基本流程
函数说明
avcodec_find_encoder：根据指定的AVCodecID查找注册的编码器。avcodec_alloc_context3：为AVCodecContext分配内存。avcodec_open2：打开编码器。avcodec_send_frame：将AVFrame⾮压缩数据给编码器。avcodec_receive_packet：获取到编码后的AVPacket数据，收到的packet需要⾃⼰释放内存。av_frame_get_buffer:为⾳频或视频帧分配新的buffer。在调⽤这个函数之前，必须在AVFame上设置好以下属性：format(视频为像素格式，⾳频为样本格式)、nb_samples(样本个数，针对⾳频)、channel_layout(通道类型，针对⾳频)、width/height(宽⾼，针对视频）。av_frame_make_writable：确保AVFrame是可写的，使⽤av_frame_make_writable()的问题是，在最坏的情况下，它会在您使⽤encode再次更改整个输⼊frame之前复制它. 如果frame不可写，av_frame_make_writable()将分配新的缓冲区，并复制这个输⼊input frame数据，避免和编码器需要缓存该帧时造成冲突。av_samples_fill_arrays填充⾳频帧
编码
/*** @projectName 08-01-encode_audio* @brief 音频编码*从本地读取PCM数据进行AAC编码* 1. 输入PCM格式问题，通过AVCodec的sample_fmts参数获取具体的格式支持* （1）默认的aac编码器输入的PCM格式为:AV_SAMPLE_FMT_FLTP* （2）libfdk_aac编码器输入的PCM格式为AV_SAMPLE_FMT_S16.* 2. 支持的采样率，通过AVCodec的supported_samplerates可以获取* @author Liao Qingfu* @date-04-15*/#include <stdint.h>#include <stdio.h>#include <stdlib.h>#include <libavcodec/avcodec.h>#include <libavutil/channel_layout.h>#include <libavutil/common.h>#include <libavutil/frame.h>#include <libavutil/samplefmt.h>#include <libavutil/opt.h>/* 检测该编码器是否支持该采样格式 */static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt){const enum AVSampleFormat *p = codec->sample_fmts;while (*p != AV_SAMPLE_FMT_NONE) { // 通过AV_SAMPLE_FMT_NONE作为结束符if (*p == sample_fmt)return 1;p++;}return 0;}/* 检测该编码器是否支持该采样率 */static int check_sample_rate(const AVCodec *codec, const int sample_rate){const int *p = codec->supported_samplerates;while (*p != 0) {// 0作为退出条件，比如libfdk-aacenc.c的aac_sample_ratesprintf("%s support %dhz\n", codec->name, *p);if (*p == sample_rate)return 1;p++;}return 0;}/* 检测该编码器是否支持该采样率, 该函数只是作参考 */static int check_channel_layout(const AVCodec *codec, const uint64_t channel_layout){// 不是每个codec都给出支持的channel_layoutconst uint64_t *p = codec->channel_layouts;if(!p) {printf("the codec %s no set channel_layouts\n", codec->name);return 1;}while (*p != 0) { // 0作为退出条件，比如libfdk-aacenc.c的aac_channel_layoutprintf("%s support channel_layout %d\n", codec->name, *p);if (*p == channel_layout)return 1;p++;}return 0;}static int check_codec( AVCodec *codec, AVCodecContext *codec_ctx){if (!check_sample_fmt(codec, codec_ctx->sample_fmt)) {fprintf(stderr, "Encoder does not support sample format %s",av_get_sample_fmt_name(codec_ctx->sample_fmt));return 0;}if (!check_sample_rate(codec, codec_ctx->sample_rate)) {fprintf(stderr, "Encoder does not support sample rate %d", codec_ctx->sample_rate);return 0;}if (!check_channel_layout(codec, codec_ctx->channel_layout)) {fprintf(stderr, "Encoder does not support channel layout %lu", codec_ctx->channel_layout);return 0;}printf("\n\nAudio encode config\n");printf("bit_rate:%ldkbps\n", codec_ctx->bit_rate/1024);printf("sample_rate:%d\n", codec_ctx->sample_rate);printf("sample_fmt:%s\n", av_get_sample_fmt_name(codec_ctx->sample_fmt));printf("channels:%d\n", codec_ctx->channels);// frame_size是在avcodec_open2后进行关联printf("1 frame_size:%d\n", codec_ctx->frame_size);return 1;}static void get_adts_header(AVCodecContext *ctx, uint8_t *adts_header, int aac_length){uint8_t freq_idx = 0; //0: 96000 Hz 3: 48000 Hz 4: 44100 Hzswitch (ctx->sample_rate) {case 96000: freq_idx = 0; break;case 88200: freq_idx = 1; break;case 64000: freq_idx = 2; break;case 