It will not be set if the stream codec context is not the encoding
context. Use av_get_audio_frame_duration() instead, it should work for
all audio codecs supported by the muxer.
---
libavformat/rmenc.c | 7 +++++--
1 file changed, 5 insertions(+), 2 deletions(-)
diff --git a/libavformat/rmenc.c b/libavformat/rmenc.c
index 9ff9f31..d218769 100644
--- a/libavformat/rmenc.c
+++ b/libavformat/rmenc.c
@@ -181,9 +181,10 @@ static int rv10_write_header(AVFormatContext *ctx,
if (stream->enc->codec_type == AVMEDIA_TYPE_AUDIO) {
int coded_frame_size, fscode, sample_rate;
+ int frame_size = av_get_audio_frame_duration(stream->enc, 0);
sample_rate = stream->enc->sample_rate;
coded_frame_size = (stream->enc->bit_rate *
- stream->enc->frame_size) / (8 * sample_rate);
+ frame_size) / (8 * sample_rate);
/* audio codec info */
avio_write(s, ".ra", 3);
avio_w8(s, 0xfd);
@@ -311,6 +312,7 @@ static int rm_write_header(AVFormatContext *s)
for(n=0;n<s->nb_streams;n++) {
AVStream *st = s->streams[n];
+ int frame_size;
s->streams[n]->id = n;
codec = s->streams[n]->codec;
@@ -323,7 +325,8 @@ static int rm_write_header(AVFormatContext *s)
switch(codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
rm->audio_stream = stream;
- stream->frame_rate = (float)codec->sample_rate /
(float)codec->frame_size;
+ frame_size = av_get_audio_frame_duration(codec, 0);
+ stream->frame_rate = (float)codec->sample_rate / (float)frame_size;
/* XXX: dummy values */
stream->packet_max_size = 1024;
stream->nb_packets = 0;
--
2.0.0
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