# c-media-codec **Repository Path**: ppnt/c-media-codec ## Basic Information - **Project Name**: c-media-codec - **Description**: 一个轻量级 JNI 音频编解码库,支持,G.711 PCMU,G.711 PCMA,G.722 - **Primary Language**: Unknown - **License**: MIT - **Default Branch**: main - **Homepage**: None - **GVP Project**: No ## Statistics - **Stars**: 0 - **Forks**: 0 - **Created**: 2026-03-10 - **Last Updated**: 2026-03-11 ## Categories & Tags **Categories**: Uncategorized **Tags**: None ## README # c-media-codec 一个轻量级 **JNI 音频编解码与重采样库**，提供： * **G.711 PCMU (μ-law)** * **G.711 PCMA (A-law)** * **G.722 wideband codec** * **SpeexDSP Resampler（高质量音频重采样）** 该库面向 **VoIP / RTP / SIP / WebRTC / Media Server** 场景设计，核心目标是： * **纯 C 实现** * **无额外运行时依赖** * **JNI DirectByteBuffer 零拷贝** * **高性能实时音频处理** --- # Features * G.711 PCMU / PCMA 编解码 * G.722 宽带音频编解码 * 高质量 SpeexDSP 重采样 * JNI DirectByteBuffer **zero-copy pipeline** * 支持 **实时流式处理** * 跨平台构建 支持平台： ``` Linux macOS Windows ``` 适用于： ``` VoIP SIP Server RTP Gateway WebRTC Gateway Media Server Audio Processing Pipeline ``` --- # Project Structure ``` c-media-codec ├── CMakeLists.txt ├── jni │ └── com_litongjava_media_MediaCodec.h ├── readme.md ├── src │ ├── codec │ │ ├── g711.c │ │ ├── g711.h │ │ ├── g722_decode.c │ │ ├── g722_enc_dec.h │ │ ├── g722_encode.c │ │ │ ├── resampler │ │ ├── arch.h │ │ ├── os_support.h │ │ ├── resample.c │ │ ├── resampler_compat.h │ │ ├── speex_resampler.h │ │ ├── speexdsp_types.h │ │ │ ├── jni_media_codec.c │ ├── media_codec_core.c │ ├── media_codec_core.h │ ├── media_resample.c │ └── media_resample.h ``` 模块说明： | 模块 | 说明 | | ---------------- | ----------------- | | codec | G711 / G722 编解码实现 | | resampler | SpeexDSP 重采样实现 | | media_codec_core | codec 统一封装 | | media_resample | 重采样封装 | | jni_media_codec | JNI 接口 | --- # Source Code Origin ## G.711 / G.722 来源： WebRTC third_party codec 实现 ``` modules/third_party/g711 modules/third_party/g722 ``` 原始代码： * [https://github.com/litongjava/webrtc/blob/main/modules/third_party/g711/g711.c](https://github.com/litongjava/webrtc/blob/main/modules/third_party/g711/g711.c) * [https://github.com/litongjava/webrtc/blob/main/modules/third_party/g711/g711.h](https://github.com/litongjava/webrtc/blob/main/modules/third_party/g711/g711.h) * [https://github.com/litongjava/webrtc/blob/main/modules/third_party/g722/g722_encode.c](https://github.com/litongjava/webrtc/blob/main/modules/third_party/g722/g722_encode.c) * [https://github.com/litongjava/webrtc/blob/main/modules/third_party/g722/g722_decode.c](https://github.com/litongjava/webrtc/blob/main/modules/third_party/g722/g722_decode.c) * [https://github.com/litongjava/webrtc/blob/main/modules/third_party/g722/g722_enc_dec.h](https://github.com/litongjava/webrtc/blob/main/modules/third_party/g722/g722_enc_dec.h) 功能： ``` G711 μ-law / A-law encode / decode G722 encoder / decoder ``` 支持码率： ``` 64000 56000 48000 ``` --- ## Resampler 来源： SpeexDSP ``` https://github.com/xiph/speexdsp ``` 使用文件： ``` resample.c speex_resampler.h arch.h os_support.h resampler_compat.h speexdsp_types.h ``` 提供： ``` 高质量音频重采样 低延迟 流式处理 ``` --- # Build 要求： ``` CMake >= 3.15 JDK (JNI headers) C compiler ``` --- ## 创建 build 目录 ``` cmake -S . -B cmake-build-release -DCMAKE_BUILD_TYPE=Release ``` --- ## 编译 ``` cmake --build cmake-build-release --target all ``` --- ## 输出文件 Linux： ``` libc_media_codec.so ``` macOS： ``` libc_media_codec.dylib ``` Windows： ``` c_media_codec.dll ``` --- # Java API Java 类： ``` com.litongjava.media.MediaCodec ``` 包含两部分： ``` Codec API Resampler API ``` --- # Codec API ```java public static native long createEncoder( int codecType, int sampleRate, int channels, int bitrate, int options); public static native