Android AudioRecord 到 FFMPEG 编码原生 AAC

Question

I am doing video chatting in android and i would like to port ffmpeg to stream rtsp or rtmp but now i have a try in RTSP first. Somehow the problem now is av_write_frame or av_interleaved_write_frame is fail to work or just crash. Maybe... AudioRecord Sample format is not equals to FFMPEG setting Frame receive is not equals

So code... AudioRecorder http://pastebin.com/iWtB3Jhy package com.curtis.broadcaster.Publisher;

import android.app.Activity;
import android.graphics.Bitmap;
import android.media.AudioFormat;
import android.media.AudioRecord;
import android.media.AudioRecord.OnRecordPositionUpdateListener;
import android.media.MediaRecorder;
import android.os.Bundle;
import android.util.Log;

public class Publisher extends Activity {
    private int mAudioBufferSize;
    private int mAudioBufferSampleSize;
    private AudioRecord mAudioRecord;
    private boolean inRecordMode = false;
    private short[] audioBuffer;
    private String Tag = "Publisher/Publisher.java";

    public void onCreate(Bundle savedInstanceState) {
        Log.i(Tag, "|| onCreate()");
        super.onCreate(savedInstanceState);
        initAudioRecord();
        Log.i(Tag, "-- End onCreate()");
    }

    @Override
    public void onResume() {
        Log.i(Tag, "|| onResume()");
        super.onResume();
        inRecordMode = true;
        Thread t = new Thread(new Runnable() {

            public void run() {
                Log.i(Tag, "|| Run Threat t");
                getSamples();
                Log.i(Tag, "-- End Threat t");
            }
        });
        t.start();
        Log.i(Tag, "-- End onResume()");
    }

    protected void onPause() {
        Log.i(Tag, "|| Run onPause()");
        inRecordMode = false;
        super.onPause();
        Log.i(Tag, "-- End onPause()");
    }

    @Override
    protected void onDestroy() {
        Log.i(Tag, "|| Run onDestroy()");
        if (mAudioRecord != null) {
            mAudioRecord.release();
            Log.i(Tag + " onDestroy", "mAudioRecord.release()");
        }
        jniStopAll();
        super.onDestroy();
        android.os.Process.killProcess(android.os.Process.myPid());
        Log.i(Tag, "-- End onDestroy()");
    }

    public OnRecordPositionUpdateListener mListener = new OnRecordPositionUpdateListener() {

        public void onPeriodicNotification(AudioRecord recorder) {
            Log.i(Tag + " mListener(onPeriodicNotification)", "time is "
                    + System.currentTimeMillis());
            jniSetAudioSample(audioBuffer);
        //  audioBuffer = new short[mAudioBufferSampleSize];
        }

        public void onMarkerReached(AudioRecord recorder) {
            Log.i(Tag + " mListener(onMarkerReached)",
                    "time is " + System.currentTimeMillis());
            inRecordMode = false;
            recorder.stop();
            Log.i(Tag, "recorder.stop()");
        }
    };

    private void initAudioRecord() {
        try {
            jniCheck();
            int sampleRate = 44100;
            int channelConfig = AudioFormat.CHANNEL_IN_MONO;
            int audioFormat = AudioFormat.ENCODING_PCM_16BIT;
            mAudioBufferSize = 2 * AudioRecord.getMinBufferSize(sampleRate,
                    channelConfig, audioFormat);
            mAudioBufferSampleSize = mAudioBufferSize / 2;
            Log.i(Tag, "Buffer Size " + mAudioBufferSize);
            Log.i(Tag, "new AudioRecord begin");

            mAudioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC,
                    sampleRate, channelConfig, audioFormat, mAudioBufferSize);
            Log.i(Tag, "new AudioRecord end");

            jniInitFFMpeg();
        } catch (IllegalArgumentException e) {
            Log.i(Tag, "initAudioRecord go Errors");
            e.printStackTrace();
        }

        // mAudioRecord.setNotificationMarkerPosition(10000);
        mAudioRecord.setPositionNotificationPeriod(1024);
        mAudioRecord.setRecordPositionUpdateListener(mListener);

        int audioRecordState = mAudioRecord.getState();
        if (audioRecordState != AudioRecord.STATE_INITIALIZED) {
            finish();
        }

