音频采样率转换

在音视频应用中，经常需要将音频的采样率做一些转换，比如将16000hz转成441000hz等。 ffmpeg的libresample提供的音频通道数、采样率、位数的转换，下面介绍一下使用流程，示例是将单通道、8khz、16bit转换为单通道、44.1khz、float型pcm

1，初始化context

extern "C"
{
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
}
#pragma comment(lib, "avutil.lib")
#pragma comment(lib, "swresample.lib")

SwrContext *swr_ctx_ = NULL;
		uint8_t **src_data_ = NULL, **dst_data_ = NULL;
		int src_nb_samples_, dst_nb_samples_, max_dst_nb_samples_;

int64_t src_ch_layout = AV_CH_LAYOUT_MONO, dst_ch_layout = AV_CH_LAYOUT_MONO;
	int src_rate = 8000, dst_rate = 44100;
	enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_S16, dst_sample_fmt = AV_SAMPLE_FMT_FLT;
	int ret;
	int src_nb_channels = 0, dst_nb_channels = 0;
	int src_linesize, dst_linesize;
	src_nb_samples_ = 1024;

	/* create resampler context */
	swr_ctx_ = swr_alloc();
	if (!swr_ctx_) {
		fprintf(stderr, "Could not allocate resampler context\n");
		ret = AVERROR(ENOMEM);
		return;
	}

	/* set options */
	av_opt_set_int(swr_ctx_, "in_channel_layout", src_ch_layout, 0);
	av_opt_set_int(swr_ctx_, "in_sample_rate", src_rate, 0);
	av_opt_set_sample_fmt(swr_ctx_, "in_sample_fmt", src_sample_fmt, 0);

	av_opt_set_int(swr_ctx_, "out_channel_layout", dst_ch_layout, 0);
	av_opt_set_int(swr_ctx_, "out_sample_rate", dst_rate, 0);
	av_opt_set_sample_fmt(swr_ctx_, "out_sample_fmt", dst_sample_fmt, 0);

	/* initialize the resampling context */
	if ((ret = swr_init(swr_ctx_)) < 0) {
		fprintf(stderr, "Failed to initialize the resampling context\n");
		return;
	}

	/* allocate source and destination samples buffers */
	src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
	ret = av_samples_alloc_array_and_samples(&src_data_, &src_linesize, src_nb_channels,
		src_nb_samples_, src_sample_fmt, 0);
	if (ret < 0) {
		fprintf(stderr, "Could not allocate source samples\n");
		return ;
	}

	/* compute the number of converted samples: buffering is avoided
	* ensuring that the output buffer will contain at least all the
	* converted input samples */
	max_dst_nb_samples_ = dst_nb_samples_ =
		av_rescale_rnd(src_nb_samples_, dst_rate, src_rate, AV_ROUND_UP);

	/* buffer is going to be directly written to a rawaudio file, no alignment */
	dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
	ret = av_samples_alloc_array_and_samples(&dst_data_, &dst_linesize, dst_nb_channels,
		dst_nb_samples_, dst_sample_fmt, 0);
	if (ret < 0) {
		fprintf(stderr, "Could not allocate destination samples\n");
		return;
	}

2，对输入的音频数据调用如下代码，其中pBuffer表示pcm数据，dwBufSize表示数据字节数

short *pcmbuf = (short*)src_data_[0];
		for (int i = 0; i < dwBufSize; i++)
		{
			pcmbuf[i] = pBuffer[i];
		}
                /* convert to destination format */
		int ret = swr_convert(swr_ctx_, dst_data_, dst_nb_samples_, (const uint8_t **)src_data_, dwBufSize);
		if (ret < 0) {
			fprintf(stderr, "Error while converting\n");
			return -1;
		}
		int dst_linesize, dst_nb_channels = 1;
		enum AVSampleFormat dst_sample_fmt = AV_SAMPLE_FMT_FLT;
		int dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
			ret, dst_sample_fmt, 1);
		if (dst_bufsize < 0) {
			fprintf(stderr, "Could not get sample buffer size\n");
			return -1;
		}
                //转换后的数据存在dst_data_[0]，字节数为dst_bufsize
                //fwrite(dst_data_[0],1,dst_bufsize,fp);

3，销毁对象

if (src_data_)
		av_freep(&src_data_[0]);
	av_freep(&src_data_);

	if (dst_data_)
		av_freep(&dst_data_[0]);
	av_freep(&dst_data_);

	swr_free(&swr_ctx_);