48000: freq_idx = 3; break;case 44100: freq_idx = 4; break;case 32000: freq_idx = 5; break;case 24000: freq_idx = 6; break;case 22050: freq_idx = 7; break;case 16000: freq_idx = 8; break;case 12000: freq_idx = 9; break;case 11025: freq_idx = 10; break;case 8000: freq_idx = 11; break;case 7350: freq_idx = 12; break;default: freq_idx = 4; break;}uint8_t chanCfg = ctx->channels;uint32_t frame_length = aac_length + 7;adts_header[0] = 0xFF;adts_header[1] = 0xF1;adts_header[2] = ((ctx->profile) << 6) + (freq_idx << 2) + (chanCfg >> 2);adts_header[3] = (((chanCfg & 3) << 6) + (frame_length >> 11));adts_header[4] = ((frame_length & 0x7FF) >> 3);adts_header[5] = (((frame_length & 7) << 5) + 0x1F);adts_header[6] = 0xFC;}/***/static int encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt, FILE *output){int ret;/* send the frame for encoding */ret = avcodec_send_frame(ctx, frame);if (ret < 0) {fprintf(stderr, "Error sending the frame to the encoder\n");return -1;}/* read all the available output packets (in general there may be any number of them */// 编码和解码都是一样的，都是send 1次，然后receive多次, 直到AVERROR(EAGAIN)或者AVERROR_EOFwhile (ret >= 0) {ret = avcodec_receive_packet(ctx, pkt);if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {return 0;} else if (ret < 0) {fprintf(stderr, "Error encoding audio frame\n");return -1;}uint8_t aac_header[7];get_adts_header(ctx, aac_header, pkt->size);size_t len = 0;len = fwrite(aac_header, 1, 7, output);if(len != 7) {fprintf(stderr, "fwrite aac_header failed\n");return -1;}len = fwrite(pkt->data, 1, pkt->size, output);if(len != pkt->size) {fprintf(stderr, "fwrite aac data failed\n");return -1;}/* 是否需要释放数据? avcodec_receive_packet第一个调用的就是 av_packet_unref* 所以我们不用手动去释放，这里有个问题，不能将pkt直接插入到队列，因为编码器会释放数据* 可以新分配一个pkt, 然后使用av_packet_move_ref转移pkt对应的buffer*/// av_packet_unref(pkt);}return -1;}/** 这里只支持2通道的转换*/void f32le_convert_to_fltp(float *f32le, float *fltp, int nb_samples) {float *fltp_l = fltp; // 左通道float *fltp_r = fltp + nb_samples; // 右通道for(int i = 0; i < nb_samples; i++) {fltp_l[i] = f32le[i*2];// 0 1 - 2 3fltp_r[i] = f32le[i*2+1]; // 可以尝试注释左声道或者右声道听听声音}}/** 提取测试文件：* （1）s16格式：ffmpeg -i buweishui.aac -ar 48000 -ac 2 -f s16le 48000_2_s16le.pcm* （2）flt格式：ffmpeg -i buweishui.aac -ar 48000 -ac 2 -f f32le 48000_2_f32le.pcm*ffmpeg只能提取packed格式的PCM数据，在编码时候如果输入要为fltp则需要进行转换* 测试范例:* （1）48000_2_s16le.pcm libfdk_aac.aac libfdk_aac // 如果编译的时候没有支持fdk aac则提示找不到编码器* （2）48000_2_f32le.pcm aac.aac aac // 我们这里只测试aac编码器，不测试fdkaac*/int main(int argc, char **argv){const char* in_pcm_file = "48000_2_f32le.pcm";// 输入PCM文件const char* out_aac_file = "f32.aac";// 输出的AAC文件enum AVCodecID codec_id = AV_CODEC_ID_AAC;// 1.查找编码器AVCodec *codec = avcodec_find_encoder(codec_id); // 按ID查找则缺省的aac encode为aacenc.cif (!codec) {fprintf(stderr, "Codec not found\n");exit(1);}// 2.分配内存AVCodecContext *codec_ctx = avcodec_alloc_context3(codec);if (!codec_ctx) {fprintf(stderr, "Could not allocate audio codec context\n");exit(1);}codec_ctx->codec_id = codec_id;codec_ctx->codec_type = AVMEDIA_TYPE_AUDIO;codec_ctx->bit_rate = 128*1024;codec_ctx->channel_layout = AV_CH_LAYOUT_STEREO;codec_ctx->sample_rate = 48000; //48000;codec_ctx->channels = av_get_channel_layout_nb_channels(codec_ctx->channel_layout);codec_ctx->profile = FF_PROFILE_AAC_LOW; //codec_ctx->sample_fmt = AV_SAMPLE_FMT_FLTP;// 3.