【发布时间】:2021-04-01 09:13:07
【问题描述】:
我有文件.wav,我需要在.mp3 中进行转换,我正在使用MediaFoundation。这是我使用的方法:
#include "TV_AudioEncoderMF.h"
#include <windows.h>
#include <windowsx.h>
#include <atlstr.h>
#include <comdef.h>
#include <exception>
#include <mfapi.h>
#include <mfplay.h>
#include <mfreadwrite.h>
#include <mmdeviceapi.h>
#include <Audioclient.h>
#include <mferror.h>
#include <Wmcodecdsp.h>
#pragma comment(lib, "mf.lib")
#pragma comment(lib, "mfplat.lib")
#pragma comment(lib, "mfplay.lib")
#pragma comment(lib, "mfreadwrite.lib")
#pragma comment(lib, "mfuuid.lib")
#pragma comment(lib, "wmcodecdspuuid")
TV_AudioEncoderMF::TV_AudioEncoderMF()
{
}
TV_AudioEncoderMF::~TV_AudioEncoderMF()
{
}
template <class T> void SafeRelease(T **ppT)
{
if (*ppT)
{
(*ppT)->Release();
*ppT = nullptr;
}
}
HRESULT TV_AudioEncoderMF::GetOutputMediaTypes(
GUID cAudioFormat,
UINT32 cSampleRate,
UINT32 cBitPerSample,
UINT32 cChannels,
IMFMediaType **ppType
)
{
// Enumerate all codecs except for codecs with field-of-use restrictions.
// Sort the results.
DWORD dwFlags =
(MFT_ENUM_FLAG_ALL & (~MFT_ENUM_FLAG_FIELDOFUSE)) |
MFT_ENUM_FLAG_SORTANDFILTER;
IMFCollection *pAvailableTypes = NULL; // List of audio media types.
IMFMediaType *pAudioType = NULL; // Corresponding codec.
HRESULT hr = MFTranscodeGetAudioOutputAvailableTypes(
cAudioFormat,
dwFlags,
NULL,
&pAvailableTypes
);
// Get the element count.
DWORD dwMTCount;
hr = pAvailableTypes->GetElementCount(&dwMTCount);
// Iterate through the results and check for the corresponding codec.
for (DWORD i = 0; i < dwMTCount; i++)
{
hr = pAvailableTypes->GetElement(i, (IUnknown**)&pAudioType);
GUID majorType;
hr = pAudioType->GetMajorType(&majorType);
GUID subType;
hr = pAudioType->GetGUID(MF_MT_SUBTYPE, &subType);
if (majorType != MFMediaType_Audio || subType != MFAudioFormat_FLAC)
{
continue;
}
UINT32 sampleRate = NULL;
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_SAMPLES_PER_SECOND,
&sampleRate
);
UINT32 bitRate = NULL;
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_BITS_PER_SAMPLE,
&bitRate
);
UINT32 channels = NULL;
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_NUM_CHANNELS,
&channels
);
if (sampleRate == cSampleRate
&& bitRate == cBitPerSample
&& channels == cChannels)
{
// Found the codec.
// Jump out!
break;
}
}
// Add the media type to the caller
*ppType = pAudioType;
(*ppType)->AddRef();
SafeRelease(&pAudioType);
return hr;
}
void TV_AudioEncoderMF::decode()
{
HRESULT hr = S_OK;
// Initialize com interface
CoInitializeEx(0, COINIT_MULTITHREADED);
// Start media foundation
MFStartup(MF_VERSION);
IMFMediaType *pInputType = NULL;
IMFSourceReader *pSourceReader = NULL;
IMFMediaType *pOuputMediaType = NULL;
IMFSinkWriter *pSinkWriter = NULL;
// Create source reader
hr = MFCreateSourceReaderFromURL(
L"D:\\buffer\\del\\out\\test.wav",
NULL,
&pSourceReader
);
// Create sink writer
hr = MFCreateSinkWriterFromURL(
L"D:\\buffer\\del\\out\\test_out.mp3",
NULL,
NULL,
&pSinkWriter
);
// Get media type from source reader
hr = pSourceReader->GetCurrentMediaType(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
&pInputType
);
// Get sample rate, bit rate and channels
UINT32 sampleRate = NULL;
hr = pInputType->GetUINT32(
MF_MT_AUDIO_SAMPLES_PER_SECOND,
&sampleRate
);
UINT32 bitRate = NULL;
hr = pInputType->GetUINT32(
MF_MT_AUDIO_BITS_PER_SAMPLE,
&bitRate
);
UINT32 channels = NULL;
hr = pInputType->GetUINT32(
MF_MT_AUDIO_NUM_CHANNELS,
&channels
);
// Try to find a media type that is fitting.
hr = GetOutputMediaTypes(
MFAudioFormat_MP3,
sampleRate,
bitRate,
channels,
&pOuputMediaType);
DWORD dwWriterStreamIndex = -1;
// Add the stream
hr = pSinkWriter->AddStream(
pOuputMediaType,
&dwWriterStreamIndex
);
// Set input media type
hr = pSinkWriter->SetInputMediaType(
dwWriterStreamIndex,
pInputType,
NULL
);
// Tell the sink writer to accept data
hr = pSinkWriter->BeginWriting();
// Forever alone loop
while (true)
{
DWORD nStreamIndex, nStreamFlags;
LONGLONG nTime;
IMFSample *pSample;
// Read through the samples until...
