if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE))

这段代码判断输出格式是否需要打开文件，如果需要则执行if语句中的代码，否则跳过if语句。具体来说，AVFMT_NOFILE是一个标志位，表示该输出格式不需要打开文件。因此，如果该标志位没有被设置，则需要打开文件，否则不需要。ofmt_ctx是一个输出格式上下文结构体，包含了输出格式相关的信息，包括是否需要打开文件。

相关问题

Nal 数据通过av_interleaved_write_frame 输出ts格式

将 NAL 数据输出到 TS 格式文件中，可以按照以下步骤进行：

1. 初始化 AVFormatContext 和 AVOutputFormat，设置输出文件名和格式参数。

    AVFormatContext *ofmt_ctx = NULL;
    avformat_alloc_output_context2(&ofmt_ctx, NULL, "mpegts", out_filename);
    AVOutputFormat *ofmt = ofmt_ctx->oformat;

2. 打开输出文件。

    if (!(ofmt->flags & AVFMT_NOFILE)) {
        if (avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE) < 0) {
            printf("Could not open output file '%s'", out_filename);
            return -1;
        }
    }

3. 添加视频流到输出文件中，设置编码参数，打开编码器。

    AVStream *out_stream = avformat_new_stream(ofmt_ctx, NULL);
    AVCodecParameters *codecpar = out_stream->codecpar;
    codecpar->codec_id = AV_CODEC_ID_H264;
    codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
    codecpar->format = AV_PIX_FMT_YUV420P;
    codecpar->width = width;
    codecpar->height = height;
    codecpar->extradata_size = sps_len + pps_len;
    codecpar->extradata = av_mallocz(sps_len + pps_len + AV_INPUT_BUFFER_PADDING_SIZE);
    memcpy(codecpar->extradata, sps_pps_data, sps_len + pps_len);
    out_stream->time_base = (AVRational){1, fps};
    AVCodec *codec = avcodec_find_encoder(codecpar->codec_id);
    AVCodecContext *codec_ctx = avcodec_alloc_context3(codec);
    avcodec_parameters_to_context(codec_ctx, codecpar);
    codec_ctx->time_base = out_stream->time_base;
    codec_ctx->gop_size = 10;
    codec_ctx->max_b_frames = 1;
    codec_ctx->bit_rate = bitrate;
    codec_ctx->rc_min_rate = bitrate;
    codec_ctx->rc_max_rate = bitrate;
    codec_ctx->rc_buffer_size = bitrate;
    avcodec_open2(codec_ctx, codec, NULL);

4. 循环读取 NAL 数据，将数据写入 AVPacket 中，并设置时间戳和时长。

    while (read_len > 0) {
        AVPacket pkt;
        av_init_packet(&pkt);
        pkt.data = buf;
        pkt.size = read_len;
        pkt.pts = pts++;
        pkt.dts = pkt.pts;
        pkt.duration = 1;
        pkt.stream_index = out_stream->index;
        av_interleaved_write_frame(ofmt_ctx, &pkt);
        av_packet_unref(&pkt);
        read_len = fread(buf, 1, buf_size, fp);
    }

5. 写入文件尾，并释放资源。

    av_write_trailer(ofmt_ctx);
    avcodec_close(codec_ctx);
    avcodec_free_context(&codec_ctx);
    av_freep(&codecpar->extradata);
    avio_close(ofmt_ctx->pb);
    avformat_free_context(ofmt_ctx);

完整代码示例可以参考以下链接：https://gist.github.com/mstorsjo/012b5861e1fca20f1b2b961d853e49ec

使用ffmpg 实现rtmp 推音频流的C++代码

以下是使用FFmpeg实现RTMP推送音频流的C++代码示例：

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <stdint.h>

#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include <libavutil/opt.h>
#include <libavutil/error.h>
#include <libavutil/mathematics.h>

