如何采样率从AV_SAMPLE_FMT_FLTP转换为AV_SAMPLE_FMT_S16？ [英] How to convert sample rate from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16?

问题描述

我的AAC解码与AV codec_de code_audio3的ffmpeg为PCM。然而，德codeS到AV_SAMPLE_FMT_FLTP样本格式（PCM支持32位浮点平面的），我需要AV_SAMPLE_FMT_S16（PCM 16位有符号 - S16LE）。

I am decoding aac to pcm with ffmpeg with avcodec_decode_audio3. However it decodes into AV_SAMPLE_FMT_FLTP sample format (PCM 32bit Float Planar) and i need AV_SAMPLE_FMT_S16 (PCM 16 bit signed - S16LE).

我知道ffmpeg的可以用-sample_fmt很容易做到这一点。我想要做同样的code，但我还是没能弄清楚。

I know that ffmpeg can do this easily with -sample_fmt. I want to do the same with the code but i still couldn't figure it out.

audio_resample没有为工作的：它失败，错误消息：....转换失败

audio_resample did not work for: it fails with error message: .... conversion failed.

推荐答案

修改2013年4月9日：制定了如何使用libswresample做到这一点...更快

EDIT 9th April 2013: Worked out how to use libswresample to do this... much faster!

在过去的2 - 3年内某些时候FFmpeg的AAC德codeR的输出格式从AV_SAMPLE_FMT_S16改为AV_SAMPLE_FMT_FLTP。这意味着，每个音频信道具有它自己的缓冲器，并且每个采样值是从-1.0缩放+1.0 32位浮点值。

At some point in the last 2-3 years FFmpeg's AAC decoder's output format changed from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_FLTP. This means that each audio channel has it's own buffer, and each sample value is a 32-bit floating point value scaled from -1.0 to +1.0.

而与AV_SAMPLE_FMT_S16数据是在一个单一的缓冲器，与样品交错，并且每个样本是从-32767到+32767一个符号整数。

Whereas with AV_SAMPLE_FMT_S16 the data is in a single buffer, with the samples interleaved, and each sample is a signed integer from -32767 to +32767.

如果你真的需要你的音频作为AV_SAMPLE_FMT_S16，那么你必须自己做转换。我想通了两种方式来做到这一点：

And if you really need your audio as AV_SAMPLE_FMT_S16, then you have to do the conversion yourself. I figured out two ways to do it:

1。使用libswresample （推荐）

#include "libswresample/swresample.h"

...

SwrContext *swr;

...

// Set up SWR context once you've got codec information
swr = swr_alloc();
av_opt_set_int(swr, "in_channel_layout",  audioCodec->channel_layout, 0);
av_opt_set_int(swr, "out_channel_layout", audioCodec->channel_layout,  0);
av_opt_set_int(swr, "in_sample_rate",     audioCodec->sample_rate, 0);
av_opt_set_int(swr, "out_sample_rate",    audioCodec->sample_rate, 0);
av_opt_set_sample_fmt(swr, "in_sample_fmt",  AV_SAMPLE_FMT_FLTP, 0);
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);
swr_init(swr);

...

// In your decoder loop, after decoding an audio frame:
AVFrame *audioFrame = ...;
int16_t* outputBuffer = ...;
swr_convert(&outputBuffer, audioFrame->nb_samples, audioFrame->extended_data, audioFrame->nb_samples);   

这就是所有你需要做的！

And that's all you have to do!

2。做手工在C （原来的答案，不推荐）

2. Do it by hand in C (original answer, not recommended)

因此​​，在你去code循环，当你有一个音频数据包，你去code这样的：

So in your decode loop, when you've got an audio packet you decode it like this:

AVCodecContext *audioCodec;   // init'd elsewhere
AVFrame *audioFrame;          // init'd elsewhere
AVPacket packet;              // init'd elsewhere
int16_t* outputBuffer;        // init'd elsewhere
int out_size = 0;
...
int len = avcodec_decode_audio4(audioCodec, audioFrame, &out_size, &packet);

然后，如果你有声音的完整框架，可以很容易将其转换：

And then, if you've got a full frame of audio, you can convert it fairly easily:

    // Convert from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16
    int in_samples = audioFrame->nb_samples;
    int in_linesize = audioFrame->linesize[0];
    int i=0;
    float* inputChannel0 = (float*)audioFrame->extended_data[0];
    // Mono
    if (audioFrame->channels==1) {
        for (i=0 ; i<in_samples ; i++) {
            float sample = *inputChannel0++;
            if (sample<-1.0f) sample=-1.0f; else if (sample>1.0f) sample=1.0f;
            outputBuffer[i] = (int16_t) (sample * 32767.0f);
        }
    }
    // Stereo
    else {
        float* inputChannel1 = (float*)audioFrame->extended_data[1];
        for (i=0 ; i<in_samples ; i++) {
             outputBuffer[i*2] = (int16_t) ((*inputChannel0++) * 32767.0f);
             outputBuffer[i*2+1] = (int16_t) ((*inputChannel1++) * 32767.0f);
        }
    }
    // outputBuffer now contains 16-bit PCM!

我留下了几件事情出清晰......在单路夹紧最好应在立体声路径复制。而code可以很容易地优化。

I've left a couple of things out for clarity... the clamping in the mono path should ideally be duplicated in the stereo path. And the code can be easily optimized.

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