基于ffmpeg的转码代码(转码为hls)

简介

本文基于ffmpeg api将视频转码为hls,其中可以设置hls_time,hls_list_size等参数。在此份代码中,也涉及了音视频转码的一些转码流程以及参数设置,供大家借鉴。

本份代码的运行环境为:ubuntu14.04。ffmpeg版本为2.6.1。大家也可改造一下,转码成不同的格式。

代码

hls.c
#include <stdio.h>

#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
#include <libswresample/swresample.h>
#include <libavutil/fifo.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/avutil.h>

#define BUF_SIZE_20K 2048000
#define BUF_SIZE_1K 1024000

static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
static SwrContext* pSwrCtx = NULL;
AVBitStreamFilterContext* aacbsfc = NULL;

void initSwr(int audio_index)  
{  
    if (ofmt_ctx->streams[0]->codec->channels != ifmt_ctx->streams[audio_index]->codec->channels  
        || ofmt_ctx->streams[0]->codec->sample_rate != ifmt_ctx->streams[audio_index]->codec->sample_rate  
        || ofmt_ctx->streams[0]->codec->sample_fmt != ifmt_ctx->streams[audio_index]->codec->sample_fmt)  
    {  
        if ( NULL == pSwrCtx )   
        {  
            pSwrCtx = swr_alloc();  
        }  
// #if LIBSWRESAMPLE_VERSION_MINOR >= 17    // 根据版本不同,选用适当函数  
        // av_opt_set_int(pSwrCtx, "ich", ifmt_ctx->streams[audio_index]->codec->channels, 0);  
        // av_opt_set_int(pSwrCtx, "och", ofmt_ctx->streams[audio_index]->codec->channels, 0);  
        // av_opt_set_int(pSwrCtx, "in_sample_rate",  ifmt_ctx->streams[audio_index]->codec->sample_rate, 0);  
        // av_opt_set_int(pSwrCtx, "out_sample_rate",  ofmt_ctx->streams[audio_index]->codec->sample_rate, 0);  
        // av_opt_set_sample_fmt(pSwrCtx, "in_sample_fmt", ifmt_ctx->streams[audio_index]->codec->sample_fmt, 0);  
        // av_opt_set_sample_fmt(pSwrCtx, "out_sample_fmt", ofmt_ctx->streams[audio_index]->codec->sample_fmt, 0);  
  
// #else  
        pSwrCtx = swr_alloc_set_opts(NULL,             
            ofmt_ctx->streams[audio_index]->codec->channel_layout,   
            ofmt_ctx->streams[audio_index]->codec->sample_fmt,   
            ofmt_ctx->streams[audio_index]->codec->sample_rate,  
            ifmt_ctx->streams[audio_index]->codec->channel_layout,   
            ifmt_ctx->streams[audio_index]->codec->sample_fmt,   
            ifmt_ctx->streams[audio_index]->codec->sample_rate,  
            0, NULL);  
// #endif  
        swr_init(pSwrCtx);  
    }  
}  
  
//setup_array函数摘自ffmpeg例程  
static void setup_array(uint8_t* out[32], AVFrame* in_frame, int format, int samples)  
{  
    if (av_sample_fmt_is_planar(format))   
    {  
        int i;
        int plane_size = av_get_bytes_per_sample((format & 0xFF)) * samples;
        format &= 0xFF;  
          
        //从decoder出来的frame中的data数据不是连续分布的,所以不能这样写:  
        in_frame->data[0]+i*plane_size;  
        for (i = 0; i < in_frame->channels; i++)  
        {  
            out[i] = in_frame->data[i];  
        }  
    }   
    else  
    {  
        out[0] = in_frame->data[0];  
    }  
}

int TransSample(AVFrame *in_frame, AVFrame *out_frame, int audio_index)  
{  
    int ret;  
    int max_dst_nb_samples = 4096;  
    //int64_t dst_nb_samples;  
    int64_t src_nb_samples = in_frame->nb_samples;  
    out_frame->pts = in_frame->pts;  
    uint8_t* paudiobuf;  
    int decode_size, input_size, len;  
    if (pSwrCtx != NULL)   
    {  
        out_frame->nb_samples = av_rescale_rnd(swr_get_delay(pSwrCtx, ofmt_ctx->streams[audio_index]->codec->sample_rate) + src_nb_samples,  
            ofmt_ctx->streams[audio_index]->codec->sample_rate, ifmt_ctx->streams[audio_index]->codec->sample_rate, AV_ROUND_UP);  
  
        ret = av_samples_alloc(out_frame->data,   
            &out_frame->linesize[0],   
            ofmt_ctx->streams[audio_index]->codec->channels,   
            out_frame->nb_samples,   
            ofmt_ctx->streams[audio_index]->codec->sample_fmt, 0);  
          
        if (ret < 0)  
        {  
            av_log(NULL, AV_LOG_WARNING, "[%s.%d %s() Could not allocate samples Buffer\n", __FILE__, __LINE__, __FUNCTION__);  
            return -1;  
        }  
          
        max_dst_nb_samples = out_frame->nb_samples;  
        //输入也可能是分平面的,所以要做如下处理  
        uint8_t* m_ain[32];  
        setup_array(m_ain, in_frame, ifmt_ctx->streams[audio_index]->codec->sample_fmt, src_nb_samples);  
          
