c++ - 使用 FFmpeg 库、C++ 转码为 vorbis

Question

我制作了一个测试应用程序以转码为 vorbis 格式(webm 容器)。

到目前为止，基于 FFmpeg 示例，一切正常，输出文件正常播放，但右声道声音丢失。我尝试寻找不同的可能性，但到目前为止找不到任何答案。

作为引用，这是我正在使用的代码:

#include "stdafx.h"
#define MAX_AUDIO_PACKET_SIZE (128 * 1024)

#include <iostream>
#include <fstream>

#include <string>
#include <vector>
#include <map>

#include <deque>
#include <queue>

#include <math.h>
#include <stdlib.h>
#include <stdio.h>
#include <conio.h>

extern "C"
{
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
#include "libavutil/dict.h"
#include "libavutil/error.h"
#include "libavutil/opt.h"
#include <libavutil/fifo.h>
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
}
AVCodecID           outputAudioFormat = AV_CODEC_ID_VORBIS;


static int sws_flags = SWS_BICUBIC;
#define STREAM_DURATION   50.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */


AVFormatContext*    fmt_ctx= NULL;
int                    audio_stream_index = -1;
AVCodecContext *    codec_ctx_audio = NULL;
AVCodec*            codec_audio = NULL;
AVFrame*            decoded_frame = NULL;
uint8_t**            audio_dst_data = NULL;
int                    got_frame = 0;
int                    audiobufsize = 0;
AVPacket            input_packet;
int                    audio_dst_linesize = 0;
int                    audio_dst_bufsize = 0;
SwrContext *        swrContext = NULL;

AVOutputFormat *    output_format = NULL ;
AVFormatContext *    output_fmt_ctx= NULL;
AVStream *            audio_st = NULL;
AVStream*           video_st = NULL;
AVCodec *            audio_codec = NULL;
AVCodec*            video_codec = NULL;
double                audio_pts = 0.0;
AVFrame *            out_frame = avcodec_alloc_frame();

int                    audio_input_frame_size = 64;

uint8_t *            audio_data_buf = NULL;
uint8_t *            audio_out = NULL;
int                    audio_bit_rate;
int                    audio_sample_rate;
int                    audio_channels;
int                 sourceSampleRate=0;
int                 destSampleRate = 0;

int                 dst_nb_samples = 0;
int                 pivotIndex = 0;
int                 max_dst_nb_samples = 0;
int                 samples_count=0;


int decode_packet();
int open_audio_input(char* src_filename);
int decode_frame();

int open_encoder(char* output_filename);
AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
    enum AVCodecID codec_id);
int open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st);
void close_audio(AVFormatContext *oc, AVStream *st);
void write_audio_frame(uint8_t ** audio_src_data, int audio_src_bufsize);

static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                            enum AVCodecID codec_id)
{
    AVCodecContext *c;
    AVStream *st;

    /* find the encoder */
    *codec = avcodec_find_encoder(codec_id);
    if (!(*codec)) {
        fprintf(stderr, "Could not find encoder for '%s'\n",
                avcodec_get_name(codec_id));
        exit(1);
    }

    st = avformat_new_stream(oc, *codec);
    if (!st) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
    st->id = oc->nb_streams-1;
    c = st->codec;

    switch ((*codec)->type) {
    case AVMEDIA_TYPE_AUDIO:
        c->sample_fmt  = (*codec)->sample_fmts ?
            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
        c->bit_rate    = 64000;
        c->sample_rate = 44100;
        c->channels    = 2;
        break;

    case AVMEDIA_TYPE_VIDEO:
        c->codec_id = codec_id;

        c->bit_rate = 400000;
        /* Resolution must be a multiple of two. */
        c->width    = 352;
        c->height   = 288;
        /* timebase: This is the fundamental unit of time (in seconds) in terms
         * of which frame timestamps are represented. For fixed-fps content,
         * timebase should be 1/framerate and timestamp increments should be
         * identical to 1. */
        c->time_base.den = STREAM_FRAME_RATE;
        c->time_base.num = 1;
        c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
        c->pix_fmt       = STREAM_PIX_FMT;
        if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
            /* just for testing, we also add B frames */
            c->max_b_frames = 2;
        }
        if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
            /* Needed to avoid using macroblocks in which some coeffs overflow.
             * This does not happen with normal video, it just happens here as
             * the motion of the chroma plane does not match the luma plane. */
            c->mb_decision = 2;
        }
    break;

    default:
        break;
    }

    /* Some formats want stream headers to be separate. */
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;

    return st;
}


static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
    int ret;
    AVCodecContext *c = st->codec;