#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096

int main(int argc, char **argv)
{
    AVOutputFormat *ofmt = NULL;
    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
    AVPacket pkt;
    AVCodecContext *dec_ctx = NULL, *enc_ctx = NULL;
    AVCodec *encoder = NULL;
    int stream_index = 0;
    int ret = 0;
    int64_t cur_pts = 0;
    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    uint8_t *data = NULL;
    int data_size = 0;

    const char *out_filename = "rtmp://localhost:1935/live/stream";
    const char *in_filename = "test.mp3";

    av_register_all();

    // Open input file context
    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, NULL, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
        goto end;
    }

    // Retrieve stream information
    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
        goto end;
    }

    // Dump input information
    av_dump_format(ifmt_ctx, 0, in_filename, 0);

    // Open output file context
    if ((ret = avformat_alloc_output_context2(&ofmt_ctx, NULL, "flv", out_filename)) < 0) {
        av_log(NULL, AV_LOG_ERROR, "Cannot create output context\n");
        goto end;
    }

    ofmt = ofmt_ctx->oformat;

    // Add audio stream to output context
    for (int i = 0; i < ifmt_ctx->nb_streams; i++) {
        if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
            AVStream *in_stream = ifmt_ctx->streams[i];
            dec_ctx = avcodec_alloc_context3(NULL);
            if (!dec_ctx) {
                av_log(NULL, AV_LOG_ERROR, "Failed to allocate decoder context\n");
                goto end;
            }
            ret = avcodec_parameters_to_context(dec_ctx, in_stream->codecpar);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context\n");
                goto end;
            }
            encoder = avcodec_find_encoder(AV_CODEC_ID_AAC);
            if (!encoder) {
                av_log(NULL, AV_LOG_ERROR, "Codec not found\n");
                goto end;
            }
            enc_ctx = avcodec_alloc_context3(encoder);
            if (!enc_ctx) {
                av_log(NULL, AV_LOG_ERROR, "Failed to allocate encoder context\n");
                goto end;
            }
            enc_ctx->channels = dec_ctx->channels;
            enc_ctx->channel_layout = dec_ctx->channel_layout;
            enc_ctx->sample_rate = dec_ctx->sample_rate;
            enc_ctx->sample_fmt = encoder->sample_fmts[0];
            enc_ctx->bit_rate = 128000;
            if (ofmt->flags & AVFMT_GLOBALHEADER) {
                enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
            }
            ret = avcodec_open2(enc_ctx, encoder, NULL);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Failed to open encoder\n");
                goto end;
            }
            AVStream *out_stream = avformat_new_stream(ofmt_ctx, encoder);
            if (!out_stream) {
                av_log(NULL, AV_LOG_ERROR, "Failed to allocate output stream\n");
                goto end;
            }
            ret = avcodec_parameters_copy(out_stream->codecpar, enc_ctx->codecpar);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream\n");
                goto end;
            }
            out_stream->codecpar->codec_tag = 0;
            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER) {
                out_stream->codecpar->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
            }
            stream_index = out_stream->index;
            break;
        }
    }

    // Dump output information
    av_dump_format(ofmt_ctx, 0, out_filename, 1);

    // Open output file
    if (!(ofmt->flags & AVFMT_NOFILE)) {
        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Cannot open output file\n");
            goto end;
        }
    }

    // Write header to output file
    ret = avformat_write_header(ofmt_ctx, NULL);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
        goto end;
    }

    // Read frames from input file and write to output file
    while (1) {
        AVStream *in_stream, *out_stream;

        // Read packet from input file
        ret = av_read_frame(ifmt_ctx, &pkt);
        if (ret < 0) {
            break;
        }

        // Ignore packets that do not belong to the audio stream
        if (pkt.stream_index != stream_index) {
            av_packet_unref(&pkt);
            continue;
        }

        in_stream = ifmt_ctx->streams[pkt.stream_index];
        out_stream = ofmt_ctx->streams[pkt.stream_index];

