android ffmpeg编码 ffmpeg android硬件解码

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mob64ca1419a401 2023-10-16 00:38:57

文章标签 android ffmpeg编码音视频实时音视频视频编解码 webrtc 文章分类 Android 移动开发

上一篇我们了解了FFmpeg解码流程、关键函数和结构体，实现了视频解码器。这篇我们来实现下音频的解码器。解码流程和视频的基本一致。FFmpeg解码的音频裸数据是PCM格式，android上播放PCM音频数据可以通过AudioTrack和OpenSL ES来实现。

下面我们下来看下解码的流程

一、音频解码流程

android ffmpeg编码 ffmpeg android硬件解码_android ffmpeg编码

和上一篇的视频解码流程基本一致。需要注意的是音频对音频的重采样，以及不同样本格式的数据的排列方式

1.1 音频解码流程

avformat_open_input 打开媒体文件
avformat_find_stream_info 初始化AVFormatContext_
匹配到音频流的index
avcodec_find_decoder 根据音频流信息的codec_id找到对应的解码器_
avcodec_open2 使用给定的AVCodec初始化AVCodecContext_
初始化输出文件、解码AVPacket和AVFrame结构体
申请重采样SwrContext上下文并进行重采样初始化
av_read_frame 开始一帧一帧读取
avcodec_send_packet
avcodec_receive_frame
swr_convert重采样
写入到PCM文件或者使用AudioTrack、OpenSL ES进行播放
释放资源

android ffmpeg编码 ffmpeg android硬件解码_实时音视频_02

1.2 补充知识

音频采样格式

enum AVSampleFormat {
        AV_SAMPLE_FMT_NONE = -1,
        AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
        AV_SAMPLE_FMT_S16,         ///< signed 16 bits
        AV_SAMPLE_FMT_S32,         ///< signed 32 bits
        AV_SAMPLE_FMT_FLT,         ///< float
        AV_SAMPLE_FMT_DBL,         ///< double
    
        AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
        AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
        AV_SAMPLE_FMT_S32P,        ///< signed 32 bits, planar
        AV_SAMPLE_FMT_FLTP,        ///< float, planar
        AV_SAMPLE_FMT_DBLP,        ///< double, planar
        AV_SAMPLE_FMT_S64,         ///< signed 64 bits
        AV_SAMPLE_FMT_S64P,        ///< signed 64 bits, planar
    
        AV_SAMPLE_FMT_NB           ///< Number of sample formats. DO NOT USE if linking dynamically
    };

带P和不带P，关系到了AVFrame中的data的数据排列，不带P，则是LRLRLRLRLR排列，带P则是LLLLLRRRRR排列，若是双通道则带P则意味着data[0]全是L，data[1]全是R（注意：这是采样点不是字节），PCM播放器播放的文件需要的是LRLRLRLR的。

二、解码pcm代码实现

具体实现见代码和详细注释

#include <jni.h>
#include <string>
#include <unistd.h>


extern "C" {
#include "include/libavcodec/avcodec.h"
#include "include/libavformat/avformat.h"
#include "include/log.h"
#include <libswscale/swscale.h>
#include <libavutil/imgutils.h>
#include <libswresample/swresample.h>

}



extern "C"
JNIEXPORT jint JNICALL
Java_android_spport_mylibrary2_Demo_decodeAudio(JNIEnv *env, jobject thiz, jstring video_path,
                                                jstring pcm_path) {

    //申请avFormatContext空间，记得要释放
    AVFormatContext *pFormatContext = avformat_alloc_context();

    const char *url = env->GetStringUTFChars(video_path, 0);

    //1. 打开媒体文件
    int result = avformat_open_input(&pFormatContext, url, NULL, NULL);
    if (result != 0) {
        LOGE("open input error url =%s,result=%d", url, result);
        return -1;
    }
    //2.读取媒体文件信息，给avFormatContext赋值
    result = avformat_find_stream_info(pFormatContext, NULL);
    if (result < 0) {
        LOGE("open input avformat_find_stream_info,result=%d", result);
        return -1;
    }
    3. 匹配到音频流的index
    int audioIndex = -1;
    for (int i = 0; i < pFormatContext->nb_streams; ++i) {
        AVMediaType codecType = pFormatContext->streams[i]->codecpar->codec_type;
        if (AVMEDIA_TYPE_AUDIO == codecType) {
            audioIndex = i;
            break;
        }
    }
    if (audioIndex == -1) {
        LOGE("not find a audio stream");
        return -1;
    }

    AVCodecParameters *pCodecParameters = pFormatContext->streams[audioIndex]->codecpar;

    //4. 根据流信息的codec_id找到对应的解码器
    AVCodec *pCodec = avcodec_find_decoder(pCodecParameters->codec_id);

    if (pCodec == NULL) {
        LOGE("Couldn`t find Codec");
        return -1;
    }