        //注意这里,out_count和in_count是samples单位,不是byte  
        //所以这样av_get_bytes_per_sample(ifmt_ctx->streams[audio_index]->codec->sample_fmt) * src_nb_samples是错的  
        len = swr_convert(pSwrCtx, out_frame->data, out_frame->nb_samples, (const uint8_t**)in_frame->data, src_nb_samples);  
        if (len < 0)  
        {  
            char errmsg[BUF_SIZE_1K];  
            av_strerror(len, errmsg, sizeof(errmsg));  
            av_log(NULL, AV_LOG_WARNING, "[%s:%d] swr_convert!(%d)(%s)", __FILE__, __LINE__, len, errmsg);  
            return -1;  
        }  
    }  
    else  
    {  
        printf("pSwrCtx with out init!\n");  
        return -1;  
    }  
    return 0;  
}  

static int open_input_file(const char *filename)
{
	int ret;
	unsigned int i;

	ifmt_ctx = NULL;
	if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
		av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
		return ret;
	}

	if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
		av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
		return ret;
	}

	for (i = 0; i < ifmt_ctx->nb_streams; i++) {
		AVStream *stream;
		AVCodecContext *codec_ctx;
		stream = ifmt_ctx->streams[i];
		codec_ctx = stream->codec;
		/* Reencode video & audio and remux subtitles etc. */
		if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
			|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
			/* Open decoder */
			ret = avcodec_open2(codec_ctx,
				avcodec_find_decoder(codec_ctx->codec_id), NULL);
			if (ret < 0) {
				av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
				return ret;
			}
		}
	}

	av_dump_format(ifmt_ctx, 0, filename, 0);
	return 0;
}

static int open_output_file(const char *filename)
{
	AVStream *out_stream;
	AVStream *in_stream;
	AVCodecContext *dec_ctx, *enc_ctx;
	AVCodec *encoder;
	int ret;
	unsigned int i;

	ofmt_ctx = NULL;
	avformat_alloc_output_context2(&ofmt_ctx, NULL, "hls", filename);
	if (!ofmt_ctx) {
		av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
		return AVERROR_UNKNOWN;
	}

	for (i = 0; i < ifmt_ctx->nb_streams; i++) {
		out_stream = avformat_new_stream(ofmt_ctx, NULL);
		if (!out_stream) {
			av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
			return AVERROR_UNKNOWN;
		}

		in_stream = ifmt_ctx->streams[i];
		dec_ctx = in_stream->codec;
		enc_ctx = out_stream->codec;

		if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) 
		{
			encoder = avcodec_find_encoder(AV_CODEC_ID_H264);
			if (!encoder) {
				av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
				return AVERROR_INVALIDDATA;
			}

			enc_ctx->height = dec_ctx->height;
			enc_ctx->width = dec_ctx->width;
			enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
				
			enc_ctx->pix_fmt = encoder->pix_fmts[0];
				
			enc_ctx->time_base = dec_ctx->time_base;

			// enc_ctx->me_range = 25;
			// enc_ctx->max_qdiff = 4;
			enc_ctx->qmin = 10;
			enc_ctx->qmax = 51;
			// enc_ctx->qcompress = 0.6;
			// enc_ctx->refs = 3;
			enc_ctx->max_b_frames = 3;
			enc_ctx->gop_size = 250;
			enc_ctx->bit_rate = 500000;
            enc_ctx->time_base.num = dec_ctx->time_base.num;
            enc_ctx->time_base.den = dec_ctx->time_base.den;
			
			ret = avcodec_open2(enc_ctx, encoder, NULL);
			if (ret < 0) {
				av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
				return ret;
			}
            
            av_opt_set(ofmt_ctx->priv_data, "preset", "superfast", 0);
            av_opt_set(ofmt_ctx->priv_data, "tune", "zerolatency", 0);
            // av_opt_set_int(ofmt_ctx->priv_data, "hls_time", 5, AV_OPT_SEARCH_CHILDREN);
            // av_opt_set_int(ofmt_ctx->priv_data, "hls_list_size", 10, AV_OPT_SEARCH_CHILDREN);
		}
		else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
			av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
			return AVERROR_INVALIDDATA;
		}
		else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO)
		{
			encoder = avcodec_find_encoder(AV_CODEC_ID_AAC);
			enc_ctx->sample_rate = dec_ctx->sample_rate;
			enc_ctx->channel_layout = dec_ctx->channel_layout;
			enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
			enc_ctx->sample_fmt = encoder->sample_fmts[0];
			AVRational ar = { 1, enc_ctx->sample_rate };
			enc_ctx->time_base = ar;

			ret = avcodec_open2(enc_ctx, encoder, NULL);
			if (ret < 0) {
				av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
				return ret;
			}
		}
		else {
			ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
				ifmt_ctx->streams[i]->codec);
			if (ret < 0) {
				av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
				return ret;
			}
		}

		if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
			enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;

