Question

我想使用函数SELECT x.* FROM salaries x JOIN ( SELECT emp_no , MAX(to_date) max_to_date FROM salaries GROUP BY emp_no ) y ON y.emp_no = x.emp_no AND y.max_to_date = x.to_date;更新AV音频编码器到avcodec_encode_audio (deprecated)，而不修改现有编码器的结构：

avcodec_encode_audio2

其中：

1）m_handle AVCodecContext

2）dst，uint8_t *目标缓冲区

3）sizeBytes，uint32_t目标缓冲区的大小

4）m_samBuf void *到要编码的数据的输入块（这被转换为：const short int *）

有一种简单的方法吗？

我试着用：

 outBytes = avcodec_encode_audio(m_handle, dst, sizeBytes, (const short int*)m_samBuf);

它编译，但它没有编码任何东西，如果我在输出流管道输出流，我不会在输出音频，这是在升级之前的warking。

我做错了什么？

编码器只接受两种样本格式：

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);

这里是缓冲区的分配方式：

AV_SAMPLE_FMT_S16,         ///< signed 16 bits
AV_SAMPLE_FMT_FLT,         ///< float

Answer 1

avcodec_fill_audio_frame应该让你到那里

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);

FFmpeg音频编码器新编码功能

1 个答案: