SIGABRT: прервать попытку кодирования PCM в AAC

Question

Я пытаюсь кодировать входящие необработанные аудиоданные PCM в аудиофайл в кодировке AA C. Следующее аварийное завершение работы SIGABRT при вызове avcodec_encode_audio2:

aac_encoding. c

#include <stdio.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>

typedef struct AACEncoder {
    AVFormatContext* pFormatCtx;
    AVStream* audio_st;
    AVCodecContext* pCodecCtx;
    AVFrame* pFrame;
    AVPacket* pkt;
    uint8_t* frame_buf;
} AACEncoder;

AACEncoder *openEncoder(char* out_file) {
    AACEncoder* encoder = (AACEncoder*)malloc(sizeof(AACEncoder*));

    av_register_all();

    AVFormatContext* pFormatCtx = avformat_alloc_context();
    encoder->pFormatCtx = pFormatCtx;

    AVOutputFormat* outFormat = av_guess_format(NULL, out_file, NULL);
    pFormatCtx->oformat = outFormat;

    if (avio_open(&pFormatCtx->pb, out_file, AVIO_FLAG_READ_WRITE) < 0) {
        printf("Failed to open output file!\n");
        return NULL;
    }

    AVStream* audio_st = avformat_new_stream(pFormatCtx, 0);
    if (audio_st==NULL){
        return NULL;
    }
    encoder->audio_st;

    AVCodecContext* pCodecCtx = audio_st->codec;
    encoder->pCodecCtx = pCodecCtx;

    pCodecCtx->codec_id = outFormat->audio_codec;
    pCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
    pCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
    pCodecCtx->sample_rate= 48000;
    pCodecCtx->channel_layout = AV_CH_LAYOUT_MONO;
    pCodecCtx->channels = av_get_channel_layout_nb_channels(pCodecCtx->channel_layout);
    pCodecCtx->bit_rate = 64000;

    av_dump_format(pFormatCtx, 0, out_file, 1);

    AVCodec* pCodec = avcodec_find_encoder(pCodecCtx->codec_id);
    if (!pCodec){
        printf("Can not find encoder!\n");
        return NULL;
    }

    if (avcodec_open2(pCodecCtx, pCodec,NULL) < 0){
        printf("Failed to open encoder!\n");
        return NULL;
    }

    AVFrame* pFrame = av_frame_alloc();
    encoder->pFrame = pFrame;

    pFrame->nb_samples= pCodecCtx->frame_size;
    pFrame->format= pCodecCtx->sample_fmt;

    int size = av_samples_get_buffer_size(NULL, pCodecCtx->channels,pCodecCtx->frame_size,pCodecCtx->sample_fmt, 1);
    uint8_t* frame_buf = (uint8_t *)av_malloc(size);
    encoder->frame_buf = frame_buf;

    avcodec_fill_audio_frame(pFrame, pCodecCtx->channels, pCodecCtx->sample_fmt,(const uint8_t*)frame_buf, size, 1);

    //Write Header
    avformat_write_header(pFormatCtx,NULL);

    AVPacket pkt;
    encoder->pkt = &pkt;

    av_new_packet(&pkt,size);

    return encoder;
}

int writePCM(AACEncoder* encoder, int16_t* pcmData, size_t pcmSize) {
    SwrContext* swr = swr_alloc();

    av_opt_set_int(swr, "in_channel_layout",  encoder->pCodecCtx->channel_layout, 0);
    av_opt_set_int(swr, "out_channel_layout", encoder->pCodecCtx->channel_layout,  0);
    av_opt_set_int(swr, "in_sample_rate",     encoder->pCodecCtx->sample_rate, 0);
    av_opt_set_int(swr, "out_sample_rate",    encoder->pCodecCtx->sample_rate, 0);
    av_opt_set_sample_fmt(swr, "in_sample_fmt",  AV_SAMPLE_FMT_S16, 0);
    av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLT,  0);
    swr_init(swr);

    printf("Initialized SwrContext\n");
    fflush(stdout);

    int ret = swr_convert(swr, encoder->pFrame->extended_data, pcmSize,  &pcmData,  pcmSize);
    int got_frame=0;
    if(ret < 0){
        printf("Failed to resample!\n");
        return -1;
    }

    //Encode
    ret = avcodec_encode_audio2(encoder->pCodecCtx, encoder->pkt, encoder->pFrame, &got_frame);
    printf("Encoded audio using AAC\n");
    fflush(stdout);

    swr_free(&swr);
    if(ret < 0){
        printf("Failed to encode!\n");
        return -1;
    }
    if (got_frame==1){
        printf("Succeed to encode 1 frame! \tsize:%5d\n", encoder->pkt->size);
        encoder->pkt->stream_index = encoder->audio_st->index;
        ret = av_write_frame(encoder->pFormatCtx, encoder->pkt);
        av_free_packet(encoder->pkt);
    }   
}