FFmpeg —— 15.示例程序（九）：音频编码器（PCM编码为MP3）

代码和上篇几乎一模一样，只需将输出文件的后缀改为mp3即可。

目录

程序源码

看到几篇不错的博文

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程序源码

/**
 *
 *本程序实现了音频PCM采样数据编码为压缩码流（MP3，WMA，AAC等）。
 *
 */

#include <stdio.h>

#define __STDC_CONSTANT_MACROS


extern "C"
{
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavdevice/avdevice.h"
#include "libavcodec/avcodec.h"
#include "libswscale/swscale.h"
#include "libswresample/swresample.h"
#include "libavutil/imgutils.h"
}



int flush_encoder(AVFormatContext *fmt_ctx, unsigned int stream_index)
{
	int ret;
	int got_frame;
	AVPacket enc_pkt;
	if (!(fmt_ctx->streams[stream_index]->codec->codec->capabilities &
	AV_CODEC_CAP_DELAY))
		return 0;
	while (1)
	{
		enc_pkt.data = NULL;
		enc_pkt.size = 0;
		av_init_packet(&enc_pkt);

		ret = avcodec_encode_audio2(fmt_ctx->streams[stream_index]->codec, &enc_pkt,
		NULL, &got_frame);
		av_frame_free(NULL);
		if (ret < 0)
			break;
		if (!got_frame)
		{
			ret = 0;
			break;
		}
		printf("Flush Encoder: Succeed to encode 1 frame!\tsize:%5d\n", enc_pkt.size);
		/* mux encoded frame */
		ret = av_write_frame(fmt_ctx, &enc_pkt);
		if (ret < 0)
			break;
	}
	return ret;
}

int main(int argc, char *argv[])
{
	AVFormatContext *pFormatCtx = NULL;
	AVCodecContext *pCodecCtx = NULL;
	AVCodec *pCodec = NULL;
	AVFrame *pFrame = NULL;
	AVPacket pkt;
	int i = 0;

	const char *inFilename = "s16le.pcm";
	const char *outFilename = "output.mp3";

	avdevice_register_all();

	avformat_alloc_output_context2(&pFormatCtx, NULL, NULL, outFilename);

	if (avio_open(&pFormatCtx->pb, outFilename, AVIO_FLAG_READ_WRITE) < 0)
	{
		printf("can't open output file\n");
		return -1;
	}

	AVStream *stream = avformat_new_stream(pFormatCtx, NULL);
	if (!stream)
	{
		printf("can't allocate new stream\n");
		return -1;
	}

	//设置参数
	pCodecCtx = stream->codec;
	pCodecCtx->codec_id = pFormatCtx->oformat->audio_codec;
	pCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
	pCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
	pCodecCtx->sample_rate = 44100;
	pCodecCtx->channel_layout = AV_CH_LAYOUT_STEREO;
	pCodecCtx->channels = av_get_channel_layout_nb_channels(pCodecCtx->channel_layout);
	pCodecCtx->bit_rate = 128000;
//	pCodecCtx->frame_size = 1024;

//查找编码器
	pCodec = avcodec_find_encoder(pCodecCtx->codec_id);
//	pCodec = avcodec_find_encoder_by_name("libfdk_aac");
	if (!pCodec)
	{
		printf("can't find encoder\n");
		return -1;
	}

	//打开编码器
	if (avcodec_open2(pCodecCtx, pCodec, NULL) < 0)
	{
		printf("can't open encoder\n");
		return -1;
	}

	pFrame = av_frame_alloc();
	if (!pFrame)
	{
		printf("can't alloc frame\n");
		return -1;
	}

	pFrame->nb_samples = pCodecCtx->frame_size;
	pFrame->format = pCodecCtx->sample_fmt;
	pFrame->channels = 2;

	//PCM重采样
	SwrContext *swrCtx = swr_alloc();

	swr_alloc_set_opts(swrCtx, av_get_default_channel_layout(pCodecCtx->channels), pCodecCtx->sample_fmt,
			pCodecCtx->sample_rate, av_get_default_channel_layout(pFrame->channels), AV_SAMPLE_FMT_S16, //PCM源文件的采样格式
			44100, 0, NULL);
	swr_init(swrCtx);

	/* 分配空间 */
	uint8_t **convert_data = (uint8_t**) calloc(pCodecCtx->channels, sizeof(*convert_data));
	av_samples_alloc(convert_data, NULL, pCodecCtx->channels, pCodecCtx->frame_size, pCodecCtx->sample_fmt, 0);


	int size = av_samples_get_buffer_size(NULL, pCodecCtx->channels, pCodecCtx->frame_size, pCodecCtx->sample_fmt, 1);
	uint8_t *frameBuf = (uint8_t*) av_malloc(size);
	avcodec_fill_audio_frame(pFrame, pCodecCtx->channels, pCodecCtx->sample_fmt, (const uint8_t*) frameBuf, size, 1);

	//写帧头
	avformat_write_header(pFormatCtx, NULL);

	FILE *inFile = fopen(inFilename, "rb");

	av_init_packet(&pkt);
	pkt.data = NULL;
	pkt.size = 0;

	for (i = 0;; i++)
	{
		//输入一帧数据的长度
		int length = pFrame->nb_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * pFrame->channels;
		printf("length = %d\n", length);

		//读PCM：特意注意读取的长度，否则可能出现转码之后声音变快或者变慢
		if (fread(frameBuf, 1, length, inFile) <= 0)
		{
			printf("failed to read raw data\n");
			return -1;
		}
		else if (feof(inFile))
		{
			break;
		}

		swr_convert(swrCtx, convert_data, pCodecCtx->frame_size, (const uint8_t**) pFrame->data, pFrame->nb_samples);

		//输出一帧数据的长度
		length = pCodecCtx->frame_size * av_get_bytes_per_sample(pCodecCtx->sample_fmt);
		//双通道赋值（输出的AAC为双通道）
		memcpy(pFrame->data[0], convert_data[0], length);
		memcpy(pFrame->data[1], convert_data[1], length);

		pFrame->pts = i * 100;

		if (avcodec_send_frame(pCodecCtx, pFrame) < 0)
		{
			printf("can't send frame for encoding\n");
			break;
		}

		if (avcodec_receive_packet(pCodecCtx, &pkt) >= 0)
		{
			pkt.stream_index = stream->index;
			printf("write %4d frame, size = %d, length = %d\n", i, size, length);
			av_write_frame(pFormatCtx, &pkt);
		}

		av_packet_unref(&pkt);
	}

	//flush encoder
	if (flush_encoder(pFormatCtx, 0) < 0)
	{
		printf("flushing encoder failed\n");
		return -1;
	}

	//write trailer
	av_write_trailer(pFormatCtx);

	avcodec_close(stream->codec);
	av_free(pFrame);
	av_free(frameBuf);

	avio_close(pFormatCtx->pb);
	avformat_free_context(pFormatCtx);

	fclose(inFile);

	return 0;
}

看到几篇不错的博文

音频重采样

音频编码为AAC/MP3

常见音频编码格式