#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
extern"C"
{
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/avutil.h>
#include <libswscale/swscale.h>
#include <libavdevice/avdevice.h>
}

int main() {
avdevice_register_all();

constAVInputFormat*input_format=av_find_input_format("alsa");
AVDeviceInfoList*device_list=NULL;

// Get the list of input sources
if (avdevice_list_input_sources(input_format, NULL, NULL, &device_list) >=0) {
// Iterate through the device list
for (inti=0; i<device_list->nb_devices; i++) {
AVDeviceInfo*device_info=device_list->devices[i];
printf("Device %d: %s%s\n", i, device_info->device_name, device_info->device_description);
}

// Free the device list
avdevice_free_list_devices(&device_list);
}
return0;
}

#include <stdio.h>
#include "libavutil/avutil.h"
#include "libavdevice/avdevice.h"  //打开音频设备相关的头文件
#include "libavformat/avformat.h"  //ffmpeg下的所有文件都是以格式来呈现的


void main(int argc, char **argv)
{
    int ret = 0;
    char errors[1024] = {0};
    //context
    AVFormatContext *fmt_ctx = NULL;   //ffmpeg下的“文件描述符”

    //paket
    int count = 0;
    AVPacket pkt;

    //create file
    char *out = "./audio_old.pcm";
    FILE *outfile = fopen(out,"wb+");

    char *devicename = "hw:1,0";
    //register audio device
    avdevice_register_all();

    //get format
    AVInputFormat *iformat = av_find_input_format("alsa");

    //open audio
    if( (ret = avformat_open_input(&fmt_ctx, devicename, iformat, NULL)) < 0)
    {
        av_strerror(ret, errors, 1024);
        printf("Failed to open audio device, [%d]%s\n", ret, errors);
        return;
    };

    av_init_packet(&pkt);
    //read data form audio
    while(ret = (av_read_frame(fmt_ctx, &pkt))== 0&& 
        count++ < 1000) {
        av_log(NULL, AV_LOG_INFO, "pkt size is %d(%p), count=%d\n",
            pkt.size,pkt.data, count);
        fwrite(pkt.data, 1, pkt.size, outfile);
        fflush(outfile);
        av_packet_unref(&pkt);//release pkt
    }

    fclose(outfile);
    avformat_close_input(&fmt_ctx);//releas ctx
    return;
}

#include <stdio.h>
#include <stdlib.h>
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
 
// 只编码，不重采样
void audio_encode(const char *pcmfile, const char *aacfile) {
    FILE *sfp = fopen(pcmfile, "r");
    if (sfp == NULL) {
        printf("fail to open pcmile\n");
        goto _Error;
    }
    FILE *dfp = fopen(aacfile, "w+");
    if (dfp == NULL) {
        printf("failed to open aacfile\n");
        goto _Error;
    }
    
    // 1. 初始化pcm_frame，保存每帧pcm数据
    AVFrame *pcm_frame;
    pcm_frame = av_frame_alloc();
    pcm_frame->format = AV_SAMPLE_FMT_S16;
    pcm_frame->channel_layout = AV_CH_LAYOUT_STEREO;
    pcm_frame->sample_rate = 44100;
    pcm_frame->channels = 2;
    // nb_samples的值与具体的编码协议有关
    pcm_frame->nb_samples = 1024;    // 一帧中有多少个采样点
    av_frame_get_buffer(pcm_frame, 0);
    
    // 2. 找到合适的编码器
    AVCodec *cod = avcodec_find_encoder_by_name("libfdk_aac");
    if (cod == NULL) {
        av_log(NULL, AV_LOG_ERROR, "fail to find codec\n");
        goto _Error;
    }
    
    // 3. 设置其编码选项
    AVCodecContext *cod_ctx = avcodec_alloc_context3(cod);
    //cod_ctx->profile = FF_PROFILE_AAC_HE_V2;        // 编码协议
    cod_ctx->codec_type = AVMEDIA_TYPE_AUDIO;       // 音频编码
    cod_ctx->sample_fmt = pcm_frame->format;        
    cod_ctx->channel_layout = pcm_frame->channel_layout;
    cod_ctx->channels = pcm_frame->channels;
    cod_ctx->sample_rate = pcm_frame->sample_rate;  // 采样率
 