    AVCodecContext *pCodecContext = pFormatContext->streams[audioIndex]->codec;

    //5.使用给定的AVCodec初始化AVCodecContext
    int openResult = avcodec_open2(pCodecContext, pCodec, NULL);
    if (openResult < 0) {
        LOGE("avcodec open2 result %d", openResult);
        return -1;
    }

    const char *pcmPathStr = env->GetStringUTFChars(pcm_path, NULL);

    //新建一个二进制文件，已存在的文件将内容清空，允许读写
    FILE *pcmFile = fopen(pcmPathStr, "wb+");
    if (pcmFile == NULL) {
        LOGE(" fopen outPut file error");
        return -1;
    }

    //6. 初始化输出文件、解码AVPacket和AVFrame结构体
    auto *packet = (AVPacket *) av_malloc(sizeof(AVPacket));

    AVFrame *pFrame = av_frame_alloc();

    //7. 申请重采样SwrContext上下文
    SwrContext *swrContext = swr_alloc();

    int numBytes = 0;
    uint8_t *outData[2] = {0};
    int dstNbSamples = 0;                           // 解码目标的采样率

    int outChannel = 2;                             // 重采样后输出的通道
    //带P和不带P，关系到了AVFrame中的data的数据排列，不带P，则是LRLRLRLRLR排列，带P则是LLLLLRRRRR排列，
    // 若是双通道则带P则意味着data[0]全是L，data[1]全是R（注意：这是采样点不是字节），PCM播放器播放的文件需要的是LRLRLRLR的。
    //P表示Planar（平面），其数据格式排列方式为 (特别记住，该处是以点nb_samples采样点来交错，不是以字节交错）:
    //                    LLLLLLRRRRRRLLLLLLRRRRRRLLLLLLRRRRRRL...（每个LLLLLLRRRRRR为一个音频帧）
    //                    而不带P的数据格式（即交错排列）排列方式为：
    //                    LRLRLRLRLRLRLRLRLRLRLRLRLRLRLRLRLRLRL...（每个LR为一个音频样本）

    AVSampleFormat outFormat = AV_SAMPLE_FMT_S16P;  // 重采样后输出的格式
    int outSampleRate = 44100;                          // 重采样后输出的采样率

    // 通道布局与通道数据的枚举值是不同的，需要av_get_default_channel_layout转换
    swrContext = swr_alloc_set_opts(0,                                 // 输入为空，则会分配
                                    av_get_default_channel_layout(outChannel),
                                    outFormat,                         // 输出的采样频率
                                    outSampleRate,                     // 输出的格式
                                    av_get_default_channel_layout(pCodecContext->channels),
                                    pCodecContext->sample_fmt,       // 输入的格式
                                    pCodecContext->sample_rate,      // 输入的采样率
                                    0,
                                    0);

    //重采样初始化
    int swrInit = swr_init(swrContext);
    if (swrInit < 0) {
        LOGE("swr init error swrInit=%d", swrInit);
        return -1;
    }

    auto *outPcmBuffer = (uint8_t *) av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);

    int frame_cnt = 0;

    outData[0] = (uint8_t *) av_malloc(1152 * 8);
    outData[1] = (uint8_t *) av_malloc(1152 * 8);

    //8. 开始一帧一帧读取
    while (av_read_frame(pFormatContext, packet) >= 0) {
        if (packet->stream_index == audioIndex) {
            //9。将封装包发往解码器
            int ret = avcodec_send_packet(pCodecContext, packet);
            if (ret) {
                LOGE("Failed to avcodec_send_packet(pAVCodecContext, pAVPacket) ,ret =%d", ret);
                break;
            }
//            LOGI("av_read_frame");
            // 10. 从解码器循环拿取数据帧
            while (!avcodec_receive_frame(pCodecContext, pFrame)) {
                // nb_samples并不是每个包都相同，遇见过第一个包为47，第二个包开始为1152的

