c - libav:编码/解码错误:get_buffer() 失败

Question

我正在尝试编辑位于 libav-11.4/doc/examples/avcodec.c 中的 libav 的给定音频编码/解码示例。

我只是想将 .wav 文件编码为 aac，然后解码回 .wav

但是在解码步骤中我总是遇到以下错误:

[aac @ 0x...] get_buffer() failed
Error while decoding

编辑后的示例代码:

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

#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif

#include "libavcodec/avcodec.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/imgutils.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"


#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096

/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
    const enum AVSampleFormat *p = codec->sample_fmts;

    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
}

/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
    const int *p;
    int best_samplerate = 0;

    if (!codec->supported_samplerates)
        return 44100;

    p = codec->supported_samplerates;
    while (*p) {
        best_samplerate = FFMAX(*p, best_samplerate);
        p++;
    }
    return best_samplerate;
}

/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channels   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;

    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
}

/*
 * Audio encoding example
 */
static void audio_encode_example(const char *outfilename, const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    FILE *f, *outfile;
    uint16_t *samples;
    float t, tincr;

    printf("Audio encoding\n");

    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);

    /* put sample parameters */
    c->bit_rate = 64000;

    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "encoder does not support %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }

    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);


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

    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }

    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }

    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "could not allocate audio frame\n");
        exit(1);
    }

    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    frame->channel_layout = c->channel_layout;

    /* the codec gives us the frame size, in samples,
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
    samples = av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
                buffer_size);
        exit(1);
    }
    /* setup the data pointers in the AVFrame */
    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "could not setup audio frame\n");
        exit(1);
    }

    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for(i=0;i<200;i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;

        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);

            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
        /* encode the samples */
        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, outfile);
            av_free_packet(&pkt);
        }
    }
    fclose(f);
    fclose(outfile);

    av_freep(&samples);
    av_frame_free(&frame);
    avcodec_close(c);
    av_free(c);
}

/*
 * Audio decoding.
 */
static void audio_decode_example(const char *outfilename, const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;

    av_init_packet(&avpkt);

    printf("Audio decoding\n");

    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);

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

    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }
    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }

    /* decode until eof */
    avpkt.data = inbuf;
    avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);

    while (avpkt.size > 0) {
        int got_frame = 0;

        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "out of memory\n");
                exit(1);
            }
        }

        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
            fprintf(stderr, "Error while decoding\n");
            exit(1);
        }
        if (got_frame) {
            /* if a frame has been decoded, output it */
            int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                       decoded_frame->nb_samples,
                                                       c->sample_fmt, 1);
            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }
        avpkt.size -= len;
        avpkt.data += len;
        if (avpkt.size < AUDIO_REFILL_THRESH) {
            /* Refill the input buffer, to avoid trying to decode
             * incomplete frames. Instead of this, one could also use
             * a parser, or use a proper container format through
             * libavformat. */
            memmove(inbuf, avpkt.data, avpkt.size);
            avpkt.data = inbuf;
            len = fread(avpkt.data + avpkt.size, 1,
                        AUDIO_INBUF_SIZE - avpkt.size, f);
            if (len > 0)
                avpkt.size += len;
        }
    }

    fclose(outfile);
    fclose(f);

    avcodec_close(c);
    av_free(c);
    av_frame_free(&decoded_frame);
}

int main(int argc, char **argv)
{
    const char *filename;

    /* register all the codecs */
    avcodec_register_all();


   filename = argv[1];


    audio_encode_example("test.aac", filename);
    audio_decode_example("out.wav", "test.aac");

    return 0;
}

编辑:

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

#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif

#include "libavcodec/avcodec.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/imgutils.h"
#include "libavutil/mathematics.h"
#include "libavutil/samplefmt.h"

#include "libavformat/avformat.h"


#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096

/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
    const enum AVSampleFormat *p = codec->sample_fmts;

    while (*p != AV_SAMPLE_FMT_NONE) {
        if (*p == sample_fmt)
            return 1;
        p++;
    }
    return 0;
}

/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
    const int *p;
    int best_samplerate = 0;

    if (!codec->supported_samplerates)
        return 44100;

    p = codec->supported_samplerates;
    while (*p) {
        best_samplerate = FFMAX(*p, best_samplerate);
        p++;
    }
    return best_samplerate;
}

/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
    const uint64_t *p;
    uint64_t best_ch_layout = 0;
    int best_nb_channels   = 0;

    if (!codec->channel_layouts)
        return AV_CH_LAYOUT_STEREO;

    p = codec->channel_layouts;
    while (*p) {
        int nb_channels = av_get_channel_layout_nb_channels(*p);

        if (nb_channels > best_nb_channels) {
            best_ch_layout    = *p;
            best_nb_channels = nb_channels;
        }
        p++;
    }
    return best_ch_layout;
}

/*
 * Audio encoding example
 */
static void audio_encode_example(const char *outfilename, const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    AVFrame *frame;
    AVPacket pkt;
    int i, j, k, ret, got_output;
    int buffer_size;
    FILE *f, *outfile;
    uint16_t *samples;
    float t, tincr;

    printf("Audio encoding\n");

    /* find the MP2 encoder */
    codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);

    /* put sample parameters */
    c->bit_rate = 64000;