    /* open the codec */
    ret = avcodec_open2(c, codec, NULL);
    if (ret < 0) {
        //fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
        exit(1);
    }

    /* allocate and init a re-usable frame */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate video frame\n");
        exit(1);
    }
    frame->format = c->pix_fmt;
    frame->width = c->width;
    frame->height = c->height;

    /* Allocate the encoded raw picture. */
    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
    if (ret < 0) {
        //fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
        exit(1);
    }

    /* If the output format is not YUV420P, then a temporary YUV420P
     * picture is needed too. It is then converted to the required
     * output format. */
    if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
        ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
        if (ret < 0) {
            //fprintf(stderr, "Could not allocate temporary picture: %s\n",
            //        av_err2str(ret));
            exit(1);
        }
    }

    /* copy data and linesize picture pointers to frame */
    *((AVPicture *)frame) = dst_picture;
}

int open_audio_input(char* src_filename)
{
    int i =0;
    /* open input file, and allocate format context */
    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0)
    {
        fprintf(stderr, "Could not open source file %s\n", src_filename);
        exit(1);
    }

    // Retrieve stream information
    if(avformat_find_stream_info(fmt_ctx, NULL)<0)
        return -1; // Couldn't find stream information

    // Dump information about file onto standard error
    av_dump_format(fmt_ctx, 0, src_filename, 0);

    // Find the first video stream
    for(i=0; i<fmt_ctx->nb_streams; i++)
    {
        if(fmt_ctx->streams[i]->codec->codec_type==AVMEDIA_TYPE_AUDIO)
        {
            audio_stream_index=i;
            break;
        }
    }
    if ( audio_stream_index != -1 )
    {
        // Get a pointer to the codec context for the audio stream
        codec_ctx_audio=fmt_ctx->streams[audio_stream_index]->codec;

        // Find the decoder for the video stream
        codec_audio=avcodec_find_decoder(codec_ctx_audio->codec_id);
        if(codec_audio==NULL) {
            fprintf(stderr, "Unsupported audio codec!\n");
            return -1; // Codec not found
        }

        // Open codec
        AVDictionary *codecDictOptions = NULL;
        if(avcodec_open2(codec_ctx_audio, codec_audio, &codecDictOptions)<0)
            return -1; // Could not open codec

        // Set up SWR context once you've got codec information
        swrContext = swr_alloc();
        av_opt_set_int(swrContext, "in_channel_layout",  codec_ctx_audio->channel_layout, 0);
        av_opt_set_int(swrContext, "out_channel_layout", codec_ctx_audio->channel_layout,  0);
        av_opt_set_int(swrContext, "in_sample_rate",     codec_ctx_audio->sample_rate, 0);
        av_opt_set_int(swrContext, "out_sample_rate",    codec_ctx_audio->sample_rate, 0);
        av_opt_set_sample_fmt(swrContext, "in_sample_fmt",  codec_ctx_audio->sample_fmt, 0);
        if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
        {
            av_opt_set_sample_fmt(swrContext, "out_sample_fmt", AV_SAMPLE_FMT_FLTP,  0);
        }
        else
        {
            av_opt_set_sample_fmt(swrContext, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);
        }
        int rv = swr_init(swrContext);

        sourceSampleRate    =   destSampleRate = codec_ctx_audio->sample_rate;

        // Allocate audio frame
        if ( decoded_frame == NULL ) decoded_frame = avcodec_alloc_frame();
        int nb_planes = 0;
        AVStream* audio_stream = fmt_ctx->streams[audio_stream_index];
        nb_planes = av_sample_fmt_is_planar(codec_ctx_audio->sample_fmt) ? codec_ctx_audio->channels : 1;
        int tempSize =  sizeof(uint8_t *) * nb_planes;
        audio_dst_data = (uint8_t**)av_mallocz(tempSize);
        if (!audio_dst_data)
        {
            fprintf(stderr, "Could not allocate audio data buffers\n");
        }
        else
        {
            for ( int i = 0 ; i < nb_planes ; i ++ )
            {
                audio_dst_data[i] = NULL;
            }
        }
    }
}

int decode_frame()
{
    int rv = 0;
    got_frame = 0;
    if ( fmt_ctx == NULL  )
    {
        return rv;
    }
    int ret = 0;
    audiobufsize = 0;
    rv = av_read_frame(fmt_ctx, &input_packet);
    if ( rv < 0 )
    {
        return rv;
    }
    rv = decode_packet();
    // Free the input_packet that was allocated by av_read_frame
    av_free_packet(&input_packet);
    return rv;
}

int decode_packet()
{
    int rv = 0;
    int ret = 0;