        // Calculate packet duration
        AVRational time_base = in_stream->time_base;
        int64_t pts_time = av_rescale_q(pkt.pts, time_base, AV_TIME_BASE_Q);
        int64_t now_time = av_gettime() - cur_pts;
        if (now_time < pts_time) {
            av_usleep(pts_time - now_time);
        }

        // Decode audio frame
        ret = avcodec_send_packet(dec_ctx, &pkt);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error sending packet to decoder\n");
            goto end;
        }
        while (ret >= 0) {
            AVFrame *decoded_frame = av_frame_alloc();
            if (!decoded_frame) {
                av_log(NULL, AV_LOG_ERROR, "Failed to allocate memory for decoded frame\n");
                goto end;
            }
            ret = avcodec_receive_frame(dec_ctx, decoded_frame);
            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                av_frame_free(&decoded_frame);
                break;
            } else if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error receiving frame from decoder\n");
                goto end;
            }

            // Encode audio frame
            AVPacket enc_pkt;
            av_init_packet(&enc_pkt);
            enc_pkt.data = NULL;
            enc_pkt.size = 0;
            ret = avcodec_send_frame(enc_ctx, decoded_frame);
            if (ret < 0) {
                av_log(NULL, AV_LOG_ERROR, "Error sending frame to encoder\n");
                goto end;
            }
            while (ret >= 0) {
                ret = avcodec_receive_packet(enc_ctx, &enc_pkt);
                if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
                    break;
                } else if (ret < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error receiving packet from encoder\n");
                    goto end;
                }

                // Write packet to output file
                av_packet_rescale_ts(&enc_pkt, enc_ctx->time_base, out_stream->time_base);
                enc_pkt.stream_index = stream_index;
                ret = av_write_frame(ofmt_ctx, &enc_pkt);
                if (ret < 0) {
                    av_log(NULL, AV_LOG_ERROR, "Error writing audio frame\n");
                    goto end;
                }
                av_packet_unref(&enc_pkt);
            }
            av_frame_free(&decoded_frame);
        }
        av_packet_unref(&pkt);
    }

    // Flush encoders
    ret = avcodec_send_frame(enc_ctx, NULL);
    if (ret < 0) {
        av_log(NULL, AV_LOG_ERROR, "Error sending null frame to encoder\n");
        goto end;
    }
    while (ret >= 0) {
        AVPacket enc_pkt;
        av_init_packet(&enc_pkt);
        enc_pkt.data = NULL;
        enc_pkt.size = 0;
        ret = avcodec_receive_packet(enc_ctx, &enc_pkt);
        if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
            break;
        } else if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error receiving packet from encoder\n");
            goto end;
        }
        av_packet_rescale_ts(&enc_pkt, enc_ctx->time_base, ofmt_ctx->streams[stream_index]->time_base);
        enc_pkt.stream_index = stream_index;
        ret = av_write_frame(ofmt_ctx, &enc_pkt);
        if (ret < 0) {
            av_log(NULL, AV_LOG_ERROR, "Error writing audio frame\n");
            goto end;
        }
        av_packet_unref(&enc_pkt);
    }

    // Write trailer to output file
    av_write_trailer(ofmt_ctx);

end:
    avcodec_free_context(&dec_ctx);
    avcodec_free_context(&enc_ctx);
    avformat_close_input(&ifmt_ctx);
    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE)) {
        avio_closep(&ofmt_ctx->pb);
    }
    avformat_free_context(ofmt_ctx);
    if (ret < 0) {
        char errbuf[AV_ERROR_MAX_STRING_SIZE] = {0};
        av_strerror(ret, errbuf, sizeof(errbuf));
        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", errbuf);
        return 1;
    }
    return 0;
}

这段代码假设输入文件为MP3格式，输出文件为FLV格式，并且只包含一个音频流。代码将音频流从输入文件中解码，并使用AAC编码器编码成AAC格式，然后将编码后的音频流写入输出文件。在实际使用时，你需要根据你的需求进行相应的修改。