    }

    private void getSamples() {
        Log.i(Tag, "|| getSamples()");
        if (mAudioRecord == null)
            return;

        audioBuffer = new short[mAudioBufferSampleSize];
        mAudioRecord.startRecording();
        int audioRecordingState = mAudioRecord.getRecordingState();
        if (audioRecordingState != AudioRecord.RECORDSTATE_RECORDING) {
            finish();
        }
        while (inRecordMode) {
            int samplesRead = mAudioRecord.read(audioBuffer, 0,
                    mAudioBufferSampleSize);
            Log.i(Tag, "getSamples >>SamplesRead : " + samplesRead);
        }
        mAudioRecord.stop();
        Log.i(Tag, "mAudioRecord.stop()");
    }

    private native void jniCheck();

    private native void jniInitFFMpeg();

    private native void jniSetAudioSample(short[] audioBuffer);

    private native void jniStopAll();

    static {
        System.loadLibrary("ffmpeg");
        System.loadLibrary("testerv4");

    }

}

FFMPEG JNI http://pastebin.com/hgPva35b

#include <jni.h>
#include <android/log.h>
#include <android/bitmap.h>

#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <sys/time.h>
#include "libavformat/rtsp.h"

#include <libavutil/mathematics.h>
#include <libavformat/avformat.h>
#include <libavcodec/avcodec.h>
#include <libswscale/swscale.h>

#undef exit
/* Log System */
#define  LOG_TAG    "FFMPEGSample - v4a"
#define DEBUG_TAG   "FFMPEG-AUDIO PART"
#define  LOGI(...)  __android_log_print(ANDROID_LOG_INFO,LOG_TAG,__VA_ARGS__)
#define  LOGE(...)  __android_log_print(ANDROID_LOG_ERROR,LOG_TAG,__VA_ARGS__)

/* 5 seconds stream duration */
#define STREAM_DURATION   5.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT      PIX_FMT_YUV420P /* default pix_fmt */
#define VIDEO_CODEC_ID      CODEC_ID_FLV1
#define AUDIO_CODEC_ID      CODEC_ID_AAC

static int sws_flags = SWS_BICUBIC;
int mode = 1; //1 = only audio, 2 = only video, 3 = both video and audio

AVFormatContext *avForCtx;
//AVFormatContext *oc;
AVStream *audio_st, *video_st;
double audio_pts, video_pts;
int frameCount, audioFrameCount, start;
char *url;

/*Audio Declare*/
float t, tincr, tincr2;
int16_t *samples;
uint8_t *audio_outbuf;
int audio_outbuf_size;
int audio_input_frame_size;

AVFormatContext *createAVFormatContext();
AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id);
void open_video(AVFormatContext *oc, AVStream *st);
void open_audio(AVFormatContext *oc, AVStream *st);
AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id);
void write_audio_frame(AVFormatContext *oc, AVStream *st);
void write_video_frame(AVFormatContext *oc, AVStream *st);
void init();
void setAudioSample(unsigned char *inSample[]);
void stopAll();

/*/////////////////////////////////JNI Bridge////////////////////////////////////// */
void Java_com_curtis_broadcaster_Publisher_Publisher_jniCheck(JNIEnv* env,
        jobject this) {
    LOGI("-@ JNI work fine @-");
}
void Java_com_curtis_broadcaster_Publisher_Publisher_jniInitFFMpeg(JNIEnv* env,
        jobject this) {
    LOGI("-@ Init Encorder @-");

    /* initialize libavcodec, and register all codecs and formats */
    avcodec_init();
    avcodec_register_all();
    av_register_all();
    avformat_network_init(); //ERROR


    /* allocate the output media context */
    avForCtx = createAVFormatContext();
    frameCount = 1;
    audioFrameCount = 1;
    start = 0;

