frsh: Export information about the last RTP contract and VRES

[frescor/ffmpeg.git] / libavformat / audiointerleave.c

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index fac68d0bf3711b48ee53fab2ac4c87abc48ce40f..a3a6531348c44a0b2639f5d085b5b39245c55914 100644 (file)
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -23,6 +23,7 @@
 #include "libavutil/fifo.h"
 #include "avformat.h"
 #include "audiointerleave.h"
+#include "internal.h"
 
 void ff_audio_interleave_close(AVFormatContext *s)
 {
@@ -32,7 +33,7 @@ void ff_audio_interleave_close(AVFormatContext *s)
         AudioInterleaveContext *aic = st->priv_data;
 
         if (st->codec->codec_type == CODEC_TYPE_AUDIO)
-            av_fifo_free(&aic->fifo);
+            av_fifo_free(aic->fifo);
     }
 }
 
@@ -60,7 +61,8 @@ int ff_audio_interleave_init(AVFormatContext *s,
             aic->samples = aic->samples_per_frame;
             aic->time_base = time_base;
 
-            av_fifo_init(&aic->fifo, 100 * *aic->samples);
+            aic->fifo_size = 100* *aic->samples;
+            aic->fifo= av_fifo_alloc(100 * *aic->samples);
         }
     }
 
@@ -73,12 +75,12 @@ static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
     AVStream *st = s->streams[stream_index];
     AudioInterleaveContext *aic = st->priv_data;
 
-    int size = FFMIN(av_fifo_size(&aic->fifo), *aic->samples * aic->sample_size);
-    if (!size || (!flush && size == av_fifo_size(&aic->fifo)))
+    int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
+    if (!size || (!flush && size == av_fifo_size(aic->fifo)))
         return 0;
 
     av_new_packet(pkt, size);
-    av_fifo_read(&aic->fifo, pkt->data, size);
+    av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
 
     pkt->dts = pkt->pts = aic->dts;
     pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
@@ -92,7 +94,7 @@ static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
     return size;
 }
 
-int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
+int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
                         int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
                         int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
 {
@@ -102,7 +104,13 @@ int ff_audio_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int fl
         AVStream *st = s->streams[pkt->stream_index];
         AudioInterleaveContext *aic = st->priv_data;
         if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
-            av_fifo_generic_write(&aic->fifo, pkt->data, pkt->size, NULL);
+            unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
+            if (new_size > aic->fifo_size) {
+                if (av_fifo_realloc2(aic->fifo, new_size) < 0)
+                    return -1;
+                aic->fifo_size = new_size;
+            }
+            av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
         } else {
             // rewrite pts and dts to be decoded time line position
             pkt->pts = pkt->dts = aic->dts;