    /* check that the encoder supports s16 pcm input */
    c->sample_fmt = AV_SAMPLE_FMT_S16;
    if (!check_sample_fmt(codec, c->sample_fmt)) {
        fprintf(stderr, "encoder does not support %s",
                av_get_sample_fmt_name(c->sample_fmt));
        exit(1);
    }

    /* select other audio parameters supported by the encoder */
    c->sample_rate    = select_sample_rate(codec);
    c->channel_layout = select_channel_layout(codec);
    c->channels       = av_get_channel_layout_nb_channels(c->channel_layout);


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

    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }

    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }

    /* frame containing input raw audio */
    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "could not allocate audio frame\n");
        exit(1);
    }

    frame->nb_samples     = c->frame_size;
    frame->format         = c->sample_fmt;
    frame->channel_layout = c->channel_layout;

    /* the codec gives us the frame size, in samples,
     * we calculate the size of the samples buffer in bytes */
    buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                             c->sample_fmt, 0);
    samples = av_malloc(buffer_size);
    if (!samples) {
        fprintf(stderr, "could not allocate %d bytes for samples buffer\n",
                buffer_size);
        exit(1);
    }
    /* setup the data pointers in the AVFrame */
    ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
                                   (const uint8_t*)samples, buffer_size, 0);
    if (ret < 0) {
        fprintf(stderr, "could not setup audio frame\n");
        exit(1);
    }

    /* encode a single tone sound */
    t = 0;
    tincr = 2 * M_PI * 440.0 / c->sample_rate;
    for(i=0;i<200;i++) {
        av_init_packet(&pkt);
        pkt.data = NULL; // packet data will be allocated by the encoder
        pkt.size = 0;

        for (j = 0; j < c->frame_size; j++) {
            samples[2*j] = (int)(sin(t) * 10000);

            for (k = 1; k < c->channels; k++)
                samples[2*j + k] = samples[2*j];
            t += tincr;
        }
        /* encode the samples */
        ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
        if (ret < 0) {
            fprintf(stderr, "error encoding audio frame\n");
            exit(1);
        }
        if (got_output) {
            fwrite(pkt.data, 1, pkt.size, outfile);
            av_free_packet(&pkt);
        }
    }
    fclose(f);
    fclose(outfile);

    av_freep(&samples);
    av_frame_free(&frame);
    avcodec_close(c);
    av_free(c);
}

/*
 * Audio decoding.
 */
static void audio_decode_example(const char *outfilename, const char *filename)
{
    AVCodec *codec;
    AVCodecContext *c= NULL;
    int len;
    FILE *f, *outfile;
    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
    AVPacket avpkt;
    AVFrame *decoded_frame = NULL;

    av_init_packet(&avpkt);

    printf("Audio decoding\n");

    /* find the mpeg audio decoder */
    codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }

    c = avcodec_alloc_context3(codec);

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

    /*
    f = fopen(filename, "rb");
    if (!f) {
        fprintf(stderr, "could not open %s\n", filename);
        exit(1);
    }
    */


    AVFormatContext *s = avformat_alloc_context();
    int ret = avformat_open_input(&s, filename, NULL, NULL);
    if (ret != 0)
    {
        abort();
    }

    // Retrieve stream information
    printf("sdfsdf\n");

    if(avformat_find_stream_info(s, NULL)<0)
    {
        printf("Could not find stream info");// Couldn't find stream information
    }
    // Dump information about file into standard error
    av_dump_format(s, 0, filename, 0);


    outfile = fopen(outfilename, "wb");
    if (!outfile) {
        av_free(c);
        exit(1);
    }

    /* decode until eof */
    //avpkt.data = inbuf;
    //avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);

    int read_ok = av_read_frame(s,&avpkt);

    while (read_ok) 
    {
read_ok = av_read_frame(s,&avpkt);
        int got_frame = 0;

        if (!decoded_frame) {
            if (!(decoded_frame = av_frame_alloc())) {
                fprintf(stderr, "out of memory\n");
                exit(1);
            }
        }

        len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
        if (len < 0) {
            fprintf(stderr, "Error while decoding\n");
            exit(1);
        }
        if (got_frame) {
            /* if a frame has been decoded, output it */
            int data_size = av_samples_get_buffer_size(NULL, c->channels,
                                                       decoded_frame->nb_samples,
                                                       c->sample_fmt, 1);
            fwrite(decoded_frame->data[0], 1, data_size, outfile);
        }

        /*
        avpkt.size -= len;
        avpkt.data += len;
        if (avpkt.size < AUDIO_REFILL_THRESH) {

            memmove(inbuf, avpkt.data, avpkt.size);
            avpkt.data = inbuf;

            len = fread(avpkt.data + avpkt.size, 1,
                        AUDIO_INBUF_SIZE - avpkt.size, f);
            if (len > 0)
                avpkt.size += len;
        }*/
    }

    fclose(outfile);
   // fclose(f);

    avcodec_close(c);
    av_free(c);
    av_frame_free(&decoded_frame);
}

int main(int argc, char **argv)
{
    const char *filename;

    /* register all the codecs */
    avcodec_register_all();
    av_register_all();




   filename = argv[1];


    audio_encode_example("test.aac", filename);
    audio_decode_example("out.wav", "test.aac");

    return 0;
}

Answer 1

参见this post，您不能使用 fopen/fread 将原始 AAC 数据输入解码器，您需要使用 av_read_frame()。