FFmpeg PCM编码为AAC

使用FFmpeg库把PCM文件编码为AAC文件，FFmpeg版本为4.4.2-0

代码如下：

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <libavutil/mem.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>int main(int argc, char *argv[])
{int ret = -1;int64_t pts = 0;const char *input_file = argv[1];const char *output_file = argv[2];FILE *input_pcm = NULL;AVFormatContext *format_context = NULL;AVCodecContext *codec_context = NULL;AVStream *stream = NULL;AVCodec *codec = NULL;AVFrame *frame = NULL;AVPacket *packet = NULL;struct SwrContext *swr_ctx = NULL;enum AVSampleFormat in_sample_fmt = AV_SAMPLE_FMT_S16; // 输入pcm文件的采样格式int in_sample_rate = 44100;                            // 输入pcm文件的采样率uint64_t in_channel_layout = AV_CH_LAYOUT_STEREO;      // 输入pcm文件的声道布局if (argc < 3){fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);return -1;}input_pcm = fopen(input_file, "rb");if (!input_pcm){fprintf(stderr, "Could not open input file %s\n", input_file);goto end;}// 分配输出格式上下文avformat_alloc_output_context2(&format_context, NULL, NULL, output_file);if (!format_context){fprintf(stderr, "Could not allocate output format context\n");goto end;}// 查找编码器, 这里使用ffmpeg内置的aac编码器,// 如果需要使用其他编码器, 可以使用 avcodec_find_encoder_by_name 按名称查找codec = avcodec_find_encoder(AV_CODEC_ID_AAC);if (!codec){fprintf(stderr, "Codec not found\n");goto end;}// 创建新的音频流stream = avformat_new_stream(format_context, NULL);if (!stream){fprintf(stderr, "Could not allocate stream\n");goto end;}// 分配编码器上下文codec_context = avcodec_alloc_context3(codec);if (!codec_context){fprintf(stderr, "Could not allocate codec context\n");goto end;}/* 设置编码器参数 */// 编码器IDcodec_context->codec_id = AV_CODEC_ID_AAC;// 媒体类型codec_context->codec_type = AVMEDIA_TYPE_AUDIO;// 设置编码器接收的输入音频样本的采样格式，这里设置为内置aac编码器支持的采样格式，// 不同的编码器支持的采样格式可能不同。codec_context->sample_fmt = AV_SAMPLE_FMT_FLTP;// 设置采样率codec_context->sample_rate = in_sample_rate;// 设置声道布局codec_context->channel_layout = in_channel_layout;// 设置声道数codec_context->channels = av_get_channel_layout_nb_channels(codec_context->channel_layout);// 设置比特率codec_context->bit_rate = 128000;// 设置AAC的profilecodec_context->profile = FF_PROFILE_AAC_LOW;// 打开编码器if (avcodec_open2(codec_context, codec, NULL) < 0){fprintf(stderr, "Could not open codec\n");goto end;}// 将编码器参数复制到流int result = avcodec_parameters_from_context(stream->codecpar, codec_context);if (result < 0){fprintf(stderr, "Could not copy codec parameters\n");goto end;}// 打开输出文件if (!(format_context->oformat->flags & AVFMT_NOFILE)) // 检查输出格式是否需要文件存储{result = avio_open(&format_context->pb, output_file, AVIO_FLAG_WRITE);if (result < 0){fprintf(stderr, "Could not open output file %s\n", output_file);goto end;}}// 写文件头result = avformat_write_header(format_context, NULL);if (result < 0){fprintf(stderr, "Error occurred when opening output file\n");goto end;}frame = av_frame_alloc();packet = av_packet_alloc();if (!frame || !packet){fprintf(stderr, "Could not allocate frame or packet\n");goto end;}// 设置帧参数frame->nb_samples = codec_context->frame_size; // 一帧音频数据的采样个数，AAC编码器的frame_size默认为1024frame->format = codec_context->sample_fmt;frame->channel_layout = codec_context->channel_layout;// 分配帧数据缓冲区result = av_frame_get_buffer(frame, 0);if (result < 0){fprintf(stderr, "Could not allocate audio data buffers\n");goto end;}// 初始化重采样上下文swr_ctx = swr_alloc_set_opts(NULL,codec_context->channel_layout, codec_context->sample_fmt, codec_context->sample_rate,codec_context->channel_layout, in_sample_fmt, codec_context->sample_rate,0, NULL);if (!swr_ctx || swr_init(swr_ctx) < 0){fprintf(stderr, "Could not allocate resampler context\n");goto end;}// 计算每帧音频数据的大小 = 采样个数 * 采样格式大小 * 声道数int fsize = frame->nb_samples * av_get_bytes_per_sample(in_sample_fmt) * codec_context->channels;// 分配缓冲区uint8_t *input_buffer = (uint8_t *)av_malloc(fsize);if (!input_buffer){fprintf(stderr, "Could not allocate input buffer\n");goto end;}// 编码循环while (1){// 读取PCM数据到缓冲区result = fread(input_buffer, 1, fsize, input_pcm);if (result <= 0){break;}// 处理不足一帧的情况，不足一帧的数据用0填充if (result < fsize){memset(input_buffer + result, 0, fsize - result);}// 重采样result = swr_convert(swr_ctx, frame->data, frame->nb_samples,(const uint8_t **)&input_buffer, frame->nb_samples);if (result < 0){fprintf(stderr, "Error while converting\n");goto end;}frame->pts = pts;pts += frame->nb_samples; // 计算下一帧的pts，每个帧的时间戳应该是前一个帧的时间戳加上该帧的样本数// 发送帧到编码器result = avcodec_send_frame(codec_context, frame);if (result < 0){fprintf(stderr, "Error sending frame to codec\n");goto end;}// 接收编码后的数据包while (result >= 0){result = avcodec_receive_packet(codec_context, packet);if (result == AVERROR(EAGAIN) || result == AVERROR_EOF){break;}else if (result < 0){fprintf(stderr, "avcodec_receive_packet failed (errmsg '%s')\n", av_err2str(result));goto end;}packet->stream_index = stream->index;// 将时间戳从编码器时间基转换到流时间基av_packet_rescale_ts(packet, codec_context->time_base, stream->time_base);// 写数据包到输出文件result = av_interleaved_write_frame(format_context, packet);if (result < 0){fprintf(stderr, "Error writing audio packet\n");av_packet_unref(packet);goto end;}av_packet_unref(packet);}}av_free(input_buffer);// 发送NULL帧到编码器，刷新编码器内部缓冲区result = avcodec_send_frame(codec_context, NULL);while (result >= 0){result = avcodec_receive_packet(codec_context, packet);if (result == AVERROR_EOF){break;}else if (result < 0){fprintf(stderr, "avcodec_receive_packet failed (errmsg '%s')\n", av_err2str(result));goto end;}packet->stream_index = stream->index;av_packet_rescale_ts(packet, codec_context->time_base, stream->time_base);result = av_interleaved_write_frame(format_context, packet);if (result < 0){fprintf(stderr, "Error writing audio packet\n");av_packet_unref(packet);goto end;}av_packet_unref(packet);}// 写文件尾av_write_trailer(format_context);ret = 0;end:if (frame)av_frame_free(&frame);if (packet)av_packet_free(&packet);if (codec_context)avcodec_free_context(&codec_context);if (format_context)avformat_free_context(format_context);if (swr_ctx)swr_free(&swr_ctx);if (input_pcm)fclose(input_pcm);return ret;
}