检测支持采样格式支持情况if (!check_codec(codec, codec_ctx)) {exit(1);}// 4.将编码器上下文和编码器进行关联if (avcodec_open2(codec_ctx, codec, NULL) < 0) {fprintf(stderr, "Could not open codec\n");exit(1);}printf("2 frame_size:%d\n\n", codec_ctx->frame_size); // 决定每次到底送多少个采样点// 5.打开输入和输出文件FILE *infile = fopen(in_pcm_file, "rb");if (!infile) {fprintf(stderr, "Could not open %s\n", in_pcm_file);exit(1);}FILE *outfile = fopen(out_aac_file, "wb");if (!outfile) {fprintf(stderr, "Could not open %s\n", out_aac_file);exit(1);}// 6.分配packetAVPacket *pkt = av_packet_alloc();if (!pkt) {fprintf(stderr, "could not allocate the packet\n");exit(1);}// 7.分配frameAVFrame *frame = av_frame_alloc();if (!frame) {fprintf(stderr, "Could not allocate audio frame\n");exit(1);}/* 每次送多少数据给编码器由：* (1)frame_size(每帧单个通道的采样点数);* (2)sample_fmt(采样点格式);* (3)channel_layout(通道布局情况);* 3要素决定*/frame->nb_samples= codec_ctx->frame_size;frame->format = codec_ctx->sample_fmt;frame->channel_layout = codec_ctx->channel_layout;frame->channels = av_get_channel_layout_nb_channels(frame->channel_layout);printf("frame nb_samples:%d\n", frame->nb_samples);printf("frame sample_fmt:%d\n", frame->format);printf("frame channel_layout:%lu\n\n", frame->channel_layout);// 8.为frame分配bufferint ret = av_frame_get_buffer(frame, 0);if (ret < 0) {fprintf(stderr, "Could not allocate audio data buffers\n");exit(1);}// 9.循环读取数据// 计算出每一帧的数据 单个采样点的字节 * 通道数目 * 每帧采样点数量int frame_bytes = av_get_bytes_per_sample(frame->format) \* frame->channels \* frame->nb_samples;printf("frame_bytes %d\n", frame_bytes);uint8_t *pcm_buf = (uint8_t *)malloc(frame_bytes);if(!pcm_buf) {printf("pcm_buf malloc failed\n");return 1;}uint8_t *pcm_temp_buf = (uint8_t *)malloc(frame_bytes);if(!pcm_temp_buf) {printf("pcm_temp_buf malloc failed\n");return 1;}int64_t pts = 0;printf("start enode\n");for (;;) {memset(pcm_buf, 0, frame_bytes);size_t read_bytes = fread(pcm_buf, 1, frame_bytes, infile);if(read_bytes <= 0) {printf("read file finish\n");break;}// 10.确保该frame可写, 如果编码器内部保持了内存参考计数，则需要重新拷贝一个备份 目的是新写入的数据和编码器保存的数据不能产生冲突ret = av_frame_make_writable(frame);if(ret != 0)printf("av_frame_make_writable failed, ret = %d\n", ret);// 11.填充音频帧if(AV_SAMPLE_FMT_S16 == frame->format) {// 将读取到的PCM数据填充到frame去，但要注意格式的匹配, 是planar还是packed都要区分清楚ret = av_samples_fill_arrays(frame->data, frame->linesize,pcm_buf, frame->channels,frame->nb_samples, frame->format, 0);} else {// 将读取到的PCM数据填充到frame去，但要注意格式的匹配, 是planar还是packed都要区分清楚// 将本地的f32le packed模式的数据转为float palanarmemset(pcm_temp_buf, 0, frame_bytes);f32le_convert_to_fltp((float *)pcm_buf, (float *)pcm_temp_buf, frame->nb_samples);ret = av_samples_fill_arrays(frame->data, frame->linesize,pcm_temp_buf, frame->channels,frame->nb_samples, frame->format, 0);}// 12.编码pts += frame->nb_samples;frame->pts = pts; // 使用采样率作为pts的单位，具体换算成秒 pts*1/采样率ret = encode(codec_ctx, frame, pkt, outfile);if(ret < 0) {printf("encode failed\n");break;}}// 13.冲刷编码器encode(codec_ctx, NULL, pkt, outfile);// 14.关闭文件fclose(infile);fclose(outfile);// 15.释放内存if(pcm_buf) {free(pcm_buf);}if (pcm_temp_buf) {free(pcm_temp_buf);}av_frame_free(&frame);av_packet_free(&pkt);avcodec_free_context(&codec_ctx);printf("main finish, please enter Enter and exit\n");getchar();return 0;}