hr = pSourceReader->ReadSample(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
0,
&nStreamIndex,
&nStreamFlags,
&nTime,
&pSample);
if (pSample)
{
hr = pSinkWriter->WriteSample(
dwWriterStreamIndex,
pSample
);
}
// ... we are at the end of the stream...
if (nStreamFlags & MF_SOURCE_READERF_ENDOFSTREAM)
{
// ... and jump out.
break;
}
}
// Call finalize to finish writing.
hr = pSinkWriter->Finalize();
// Done :D
}
问题是 - 音频质量有很大差异,当我播放(通过 win 标准播放器).wav 文件时听起来不错,但是当我播放压缩到 .mp3 文件时声音听起来像是人录了他的声音在质量很差的录音机上。
这里可能存在什么问题?我没有看到任何可能的方式来设置质量,比如setOutQualityInPersent(100)
编辑
void co_AudioEncoderMF::decode()
{
HRESULT hr = S_OK;
// Initialize com interface
CoInitializeEx(0, COINIT_MULTITHREADED);
// Start media foundation
MFStartup(MF_VERSION);
IMFMediaType *pInputType = NULL;
IMFSourceReader *pSourceReader = NULL;
IMFMediaType *pOuputMediaType = NULL;
IMFSinkWriter *pSinkWriter = NULL;
// Create source reader
hr = MFCreateSourceReaderFromURL(
L"D:\\buffer\\del\\out\\test.wav",
NULL,
&pSourceReader
);
// Create sink writer
hr = MFCreateSinkWriterFromURL(
L"D:\\buffer\\del\\out\\test_out.mp3",
NULL,
NULL,
&pSinkWriter
);
// Get media type from source reader
hr = pSourceReader->GetCurrentMediaType(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
&pInputType
);
// Get sample rate, bit rate and channels
UINT32 sampleRate = NULL;
hr = pInputType->GetUINT32(
MF_MT_AUDIO_SAMPLES_PER_SECOND,
&sampleRate
);
UINT32 bitRate = NULL;
hr = pInputType->GetUINT32(
MF_MT_AUDIO_BITS_PER_SAMPLE,
&bitRate
);
UINT32 channels = NULL;
hr = pInputType->GetUINT32(
MF_MT_AUDIO_NUM_CHANNELS,
&channels
);
// Try to find a media type that is fitting.
hr = GetOutputMediaTypes(
MFAudioFormat_MP3,
sampleRate,
bitRate,
channels,
&pOuputMediaType);
bitRate = bitRate + 2; <------- This line
pOuputMediaType->SetUINT32(MF_MT_AUDIO_BITS_PER_SAMPLE, bitRate); <------- This line
DWORD dwWriterStreamIndex = -1;
// Add the stream
hr = pSinkWriter->AddStream(
pOuputMediaType,
&dwWriterStreamIndex
);
// Set input media type
hr = pSinkWriter->SetInputMediaType(
dwWriterStreamIndex,
pInputType,
NULL
);
// Tell the sink writer to accept data
hr = pSinkWriter->BeginWriting();
// Forever alone loop
while (true)
{
DWORD nStreamIndex, nStreamFlags;
LONGLONG nTime;
IMFSample *pSample;
// Read through the samples until...
hr = pSourceReader->ReadSample(
MF_SOURCE_READER_FIRST_AUDIO_STREAM,
0,
&nStreamIndex,
&nStreamFlags,
&nTime,
&pSample);
if (pSample)
{
hr = pSinkWriter->WriteSample(
dwWriterStreamIndex,
pSample
);
}
// ... we are at the end of the stream...
if (nStreamFlags & MF_SOURCE_READERF_ENDOFSTREAM)
{
// ... and jump out.
break;
}
}
// Call finalize to finish writing.
hr = pSinkWriter->Finalize();
// Done :D
}
EDIT2
有 2 个文件 - https://drive.google.com/drive/folders/1yzB2u0TvMSnwsTpYnDDPFBDkTB75ZFwM?usp=sharing
结果和来源
【问题讨论】:
-
与解码质量无关 - 增加您在音频编码代码中明确要求的编码比特率。
-
@RomanR。我只有一个地方使用
bitrate- 这里是GetOutputMediaTypes,但它看起来不像你的意思 -
@RomanR。我还尝试通过使用这种方法来提高输出比特率 -
pOuputMediaType->SetUINT32(MF_MT_AUDIO_SAMPLES_PER_SECOND, 44000);my 相反,我听到的只是短促的抽动。提高编码比特率的正确方法是什么? -
在将
pOuputMediaType对象传递给AddStream调用之前更新比特率属性(参见MF_MT_AUDIO_AVG_BYTES_PER_SECONDhere 以获取参考)。或发布完整的源代码以获取更具体的源代码编辑/建议。 -
@RomanR。哦,抱歉刚刚注意到我发布了 samperate 的更新(我也只是尝试玩这个 para)。无论如何,结果是一样的,当我尝试使用参数时,我只听到一个 tic 并且输出文件的大小从 ~5Kb 增加到 430Kb。编辑了我的问题,是你的意思吗?
标签: c++ audio codec ms-media-foundation