在使用ffmpeg libavformat將VP8 RTP流混合到webm時，如何從RTP時間戳設置AVPacket的pts和dts？

Question

我正在使用ffmpeg libavformat庫編寫僅視頻的webm文件。 我在服務器上收到VP8編碼的rtp流。 我已經成功地將rtp字節流（來自rtp有效負載）分組為單獨的幀，並構造了AVPacket。 我沒有在這里將有效負載重新編碼為VP8，因為它已經被vp8編碼了。

我正在使用av_write_interleaved（）方法將AVPacket寫入文件。 雖然我得到一個webm文件作為輸出，但它根本沒有播放。 當我使用mkv工具的'mkvinfo'命令檢查文件信息時，發現以下信息：

+ EBML head
|+ EBML version: 1
|+ EBML read version: 1
|+ EBML maximum ID length: 4
|+ EBML maximum size length: 8
|+ Doc type: webm
|+ Doc type version: 2
|+ Doc type read version: 2
+ Segment, size 2142500
|+ Seek head (subentries will be skipped)
|+ EbmlVoid (size: 170)
|+ Segment information
| + Timestamp scale: 1000000
| + Multiplexing application: Lavf58.0.100
| + Writing application: Lavf58.0.100
| + Duration: 78918744.480s (21921:52:24.480)
|+ Segment tracks
| + A track
|  + Track number: 1 (track ID for mkvmerge & mkvextract: 0)
|  + Track UID: 1
|  + Lacing flag: 0
|  + Name: Video Track
|  + Language: eng
|  + Codec ID: V_VP8
|  + Track type: video
|  + Default duration: 1.000ms (1000.000 frames/fields per second for a 
video track)
|  + Video track
|   + Pixel width: 640
|   + Pixel height: 480
|+ Tags
| + Tag
|  + Targets
|  + Simple
|   + Name: ENCODER
|   + String: Lavf58.0.100
| + Tag
|  + Targets
|   + TrackUID: 1
|  + Simple
|   + Name: DURATION
|   + String: 21921:52:24.4800000
|+ Cluster

我們可以看到，流的持續時間非常高。 （我的有效流時長應該在8到10秒左右）。 而且，曲目信息中的幀頻也不是我要設置的。 我將幀頻設置為25 fps。

我正在應用av_scale_q（rtpTimeStamp，codec_timebase，stream_timebase）並將重新縮放的rtpTimeStamp設置為pts和dts值。 我的猜測是我設置pts和dts的方式是錯誤的。 請幫助我如何在AVPacket上設置pts和dts值，以獲取具有適當元信息的有效webm文件。

編輯：

以下是我調用該庫的代碼：

 #define STREAM_FRAME_RATE 25
 #define STREAM_PIX_FMT AV_PIX_FMT_YUV420P 

 typedef struct OutputStream {
   AVStream *st;
   AVCodecContext *enc;
   AVFrame *frame;
 } OutputStream;


 typedef struct WebMWriter {
      OutputStream *audioStream, *videoStream;
      AVFormatContext *ctx;
      AVOutputFormat *outfmt;
      AVCodec *audioCodec, *videoCodec;
 } WebMWriter;

 static OutputStream audioStream = { 0 }, videoStream = { 0 };

 WebMWriter *init(char *filename)
 {
    av_register_all();

    AVFormatContext *ctx = NULL;
    AVCodec *audioCodec = NULL, *videoCodec = NULL;
    const char *fmt_name = NULL;
    const char *file_name = filename;

    int alloc_status = avformat_alloc_output_context2(&ctx, NULL, fmt_name, file_name);

    if(!ctx)
            return NULL;

    AVOutputFormat *fmt = (*ctx).oformat;

    AVDictionary *video_opt = NULL;
    av_dict_set(&video_opt, "language", "eng", 0);
    av_dict_set(&video_opt, "title", "Video Track", 0);

    if(fmt->video_codec != AV_CODEC_ID_NONE)
    {
            addStream(&videoStream, ctx, &videoCodec, AV_CODEC_ID_VP8, video_opt);
    }

 if(videoStream.st)
            openVideo1(&videoStream, videoCodec, NULL);

    av_dump_format(ctx, 0, file_name, 1);

    int ret = -1;
    /* open the output file, if needed */
    if (!(fmt->flags & AVFMT_NOFILE)) {
            ret = avio_open(&ctx->pb, file_name, AVIO_FLAG_WRITE);
            if (ret < 0) {
                    printf("Could not open '%s': %s\n", file_name, av_err2str(ret));
                    return NULL;
            }
    }

