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c# - 使用带有 C# 的 MediaTranscoder 将 PCM 音频转码为 MP3

问题描述

我正在尝试对从 WebRTC 调用中保存的 PCM 格式的音频文件进行转码。WebRTC 报告的音频流格式为 16 位深度、1 通道和 48000 Hz 采样率。我想将音频转换为 MP3，以便之后可以将音频作为背景音轨添加到我的 Unity UWP 应用程序的屏幕录制中（使用 MediaComposition）。我在第一部分遇到问题：尝试将我的 PCM 音频文件转码为 MP3 文件。当我尝试准备转码时preparedTranscodeResult.CanTranscode正在返回。false以下是我的代码。

StorageFile remoteAudioPCMFile = await StorageFile.GetFileFromPathAsync(Path.Combine(Application.temporaryCachePath, "remote.pcm").Replace("/", "\\"));
StorageFolder tempFolder = await StorageFolder.GetFolderFromPathAsync(Application.temporaryCachePath.Replace("/", "\\"));
StorageFile remoteAudioMP3File = await tempFolder.CreateFileAsync("remote.mp3", CreationCollisionOption.ReplaceExisting);

MediaEncodingProfile profile = MediaEncodingProfile.CreateMp3(AudioEncodingQuality.Auto);
profile.Audio.BitsPerSample = 16;
profile.Audio.ChannelCount = 1;
profile.Audio.SampleRate = 48000;

MediaTranscoder transcoder = new MediaTranscoder();
var preparedTranscodeResult = await transcoder.PrepareFileTranscodeAsync(remoteAudioPCMFile, remoteAudioMP3File, profile);

if (preparedTranscodeResult.CanTranscode)
{
    await preparedTranscodeResult.TranscodeAsync();
}
else
{
    if (remoteAudioPCMFile != null)
    {
        await remoteAudioPCMFile.DeleteAsync();
    }

    if (remoteAudioMP3File != null)
    {
        await remoteAudioMP3File.DeleteAsync();
    }

    switch (preparedTranscodeResult.FailureReason)
    {
        case TranscodeFailureReason.CodecNotFound:
            Debug.LogError("Codec not found.");
            break;
        case TranscodeFailureReason.InvalidProfile:
            Debug.LogError("Invalid profile.");
            break;
        default:
            Debug.LogError("Unknown failure.");
            break;
    }
}

标签： c#unity3daudiouwpms-media-foundation

解决方案

所以我必须做的是FileStream在我开始将数据写入流之前将标题写入我的。我从这篇文章中得到它。

private void WriteWavHeader(FileStream stream, bool isFloatingPoint, ushort channelCount, ushort bitDepth, int sampleRate, int totalSampleCount)
{
    stream.Position = 0;

    // RIFF header.
    // Chunk ID.
    stream.Write(Encoding.ASCII.GetBytes("RIFF"), 0, 4);

    // Chunk size.
    stream.Write(BitConverter.GetBytes((bitDepth / 8 * totalSampleCount) + 36), 0, 4);

    // Format.
    stream.Write(Encoding.ASCII.GetBytes("WAVE"), 0, 4);



    // Sub-chunk 1.
    // Sub-chunk 1 ID.
    stream.Write(Encoding.ASCII.GetBytes("fmt "), 0, 4);

    // Sub-chunk 1 size.
    stream.Write(BitConverter.GetBytes(16), 0, 4);

    // Audio format (floating point (3) or PCM (1)). Any other format indicates compression.
    stream.Write(BitConverter.GetBytes((ushort)(isFloatingPoint ? 3 : 1)), 0, 2);

    // Channels.
    stream.Write(BitConverter.GetBytes(channelCount), 0, 2);

    // Sample rate.
    stream.Write(BitConverter.GetBytes(sampleRate), 0, 4);

    // Bytes rate.
    stream.Write(BitConverter.GetBytes(sampleRate * channelCount * (bitDepth / 8)), 0, 4);

    // Block align.
    stream.Write(BitConverter.GetBytes(channelCount * (bitDepth / 8)), 0, 2);

    // Bits per sample.
    stream.Write(BitConverter.GetBytes(bitDepth), 0, 2);



    // Sub-chunk 2.
    // Sub-chunk 2 ID.
    stream.Write(Encoding.ASCII.GetBytes("data"), 0, 4);

    // Sub-chunk 2 size.
    stream.Write(BitConverter.GetBytes(bitDepth / 8 * totalSampleCount), 0, 4);
}

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