ffmpeg实现音频resample(重采样)

ffmpeg实现音频resample(重采样)

下面代码是实现将音频的三大要素：声道，样本，采样率变更的demo

例如双声道变成单声道，44100->48000，float->s16等等。 

int AudioResampling(AVCodecContext * audio_dec_ctx,AVFrame * pAudioDecodeFrame,
					int out_sample_fmt,int out_channels ,int out_sample_rate , uint8_t * out_buf)
{
	//////////////////////////////////////////////////////////////////////////
	SwrContext * swr_ctx = NULL;
	int data_size = 0;
	int ret = 0;
	int64_t src_ch_layout = AV_CH_LAYOUT_STEREO; //初始化这样根据不同文件做调整
	int64_t dst_ch_layout = AV_CH_LAYOUT_STEREO; //这里设定ok
	int dst_nb_channels = 0;
	int dst_linesize = 0;
	int src_nb_samples = 0;
	int dst_nb_samples = 0;
	int max_dst_nb_samples = 0;
	uint8_t **dst_data = NULL;
	int resampled_data_size = 0;
	
	//重新采样
	if (swr_ctx)
	{
		swr_free(&swr_ctx);
	}
	swr_ctx = swr_alloc();
	if (!swr_ctx)
	{
		printf("swr_alloc error \n");
		return -1;
	}

	src_ch_layout = (audio_dec_ctx->channel_layout && 
		audio_dec_ctx->channels == 
		av_get_channel_layout_nb_channels(audio_dec_ctx->channel_layout)) ? 
		audio_dec_ctx->channel_layout : 
	av_get_default_channel_layout(audio_dec_ctx->channels);

	if (out_channels == 1)
	{
		dst_ch_layout = AV_CH_LAYOUT_MONO;
	}
	else if(out_channels == 2)
	{
		dst_ch_layout = AV_CH_LAYOUT_STEREO;
	}
	else
	{
		//可扩展
	}

	if (src_ch_layout <= 0)
	{
		printf("src_ch_layout error \n");
		return -1;
	}

	src_nb_samples = pAudioDecodeFrame->nb_samples;
	if (src_nb_samples <= 0)
	{
		printf("src_nb_samples error \n");
		return -1;
	}

	/* set options */
	av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_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",    dst_ch_layout, 0);
	av_opt_set_int(swr_ctx, "out_sample_rate",       out_sample_rate, 0);
	av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", (AVSampleFormat)out_sample_fmt, 0);
	swr_init(swr_ctx);

	max_dst_nb_samples = dst_nb_samples =
		av_rescale_rnd(src_nb_samples, out_sample_rate, audio_dec_ctx->sample_rate, AV_ROUND_UP);
	if (max_dst_nb_samples <= 0)
	{
		printf("av_rescale_rnd error \n");
		return -1;
	}

	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, (AVSampleFormat)out_sample_fmt, 0);
	if (ret < 0)
	{
		printf("av_samples_alloc_array_and_samples error \n");
		return -1;
	}


	dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, audio_dec_ctx->sample_rate) +
		src_nb_samples, out_sample_rate, audio_dec_ctx->sample_rate,AV_ROUND_UP);
	if (dst_nb_samples <= 0)
	{
		printf("av_rescale_rnd error \n");
		return -1;
	}
	if (dst_nb_samples > max_dst_nb_samples)
	{
		av_free(dst_data[0]);
		ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
			dst_nb_samples, (AVSampleFormat)out_sample_fmt, 1);
		max_dst_nb_samples = dst_nb_samples;
	}

	data_size = av_samples_get_buffer_size(NULL, audio_dec_ctx->channels,
		pAudioDecodeFrame->nb_samples,
		audio_dec_ctx->sample_fmt, 1);
	if (data_size <= 0)
	{
		printf("av_samples_get_buffer_size error \n");
		return -1;
	}
	resampled_data_size = data_size;
	
	if (swr_ctx)
	{
		ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, 
			(const uint8_t **)pAudioDecodeFrame->data, pAudioDecodeFrame->nb_samples);
		if (ret <= 0)
		{
			printf("swr_convert error \n");
			return -1;
		}

		resampled_data_size = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
			ret, (AVSampleFormat)out_sample_fmt, 1);
		if (resampled_data_size <= 0)
		{
			printf("av_samples_get_buffer_size error \n");
			return -1;
		}
	}
	else 
	{
		printf("swr_ctx null error \n");
		return -1;
	}

	//将值返回去
	memcpy(out_buf,dst_data[0],resampled_data_size);

	if (dst_data)
	{
		av_freep(&dst_data[0]);
	}
	av_freep(&dst_data);
	dst_data = NULL;

	if (swr_ctx)
	{
		swr_free(&swr_ctx);
	}
	return resampled_data_size;
}

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