    //audio stream?
    if(input_packet.stream_index == audio_stream_index)
    {
        avcodec_get_frame_defaults(decoded_frame);
        while( input_packet.size > 0 )
        {
            int result = avcodec_decode_audio4(codec_ctx_audio, decoded_frame, &got_frame, &input_packet);
            if ( result < 0)
            {
                fprintf(stderr, "Error decoding audio frame\n");
                //return ret;
            }
            else
            {
                if ( got_frame )
                {
                    dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(swrContext, sourceSampleRate) + decoded_frame->nb_samples, sourceSampleRate, destSampleRate, AV_ROUND_UP);
                    if ( dst_nb_samples > max_dst_nb_samples )
                    {
                        max_dst_nb_samples = dst_nb_samples;
                        if ( audio_dst_data[0] )
                        {
                            av_freep(&audio_dst_data[0]);
                            audio_dst_data[0] = NULL;
                        }
                    }
                    if ( audio_dst_data[0] == NULL )
                    {
                        if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
                        {
                            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, codec_ctx_audio->channels,
                                decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 0);
                        }
                        else
                        {
                            ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, codec_ctx_audio->channels,
                                decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_S16, 0);
                        }
                    }
                    /* TODO: extend return code of the av_samples_* functions so that this call is not needed */
                    int resampled  = swr_convert(swrContext, audio_dst_data, out_frame->nb_samples,
                        (const uint8_t **)(decoded_frame->extended_data), decoded_frame->nb_samples);
                    char str[900]="";
                    sprintf(str,"out_frame->nb_samples:\t%d; decoded_frame->nb_samples:\t%d",out_frame->nb_samples,decoded_frame->nb_samples );
                    if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
                    {
                        audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 1);
                    }
                    else
                    {
                        audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_S16, 1);
                    }

                    input_packet.size -= result;
                    input_packet.data += result;
                }
                else
                {
                    input_packet.size   =   0;
                    input_packet.data   =   NULL;
                }
            }
        }
    }
    return rv;
}

int open_encoder(char* output_filename )
{
    int rv = 0;

    /* allocate the output media context */
    AVOutputFormat *opfmt = NULL;

    avformat_alloc_output_context2(&output_fmt_ctx, opfmt, NULL, output_filename);
    if (!output_fmt_ctx) {
        printf("Could not deduce output format from file extension: using MPEG.\n");
        avformat_alloc_output_context2(&output_fmt_ctx, NULL, "mpeg", output_filename);
    }
    if (!output_fmt_ctx) {
        rv = -1;
    }
    else
    {
        output_format = output_fmt_ctx->oformat;
    }

    /* Add the audio stream using the default format codecs
    * and initialize the codecs. */
    audio_st = NULL;

    if ( output_fmt_ctx )
    {
        if (output_format->audio_codec != AV_CODEC_ID_NONE)
        {
            audio_st = add_audio_stream(output_fmt_ctx, &audio_codec, output_format->audio_codec);
        }

        /* Now that all the parameters are set, we can open the audio and
        * video codecs and allocate the necessary encode buffers. */

        if (audio_st)
        {
            rv = open_audio(output_fmt_ctx, audio_codec, audio_st);
            if ( rv < 0 ) return rv;
        }

        av_dump_format(output_fmt_ctx, 0, output_filename, 1);
        /* open the output file, if needed */
        if (!(output_format->flags & AVFMT_NOFILE))
        {
            if (avio_open(&output_fmt_ctx->pb, output_filename, AVIO_FLAG_WRITE) < 0) {
                fprintf(stderr, "Could not open '%s'\n", output_filename);
                rv = -1;
            }
            else
            {
                /* Write the stream header, if any. */
                if (avformat_write_header(output_fmt_ctx, NULL) < 0)
                {
                    fprintf(stderr, "Error occurred when opening output file\n");
                    rv = -1;
                }
            }
        }
    }

    return rv;
}

AVStream *add_audio_stream(AVFormatContext *oc, AVCodec **codec,
    enum AVCodecID codec_id)
{
    AVCodecContext *c;
    AVStream *st;