	}
	av_dump_format(ofmt_ctx, 0, filename, 1);
    
	// if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
		ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
		if (ret < 0) {
			av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
			return ret;
		}
	// }

	/* init muxer, write output file header */
	ret = avformat_write_header(ofmt_ctx, NULL);
	if (ret < 0) {
		av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
		return ret;
	}

	return 0;
}

static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
    static a_total_duration = 0;
    static v_total_duration = 0;
	int ret;
	int got_frame_local;
	AVPacket enc_pkt;
	int(*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
		(ifmt_ctx->streams[stream_index]->codec->codec_type ==
		AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;

	if (!got_frame)
		got_frame = &got_frame_local;

	av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
	/* encode filtered frame */
	enc_pkt.data = NULL;
	enc_pkt.size = 0;
	av_init_packet(&enc_pkt);
	ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
		filt_frame, got_frame);
	if (ret < 0)
		return ret;
	if (!(*got_frame))
		return 0;

    // if (ifmt_ctx->streams[stream_index]->codec->codec_type !=
		// AVMEDIA_TYPE_VIDEO)
        // av_bitstream_filter_filter(aacbsfc, ofmt_ctx->streams[stream_index]->codec, NULL, &enc_pkt.data, &enc_pkt.size, enc_pkt.data, enc_pkt.size, 0);
    
	/* prepare packet for muxing */
	enc_pkt.stream_index = stream_index;
	av_packet_rescale_ts(&enc_pkt,
		ofmt_ctx->streams[stream_index]->codec->time_base,
		ofmt_ctx->streams[stream_index]->time_base);
            
    if (ifmt_ctx->streams[stream_index]->codec->codec_type !=
		AVMEDIA_TYPE_VIDEO){
        enc_pkt.pts = enc_pkt.dts = a_total_duration;
        a_total_duration += av_rescale_q(filt_frame->nb_samples, ofmt_ctx->streams[stream_index]->codec->time_base, ofmt_ctx->streams[stream_index]->time_base);
    }
    // printf("v_total_duration: %d, a_total_duration: %d\n", v_total_duration, a_total_duration);
	av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
	/* mux encoded frame */
	ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
	return ret;
}

static int flush_encoder(unsigned int stream_index)
{
	int ret;
	int got_frame;
    AVPacket enc_pkt;

	if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
		CODEC_CAP_DELAY))
		return 0;

	while (1) {
        enc_pkt.data = NULL;
        enc_pkt.size = 0;
        av_init_packet(&enc_pkt);

		av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
		ret = encode_write_frame(NULL, stream_index, &got_frame);
		if (ret < 0)
			break;
		if (!got_frame)
			return 0;
	}
	return ret;
}

int main(int argc, char **argv)
{
	int ret;
	AVPacket packet; //= { .data = NULL, .size = 0 };
	packet.data = NULL;
	packet.size = 0;
	AVFrame *frame = NULL;
	enum AVMediaType type;
	unsigned int stream_index;
	unsigned int i;
	int got_frame;
	int(*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);

	av_register_all();
	avfilter_register_all();

	if ((ret = open_input_file("test.flv")) < 0)
		goto end;
	if ((ret = open_output_file("test.m3u8")) < 0)
		goto end;

    aacbsfc = av_bitstream_filter_init("aac_adtstoasc");
	/* read all packets */
    int count = 0;
    int flag = 1;
	while (1) {
		if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
			break;
		stream_index = packet.stream_index;
		type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
		av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
			stream_index);

        av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
        frame = av_frame_alloc();
        if (!frame) {
            ret = AVERROR(ENOMEM);
            break;
        }
        av_packet_rescale_ts(&packet,
            ifmt_ctx->streams[stream_index]->time_base,
            ifmt_ctx->streams[stream_index]->codec->time_base);
        dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
            avcodec_decode_audio4;
        ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
            &got_frame, &packet);
        if (ret < 0) {
            av_frame_free(&frame);
            av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
            break;
        }

        if (got_frame) {
            frame->pts = frame->pkt_pts;
            // frame->pts = av_frame_get_best_effort_timestamp(frame);
            // frame->pts=count;
            if (type == AVMEDIA_TYPE_VIDEO){
                ret = encode_write_frame(frame, stream_index, NULL);
            }else {
                if (flag){
                    initSwr(stream_index);
                    flag = 0;
                }
                    
