ffmpeg实现音频resample(重采样)(二)

本篇文章将增加AVFifoBuffer和音频样本是av_sample_fmt_is_planar的样式采样率讲解，下面上代码

AVFifoBuffer * m_fifo = NULL;



SwrContext * init_pcm_resample(AVFrame *in_frame, AVFrame *out_frame)
{
	SwrContext * swr_ctx = NULL;
	swr_ctx = swr_alloc();
	if (!swr_ctx)
	{
		printf("swr_alloc error \n");
		return NULL;
	}
	AVCodecContext * audio_dec_ctx = icodec->streams[audio_stream_idx]->codec;
	AVSampleFormat sample_fmt;
	sample_fmt = (AVSampleFormat)m_dwBitsPerSample; //样本
	if (audio_dec_ctx->channel_layout == 0)
	{
		audio_dec_ctx->channel_layout = av_get_default_channel_layout(icodec->streams[audio_stream_idx]->codec->channels);
	}
	/* set options */
	av_opt_set_int(swr_ctx, "in_channel_layout",    audio_dec_ctx->channel_layout, 0);
	av_opt_set_int(swr_ctx, "in_sample_rate",       audio_dec_ctx->sample_rate, 0);
	av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", audio_dec_ctx->sample_fmt, 0);

	av_opt_set_int(swr_ctx, "out_channel_layout",    audio_dec_ctx->channel_layout, 0);
	av_opt_set_int(swr_ctx, "out_sample_rate",       audio_dec_ctx->sample_rate, 0);
	av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", sample_fmt, 0);
	swr_init(swr_ctx);

	int64_t src_nb_samples = in_frame->nb_samples;
	out_frame->nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx,oaudio_st->codec->sample_rate) + src_nb_samples,
		oaudio_st->codec->sample_rate, oaudio_st->codec->sample_rate, AV_ROUND_UP);

	int ret = av_samples_alloc(out_frame->data, &out_frame->linesize[0], 
		icodec->streams[audio_stream_idx]->codec->channels, out_frame->nb_samples,oaudio_st->codec->sample_fmt,1);
	if (ret < 0)
	{
		return NULL;
	}

	//pcm分包初始化
	int buffersize = av_samples_get_buffer_size(NULL, oaudio_st->codec->channels,
		2048, oaudio_st->codec->sample_fmt, 1);
	m_fifo = av_fifo_alloc(buffersize);
	return swr_ctx;
}

int preform_pcm_resample(SwrContext * pSwrCtx,AVFrame *in_frame, AVFrame *out_frame)
{
	int ret = 0;
	if (pSwrCtx != NULL) 
	{
		ret = swr_convert(pSwrCtx, out_frame->data, out_frame->nb_samples, 
			(const uint8_t**)in_frame->data, in_frame->nb_samples);
		if (ret < 0)
		{
			return -1;
		}
		//修改分包内存
		int buffersize = av_samples_get_buffer_size(&out_frame->linesize[0], oaudio_st->codec->channels,
			ret, oaudio_st->codec->sample_fmt, 1);
		int sss = av_fifo_size(m_fifo);
		sss = av_fifo_realloc2(m_fifo, av_fifo_size(m_fifo) + out_frame->linesize[0]);
		sss = av_fifo_size(m_fifo);
		av_fifo_generic_write(m_fifo, out_frame->data[0], out_frame->linesize[0], NULL);

		out_frame->pkt_pts = in_frame->pkt_pts;
		out_frame->pkt_dts = in_frame->pkt_dts;
		//有时pkt_pts和pkt_dts不同，并且pkt_pts是编码前的dts,这里要给avframe传入pkt_dts而不能用pkt_pts
		//out_frame->pts = out_frame->pkt_pts;
		out_frame->pts = in_frame->pkt_dts;
	}
	return 0;
}
void uinit_pcm_resample(AVFrame * poutframe,SwrContext * swr_ctx)
{
	if (poutframe)
	{
		avcodec_free_frame(&poutframe);
		poutframe = NULL;
	}
	if (swr_ctx)
	{
		swr_free(&swr_ctx);
		swr_ctx = NULL;
	}
	//析构pcm分包结构
	if(m_fifo)
	{
		av_fifo_free(m_fifo);
		m_fifo = NULL;
	}
}
int perform_code(int stream_type,AVFrame * picture)
{
	AVCodecContext *cctext = NULL;
	AVPacket pkt_t;
	av_init_packet(&pkt_t);
	pkt_t.data = NULL; // packet data will be allocated by the encoder
	pkt_t.size = 0;
	int frameFinished = 0 ;