 
    // 4. 打开编码器
    if (avcodec_open2(cod_ctx, cod, NULL) < 0) {
        av_log(NULL, AV_LOG_ERROR, "fail to open codec\n");
        goto _Error;
    }
    printf("frame_size:%d\n", cod_ctx->frame_size); 
 
    // 5. 初始化packet
    int count = 0;
    AVPacket *packet = av_packet_alloc();
    int frame_size = pcm_frame->nb_samples * \
            av_get_bytes_per_sample(pcm_frame->format) * pcm_frame->channels; 
 
    while (1) {
        // AV_SAMPLE_FMT_S16是packed格式的, 声道数据LRLRLRLRLR...
        // 所以pcm_frame->data[]是一维的
        int ret = fread(pcm_frame->data[0], 1, frame_size, sfp);
        if (ret < 0) {
            printf("fail to read raw data\n");
            goto _Error;
        } else if (ret == 0) {
            break;
        }
        pcm_frame->pts = count;
 
        // 原始数据发送到编码器
        if (avcodec_send_frame(cod_ctx, pcm_frame) < 0) {
            printf("fail to send frame\n");
            goto _Error;
        }
 
        // 获取编码数据
        if (avcodec_receive_packet(cod_ctx, packet) >= 0) {
            fwrite(packet->data, 1, packet->size, dfp);
            count++;
            av_packet_unref(packet);
            printf("receive %d frame\n", count);
        }
    }
 
_Error:
    if (sfp) {
        fclose(sfp);
    }
    if (dfp) {
        fclose(dfp);
    }
    if (cod_ctx) {
        avcodec_close(cod_ctx);
        avcodec_free_context(&cod_ctx);
    }
    if (pcm_frame) {
        av_frame_free(&pcm_frame);
    }
    if (packet) {
        av_packet_free(&packet);
    }
}
 
 
int main(int argc, char const* argv[])
{
    audio_encode("./test.pcm", "./output_test.aac");
    return 0;
}

#include <stdio.h>
#include <stdlib.h>
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
 
// 只编码，不重采样
void audio_encode(const char *pcmfile, const char *aacfile, const char *originfile) {
    FILE *pcmfp = fopen(pcmfile, "w+");
    if (pcmfp == NULL) {
        printf("failed to open pcmfile\n");
        goto _Error;
    }

    FILE *dfp = fopen(aacfile, "w+");
    if (dfp == NULL) {
        printf("failed to open aacfile\n");
        goto _Error;
    }

    FILE *originfp = fopen(originfile, "w+");
    if (originfp == NULL) {
        printf("failed to open pcmfile\n");
        goto _Error;
    }
    
    // 1. 初始化pcm_frame，保存每帧pcm数据
    AVFrame *pcm_frame;
    pcm_frame = av_frame_alloc();
    pcm_frame->format = AV_SAMPLE_FMT_S16;
    pcm_frame->channel_layout = AV_CH_LAYOUT_STEREO;
    pcm_frame->sample_rate = 44100;
    pcm_frame->channels = 2;
    // nb_samples的值与具体的编码协议有关
    pcm_frame->nb_samples = 1024;    // 一帧中有多少个采样点
    av_frame_get_buffer(pcm_frame, 0);
    
    // 2. 找到合适的编码器
    AVCodec *cod = avcodec_find_encoder_by_name("libfdk_aac");
    if (cod == NULL) {
        av_log(NULL, AV_LOG_ERROR, "fail to find codec\n");
        goto _Error;
    }
    
    // 3. 设置其编码选项
    AVCodecContext *cod_ctx = avcodec_alloc_context3(cod);
    //cod_ctx->profile = FF_PROFILE_AAC_HE_V2;        // 编码协议
    cod_ctx->codec_type = AVMEDIA_TYPE_AUDIO;       // 音频编码
    cod_ctx->sample_fmt = pcm_frame->format;        
    cod_ctx->channel_layout = pcm_frame->channel_layout;
    cod_ctx->channels = pcm_frame->channels;
    cod_ctx->sample_rate = pcm_frame->sample_rate;  // 采样率
 