    /* find the audio encoder */
    *codec = avcodec_find_encoder(codec_id);
    if (!(*codec)) {
        fprintf(stderr, "Could not find codec\n");
        exit(1);
    }

    st = avformat_new_stream(oc, *codec);
    if (!st) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }
    st->id = 1;

    c = st->codec;

    /* put sample parameters */
    if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
    {
        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
    }
    else
    {
        c->sample_fmt  = AV_SAMPLE_FMT_S16;
    }

    c->bit_rate    = audio_bit_rate;
    c->sample_rate = audio_sample_rate;
    c->channels    = audio_channels;

    // some formats want stream headers to be separate
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;

    return st;
}

int open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
    int ret=0;
    AVCodecContext *c;

    st->duration = fmt_ctx->duration;
    c = st->codec;

    /* open it */
    ret = avcodec_open2(c, codec, NULL) ;
    if ( ret < 0)
    {
        fprintf(stderr, "could not open codec\n");
        return -1;
        //exit(1);
    }

    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
        audio_input_frame_size = 10000;
    else
        audio_input_frame_size = c->frame_size;
    out_frame->nb_samples = audio_input_frame_size;
    int tempSize = audio_input_frame_size *
        av_get_bytes_per_sample(c->sample_fmt) *
        c->channels;
    return ret;
}

void close_audio(AVFormatContext *oc, AVStream *st)
{
    avcodec_close(st->codec);
}

void write_audio_frame(uint8_t ** audio_dst_data, int audio_dst_bufsize)
{
    AVFormatContext *oc = output_fmt_ctx;
    AVStream *st = audio_st;
    if ( oc == NULL || st == NULL ) return;
    AVCodecContext *c;
    AVPacket pkt = { 0 }; // data and size must be 0;
    int got_packet=0, ret=0;

    av_init_packet(&pkt);
    c = st->codec;

    out_frame->nb_samples = audio_input_frame_size;

    AVRational r;
    r.num = 1;
    r.den = c->sample_rate;
    out_frame->pts = av_rescale_q(samples_count, (AVRational)r, c->time_base);
    avcodec_fill_audio_frame(out_frame, c->channels, c->sample_fmt,
                             audio_dst_data[0], audio_dst_bufsize, 0);
    samples_count += out_frame->nb_samples;

    ret = avcodec_encode_audio2(c, &pkt, out_frame, &got_packet);
    if (ret < 0) 
    {
        return;
    }

    if (!got_packet)
        return;

    /* rescale output packet timestamp values from codec to stream timebase */
    pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
    pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
    pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
    pkt.stream_index = st->index;

    char str[999]="";
    sprintf(str,"out_frame->nb_samples:\t%d",out_frame->nb_samples);
    /* Write the compressed frame to the media file. */
    ret = av_interleaved_write_frame(oc, &pkt);
    if (ret != 0) 
    {
        exit(1);
    }
    av_free_packet(&pkt);
}

void write_delayed_frames(AVFormatContext *oc, AVStream *st)
{
    AVCodecContext *c = st->codec;
    int got_output = 0;
    int ret = 0;
    AVPacket pkt;
    pkt.data = NULL;
    pkt.size = 0;
    av_init_packet(&pkt);
    int i = 0;
    for (got_output = 1; got_output; i++)
    {
        ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
        if (ret < 0)
        {
            fprintf(stderr, "error encoding frame\n");
            exit(1);
        }
        static int64_t tempPts = 0;
        static int64_t tempDts = 0;
        /* If size is zero, it means the image was buffered. */
        if (got_output)
        {
            pkt.pts = av_rescale_q_rnd(pkt.pts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
            pkt.dts = av_rescale_q_rnd(pkt.dts, c->time_base, st->time_base, (AVRounding )(AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX));
            pkt.duration = av_rescale_q(pkt.duration, c->time_base, st->time_base);
            pkt.stream_index = st->index;
            if ( c && c->coded_frame && c->coded_frame->key_frame)
                pkt.flags |= AV_PKT_FLAG_KEY;
            /* Write the compressed frame to the media file. */
            ret = av_interleaved_write_frame(oc, &pkt);
        }
        else
        {
            ret = 0;
        }
        av_free_packet(&pkt);
    }
}