    /* add the audio and video streams using the default format codecs
     and initialize the codecs */
    video_st = NULL;
    audio_st = NULL;
    if (mode == 1 || mode == 3) {
        audio_st = add_audio_stream(avForCtx, AUDIO_CODEC_ID);
        LOGI("(Init Encorder) - addAudioStream");
    }
    if (mode == 2 || mode == 3) {
        video_st = add_video_stream(avForCtx, VIDEO_CODEC_ID);
        LOGI("(Init Encorder) - addVideoStream");

    }

    //  av_dump_format(avForCtx, 0, "rtsp://192.168.1.104/live/live", 1);
    LOGI("(Init Encorder) - Waiting to call open_*");

    if (audio_st) {
        open_audio(avForCtx, audio_st);
        LOGI("(Init Encorder) - open_audio");
    }

    if (video_st) {
        open_video(avForCtx, video_st);
        LOGI("(Init Encorder) - open_video");
    }

    av_write_header(avForCtx);
    LOGI("-@ Finish Init Encorder @-");

}

void Java_com_curtis_broadcaster_Publisher_Publisher_jniSetAudioSample(
        JNIEnv* env, jobject this, unsigned char *inSample[]) {
    if (audio_st) {
        LOGI("-@ Start setAudioSample @-");
        samples = (int16_t *) inSample;

        write_audio_frame(avForCtx, audio_st);
        LOGI("-@ Finish setAudioSample @-");
    }
}

void Java_com_curtis_broadcaster_Publisher_Publisher_jniStopAll(JNIEnv* env,
        jobject this) {
    LOGI("-@ Stopping All @-");
    //close_audio(avForCtx, audio_st);
    //close_video(avForCtx, video_st);
    LOGI("-@ Stopped All @-");
}
/*/////////////////////////////END JNI Bridge////////////////////////////////////// */

/* New Added Coding */
AVFormatContext *createAVFormatContext() {
    LOGI("-@OPEN - createAVFormatContext@-");

    AVFormatContext *ctx = avformat_alloc_context();
    //  ctx->oformat = av_guess_format("flv", "rtmp://192.168.1.104/live/live",
    //      NULL);
    //  ctx->oformat = av_guess_format("flv", NULL, NULL);

    //if (!av_guess_format("flv", NULL, NULL)) {

    //LOGI("-flv Can not Guess Format-");
    //}

    ctx->oformat = av_guess_format("rtsp", NULL, NULL);

    if (!av_guess_format("rtsp", NULL, NULL)) {

        LOGI("-flv Can not Guess Format-");
    }

    /*
     LOGI("%d",avformat_alloc_output_context2(&ctx, ctx->oformat, "flv",
     "rtmp://192.168.1.104/live/live"));
     if (!ctx) {
     LOGI("-@avformat_alloc_output_context2 fail@-");
     }*/
    //   LOGI("flv %d",avformat_alloc_output_context2(&ctx, ctx->oformat, "flv",
    //   "rtmp://192.168.1.104/live/live"));
    //   LOGI("rtmp %d",avformat_alloc_output_context2(&ctx, ctx->oformat, "rtmp",
    //   "rtmp://192.168.1.104/live/live"));
    //   LOGI("mpeg4 %d",avformat_alloc_output_context2(&ctx, ctx->oformat, "mpeg4",
    //   "rtmp://192.168.1.104/live/live"));
    //   LOGI("NULL %d",avformat_alloc_output_context2(&ctx, ctx->oformat, NULL,
    //   "rtmp://192.168.1.104/live/live"));
    avformat_alloc_output_context2(&ctx, ctx->oformat, "sdp",
            "rtsp://192.168.1.104:1935/live/live");

    if (!ctx) {
        LOGI("-@avformat_alloc_output_context2 fail@-");
    }

    LOGI("-@CLOSE - createAVFormatContext@-");

    return ctx;
}

/**************************************************************/
/* audio output */

/*
 * add an audio output stream
 */
AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id) {
    LOGI("-@OPEN - add_audio_stream@-");

    AVCodecContext *c;
    AVStream *st = avformat_new_stream(oc, avcodec_find_encoder(codec_id));

    if (!st) {
        LOGI("-@add_audio_stream - Could not alloc stream@-");
        exit(1);
    }
    st->id = 1;

    c = st->codec;
    c->codec_id = AUDIO_CODEC_ID;
    c->codec_type = AVMEDIA_TYPE_AUDIO;