 
    // 4. 打开编码器
    if (avcodec_open2(cod_ctx, cod, NULL) < 0) {
        av_log(NULL, AV_LOG_ERROR, "fail to open codec\n");
        goto _Error;
    }
    printf("frame_size:%d\n", cod_ctx->frame_size); 
 
    // 5. 初始化packet
    int count = 0;
    AVPacket *packet = av_packet_alloc();
    int frame_size = pcm_frame->nb_samples * \
            av_get_bytes_per_sample(pcm_frame->format) * pcm_frame->channels; 
 
     // 6 注册设备
    avdevice_register_all();

    char *devicename = "hw:2,0";

    // 获取输入(采集）格式
    // AVInputFormat在FFmpeg中的作用是表示输入格式的数据结构，它包含了输入格式的相关信息，如名称、扩展名、MIME类型、解析函数
    AVInputFormat * iformat = av_find_input_format("alsa");  //通过使用AVInputFormat，可以将特定格式的音视频文件读取到内存中，并对其进行解码、转码等操作
    //打开输入设备
    AVFormatContext* fmt_ctx=NULL; //AVFormatContext是FFmpeg中表示音视频格式的数据结构，它包含了音视频文件的所有信息，如文件名、流信息、时长、元数据
   int ret = avformat_open_input(&fmt_ctx, devicename, iformat, NULL);
    char errors[1024];
    if(ret < 0){
        av_strerror(ret, errors, 1024);
        av_log(NULL,AV_LOG_ERROR,"Failed to open audio device,[%d]%s\n",ret,errors);
    }
    int buf_count = 0;
    int count_in = 0;
    AVPacket in_pkt;
    int pcm_buffer_size = 4096;
    while (count_in++ < 5000) {
        // AV_SAMPLE_FMT_S16是packed格式的, 声道数据LRLRLRLRLR...
        // 所以pcm_frame->data[]是一维的
        //ret = fread(pcm_frame->data[0], 1, frame_size, sfp);
        ret = av_read_frame(fmt_ctx, &in_pkt);

        if (buf_count < pcm_buffer_size){
            printf("[%d][%d][%d]\n", pcm_buffer_size, count_in, in_pkt.size);
            memcpy(pcm_frame->data[0] + buf_count, in_pkt.data, in_pkt.size);
            fwrite(in_pkt.data, 1, in_pkt.size, originfp);
            buf_count += in_pkt.size;
            continue;
        }

        if (buf_count == pcm_buffer_size){
            buf_count = 0;
        }
        fwrite(pcm_frame->data[0], 1, pcm_buffer_size, pcmfp);

        if (ret < 0) {
            printf("fail to read raw data\n");
            goto _Error;
        } 
        // else if (ret == 0) {
        //     printf("ret == 0\n");
        //     break;
        // }
        pcm_frame->pts = count;
 
        // 原始数据发送到编码器
        if (avcodec_send_frame(cod_ctx, pcm_frame) < 0) {
            printf("fail to send frame\n");
            goto _Error;
        }
 
        // 获取编码数据
        if (avcodec_receive_packet(cod_ctx, packet) >= 0) {
            fwrite(packet->data, 1, packet->size, dfp);
            count++;
            av_packet_unref(packet);
            printf("receive %d frame\n", count);
        }
        av_packet_unref(&in_pkt);//release pkt
    }
 
_Error:
    fflush(pcmfp);
    if (pcmfp){
        fclose(pcmfp);
    }
    fflush(originfp);
    if (originfp){
        fclose(originfp);
    }

    if (dfp) {
        fclose(dfp);
    }
    if (cod_ctx) {
        avcodec_close(cod_ctx);
        avcodec_free_context(&cod_ctx);
    }
    if (pcm_frame) {
        av_frame_free(&pcm_frame);
    }
    if (packet) {
        av_packet_free(&packet);
    }
}
 
 
int main(int argc, char const* argv[])
{
    audio_encode("./encode_acc.pcm", "./encode_aac.aac", "origin.pcm");
    return 0;
}