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我試圖用下面的代碼創建一個使用FFmpeg的RTP音頻流。當我的Windows 10的機器上運行,我得到如下回應:Stream#0:0:Unknown:none(pcm_s16be)
Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Audio: pcm_s16be, 8000 Hz, mono, s16, 128 kb/s
SDP dump:
=================
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 57.25.101
m=audio 8554 RTP/AVP 96
b=AS:128
a=rtpmap:96 L16/8000/1
ret = 0
,但在Linux上(#57〜14.04.1 Ubuntu的),FFmpeg的對待流爲 「未知」:
Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Unknown: none (pcm_s16be)
SDP dump:
=================
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 57.57.100
m=application 8554 RTP/AVP 3
有誰知道爲什麼會出現這種情況?任何形式的幫助將不勝感激。
#include <math.h>
extern "C"
{
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#include <libavformat/avformat.h>
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
int ret;
AVCodec *outCodec = NULL;
AVCodecContext *outCodecCtx = NULL;
AVFormatContext *outFormatCtx = NULL;
AVStream * outAudioStream = NULL;
AVFrame *outFrame = NULL;
ret = avformat_alloc_output_context2(&outFormatCtx, NULL, "rtp", filename);
if (!outFormatCtx || ret < 0)
{
fprintf(stderr, "Could not allocate output context");
}
outFormatCtx->flags |= AVFMT_FLAG_NOBUFFER | AVFMT_FLAG_FLUSH_PACKETS;
outFormatCtx->oformat->audio_codec = AV_CODEC_ID_PCM_S16BE;
// outFormatCtx->audio_codec_id = AV_CODEC_ID_PCM_S16BE;
// outFormatCtx->oformat->video_codec = AV_CODEC_ID_NONE;
// outFormatCtx->oformat->data_codec = AV_CODEC_ID_PCM_S16BE;
/* find the encoder */
outCodec = avcodec_find_encoder(outFormatCtx->oformat->audio_codec);
if (!outCodec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
outAudioStream = avformat_new_stream(outFormatCtx, outCodec);
if (!outAudioStream)
{
fprintf(stderr, "Cannot add new audio stream\n");
exit(1);
}
outAudioStream->time_base.den = 8000;
outAudioStream->time_base.num = 1;
outCodecCtx = outAudioStream->codec;
outCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
/* select other audio parameters supported by the encoder */
outCodecCtx->sample_rate = 8000;
outCodecCtx->channel_layout = AV_CH_LAYOUT_MONO;
outCodecCtx->channels = 1;
/* open it */
if (avcodec_open2(outCodecCtx, outCodec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
outCodecCtx->frame_size = 372;
av_dump_format(outFormatCtx, 0, filename, 1);
char buff[10000] = { 0 };
ret = av_sdp_create(&outFormatCtx, 1, buff, sizeof(buff));
printf("SDP dump:\n"
"=================\n"
"%s", buff);
/*
Running the program returns the following:
Output #0, rtp, to 'rtp://127.0.0.1:8554':
Stream #0:0: Unknown: none (pcm_s16be)
SDP dump:
=================
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 57.57.100
m=application 8554 RTP/AVP 3
*/
exit(1);
}
int main(int argc, char **argv)
{
const char *output;
av_register_all();
avformat_network_init(); // for network streaming
audio_encode_example("rtp://127.0.0.1:8554");
return 0;
}