    /* put sample parameters */
    c->sample_fmt = AV_SAMPLE_FMT_FLT;
    //c->sample_fmt = AV_SAMPLE_FMT_S16;
    c->bit_rate = 100000;
    c->sample_rate = 44100;
    c->channels = 1;

    // some formats want stream headers to be separate
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
    LOGI("-@Close - add_audio_stream@-");

    return st;
}

void open_audio(AVFormatContext *oc, AVStream *st) {
    LOGI("@- open_audio -@");

    AVCodecContext *c;
    AVCodec *codec;

    c = st->codec;
    c->strict_std_compliance = -2;
    /* find the audio encoder */
    codec = avcodec_find_encoder(c->codec_id);
    if (!codec) {
        LOGI("@- open_audio E:codec not found-@");
        exit(1);
    }

    /* open it */
    if (avcodec_open(c, codec) < 0) {
        LOGI("%d",avcodec_open(c, codec));
        LOGI("@- open_audio E:could not open codec-@");
        exit(1);
    }

    /* init signal generator */
    t = 0;
    tincr = 2 * M_PI * 110.0 / c->sample_rate;
    /* increment frequency by 110 Hz per second */
    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;

    audio_outbuf_size = 10000;
    audio_outbuf = av_malloc(audio_outbuf_size);

    /* ugly hack for PCM codecs (will be removed ASAP with new PCM
     support to compute the input frame size in samples */
    if (c->frame_size <= 1) {
        audio_input_frame_size = audio_outbuf_size / c->channels;
        switch (st->codec->codec_id) {
        case CODEC_ID_PCM_S16LE:
        case CODEC_ID_PCM_S16BE:
        case CODEC_ID_PCM_U16LE:
        case CODEC_ID_PCM_U16BE:
            audio_input_frame_size >>= 1;
            break;
        default:
            break;
        }
    } else {
        audio_input_frame_size = c->frame_size;
    }
    LOGI("audio_input_frame_size : %d",audio_input_frame_size);
    samples = av_malloc(audio_input_frame_size * 2 * c->channels);
    LOGI("@- Close open_audio -@");

}

/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
 'nb_channels' channels */
void get_audio_frame(int16_t *samples, int frame_size, int nb_channels) {
    LOGI("@- get_audio_frame -@");

    int j, i, v;
    int16_t *q;

    q = samples;
    for (j = 0; j < frame_size; j++) {
        v = (int) (sin(t) * 10000);
        for (i = 0; i < nb_channels; i++)
            *q++ = v;
        t += tincr;
        tincr += tincr2;
        LOGI("@- audio_frame Looping -@");
    }
    LOGI("@- CLOSE get_audio_frame -@");

}

void write_audio_frame(AVFormatContext *oc, AVStream *st) {
    LOGI("@- write_audio_frame -@");

    AVCodecContext *c;
    AVPacket pkt;
    av_init_packet(&pkt);

    c = st->codec;

    //get_audio_frame(samples, audio_input_frame_size, c->channels);
    LOGI("@- write_audio_frame : got frame from get_audio_frame -@");

    pkt.size
            = avcodec_encode_audio(c, audio_outbuf, audio_outbuf_size, samples);
    LOGI("%d",pkt.size);

    if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
        pkt.pts
                = av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
    LOGI("%d",pkt.pts);

    pkt.flags |= AV_PKT_FLAG_KEY;
    pkt.stream_index = st->index;
    pkt.data = audio_outbuf;
    LOGI("Finish PKT");

    /* write the compressed frame in the media file */
    //  if (av_interleaved_write_frame(oc, &pkt) != 0) {
    //  LOGI("@- write_audio_frame E:Error while writing audio frame -@");
    //  exit(1);
    //  }

    if (av_interleaved_write_frame(oc, &pkt) != 0) {
        LOGI("Error while writing audio frame %d\n", audioFrameCount);
    } else {
        LOGI("Writing Audio Frame %d", audioFrameCount);
    }