    /* Write the stream header, if any. */
    AVDictionary *format_opt = NULL;
    ret = avformat_write_header(ctx, &format_opt);
    if (ret < 0) {
            fprintf(stderr, "Error occurred when opening output file: %s\n",
                            av_err2str(ret));
            return NULL;
    }


    WebMWriter *webmWriter = malloc(sizeof(struct WebMWriter));
    webmWriter->ctx = ctx;
    webmWriter->outfmt = fmt;
    webmWriter->audioStream = &audioStream;
    webmWriter->videoStream = &videoStream;
    webmWriter->videoCodec = videoCodec;

    return webmWriter;
 }

以下是openVideo（）方法：

 void openVideo1(OutputStream *out_st, AVCodec *codec, AVDictionary *opt_arg)
 {       
    AVCodecContext *codec_ctx = out_st->enc;
    int ret = -1;
    AVDictionary *opt = NULL;
    if(opt_arg != NULL)
    {       
            av_dict_copy(&opt, opt_arg, 0);
            ret = avcodec_open2(codec_ctx, codec, &opt);
    }
    else
    {       
            ret = avcodec_open2(codec_ctx, codec, NULL);
    }

    /* copy the stream parameters to the muxer */
    ret = avcodec_parameters_from_context(out_st->st->codecpar, codec_ctx);
    if (ret < 0) {
            printf("Could not copy the stream parameters\n");
            exit(1);
    }

 }

以下是addStream（）方法：

 void addStream(OutputStream *out_st, AVFormatContext *ctx, AVCodec **cdc, enum AVCodecID codecId, AVDictionary *opt_arg)
 {

    (*cdc) = avcodec_find_encoder(codecId);
    if(!(*cdc)) {
            exit(1);
    }

    /*as we are passing a NULL AVCodec cdc, So AVCodecContext codec_ctx will not be allocated, we have to do it explicitly */
    AVStream *st = avformat_new_stream(ctx, *cdc);
    if(!st) {
            exit(1);
    }

    out_st->st = st;
    st->id = ctx->nb_streams-1;

    AVDictionary *opt = NULL;
    av_dict_copy(&opt, opt_arg, 0);
    st->metadata = opt;

    AVCodecContext *codec_ctx = st->codec;
    if (!codec_ctx) {
            fprintf(stderr, "Could not alloc an encoding context\n");
            exit(1);
    }
    out_st->enc = codec_ctx;

    codec_ctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

 switch ((*cdc)->type) {
            case AVMEDIA_TYPE_AUDIO:
                    codec_ctx->codec_id = codecId;
                    codec_ctx->sample_fmt  = AV_SAMPLE_FMT_FLTP;
                    codec_ctx->bit_rate    = 64000;
                    codec_ctx->sample_rate = 48000;
                    codec_ctx->channels    = 2;//1;
                    codec_ctx->channel_layout = AV_CH_LAYOUT_STEREO; 
                    codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO;
                    codec_ctx->time_base = (AVRational){1,STREAM_FRAME_RATE};


                    break;

            case AVMEDIA_TYPE_VIDEO:
                    codec_ctx->codec_id = codecId;
                    codec_ctx->bit_rate = 90000;
                    codec_ctx->width    = 640;
                    codec_ctx->height   = 480;


                    codec_ctx->time_base = (AVRational){1,STREAM_FRAME_RATE};
                    codec_ctx->gop_size = 12;
                    codec_ctx->pix_fmt = STREAM_PIX_FMT;
                    codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;

                    break;

            default:
                    break;
    }

 /* Some formats want stream headers to be separate. */
    if (ctx->oformat->flags & AVFMT_GLOBALHEADER)
            codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 }

以下是我調用的將數據幀寫入文件的代碼：

 int writeVideoStream(AVFormatContext *ctx, AVStream *st, uint8_t *data, int size, long frameTimeStamp, int isKeyFrame, AVCodecContext *codec_ctx)
 {       
    AVRational rat = st->time_base;
    AVPacket pkt = {0};
    av_init_packet(&pkt);

    void *opaque = NULL;
    int flags = AV_BUFFER_FLAG_READONLY;
    AVBufferRef *bufferRef = av_buffer_create(data, size, NULL, opaque, flags);

    pkt.buf = bufferRef;
    pkt.data = data;
    pkt.size = size;  
    pkt.stream_index  = st->index;

    pkt.pts = pkt.dts = frameTimeStamp;
    pkt.pts = av_rescale_q(pkt.pts, codec_ctx->time_base, st->time_base);
    pkt.dts = av_rescale_q(pkt.dts, codec_ctx->time_base, st->time_base);


    if(isKeyFrame == 1)
            pkt.flags |= AV_PKT_FLAG_KEY;

    int ret = av_interleaved_write_frame(ctx, &pkt);
    return ret;
 }

注意：這里的“ frameTimeStamp”是該幀的rtp數據包上的rtp timeStamp。

編輯2.0：

我使用codecpars更新的addStream（）方法更改了：

 void addStream(OutputStream *out_st, AVFormatContext *ctx, AVCodec **cdc, enum AVCodecID codecId, AVDictionary *opt_arg)
 {

