cmake编译最小ffmpeg8.1工程, 用于flac解码

CMakeLists.txt

file(GLOB FF_SOURCES 
"libavcodec/flac*.c"
"libavformat/flacdec.c"
"libavformat/flac_picture.c"
)

add_library(
    ffmpeg STATIC
    ${FF_SOURCES}
    libavcodec/packet.c
    libavcodec/allcodecs.c
    libavcodec/avcodec.c
    libavcodec/decode.c
    libavutil/samplefmt.c
    libavcodec/parser.c
    libavcodec/options.c
    libavutil/frame.c
    libavutil/opt.c
    libavcodec/utils.c
    libavcodec/codec_desc.c
    libavutil/buffer.c
    libavcodec/profiles.c
    libavutil/crc.c
    # libavutil/log.c
    libavutil/log2_tab.c
    libavutil/bprint.c
    libavutil/mem.c
    libavutil/dict.c
    libavutil/imgutils.c
    libavutil/channel_layout.c
    libavutil/refstruct.c
    libavcodec/bsf.c
    libavutil/side_data.c
    libavutil/error.c
    libavutil/mathematics.c
    libavcodec/get_buffer.c
    libavutil/avstring.c
    libavcodec/encode.c
    libavutil/pixdesc.c
    libavutil/hwcontext.c
    libavcodec/exif.c
    libavutil/display.c
    libavcodec/tiff_common.c
    libavutil/eval.c

    libavutil/rational.c
    libavcodec/codec_par.c

    libavcodec/parsers.c
    libavutil/parseutils.c
    libavutil/reverse.c
    libavutil/avsscanf.c
    libavcodec/threadprogress.c
    libavcodec/bitstream_filters.c

    libavformat/demux.c
    libavformat/avformat.c

    libavutil/utils.c

    libavformat/format.c

    libavformat/options.c

    libavformat/id3v1.c
    libavformat/id3v2.c

    libavformat/aviobuf.c

    libavformat/metadata.c

    libavformat/demux_utils.c

    libavutil/timestamp.c

    libavformat/seek.c

    libavformat/avio.c

    libavcodec/raw.c
    libavutil/iamf.c
    libavformat/utils.c
    libavformat/allformats.c
    libavformat/protocols.c
    libavformat/rawdec.c

    libavformat/oggparsevorbis.c #?
    libavformat/vorbiscomment.c #?

    libavformat/replaygain.c
    libavutil/base64.c
    # libavformat/file.c # 没有posix接口就自己写
)

codec_list.c

/*
实际上不止这个文件要手动创建,下面是所有要手动配置的文件
libavcodec/bsf_list.c
libavcodec/codec_list.c
libavcodec/parser_list.c
libavformat/demuxer_list.c
libavformat/muxer_list.c
libavformat/protocol_list.c
*/

// 这是codec_list.c的例子, 按需添加. 注意这些列表的结尾要有一个NULL
const FFCodec *codec_list[] = {
    &ff_flac_decoder,
    // &ff_mp3_decoder,
    NULL
};

config.h

//参考配置

#ifndef FFMPEG_CONFIG_H
#define FFMPEG_CONFIG_H

#define HAVE_PRAGMA_DEPRECATED (1)
#define HAVE_FAST_64BIT (1)


#define HAVE_CBRT (1)
#define HAVE_CBRTF (1)
#define HAVE_COPYSIGN (1)
#define HAVE_ERF (1)
#define HAVE_HYPOT (1)
#define HAVE_RINT (1)
#define HAVE_LRINT (1)
#define HAVE_LRINTF (1)
#define HAVE_ROUND (1)
#define HAVE_ROUNDF (1)
#define HAVE_TRUNC (1)
#define HAVE_TRUNCF (1)
#define CHECKED
#define HAVE_THREADS (0)
#define CONFIG_GRAY (1)
#define CONFIG_MEMORY_POISONING (1)
#define HAVE_SIMD_ALIGN_64 (0)