                AVFrame *frame_out = av_frame_alloc();
                if (0 != TransSample(frame, frame_out, stream_index)){
                    av_log(NULL, AV_LOG_ERROR, "convert audio failed\n");
                    ret = -1;
                }
                // frame_out->pts = frame->pkt_pts;
                ret = encode_write_frame(frame_out, stream_index, NULL);
                av_frame_free(&frame_out);
            }
            av_frame_free(&frame);
            if (ret < 0)
                goto end;
        }
        else {
            av_frame_free(&frame);
        }
		av_free_packet(&packet);
        ++count;
	}

    /* flush  encoders */
	// for (i = 0; i < ifmt_ctx->nb_streams; i++) {
		// ret = flush_encoder(i);
		// if (ret < 0) {
			// av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
			// goto end;
		// }
	// }
    av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");

	av_write_trailer(ofmt_ctx);
end:
	av_free_packet(&packet);
	av_frame_free(&frame);
    av_bitstream_filter_close(aacbsfc);
	for (i = 0; i < ifmt_ctx->nb_streams; i++) {
		avcodec_close(ifmt_ctx->streams[i]->codec);
		if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
			avcodec_close(ofmt_ctx->streams[i]->codec);
	}
	// av_free(filter_ctx);
	avformat_close_input(&ifmt_ctx);
	if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
		avio_closep(&ofmt_ctx->pb);
	avformat_free_context(ofmt_ctx);

	// if (ret < 0)
		// av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret)); //av_err2str(ret));

	return ret ? 1 : 0;
}

编译

编译命令:

       gcc hls.c -g -o hls -lavformat -lavutil -lavcodec -lswscale

重点片段

1.视频帧率最好设置成与原视频一样,否则会有音视频不同步的现象

2.音频转码需要用swr转换一下sample。否则音频转码出来是噪音

3.转码为hls不需要用aacbsfc,但是我把output的格式换成flv后,需要用aacbsfc才能正常转码音频

代码下载

除了本份代码,我还基于ffmpeg自带的转码例子改造了一份转码hls的代码。一起提交到cdsn供大家下载。(ffmpeg自带的例子用的是filter,省了自己设置各种参数和格式转换)

http://download.csdn.net/detail/yingyemin/9918532

http://download.csdn.net/detail/yingyemin/9918545

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在使用ffmpeg进行转码时,可以根据需要设置不同的参数来实现特定的转码效果。以下是一些常用的参数及其作用: 1. -i:指定输入文件,后面跟上要转码的源文件路径。 2. -r:设置输出视频的帧率,即每秒显示的帧数。 3. -qmin和-qmax:设置视频的质量范围,可以控制输出视频的清晰度。值越小,视频质量越高,但文件大小也会增加。 4. -ab:设置音频的比特率,即音频的编码质量。 5. -ar:设置音频的采样率,即每秒钟对音频信号进行采样的次数。 6. -ac:设置音频的通道数,即音频的声道数。 7. -y:强制覆盖输出文件,如果存在同名的输出文件,会自动覆盖。 此外,还有其他一些参数可供选择,具体选择哪些参数取决于你想要实现转码效果和目标格式。例如,-vcodec参数用于选择视频编码器,可选值包括copy、libx264、h263、libxvid、mpeg4、msmpeg4v2等。根据需要,你可以根据转码的要求选择适当的编码器。 需要注意的是,不同的参数组合会产生不同的转码效果,你可以根据具体需求进行调整,以达到最佳的转码结果。<span class="em">1</span><span class="em">2</span><span class="em">3</span> #### 引用[.reference_title] - *1* [ffmpeg转码hls代码](https://download.csdn.net/download/yingyemin/9918545)[target="_blank" data-report-click={"spm":"1018.2226.3001.9630","extra":{"utm_source":"vip_chatgpt_common_search_pc_result","utm_medium":"distribute.pc_search_result.none-task-cask-2~all~insert_cask~default-1-null.142^v93^chatsearchT3_1"}}] [.reference_item style="max-width: 50%"] - *2* *3* [ffmpeg压制视频命令及常用转码参数解析](https://blog.csdn.net/qq_43174722/article/details/104596197)[target="_blank" data-report-click={"spm":"1018.2226.3001.9630","extra":{"utm_source":"vip_chatgpt_common_search_pc_result","utm_medium":"distribute.pc_search_result.none-task-cask-2~all~insert_cask~default-1-null.142^v93^chatsearchT3_1"}}] [.reference_item style="max-width: 50%"] [ .reference_list ]

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