void destroyEncoder(long encoder); public static native int encodeDirect( long encoder, ByteBuffer pcm16le, int pcmSamples, ByteBuffer encodedOut); public static native long createDecoder( int codecType, int sampleRate, int channels, int bitrate, int options); public static native void destroyDecoder(long decoder); public static native int decodeDirect( long decoder, ByteBuffer encoded, int encodedLen, ByteBuffer pcmOut); ``` 支持 codec： | Codec | PayloadType | | ----- | ----------- | | PCMU | 0 | | PCMA | 8 | | G722 | 9 | --- # Resampler API ```java public static native long createResampler( int channels, int inputRate, int outputRate, int quality, int options); public static native void destroyResampler(long resampler); public static native int resampleDirect( long resampler, ByteBuffer pcm16leIn, int inputSamplesPerChannel, ByteBuffer pcm16leOut); public static native int resetResampler(long resampler); public static native int setResamplerRate( long resampler, int inputRate, int outputRate); public static native int getResamplerExpectedOutputSamples( long resampler, int inputSamplesPerChannel); ``` --- # Codec Parameters ## G711 | 参数 | 值 | | ---------- | ------ | | sampleRate | 8000 | | channels | 1 | | bitrate | ignore | 20ms frame： ``` PCM samples = 160 encoded bytes = 160 ``` --- ## G722 | 参数 | 值 | | ---------- | --------------------- | | sampleRate | 16000 | | channels | 1 | | bitrate | 64000 / 56000 / 48000 | 20ms frame： | bitrate | encoded size | | ------- | ------------ | | 64000 | 160 bytes | | 56000 | 140 bytes | | 48000 | 120 bytes | PCM： ``` 320 samples ``` --- # RTP Payload Size Table 以下以 **20ms 音频帧**为例。 ## G711 | Codec | PayloadType | SampleRate | PCM Samples | Encoded Bytes | RTP Timestamp Step | | ----- | ----------- | ---------- | ----------- | ------------- | ------------------ | | PCMU | 0 | 8000 | 160 | 160 | 160 | | PCMA | 8 | 8000 | 160 | 160 | 160 | 说明： ``` 8000 * 20ms = 160 samples ``` G711 编码： ``` 1 sample = 1 byte ``` --- ## G722 | Codec | PayloadType | PCM SampleRate | PCM Samples | Bitrate | Encoded Bytes | RTP Timestamp | | ----- | ----------- | -------------- | ----------- | ------- | ------------- | ------------- | | G722 | 9 | 16000 | 320 | 64000 | 160 | 160 | | G722 | 9 | 16000 | 320 | 56000 | 140 | 160 | | G722 | 9 | 16000 | 320 | 48000 | 120 | 160 | 说明： PCM sample： ``` 16000 * 20ms = 320 ``` RTP timestamp： ``` 8000 * 20ms = 160 ``` --- # Resampler Example 示例： ``` 24000 Hz → 8000 Hz ``` ```java int inRate = 24000; int outRate = 8000; long rs = MediaCodec.createResampler( 1, inRate, outRate, 5, 0); ByteBuffer input = ByteBuffer.allocateDirect(960 * 2); ByteBuffer output = ByteBuffer.allocateDirect(320 * 2); int outSamples = MediaCodec.resampleDirect( rs, input, 960, output); ``` --- # JNI Zero Copy JNI 使用： ``` DirectByteBuffer ``` 避免： ``` byte[] copy ``` JNI 获取地址： ``` GetDirectBufferAddress ``` 实现： ``` zero-copy audio pipeline ``` 适用于： ``` RTP VoIP SIP Server Media Gateway Audio Processing ``` --- # Important Notes ### 1 必须使用 DirectByteBuffer 正确： ``` ByteBuffer.allocateDirect(size) ``` 错误： ``` ByteBuffer.wrap(byte[]) ``` --- ### 2 PCM 格式 必须为： ``` 16-bit signed PCM little-endian ``` 建议： ```java ByteBuffer.allocateDirect(size) .order(ByteOrder.LITTLE_ENDIAN) ``` --- ### 3 Resampler 输入输出 格式： ``` PCM 16bit little-endian ``` 支持： ``` 任意采样率转换 流式处理 ``` --- # License 第三方代码来自： * WebRTC * SpeexDSP 许可证： ``` Google BSD-style license BSD-style license ``` 请参考对应项目 License 文件。 --- # Summary c-media-codec 提供： ``` JNI Audio Codec Bridge + G711 / G722 + SpeexDSP Resampler + Zero-copy DirectByteBuffer pipeline ``` 适用于： ``` VoIP RTP Server SIP Server WebRTC Gateway Media Server Audio Gateway ```