#define CONFIG_ZLIB (0)
#define CONFIG_FILE_PROTOCOL (1)
#endif

demux_decode.c

// 注意unpadded_linesize只是一个通道的大小, 还要乘上frame->ch_layout.nb_channels
/*
 * Copyright (c) 2012 Stefano Sabatini
 *
 * Permission is hereby granted, free of charge, to any person obtaining a copy
 * of this software and associated documentation files (the "Software"), to deal
 * in the Software without restriction, including without limitation the rights
 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
 * copies of the Software, and to permit persons to whom the Software is
 * furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be included in
 * all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
 * THE SOFTWARE.
 */

/**
 * @file libavformat and libavcodec demuxing and decoding API usage example
 * @example demux_decode.c
 *
 * Show how to use the libavformat and libavcodec API to demux and decode audio
 * and video data. Write the output as raw audio and input files to be played by
 * ffplay.
 */

#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>

static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *audio_dst_file = NULL;

static int audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket *pkt = NULL;
static int audio_frame_count = 0;


static int output_audio_frame(AVFrame *frame)
{
    size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
    printf("audio_frame n:%d nb_samples:%d pts:%s\n",
           audio_frame_count++, frame->nb_samples,
           av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));

    /* Write the raw audio data samples of the first plane. This works
     * fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
     * most audio decoders output planar audio, which uses a separate
     * plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
     * In other words, this code will write only the first audio channel
     * in these cases.
     * You should use libswresample or libavfilter to convert the frame
     * to packed data. */
    // fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
    // 输出到i2s或者codec
    return 0;
}

static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
{
    int ret = 0;

    // submit the packet to the decoder
    ret = avcodec_send_packet(dec, pkt);
    if (ret < 0) {
        fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
        return ret;
    }

    // get all the available frames from the decoder
    while (ret >= 0) {
        ret = avcodec_receive_frame(dec, frame);
        if (ret < 0) {
            // those two return values are special and mean there is no output
            // frame available, but there were no errors during decoding
            if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
                return 0;

            fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
            return ret;
        }

        // write the frame data to output file
        if (dec->codec->type == AVMEDIA_TYPE_AUDIO)
            ret = output_audio_frame(frame);

        av_frame_unref(frame);
    }

    return ret;
}

static int open_codec_context(int *stream_idx,
                              AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
{
    int ret, stream_index;
    AVStream *st;
    const AVCodec *dec = NULL;

    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
                av_get_media_type_string(type), src_filename);
        return ret;
    } else {
        stream_index = ret;
        st = fmt_ctx->streams[stream_index];

        /* find decoder for the stream */
        dec = avcodec_find_decoder(st->codecpar->codec_id);
        if (!dec) {
            fprintf(stderr, "Failed to find %s codec\n",
                    av_get_media_type_string(type));
            return AVERROR(EINVAL);
        }

        /* Allocate a codec context for the decoder */
        *dec_ctx = avcodec_alloc_context3(dec);
        if (!*dec_ctx) {
            fprintf(stderr, "Failed to allocate the %s codec context\n",
                    av_get_media_type_string(type));
            return AVERROR(ENOMEM);
        }

        /* Copy codec parameters from input stream to output codec context */
        if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
            fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
                    av_get_media_type_string(type));
            return ret;
        }
        // 明确输出类型
        (*dec_ctx)->request_sample_fmt = (*dec_ctx)->sample_fmt == AV_SAMPLE_FMT_S16P ? AV_SAMPLE_FMT_S16 : (*dec_ctx)->sample_fmt;
        (*dec_ctx)->request_sample_fmt = (*dec_ctx)->sample_fmt == AV_SAMPLE_FMT_S32P ? AV_SAMPLE_FMT_S32 : (*dec_ctx)->sample_fmt;
        /* Init the decoders */
        if ((ret = avcodec_open2(*dec_ctx, dec, NULL)) < 0) {
            fprintf(stderr, "Failed to open %s codec\n",
                    av_get_media_type_string(type));
            return ret;
        }
        *stream_idx = stream_index;
    }

    return 0;
}

static int get_format_from_sample_fmt(const char **fmt,
                                      enum AVSampleFormat sample_fmt)
{
    int i;
    struct sample_fmt_entry {
        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
    } sample_fmt_entries[] = {
        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },
        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },
        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },
        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
    };
    *fmt = NULL;

