我想使用函数 avcodec_encode_audio (deprecated)
更新 AV 音频编码器以avcodec_encode_audio2
,而不修改现有编码器的结构:
outBytes = avcodec_encode_audio(m_handle, dst, sizeBytes, (const short int*)m_samBuf);
哪里:
1) m_handle AVCodecContext
2) 夏令时,uint8_t * 目标缓冲区
3) 大小字节,uint32_t目标缓冲区的大小
4) m_samBuf void * 到要编码的输入数据块(这被强制转换为:const short int*)
有没有简单的方法可以做到这一点?
我尝试:
int gotPack = 1;
memset (&m_Packet, 0, sizeof (m_Packet));
m_Frame = av_frame_alloc();
av_init_packet(&m_Packet);
m_Packet.data = dst;
m_Packet.size = sizeBytes;
uint8_t* buffer = (uint8_t*)m_samBuf;
m_Frame->nb_samples = m_handle->frame_size;
avcodec_fill_audio_frame(m_Frame,m_handle->channels,m_handle->sample_fmt,buffer,m_FrameSize,1);
outBytes = avcodec_encode_audio2(m_handle, &m_Packet, m_Frame, &gotPack);
char error[256];
av_strerror(outBytes,error,256);
if (outBytes<0){
m_server->log(1,1,"Input data: %d, encode function call error: %s n",gotPack, error);
return AUDIOWRAPPER_ERROR;
}
av_frame_free(&m_Frame);
它可以编译,但它不编码任何内容,如果我在 mplayer 上通过管道传输输出流,我不会在这里输出音频,在升级之前一直在警告。
我做错了什么?
编码器仅接受两种示例格式:
AV_SAMPLE_FMT_S16, ///< signed 16 bits
AV_SAMPLE_FMT_FLT, ///< float
以下是缓冲区的分配方式:
free(m_samBuf);
int bps = 2;
if(m_handle->codec->sample_fmts[0] == AV_SAMPLE_FMT_FLT) {
bps = 4;
}
m_FrameSize = bps * m_handle->frame_size * m_handle->channels;
m_samBuf = malloc(m_FrameSize);
m_numSam = 0;
avcodec_fill_audio_frame应该能让你到达那里
memset (&m_Packet, 0, sizeof (m_Packet));
memset (&m_Frame, 0, sizeof (m_Frame));
av_init_packet(&m_Packet);
m_Packet.data = dst;
m_Packet.size = sizeBytes;
m_Frame->nb_samples = //you need to get this value from somewhere, it is the number of samples (per channel) this frame represents
avcodec_fill_audio_frame(m_Frame, m_handle->channels, m_handle->sample_fmt,
buffer,
sizeBytes, 1);
int gotPack = 1;
avcodec_encode_audio2(m_handle, &m_Packet, &m_Frame, &gotPack);