[ffmpeg]How to use libfdk_aac to encode pcm to constant bitrate

[xengine-qyt]

#include <stdio.h>
#include <stdlib.h>
#include <windows.h>
extern "C"
{
#include <libavcodec/avcodec.h>
#include <libavdevice/avdevice.h>
#include <libavfilter/avfilter.h>
#include <libavfilter/buffersrc.h>
#include <libavfilter/buffersink.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include <libavutil/imgutils.h>
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>   
#include <libswresample/swresample.h>
#include <libswscale/swscale.h>
}

int main(int argc, char** argv) {
    AVCodecContext* codec_context = NULL;
    const AVCodec* codec = NULL;
    AVFormatContext* format_context = NULL;
    AVStream* audio_stream = NULL;
    AVFrame* frame = NULL;
    AVPacket* pkt = NULL;
    FILE* input_file = NULL;
    int ret;
    int64_t next_pts = 0;

    // Open input file
    const char* input_filename = "D:\\audio\\b.pcm";
    const char* output_filename = "D:\\audio\\input.mp4";
    input_file = fopen(input_filename, "rb");
    if (!input_file) {
        fprintf(stderr, "Could not open input file\n");
        exit(1);
    }

    // Find the AAC encoder
    codec = avcodec_find_encoder_by_name("libfdk_aac");
    if (!codec) {
        fprintf(stderr, "Codec not found\n");
        exit(1);
    }

    // Allocate AVFormatContext for MP4 output
    avformat_alloc_output_context2(&format_context, NULL, NULL, output_filename);
    if (!format_context) {
        fprintf(stderr, "Could not allocate output context\n");
        exit(1);
    }

    // Add audio stream to the output file
    audio_stream = avformat_new_stream(format_context, codec);
    if (!audio_stream) {
        fprintf(stderr, "Could not allocate stream\n");
        exit(1);
    }

    codec_context = avcodec_alloc_context3(codec);
    if (!codec_context) {
        fprintf(stderr, "Could not allocate audio codec context\n");
        exit(1);
    }
    audio_stream->codecpar->frame_size = codec_context->frame_size = 1024;

    // Set codec parameters
    codec_context->sample_fmt = AV_SAMPLE_FMT_S16;
    codec_context->sample_rate = 44100;
    codec_context->bit_rate = 256000;
    codec_context->rc_buffer_size = codec_context->bit_rate;
    codec_context->rc_min_rate = codec_context->bit_rate;
    codec_context->rc_max_rate = codec_context->bit_rate;
    av_channel_layout_default(&codec_context->ch_layout, 2);

    // Copy settings to stream
    ret = avcodec_parameters_from_context(audio_stream->codecpar, codec_context);
    if (ret < 0) {
        fprintf(stderr, "Failed to copy codec parameters to stream\n");
        exit(1);
    }

    // Open codec
    if (avcodec_open2(codec_context, codec, NULL) < 0) {
        fprintf(stderr, "Could not open codec\n");
        exit(1);
    }

    // Open output file
    if (!(format_context->oformat->flags & AVFMT_NOFILE)) {
        ret = avio_open(&format_context->pb, output_filename, AVIO_FLAG_WRITE);
        if (ret < 0) {
            fprintf(stderr, "Could not open output file '%s'\n", output_filename);
            exit(1);
        }
    }

    // Write file header
    ret = avformat_write_header(format_context, NULL);
    if (ret < 0) {
        fprintf(stderr, "Error occurred when opening output file\n");
        exit(1);
    }

    // Initialize packet and frame
    pkt = av_packet_alloc();
    if (!pkt) {
        fprintf(stderr, "Could not allocate AVPacket\n");
        exit(1);
    }

    frame = av_frame_alloc();
    frame->nb_samples = codec_context->frame_size;
    frame->format = codec_context->sample_fmt;
    frame->ch_layout = codec_context->ch_layout;

    // Allocate the data buffers
    ret = av_frame_get_buffer(frame, 0);
    if (ret < 0) {
        fprintf(stderr, "Could not allocate audio data buffers\n");
        exit(1);
    }

    // Main loop: read from the input file, encode, write to the output file
    while (fread(frame->data[0], 1, frame->linesize[0], input_file) == frame->linesize[0]) {
        frame->pts = next_pts;  // Set PTS for the frame
        next_pts += frame->nb_samples;  // Increment the next PTS
        // Send the frame to the encoder
        if (avcodec_send_frame(codec_context, frame) < 0) {
            fprintf(stderr, "Error sending frame to codec\n");
            exit(1);
        }

        // Get the encoded packet
        while (avcodec_receive_packet(codec_context, pkt) == 0) {
            pkt->pts = pkt->dts = frame->pts;
            av_packet_rescale_ts(pkt, codec_context->time_base, audio_stream->time_base);
            pkt->stream_index = audio_stream->index;
            av_interleaved_write_frame(format_context, pkt);
            av_packet_unref(pkt);
        }
    }

    // Flush the encoder
    avcodec_send_frame(codec_context, NULL);
    while (avcodec_receive_packet(codec_context, pkt) == 0) {
        pkt->pts = pkt->dts = next_pts;
        av_packet_rescale_ts(pkt, codec_context->time_base, audio_stream->time_base);
        pkt->stream_index = audio_stream->index;
        av_interleaved_write_frame(format_context, pkt);
        av_packet_unref(pkt);
        next_pts += pkt->duration;
    }

    // Write file trailer
    av_write_trailer(format_context);

    // Clean up
    fclose(input_file);
    av_frame_free(&frame);
    av_packet_free(&pkt);
    avcodec_free_context(&codec_context);
    avio_closep(&format_context->pb);
    avformat_free_context(format_context);

    return 0;
}

It can be seen through mediainfo that it is dynamic, but the command rotation of ffmepg is constant. This is why
pcm is s16 2 44100
https://drive.google.com/file/d/1Udrtpljmu_7p6bFd2TxYr04iFGeZ6UTE/view