ffmpeg rtp接收pcm音频流代码

以下是使用FFmpeg接收RTP传输的PCM音频流的示例代码：

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <netinet/in.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>

#define AUDIO_BUFFER_SIZE 1024

int main(int argc, char *argv[]) {
    if (argc < 2) {
        printf("Usage: %s [RTP URL]\n", argv[0]);
        return 1;
    }
    
    av_register_all();
    avcodec_register_all();
    
    AVFormatContext *formatCtx = NULL;
    AVCodecContext *codecCtx = NULL;
    AVCodec *codec = NULL;
    AVPacket packet;
    AVFrame *frame = NULL;
    int streamIndex = -1;
    uint8_t inBuffer[AUDIO_BUFFER_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
    int inBufferLen;
    uint8_t *data = NULL;
    int dataSize = 0;
    int ret;
    
    if ((ret = avformat_open_input(&formatCtx, argv[1], NULL, NULL)) < 0) {
        printf("Failed to open input: %s\n", av_err2str(ret));
        return 1;
    }
    
    if ((ret = avformat_find_stream_info(formatCtx, NULL)) < 0) {
        printf("Failed to find stream info: %s\n", av_err2str(ret));
        return 1;
    }
    
    for (int i = 0; i < formatCtx->nb_streams; i++) {
        if (formatCtx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
            streamIndex = i;
            break;
        }
    }
    
    if (streamIndex == -1) {
        printf("Failed to find audio stream\n");
        return 1;
    }
    
    codec = avcodec_find_decoder(formatCtx->streams[streamIndex]->codecpar->codec_id);
    if (codec == NULL) {
        printf("Failed to find codec\n");
        return 1;
    }
    
    codecCtx = avcodec_alloc_context3(codec);
    if (codecCtx == NULL) {
        printf("Failed to allocate codec context\n");
        return 1;
    }
    
    if ((ret = avcodec_parameters_to_context(codecCtx, formatCtx->streams[streamIndex]->codecpar)) < 0) {
        printf("Failed to copy codec parameters to context: %s\n", av_err2str(ret));
        return 1;
    }
    
    if ((ret = avcodec_open2(codecCtx, codec, NULL)) < 0) {
        printf("Failed to open codec: %s\n", av_err2str(ret));
        return 1;
    }
    
    frame = av_frame_alloc();
    if (frame == NULL) {
        printf("Failed to allocate frame\n");
        return 1;
    }
    
    av_init_packet(&packet);
    
    AVIOContext *avioCtx = NULL;
    struct sockaddr_in servAddr;
    int sock = socket(AF_INET, SOCK_DGRAM, 0);
    if (sock < 0) {
        printf("Failed to create socket\n");
        return 1;
    }
    
    memset(&servAddr, 0, sizeof(servAddr));
    servAddr.sin_family = AF_INET;
    servAddr.sin_addr.s_addr = INADDR_ANY;
    servAddr.sin_port = htons(1234); // replace with your RTP port
    
    if (bind(sock, (struct sockaddr *)&servAddr, sizeof(servAddr)) < 0) {
        printf("Failed to bind socket\n");
        return 1;
    }
    
    avioCtx = avio_alloc_context(inBuffer, sizeof(inBuffer), 0, sock, NULL, NULL, NULL);
    avioCtx->seekable = 0;
    formatCtx->pb = avioCtx;
    
    while (av_read_frame(formatCtx, &packet) >= 0) {
        if (packet.stream_index == streamIndex) {
            ret = avcodec_send_packet(codecCtx, &packet);
            if (ret < 0) {
                printf("Failed to send packet to codec: %s\n", av_err2str(ret));
                break;
            }
            
            while (ret >= 0) {
                ret = avcodec_receive_frame(codecCtx, frame);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                    break;
                } else if (ret < 0) {
                    printf("Failed to receive frame from codec: %s\n", av_err2str(ret));
                    goto end;
                }
                
                dataSize = av_samples_get_buffer_size(NULL, codecCtx->channels, frame->nb_samples, codecCtx->sample_fmt, 1);
                data = frame->data[0];
                
                // TODO: Process audio data here
                
                av_frame_unref(frame);
            }
        }
        av_packet_unref(&packet);
    }
    
end:
    avcodec_free_context(&codecCtx);
    avformat_close_input(&formatCtx);
    av_frame_free(&frame);
    avio_context_free(&avioCtx);
    close(sock);
    
    return 0;
}

这是一个简单的示例，它假定您已经了解如何使用FFmpeg解码音频流并处理音频数据。您需要将代码中的RTP端口号替换为您要使用的端口号，并且需要编写自己的音频处理代码，以便将音频数据保存到文件或以其他方式处理它。