[英]Playback duration from MF SinkWriter mp4 file is half the time when adding an audio sample also the playback speed of the images is twice as fast
我为我的c#项目创建了一个托管c ++库,以基于MSDN教程SinkWriter将图像和音频编码到mp4容器中。 为了测试结果是否正确,我创建了一种提供600帧的方法。 此帧代表每秒60帧的10秒视频。
我提供的图像每秒变化一次,我的音频文件包含的声音数达到10。
我面临的问题是输出视频实际上只有5秒长。 视频的元数据显示是10秒,但不是。 而且声音几乎不超过5。
如果我只写没有音频部分的图像样本,则视频的持续时间为预期的10秒。
我在这里想念什么?
这是我的应用程序的某些部分。
这是我用来创建600帧的c#部分,然后在c#部分中也调用PushFrame方法。
var videoFrameCount = 10 * FPS;
SetBinaryImage();
for (int i = 0; i <= videoFrameCount; i++)
{
// New picture every second
if (i > 0 && i % FPS == 0)
{
SetBinaryImage();
}
PushFrame();
}
PushFrame方法将图像和音频数据复制到SinkWriter提供的指针。 然后,我调用SinkWriter的PushFrame方法。
private void PushFrame()
{
try
{
encodeStopwatch.Reset();
encodeStopwatch.Start();
// Video
var frameBufferHandler = GCHandle.Alloc(frameBuffer, GCHandleType.Pinned);
frameBufferPtr = frameBufferHandler.AddrOfPinnedObject();
CopyImageDataToPointer(BinaryImage, ScreenWidth, ScreenHeight, frameBufferPtr);
// Audio
var audioBufferHandler = GCHandle.Alloc(audioBuffer, GCHandleType.Pinned);
audioBufferPtr = audioBufferHandler.AddrOfPinnedObject();
var readLength = audioBuffer.Length;
if (BinaryAudio.Length - (audioOffset + audioBuffer.Length) < 0)
{
readLength = BinaryAudio.Length - audioOffset;
}
if (!EndOfFile)
{
Marshal.Copy(BinaryAudio, audioOffset, (IntPtr)audioBufferPtr, readLength);
audioOffset += audioBuffer.Length;
}
if (readLength < audioBuffer.Length && !EndOfFile)
{
EndOfFile = true;
}
unsafe
{
// Copy video data
var yuv = SinkWriter.VideoCapturerBuffer();
SinkWriter.Encode((byte*)frameBufferPtr, ScreenWidth, ScreenHeight, (int)SWPF.SWPF_RGB, yuv);
// Copy audio data
var audioDestPtr = SinkWriter.AudioCapturerBuffer();
SinkWriter.EncodeAudio((byte*)audioBufferPtr, audioDestPtr);
SinkWriter.PushFrame();
}
encodeStopwatch.Stop();
Console.WriteLine($"YUV frame generated in: {encodeStopwatch.TakeTotalMilliseconds()} ms");
}
catch (Exception ex)
{
}
}
这是我在c ++中添加到SinkWriter中的一些部分。 我猜音频部分的MediaType是可以的,因为音频的播放有效。
rtStart和rtDuration的定义如下:
LONGLONG rtStart = 0;
UINT64 rtDuration;
MFFrameRateToAverageTimePerFrame(fps, 1, &rtDuration);
像这样使用编码器的两个缓冲区
int SinkWriter::Encode(Byte * rgbBuf, int w, int h, int pxFormat, Byte * yufBuf)
{
const LONG cbWidth = 4 * VIDEO_WIDTH;
const DWORD cbBuffer = cbWidth * VIDEO_HEIGHT;
// Create a new memory buffer.
HRESULT hr = MFCreateMemoryBuffer(cbBuffer, &pFrameBuffer);
// Lock the buffer and copy the video frame to the buffer.
if (SUCCEEDED(hr))
{
hr = pFrameBuffer->Lock(&yufBuf, NULL, NULL);
}
if (SUCCEEDED(hr))
{
// Calculate the stride
DWORD bitsPerPixel = GetBitsPerPixel(pxFormat);
DWORD bytesPerPixel = bitsPerPixel / 8;
DWORD stride = w * bytesPerPixel;
// Copy image in yuv pointer
hr = MFCopyImage(
yufBuf, // Destination buffer.
stride, // Destination stride.
rgbBuf, // First row in source image.
stride, // Source stride.
stride, // Image width in bytes.
h // Image height in pixels.