                // 获取每个采样点的字节大小
                numBytes = av_get_bytes_per_sample(outFormat);
                //修改采样率参数后，需要重新获取采样点的样本个数
                dstNbSamples = av_rescale_rnd(pFrame->nb_samples,
                                              outSampleRate,
                                              pCodecContext->sample_rate,
                                              AV_ROUND_ZERO);
                // 重采样
                swr_convert(swrContext,
                            outData,
                            dstNbSamples,
                            (const uint8_t **) pFrame->data,
                            pFrame->nb_samples);
                LOGI("avcodec_receive_frame");
                // 第一次显示
                static bool show = true;
                if (show) {
                    LOGE("numBytes pFrame->nb_samples=%d dstNbSamples=%d,numBytes=%d,pCodecContext->sample_rate=%d,outSampleRate=%d", pFrame->nb_samples,
                         dstNbSamples,numBytes,pCodecContext->sample_rate,outSampleRate);
                    show = false;
                }
                // 使用LRLRLRLRLRL（采样点为单位，采样点有几个字节，交替存储到文件，可使用pcm播放器播放）
                for (int index = 0; index < dstNbSamples; index++) {
                    // // 交错的方式写入, 大部分float的格式输出 符合LRLRLRLR点交错模式
                    for (int channel = 0;channel < pCodecContext->channels; channel++)
                    {
                        fwrite((char *) outData[channel] + numBytes * index, 1, numBytes, pcmFile);
                    }
                }
                av_packet_unref(packet);
            }
            frame_cnt++;
        }
    }

    LOGI("frame count is %d", frame_cnt);

    swr_free(&swrContext);
    av_free(outPcmBuffer);
    avcodec_close(pCodecContext);
    avformat_close_input(&pFormatContext);

    env->ReleaseStringUTFChars(video_path, url);

    env->ReleaseStringUTFChars(pcm_path, pcmPathStr);

    return 0;
}

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三、使用AudioTrack播放PCM音频

这一小节我们再上一小节解码输出PCM音频数据的基础上，再Native层调用Java层的AudioTrack进行完成音频的播放。

我们简单回顾下。

public AudioTrack(int streamType, int sampleRateInHz, int channelConfig, int audioFormat,
int bufferSizeInBytes, int mode)
其中采样率sampleRateInHz、声道数channelConfig、音频格式audioFormat以及音频缓冲区大小bufferSizeInBytes 

来看参数streamType以及mode

streamType音频流的类型，有如下几种
AudioManager#STREAM_VOICE_CALL：电话声音AudioManager#STREAM_SYSTEM：系统声音
AudioManager#STREAM_RING：铃声
AudioManager#STREAM_MUSIC：音乐声
AudioManager#STREAM_ALARM：闹铃声
AudioManager#STREAM_NOTIFICATION：通知声

这里我们使用的是AudioManager#STREAM_MUSIC。

下面我们重点看下mode
@param mode streaming or static buffer.
MODE_STATIC and MODE_STREAM

STATIC模式：一次性将所有的数据放到一个固定的buffer，然后直接传送给AudioTrack，简单有效，通常应用于播放铃声或者系统提示音等，占用内存较少的音频数据

STREAM模式：一次一次的将音频数据流写入到AudioTrack对象中，并持续处于阻塞状态，当数据从Java层到Native层执行播放完毕后才返回，这种方式可以避免由于音频过大导致内存占用过多。当然对应的不足就是总是在java和native层进行交互，并且阻塞直到播放完毕，效率损失较大。

我们这里使用STREAM模式相关的方法类如下

package android.spport.mylibrary2;

import android.media.AudioFormat;
import android.media.AudioManager;
import android.media.AudioTrack;
import android.util.Log;

import java.io.File;
import java.io.FileInputStream;
import java.io.IOException;

public class AudioTrackStreamHelper {

    private static final String TAG = "AudioTrackStreamHelper";
    private AudioTrack audioTrack;
    private int sampleRateInHz;
    private int channels;
    private int audioFormat;
    private int bufferSize;
    private int mode = -1;

    private boolean hasPcmFile = false;
    private File pcmFile;
    private Thread audioTrackThread;


    public void initAudioTrackParams(String path) {
        sampleRateInHz = 44100;
        channels = AudioFormat.CHANNEL_OUT_STEREO;
        audioFormat = AudioFormat.ENCODING_PCM_16BIT;
        bufferSize = AudioTrack.getMinBufferSize(sampleRateInHz, channels, audioFormat);

        pcmFile = new File(path);//"raw.pcm"
        if (pcmFile.exists()) {
            hasPcmFile = true;
        }
    }

    private int initAudioTrackWithMode(int mode, int bufferSize) {
        if (audioTrack != null) {
            audioTrack.release();
            audioTrack.setPlaybackPositionUpdateListener(null);
            audioTrack = null;
        }

        audioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, sampleRateInHz, channels, audioFormat, bufferSize, mode);
        if (audioTrack != null) {
            Log.i(TAG, "initAudioTrackWithMode: state="+audioTrack.getState()+" playState="+audioTrack.getPlayState());
            return audioTrack.getState();
        }
        return AudioTrack.STATE_UNINITIALIZED;
    }

    public boolean isHasPcmFile() {
        return hasPcmFile;
    }

    public void play() {
        releaseAudioTrack();

        int state = initAudioTrackWithMode(AudioTrack.MODE_STREAM, bufferSize);
        if (state == AudioTrack.STATE_UNINITIALIZED) {
            Log.e(TAG, "run: state is uninit");
            return;
        }