    LOGI("@- CLOSE write_audio_frame -@");
    audioFrameCount++;
    av_free_packet(&pkt);
}

void close_audio(AVFormatContext *oc, AVStream *st) {
    avcodec_close(st->codec);

    av_free(samples);
    av_free(audio_outbuf);
}

/**************************************************************/
/* video output */

AVFrame *picture, *tmp_picture;
uint8_t *video_outbuf;
int frame_count, video_outbuf_size;

/* add a video output stream */
AVStream *add_video_stream(AVFormatContext *oc, enum CodecID codec_id) {
    AVCodecContext *c;
    AVStream *st;
    AVCodec *codec;

    st = avformat_new_stream(oc, NULL);
    if (!st) {
        fprintf(stderr, "Could not alloc stream\n");
        exit(1);
    }

    c = st->codec;

    /* find the video encoder */
    codec = avcodec_find_encoder(codec_id);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }
    avcodec_get_context_defaults3(c, codec);

    c->codec_id = codec_id;

    /* put sample parameters */
    c->bit_rate = 400000;
    /* resolution must be a multiple of two */
    c->width = 352;
    c->height = 288;
    /* time base: this is the fundamental unit of time (in seconds) in terms
     of which frame timestamps are represented. for fixed-fps content,
     timebase should be 1/framerate and timestamp increments should be
     identically 1. */
    c->time_base.den = STREAM_FRAME_RATE;
    c->time_base.num = 1;
    c->gop_size = 12; /* emit one intra frame every twelve frames at most */
    c->pix_fmt = STREAM_PIX_FMT;
    if (c->codec_id == CODEC_ID_MPEG2VIDEO) {
        /* just for testing, we also add B frames */
        c->max_b_frames = 2;
    }
    if (c->codec_id == CODEC_ID_MPEG1VIDEO) {
        /* Needed to avoid using macroblocks in which some coeffs overflow.
         This does not happen with normal video, it just happens here as
         the motion of the chroma plane does not match the luma plane. */
        c->mb_decision = 2;
    }
    // some formats want stream headers to be separate
    if (oc->oformat->flags & AVFMT_GLOBALHEADER)
        c->flags |= CODEC_FLAG_GLOBAL_HEADER;

    return st;
}

AVFrame *alloc_picture(enum PixelFormat pix_fmt, int width, int height) {
    AVFrame * picture;
    uint8_t *picture_buf;
    int size;

    picture = avcodec_alloc_frame();
    if (!picture)
        return NULL;
    size = avpicture_get_size(pix_fmt, width, height);
    picture_buf = av_malloc(size);
    if (!picture_buf) {
        av_free(picture);
        return NULL;
    }
    avpicture_fill((AVPicture *) picture, picture_buf, pix_fmt, width, height);
    return picture;
}

void open_video(AVFormatContext *oc, AVStream *st) {
    AVCodec *codec;
    AVCodecContext *c;

    c = st->codec;

    /* find the video encoder */
    codec = avcodec_find_encoder(c->codec_id);
    if (!codec) {
        fprintf(stderr, "codec not found\n");
        exit(1);
    }

    /* open the codec */
    if (avcodec_open(c, codec) < 0) {
        fprintf(stderr, "could not open codec\n");
        exit(1);
    }

    video_outbuf = NULL;
    if (!(oc->oformat->flags & AVFMT_RAWPICTURE)) {
        /* allocate output buffer */
        /* XXX: API change will be done */
        /* buffers passed into lav* can be allocated any way you prefer,
         as long as they're aligned enough for the architecture, and
         they're freed appropriately (such as using av_free for buffers
         allocated with av_malloc) */
        video_outbuf_size = 200000;
        video_outbuf = av_malloc(video_outbuf_size);
    }

    /* allocate the encoded raw picture */
    picture = alloc_picture(c->pix_fmt, c->width, c->height);
    if (!picture) {
        fprintf(stderr, "Could not allocate picture\n");
        exit(1);
    }

    /* if the output format is not YUV420P, then a temporary YUV420P
     picture is needed too. It is then converted to the required
     output format */
    tmp_picture = NULL;
    if (c->pix_fmt != PIX_FMT_YUV420P) {
        tmp_picture = alloc_picture(PIX_FMT_YUV420P, c->width, c->height);
        if (!tmp_picture) {
            fprintf(stderr, "Could not allocate temporary picture\n");
            exit(1);
        }
    }
}

