使用FFmpeg编写音乐播放器

本文介绍了用ffmpeg编写一个简单的音乐播放器, 适合初学者入门一下.

本文用FFmpeg进行音频解码,用PortAudio播放声音.


/// main.cpp

// 这里是ffmpeg的头文件, 如果是用c++, 必须加上extern "C", 否则可能导致链接时出错.
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
}

// 这里是PortAudio的头文件
#include <portaudio.h>
#include <assert.h>
#include <iostream>

struct AudioContext {
	AVCodecContext* codecContext;
	SwrContext* swrContext;
	ReSampleContext* resamplerContext;
};

static
void audio_copy(AudioContext *context,  AVFrame *dst, AVFrame* src)
{
	int nb_sample;
	int dst_buf_size;
	int out_channels;
	int bytes_per_sample = 0;

	dst->linesize[0] = src->linesize[0];
	*dst = *src;
	dst->data[0] = NULL;
	dst->type = 0;

	/* 备注: FFMIN(codecContext->channels, 2); 会有问题, 因为swr_alloc_set_opts的out_channel_layout参数. */
	out_channels = context->codecContext->channels;

	bytes_per_sample = av_get_bytes_per_sample(context->codecContext->sample_fmt);
	/* 备注: 由于 src->linesize[0] 可能是错误的, 所以计算得到的nb_sample会不正确, 直接使用src->nb_samples即可. */
	nb_sample = src->nb_samples;/* src->linesize[0] / codecContext->channels / bytes_per_sample; */
	bytes_per_sample = av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
	dst_buf_size = nb_sample * bytes_per_sample * out_channels;
	dst->data[0] = (uint8_t*) av_malloc(dst_buf_size);
	assert(dst->data[0]);
	avcodec_fill_audio_frame(dst, out_channels, AV_SAMPLE_FMT_S16, dst->data[0], dst_buf_size, 0);

	/* 重采样到AV_SAMPLE_FMT_S16格式. */
	if (context->codecContext->sample_fmt != AV_SAMPLE_FMT_S16)
	{
		if (!context->swrContext)
		{
			uint64_t in_channel_layout = av_get_default_channel_layout(context->codecContext->channels);
			uint64_t out_channel_layout = av_get_default_channel_layout(out_channels);
			context->swrContext = swr_alloc_set_opts(NULL,
				out_channel_layout, AV_SAMPLE_FMT_S16, context->codecContext->sample_rate,
				in_channel_layout, context->codecContext->sample_fmt, context->codecContext->sample_rate,
				0, NULL);
			swr_init(context->swrContext);
		}

		if (context->swrContext)
		{
			int ret, out_count;
			out_count = dst_buf_size / out_channels / av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
			ret = swr_convert(context->swrContext, dst->data, out_count, const_cast<const uint8_t**>(src->data), nb_sample);
			if (ret < 0)
				assert(0);
			src->linesize[0] = dst->linesize[0] = ret * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * out_channels;
			memcpy(src->data[0], dst->data[0], src->linesize[0]);
		}
	}

	/* 重采样到双声道. */
	if (context->codecContext->channels > 2)
	{
		if (!context->resamplerContext)
		{
			context->resamplerContext = av_audio_resample_init(
					FFMIN(2, context->codecContext->channels),
					context->codecContext->channels, context->codecContext->sample_rate,
					context->codecContext->sample_rate, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16,
					16, 10, 0, 0.f);
		}

		if (context->resamplerContext)
		{
			int samples = src->linesize[0] / (av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * context->codecContext->channels);
			dst->linesize[0] = audio_resample(context->resamplerContext,
				(short *) dst->data[0], (short *) src->data[0], samples) * 4;
		}
	}
	else
	{
		dst->linesize[0] = dst->linesize[0];
		memcpy(dst->data[0], src->data[0], dst->linesize[0]);
	}
}

int main(int argc, char* argv[]) 
{
    // 将要打开的音频文件(视频文件也可以支持).
    const char* filename = argc > 1 ? argv[1] : "1.mp3";
    
    // 初始化libavformat,并注册所有的模块
    av_register_all();
    
    // 这里一定要设置成NULL, 或者调用avformat_alloc_context分配内存, 否则可能崩溃.
    AVFormatContext *formatContext = NULL;
    
    // 打开输入文件.
    if( avformat_open_input(&formatContext, filename, NULL, NULL) < 0) {
        std::cerr << "cannot open file" << std::endl;
        return -1;
    }

    // 探测文件里面的音视频流信息.
    if( avformat_find_stream_info(formatContext, NULL) < 0) {
        std::cerr << "cannot find stream info" << std::endl;
        return -1;
    }

    // 输出来看看.
    av_dump_format(formatContext, 0, 0, 0);

    // 找到音频流的索引(如果是视频的话,可能存在多个流).
    int audioIndex;
    if((audioIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, 0, 0, NULL, 0)) < 0) {
        std::cerr << "cannot find audio stream" << std::endl;
        return -1;
    }

    AVCodecContext *codecContext = formatContext->streams[audioIndex]->codec;
    // 找到解码器.
    AVCodec *codec = avcodec_find_decoder(codecContext->codec_id);
    if(codec == NULL) {
        std::cerr << "cannot find decoder for " << codecContext->codec_name << std::endl;
    }

    // 打开解码器.
    if( avcodec_open2(codecContext, codec, NULL) < 0) {
        std::cerr << "cannot open decoder" << std::endl;
        return -1;
    }
    
    // AVPacket是解码前的数据, AVFrame是解码后的数据.
    AVPacket packet;
    AVFrame *frame = avcodec_alloc_frame();
    int got;
	AudioContext context;
	context.resamplerContext = NULL;
	context.swrContext = NULL;
	context.codecContext = codecContext;

    // 下面是初始化PortAudio, 用PortAudio的Blocking API比较简单.
    PaStream *stream;
    Pa_Initialize();
    Pa_OpenDefaultStream(&stream, 0, codecContext->channels, 
        paInt16, codecContext->sample_rate, 
        1024, NULL, NULL);
    Pa_StartStream(stream);
	codecContext->sample_fmt;
    int size = 4092;
    uint8_t* buffer = new uint8_t[size * 2];
    int channels = codecContext->channels;
    while(true) {
        // 从文件中读取一帧.
        if(av_read_frame(formatContext, &packet) < 0) {
            // 文件读完了.
            break;
        }

        // 解码.
        if( avcodec_decode_audio4(codecContext, frame, &got, &packet) < 0) {
            std::cerr << "cannot decode" << std::endl;
            // 偶尔会出错,一般都可以原谅的...
            // break;
        }

        // 解码出来了一帧
        if(got) {
            // 因为frame->data[0]表示的是左声道LLL....,frame->data[1]表示右声道RRR...
            // 而PortAudio要求的是LRLRLR....这样的数据排布, 所以这里用循环重新将数据复制到buffer中
			AVFrame *dst = avcodec_alloc_frame();
			audio_copy(&context, dst, frame);
            Pa_WriteStream(stream, reinterpret_cast<int16_t*>(dst->data[0]), dst->nb_samples);
        }

    }
    delete buffer;
    return 0; 
}

Published: July 10 2013

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