        audioTrackThread = new Thread(new Runnable() {
            @Override
            public void run() {
                FileInputStream fileInputStream = null;
                try {
                    fileInputStream = new FileInputStream(pcmFile);
                    byte[] buffer = new byte[bufferSize / 2];
                    int readCount;
                    Log.d(TAG, "run: ThreadId=" + Thread.currentThread() + " playState=" + audioTrack.getPlayState());
                    //stream模式，可以先调用play
                    audioTrack.play();
                    while (fileInputStream.available() > 0) {
                        readCount = fileInputStream.read(buffer);
                        if (readCount == AudioTrack.ERROR_BAD_VALUE || readCount == AudioTrack.ERROR_INVALID_OPERATION) {
                            continue;
                        }
                        if (audioTrack == null) {
                            return;
                        } else {
                            Log.i(TAG, "run: audioTrack.getState()" + audioTrack.getState() + " audioTrack.getPlayState()=" + audioTrack.getPlayState());
                        }
//                        audioTrack.getPlayState()
                        //一次一次的写入pcm数据到audioTrack.由于是在子线程中进行write，快速连续点击可能主线程触发了stop或者release，导致子线程write异常：IllegalStateException: Unable to retrieve AudioTrack pointer for write()
                        //所以加playstate的判断
                        if (readCount > 0 && audioTrack != null && audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING && audioTrack.getState() == AudioTrack.STATE_INITIALIZED) {
                            audioTrack.write(buffer, 0, readCount);
                        }
                    }

                } catch (IOException | IllegalStateException e) {
                    e.printStackTrace();
                    Log.e(TAG, "play: " + e.getMessage());
                } finally {
                    if (fileInputStream != null) {
                        try {
                            fileInputStream.close();
                        } catch (IOException e) {
                            e.printStackTrace();
                        }
                    }
                    Log.d(TAG, "playWithStreamMode: end  ThreadID=" + Thread.currentThread());
                }
            }
        });
        audioTrackThread.start();
    }


    public void pausePlay() {
        if (audioTrack != null) {
            if (audioTrack.getState() > AudioTrack.STATE_UNINITIALIZED) {
                audioTrack.pause();
            }
            Log.d(TAG, "pausePlay: isPlaying false getPlayState= " + audioTrack.getPlayState());
        }
        if (audioTrackThread != null) {
            audioTrackThread.interrupt();
        }
    }

    private void releaseAudioTrack() {
        if (audioTrack != null && audioTrack.getState() == AudioTrack.STATE_INITIALIZED) {
            audioTrack.stop();
            audioTrack.release();
            Log.d(TAG, "pausePlay: isPlaying false");
        }
        if (audioTrackThread != null) {
            audioTrackThread.interrupt();
        }
    }

    public void destroy() {
        if (audioTrack != null) {
            audioTrack.release();
            audioTrack = null;
        }
        if (audioTrackThread != null) {
            audioTrackThread.interrupt();
            audioTrackThread = null;
        }
    }
}

由于是Java代码，可以在java层在直接调用，省去了JNI的消耗。

public class MainActivity extends AppCompatActivity {


    private Demo demo;
    AudioTrackStaticModeHelper audioTrackHelper;
    AudioTrackStreamHelper audioTrackStreamHelper;
    @Override
    protected void onCreate(Bundle savedInstanceState) {
        super.onCreate(savedInstanceState);
        setContentView(R.layout.activity_main);

        // Example of a call to a native method
        TextView tv = findViewById(R.id.sample_text);

        checkPermission();

        demo = new Demo();
        tv.setText(demo.stringFromJNI());
        String folderurl= Environment.getExternalStorageDirectory().getPath();
        File externalFilesDir = getExternalFilesDir(null);
        Log.i("MainActivity", "externalFilesDir: "+externalFilesDir);

//        demo.decodeVideo(folderurl+"/input.mp4", externalFilesDir+"/output7.yuv");

        demo.decodeAudio(folderurl+"/input.mp4", externalFilesDir+"/audio.pcm");

        initAudioTrackStreamMode(externalFilesDir);

    }

    private void initAudioTrackStreamMode(File externalFilesDir) {
        audioTrackStreamHelper = new AudioTrackStreamHelper();

        audioTrackStreamHelper.initAudioTrackParams(externalFilesDir+"/audio.pcm");
        audioTrackStreamHelper.play();
    }
}

由于我们FFmpeg解码时同步的，所以可以采用这种方式，但是解码本事是耗时操作，应该创建解码线程，然后播放PCM时也可以直接送给AudioTrack进行播放，而不用先写入到PCM文件再设置播放。这些都是可优化点。我们在后续音视频同步时再进行优化。