int main(int argc, char **argv)
{
    /* register all formats and codecs */
    av_register_all();
    avcodec_register_all();
    avformat_network_init();
    avdevice_register_all();
    int i =0;
    int ret=0;
    char src_filename[90] = "test.mp2";
    char dst_filename[90] = "output.webm";
    outputAudioFormat = AV_CODEC_ID_VORBIS;
    open_audio_input(src_filename);
    if ( codec_ctx_audio->bit_rate == 0 ) codec_ctx_audio->bit_rate = 112000;
    audio_bit_rate        = codec_ctx_audio->bit_rate;
    audio_sample_rate    = codec_ctx_audio->sample_rate;
    audio_channels        = codec_ctx_audio->channels;
    open_encoder( dst_filename );
    int frames= 0;
    while(1)
    {
        int rv = decode_frame();
        if ( rv < 0 )
        {
            break;
        }

        if (audio_st)
        {
            audio_pts = audio_st->pts.val * av_q2d(audio_st->time_base);
        }
        else
        {
            audio_pts = 0.0;
        }
        if ( codec_ctx_audio )
        {
            if ( got_frame )
            {
                write_audio_frame( audio_dst_data, audio_dst_bufsize );
                frames++;
            }
        }
        printf("\naudio_pts: %f", audio_pts);
    }
    while(1)
    {
        dst_nb_samples = (int)av_rescale_rnd(swr_get_delay(swrContext, sourceSampleRate) + decoded_frame->nb_samples, sourceSampleRate, destSampleRate, AV_ROUND_UP);
        if ( dst_nb_samples > max_dst_nb_samples )
        {
            max_dst_nb_samples = dst_nb_samples;
            if ( audio_dst_data[0] )
            {
                av_freep(&audio_dst_data[0]);
                audio_dst_data[0] = NULL;
            }
        }
        if ( audio_dst_data[0] == NULL )
        {
            if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
            {
                ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
                    decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 0);
            }
            else
            {
                ret = av_samples_alloc(audio_dst_data, NULL, codec_ctx_audio->channels,
                    decoded_frame->nb_samples, (AVSampleFormat)AV_SAMPLE_FMT_S16, 0);
            }
        }
        int resampled = swr_convert(swrContext, audio_dst_data, out_frame->nb_samples,NULL, 0);
        if ( outputAudioFormat == AV_CODEC_ID_VORBIS )
        {
            audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_FLTP, 1);
        }
        else
        {
            audio_dst_bufsize  = av_samples_get_buffer_size(&audio_dst_linesize, decoded_frame->channels, resampled, (AVSampleFormat)AV_SAMPLE_FMT_S16, 1);
        }
        if ( audio_dst_bufsize <= 0 ) break;
        audio_pts = audio_st->pts.val * av_q2d(audio_st->time_base);
        printf("\naudio_pts: %f", audio_pts);
        write_audio_frame( audio_dst_data, audio_dst_bufsize );
    }
    write_delayed_frames( output_fmt_ctx, audio_st );
    av_write_trailer(output_fmt_ctx);
    close_audio( output_fmt_ctx, audio_st);
    swr_free(&swrContext);
    avcodec_free_frame(&out_frame);
    getch();
    return 0;
}

在 Windows 7 下工作，Zeranoe FFmpeg 32 位构建:

libavutil      52. 62.100 / 52. 62.100
libavcodec     55. 47.101 / 55. 47.101
libavformat    55. 22.103 / 55. 22.103
libavdevice    55.  5.102 / 55.  5.102
libavfilter     4.  1.100 /  4.  1.100
libswscale      2.  5.101 /  2.  5.101
libswresample   0. 17.104 /  0. 17.104
libpostproc    52.  3.100 / 52.  3.100

谁能指出我可能误解的地方？

感谢您提前提供任何指导!

Answer 1

重采样器很可能未初始化或未正确使用。你能改变一下我在这里使用它的方式吗:https://sourceforge.net/p/karlyriceditor/code/HEAD/tree/src/ffmpegvideoencoder.cpp ?