用c写一个代码，使用alsa库采集音频，通过ortp库打包发送

好的，以下是一个使用alsa库采集音频并使用ortp库打包发送的示例代码，您可以参考：

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
#include <ortp/ortp.h>

// 定义音频参数
#define AUDIO_FORMAT SND_PCM_FORMAT_S16_LE
#define AUDIO_CHANNELS 1
#define AUDIO_RATE 44100
#define AUDIO_FRAME_SIZE (AUDIO_CHANNELS * snd_pcm_format_width(AUDIO_FORMAT) / 8)

int main(int argc, char *argv[]) {
    // 初始化ortp库
    ortp_init();

    // 创建rtp会话
    RtpSession *session = rtp_session_new(RTP_SESSION_SENDONLY);
    if (session == NULL) {
        fprintf(stderr, "Failed to create rtp session\n");
        return -1;
    }

    // 设置rtp会话参数
    rtp_session_set_scheduling_mode(session, RTP_SCHEDULER_TIME);
    rtp_session_set_blocking_mode(session, 0);
    rtp_session_set_payload_type(session, 0);
    rtp_session_set_tx_timestamp(session, 1);
    rtp_session_enable_adaptive_jitter_compensation(session, 1);

    // 解析命令行参数获取目标IP和端口号
    if (argc < 3) {
        fprintf(stderr, "Usage: %s <dest_ip> <dest_port>\n", argv[0]);
        return -1;
    }
    const char *dest_ip = argv[1];
    int dest_port = atoi(argv[2]);

    // 创建rtp传输地址
    RtpAddress *addr = rtp_address_new(dest_ip, dest_port, NULL);
    if (addr == NULL) {
        fprintf(stderr, "Failed to create rtp address\n");
        return -1;
    }

    // 打开默认alsa音频设备
    snd_pcm_t *handle;
    if (snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0) < 0) {
        fprintf(stderr, "Failed to open audio device\n");
        return -1;
    }

    // 配置alsa音频设备参数
    if (snd_pcm_set_params(handle,
                            AUDIO_FORMAT,
                            SND_PCM_ACCESS_RW_INTERLEAVED,
                            AUDIO_CHANNELS,
                            AUDIO_RATE,
                            1,
                            500000) < 0) {
        fprintf(stderr, "Failed to set audio device parameters\n");
        return -1;
    }

    // 循环读取音频数据并发送rtp包
    char audio_buf[AUDIO_FRAME_SIZE * 1024];
    int frame_count = 0;
    while (1) {
        // 从alsa设备读取音频数据
        int read_count = snd_pcm_readi(handle, audio_buf, 1024);
        if (read_count < 0) {
            read_count = snd_pcm_recover(handle, read_count, 0);
            if (read_count < 0) {
                fprintf(stderr, "Failed to read audio data\n");
                break;
            }
        }

        // 创建rtp包并设置其数据
        RtpPacket *packet = rtp_packet_new(AUDIO_FRAME_SIZE * read_count);
        packet->payloadtype = 0;
        memcpy(packet->payload, audio_buf, AUDIO_FRAME_SIZE * read_count);
        packet->timestamp = frame_count * AUDIO_FRAME_SIZE / AUDIO_CHANNELS;
        packet->marker = 0;

        // 将rtp包发送到目标地址
        if (rtp_session_sendto(session, packet->payload, packet->payload_len, addr) < 0) {
            fprintf(stderr, "Failed to send rtp packet\n");
            rtp_packet_free(packet);
            break;
        }

        // 释放rtp包资源
        rtp_packet_free(packet);

        // 增加音频帧计数
        frame_count += read_count;
    }

    // 关闭alsa设备
    snd_pcm_close(handle);

    // 释放rtp地址和会话资源
    rtp_session_destroy(session);
    rtp_address_destroy(addr);

    // 清理ortp库资源
    ortp_exit();

    return 0;
}

请注意，此示例代码仅用于演示目的，实际使用时需要根据您的具体需求进行修改和优化。