	if (stream_type == AUDIO_ID)
	{
		cctext = oaudio_st->codec;
		//如果进和出的的声道，样本，采样率不同,需要重采样
		if(icodec->streams[audio_stream_idx]->codec->sample_fmt != (AVSampleFormat)m_dwBitsPerSample ||
			icodec->streams[audio_stream_idx]->codec->channels != m_dwChannelCount ||
			icodec->streams[audio_stream_idx]->codec->sample_rate != m_dwFrequency)
		{
			int64_t pts_t = picture->pts;
			int duration_t = (double)cctext->frame_size * (icodec->streams[audio_stream_idx]->time_base.den /icodec->streams[audio_stream_idx]->time_base.num)/ 
				icodec->streams[audio_stream_idx]->codec->sample_rate;

			int frame_bytes = cctext->frame_size * av_get_bytes_per_sample(cctext->sample_fmt)* cctext->channels;
			AVFrame * pFrameResample = avcodec_alloc_frame();
			uint8_t * readbuff = new uint8_t[frame_bytes];

			if(av_sample_fmt_is_planar(cctext->sample_fmt))
			{
				frame_bytes /= cctext->channels;
			}

			while (av_fifo_size(m_fifo) >= frame_bytes) //取出写入的未读的包
			{
				pFrameResample->nb_samples = cctext->frame_size;
				av_fifo_generic_read(m_fifo, readbuff, frame_bytes, NULL);

				//这里一定要考虑音频分片的问题
				//如果是分片的avcodec_fill_audio_frame传入的buf是单声道的，但是buf_size 是两个声道加一起的数据量
				//如果不是分片的avcodec_fill_audio_frame传入的buf是双声道的，buf_size 是两个声道加一起的数据量
				if(av_sample_fmt_is_planar(cctext->sample_fmt))
				{
					avcodec_fill_audio_frame(pFrameResample,cctext->channels,cctext->sample_fmt,readbuff,frame_bytes * cctext->channels,1);
				}
				else
				{					
					avcodec_fill_audio_frame(pFrameResample,cctext->channels,cctext->sample_fmt,readbuff,frame_bytes,0);
				}

				if(m_is_first_audio_pts == 0)
				{
					m_first_audio_pts = pts_t;
					m_is_first_audio_pts = 1;
				}
				pFrameResample->pts = m_first_audio_pts;
				m_first_audio_pts += duration_t;


				pFrameResample->pts = av_rescale_q_rnd(pFrameResample->pts, icodec->streams[audio_stream_idx]->codec->time_base, oaudio_st->codec->time_base, AV_ROUND_NEAR_INF);
				nRet = avcodec_encode_audio2(cctext,&pkt_t,pFrameResample,&frameFinished);
				if (nRet>=0 && frameFinished)
				{
					write_frame(ocodec,AUDIO_ID,pkt_t);
					av_free_packet(&pkt_t);
				}
			}
			if (readbuff)
			{
				delete []readbuff;
			}
			if (pFrameResample)
			{
				av_free(pFrameResample);
				pFrameResample = NULL;
			}
		}
		else
		{
			nRet = avcodec_encode_audio2(cctext,&pkt_t,picture,&frameFinished);
			if (nRet>=0 && frameFinished)
			{
				write_frame(ocodec,AUDIO_ID,pkt_t);
				av_free_packet(&pkt_t);
			}
		}
	}
	else if (stream_type == VIDEO_ID)
	{
		cctext = ovideo_st->codec;
		if(icodec->streams[video_stream_idx]->codec->ticks_per_frame != 1)
		{
			AVRational time_base_video_t;
			time_base_video_t.num = icodec->streams[video_stream_idx]->codec->time_base.num;
			time_base_video_t.den = icodec->streams[video_stream_idx]->codec->time_base.den /icodec->streams[video_stream_idx]->codec->ticks_per_frame;
			picture->pts = av_rescale_q_rnd(picture->pts, time_base_video_t, ovideo_st->codec->time_base, AV_ROUND_NEAR_INF);
		}
		else
		{
			picture->pts = av_rescale_q_rnd(picture->pts, icodec->streams[video_stream_idx]->codec->time_base, ovideo_st->codec->time_base, AV_ROUND_NEAR_INF);
		}
		avcodec_encode_video2(cctext,&pkt_t,picture,&frameFinished);
		picture->pts++;
		if (frameFinished)
		{
			write_frame(ocodec,VIDEO_ID,pkt_t);
			av_free_packet(&pkt_t);
		}
	}
	return 1;
}

1：由于mp3的sample是1152 aac是1024 有时候将解码的mp3编码成aac时如果不做AVFifoBuffer操作，编码的aac音频sample会比原来的少很多，生成的音频会一卡一卡的明显少声音。

2：当要编码的音频样本是av_sample_fmt_is_planar分片的时候需要将解码后的视频添加到AVFrame结构体中：但是如图

不知道ffmpeg什么这么设计或者我用的不对。不过这样用是成功的。

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