avpkt->data, avpkt->size);
 
-    for (i = 0; i < avctx->frame_size; i++)
+    for (i = 0; i < frame->nb_samples; i++)
         put_bits(&pb, c->code_size, g726_encode(c, *samples++));
 
     flush_put_bits(&pb);
 
-    return put_bits_count(&pb)>>3;
+    avpkt->size = out_size;
+    *got_packet_ptr = 1;
+    return 0;
 }
 
 #define OFFSET(x) offsetof(G726Context, x)
@@ -388,14 +399,17 @@ static const AVCodecDefault defaults[] = {
 AVCodec ff_adpcm_g726_encoder = {
     .name           = "g726",
     .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_ADPCM_G726,
+    .id             = AV_CODEC_ID_ADPCM_G726,
     .priv_data_size = sizeof(G726Context),
     .init           = g726_encode_init,
-    .encode         = g726_encode_frame,
+    .encode2        = g726_encode_frame,
+#if FF_API_OLD_ENCODE_AUDIO
     .close          = g726_encode_close,
-    .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
-    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
-    .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+#endif
+    .capabilities   = CODEC_CAP_SMALL_LAST_FRAME,
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+                                                     AV_SAMPLE_FMT_NONE },
+    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
     .priv_class     = &class,
     .defaults       = defaults,
 };
@@ -477,12 +491,12 @@ static void g726_decode_flush(AVCodecContext *avctx)
 AVCodec ff_adpcm_g726_decoder = {
     .name           = "g726",
     .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = CODEC_ID_ADPCM_G726,
+    .id             = AV_CODEC_ID_ADPCM_G726,
     .priv_data_size = sizeof(G726Context),
     .init           = g726_decode_init,
     .decode         = g726_decode_frame,
     .flush          = g726_decode_flush,
     .capabilities   = CODEC_CAP_DR1,
-    .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
+    .long_name      = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
 };
 #endif