    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
        struct sample_fmt_entry *entry = &sample_fmt_entries[i];
        if (sample_fmt == entry->sample_fmt) {
            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);
            return 0;
        }
    }

    fprintf(stderr,
            "sample format %s is not supported as output format\n",
            av_get_sample_fmt_name(sample_fmt));
    return -1;
}

int main (int argc, char **argv)
{
    int ret = 0;

    // if (argc != 4) {
    //     fprintf(stderr, "usage: %s  input_file video_output_file audio_output_file\n"
    //             "API example program to show how to read frames from an input file.\n"
    //             "This program reads frames from a file, decodes them, and writes decoded\n"
    //             "video frames to a rawvideo file named video_output_file, and decoded\n"
    //             "audio frames to a rawaudio file named audio_output_file.\n",
    //             argv[0]);
    //     exit(1);
    // }
    src_filename = argv[1];
    audio_dst_filename = argv[2];

    /* 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) {
        fprintf(stderr, "Could not find stream information\n");
        exit(1);
    }


    if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
        audio_stream = fmt_ctx->streams[audio_stream_idx];
        // audio_dst_file = fopen(audio_dst_filename, "wb");
        // if (!audio_dst_file) {
        //     fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
        //     ret = 1;
        //     goto end;
        // }
    }
    printf("nb_streams %d\n", fmt_ctx->nb_streams);
    /* dump input information to stderr */
    // av_dump_format(fmt_ctx, 0, src_filename, 0);

    if (!audio_stream) {
        fprintf(stderr, "Could not find audio stream in the input, aborting\n");
        ret = 1;
        goto end;
    }

    frame = av_frame_alloc();
    if (!frame) {
        fprintf(stderr, "Could not allocate frame\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }

    pkt = av_packet_alloc();
    if (!pkt) {
        fprintf(stderr, "Could not allocate packet\n");
        ret = AVERROR(ENOMEM);
        goto end;
    }

    if (audio_stream)
        printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);

    /* read frames from the file */
    while (av_read_frame(fmt_ctx, pkt) >= 0) 
    {
        // check if the packet belongs to a stream we are interested in, otherwise
        // skip it
        if (pkt->stream_index == audio_stream_idx)
            ret = decode_packet(audio_dec_ctx, pkt);
        av_packet_unref(pkt);
        if (ret < 0)
            break;
    }

    /* flush the decoders */

    if (audio_dec_ctx)
        decode_packet(audio_dec_ctx, NULL);

    printf("Demuxing succeeded.\n");

    if (audio_stream) {
        enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
        int n_channels = audio_dec_ctx->ch_layout.nb_channels;
        const char *fmt;

        if (av_sample_fmt_is_planar(sfmt)) {
            const char *packed = av_get_sample_fmt_name(sfmt);
            printf("Warning: the sample format the decoder produced is planar "
                   "(%s). This example will output the first channel only.\n",
                   packed ? packed : "?");
            sfmt = av_get_packed_sample_fmt(sfmt);
            n_channels = 1;
        }

        if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
            goto end;

        printf("Play the output audio file with the command:\n"
               "ffplay -f %s -ac %d -ar %d %s\n",
               fmt, n_channels, audio_dec_ctx->sample_rate,
               audio_dst_filename);
    }

end:
    avcodec_free_context(&audio_dec_ctx);
    avformat_close_input(&fmt_ctx);

    if (audio_dst_file)
        fclose(audio_dst_file);
    av_packet_free(&pkt);
    av_frame_free(&frame);

    return ret < 0;
}

// 补全依赖

static char pbuffer[4096];
void av_log(void *avcl, int level, const char *fmt, ...)
{
    va_list args;

    va_start(args, fmt);
    vsnprintf(pbuffer, 4096, fmt, args);
    va_end(args);

    puts(pbuffer);
}
int av_log_get_level(void)
{
    return 0;
}

int64_t av_gettime_relative(void)
{
    return 0;
}
int av_usleep(unsigned usec)
{
    return 0;
}

int64_t av_gettime(void)
{
    return 0;
}
const char *av_default_item_name(void *ptr)
{
    return (*(AVClass **)ptr)->class_name;
}

uint32_t av_get_random_seed()
{
    return 0;
}