);
}
if (pFrameBuffer)
{
pFrameBuffer->Unlock();
}
// Set the data length of the buffer.
if (SUCCEEDED(hr))
{
hr = pFrameBuffer->SetCurrentLength(cbBuffer);
}
if (SUCCEEDED(hr))
{
return 0;
}
else
{
return -1;
}
return 0;
}
int SinkWriter::EncodeAudio(Byte * src, Byte * dest)
{
DWORD samplePerSecond = AUDIO_SAMPLES_PER_SECOND * AUDIO_BITS_PER_SAMPLE * AUDIO_NUM_CHANNELS;
DWORD cbBuffer = samplePerSecond / 1000;
// Create a new memory buffer.
HRESULT hr = MFCreateMemoryBuffer(cbBuffer, &pAudioBuffer);
// Lock the buffer and copy the video frame to the buffer.
if (SUCCEEDED(hr))
{
hr = pAudioBuffer->Lock(&dest, NULL, NULL);
}
CopyMemory(dest, src, cbBuffer);
if (pAudioBuffer)
{
pAudioBuffer->Unlock();
}
// Set the data length of the buffer.
if (SUCCEEDED(hr))
{
hr = pAudioBuffer->SetCurrentLength(cbBuffer);
}
if (SUCCEEDED(hr))
{
return 0;
}
else
{
return -1;
}
return 0;
}
这是SinkWriter的PushFrame方法,它将SinkWriter,streamIndex,audioIndex,rtStart和rtDuration传递给WriteFrame方法。
int SinkWriter::PushFrame()
{
if (initialized)
{
HRESULT hr = WriteFrame(ptrSinkWriter, stream, audio, rtStart, rtDuration);
if (FAILED(hr))
{
return -1;
}
rtStart += rtDuration;
return 0;
}
return -1;
}
这是结合视频和音频样本的WriteFrame方法。
HRESULT SinkWriter::WriteFrame(IMFSinkWriter *pWriter, DWORD streamIndex, DWORD audioStreamIndex, const LONGLONG& rtStart, const LONGLONG& rtDuration)
{
IMFSample *pVideoSample = NULL;
// Create a media sample and add the buffer to the sample.
HRESULT hr = MFCreateSample(&pVideoSample);
if (SUCCEEDED(hr))
{
hr = pVideoSample->AddBuffer(pFrameBuffer);
}
if (SUCCEEDED(hr))
{
pVideoSample->SetUINT32(MFSampleExtension_Discontinuity, FALSE);
}
// Set the time stamp and the duration.
if (SUCCEEDED(hr))
{
hr = pVideoSample->SetSampleTime(rtStart);
}
if (SUCCEEDED(hr))
{
hr = pVideoSample->SetSampleDuration(rtDuration);
}
// Send the sample to the Sink Writer.
if (SUCCEEDED(hr))
{
hr = pWriter->WriteSample(streamIndex, pVideoSample);
}
// Audio
IMFSample *pAudioSample = NULL;
if (SUCCEEDED(hr))
{
hr = MFCreateSample(&pAudioSample);
}
if (SUCCEEDED(hr))
{
hr = pAudioSample->AddBuffer(pAudioBuffer);
}
// Set the time stamp and the duration.
if (SUCCEEDED(hr))
{
hr = pAudioSample->SetSampleTime(rtStart);
}
if (SUCCEEDED(hr))
{
hr = pAudioSample->SetSampleDuration(rtDuration);
}
// Send the sample to the Sink Writer.
if (SUCCEEDED(hr))
{
hr = pWriter->WriteSample(audioStreamIndex, pAudioSample);
}
SafeRelease(&pVideoSample);
SafeRelease(&pFrameBuffer);
SafeRelease(&pAudioSample);
SafeRelease(&pAudioBuffer);
return hr;
}
问题是音频缓冲区大小的计算错误。 这是正确的计算:
var avgBytesPerSecond = sampleRate * 2 * channels;
var avgBytesPerMillisecond = avgBytesPerSecond / 1000;
var bufferSize = avgBytesPerMillisecond * (1000 / 60);
audioBuffer = new byte[bufferSize];
在我的问题中,我的缓冲区大小为一毫秒。 因此,MF Framework似乎可以加快图像处理速度,因此声音听起来不错。 固定缓冲区大小后,视频将具有我期望的持续时间,声音也没有错误。
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