    (*cdc) = avcodec_find_encoder(codecId);
    if(!(*cdc)) {
            printf("@@@@@ couldnt find codec \n");
            exit(1);
    }

    AVStream *st = avformat_new_stream(ctx, *cdc);
    if(!st) {
            printf("@@@@@ couldnt init stream\n");
            exit(1);
    }

    out_st->st = st;
    st->id = ctx->nb_streams-1;
    AVCodecParameters *codecpars = st->codecpar;
    codecpars->codec_id = codecId;
    codecpars->codec_type = (*cdc)->type;

    AVDictionary *opt = NULL;
    av_dict_copy(&opt, opt_arg, 0);
    st->metadata = opt;
    //av_dict_free(&opt);

    AVCodecContext *codec_ctx = st->codec;
    if (!codec_ctx) {
            fprintf(stderr, "Could not alloc an encoding context\n");
            exit(1);
    }
    out_st->enc = codec_ctx;

    //since opus is experimental codec
    //codec_ctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;

 switch ((*cdc)->type) {
            case AVMEDIA_TYPE_AUDIO:
                    codec_ctx->codec_id = codecId;
                    codec_ctx->sample_fmt  = AV_SAMPLE_FMT_FLTP;//AV_SAMPLE_FMT_U8 or AV_SAMPLE_FMT_S16;
                    codec_ctx->bit_rate    = 64000;
                    codec_ctx->sample_rate = 48000;
                    codec_ctx->channels    = 2;//1;
                    codec_ctx->channel_layout = AV_CH_LAYOUT_STEREO; //AV_CH_LAYOUT_MONO;
                    codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO;
                    codec_ctx->time_base = (AVRational){1,STREAM_FRAME_RATE};

                    codecpars->format = codec_ctx->sample_fmt;
                    codecpars->channels = codec_ctx->channels;
                    codecpars->sample_rate = codec_ctx->sample_rate;

                    break;

            case AVMEDIA_TYPE_VIDEO:
                    codec_ctx->codec_id = codecId;
                    codec_ctx->bit_rate = 90000;
                    codec_ctx->width    = 640;
                    codec_ctx->height   = 480;

                    codec_ctx->time_base = (AVRational){1,STREAM_FRAME_RATE};
                    codec_ctx->gop_size = 12;
                    codec_ctx->pix_fmt = STREAM_PIX_FMT;
                    //codec_ctx->max_b_frames = 1;
                    codec_ctx->codec_type = AVMEDIA_TYPE_VIDEO;
                    codec_ctx->framerate = av_inv_q(codec_ctx->time_base);
                    st->avg_frame_rate = codec_ctx->framerate;//(AVRational){25000, 1000};

                    codecpars->format = codec_ctx->pix_fmt;
                    codecpars->width = codec_ctx->width;
                    codecpars->height = codec_ctx->height;
                    codecpars->sample_aspect_ratio = (AVRational){codec_ctx->width, codec_ctx->height};

                    break;

            default:
                    break;
    }      
    codecpars->bit_rate = codec_ctx->bit_rate;

    int ret = avcodec_parameters_to_context(codec_ctx, codecpars);
    if (ret < 0) {
            printf("Could not copy the stream parameters\n");
            exit(1);
    }

    /* Some formats want stream headers to be separate. */
    if (ctx->oformat->flags & AVFMT_GLOBALHEADER)
            codec_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 }

Answer 1

我認為您對計算pts / dts是正確的問題，請使用此公式手動計算時間戳，看看是否av_rescale_q ，然后可以使用av_rescale_q 。

這是我測試過的公式（用於原始（yuv）輸出）：

int64_t frameTime;
int64_t frameDuration;

frameDuration = video_st->time_base.den / video_fps; // i.e. 25
frameTime     = frame_count * frameDuration;
pkt->pts      = frameTime / video_st->time_base.num;
pkt->duration = frameDuration;

pkt->dts          = pkt->pts;
pkt->stream_index = video_st->index;

在av_interleaved_write_frame之前使用它。
注意：這里的frame_count是一個計數器，該計數器在每個視頻幀輸出（帶有av_interleaved_write_frame）之后增加。