/* prepare a dummy image */
void fill_yuv_image(AVFrame *pict, int frame_index, int width, int height) {
    int x, y, i;

    i = frame_index;

    /* Y */
    for (y = 0; y < height; y++) {
        for (x = 0; x < width; x++) {
            pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
        }
    }

    /* Cb and Cr */
    for (y = 0; y < height / 2; y++) {
        for (x = 0; x < width / 2; x++) {
            pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
            pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
        }
    }
}

void write_video_frame(AVFormatContext *oc, AVStream *st) {
    int out_size, ret;
    AVCodecContext *c;
    struct SwsContext *img_convert_ctx;

    c = st->codec;

    if (frame_count >= STREAM_NB_FRAMES) {
        /* no more frame to compress. The codec has a latency of a few
         frames if using B frames, so we get the last frames by
         passing the same picture again */
    } else {
        if (c->pix_fmt != PIX_FMT_YUV420P) {
            /* as we only generate a YUV420P picture, we must convert it
             to the codec pixel format if needed */
            if (img_convert_ctx == NULL) {
                img_convert_ctx = sws_getContext(c->width, c->height,
                        PIX_FMT_YUV420P, c->width, c->height, c->pix_fmt,
                        sws_flags, NULL, NULL, NULL);
                if (img_convert_ctx == NULL) {
                    fprintf(stderr,
                            "Cannot initialize the conversion context\n");
                    exit(1);
                }
            }
            fill_yuv_image(tmp_picture, frame_count, c->width, c->height);
            sws_scale(img_convert_ctx, tmp_picture->data,
                    tmp_picture->linesize, 0, c->height, picture->data,
                    picture->linesize);
        } else {
            fill_yuv_image(picture, frame_count, c->width, c->height);
        }
    }

    if (oc->oformat->flags & AVFMT_RAWPICTURE) {
        /* raw video case. The API will change slightly in the near
         future for that. */
        AVPacket pkt;
        av_init_packet(&pkt);

        pkt.flags |= AV_PKT_FLAG_KEY;
        pkt.stream_index = st->index;
        pkt.data = (uint8_t *) picture;
        pkt.size = sizeof(AVPicture);

        ret = av_interleaved_write_frame(oc, &pkt);
    } else {
        /* encode the image */
        out_size = avcodec_encode_video(c, video_outbuf, video_outbuf_size,
                picture);
        /* if zero size, it means the image was buffered */
        if (out_size > 0) {
            AVPacket pkt;
            av_init_packet(&pkt);

            if (c->coded_frame->pts != AV_NOPTS_VALUE)
                pkt.pts = av_rescale_q(c->coded_frame->pts, c->time_base,
                        st->time_base);
            if (c->coded_frame->key_frame)
                pkt.flags |= AV_PKT_FLAG_KEY;
            pkt.stream_index = st->index;
            pkt.data = video_outbuf;
            pkt.size = out_size;

            /* write the compressed frame in the media file */
            ret = av_interleaved_write_frame(oc, &pkt);
        } else {
            ret = 0;
        }
    }
    if (ret != 0) {
        fprintf(stderr, "Error while writing video frame\n");
        exit(1);
    }
    frame_count++;
}

void close_video(AVFormatContext *oc, AVStream *st) {
    avcodec_close(st->codec);
    av_free(picture->data[0]);
    av_free(picture);
    if (tmp_picture) {
        av_free(tmp_picture->data[0]);
        av_free(tmp_picture);
    }
    av_free(video_outbuf);
}

Android Manifest has been set and init everything. Please give me some ideas.. Some log message to yours http://pastebin.com/uPD5LyH2

Answer 1

I know its probably too late to answer this question but just in case if it helps you or someone else who stumbles upon the same question in the future, here's a workaround.

I had been working on a similar kind of project but the difference is that rather than using JNI and compiled Native FFmpeg Shared Libraries, I opted for a compiled native binary and used the Java Process API to communicate with the binary.

FFmpeg needs to be prompted of the nature of the incoming data. The audio frame created by AudioRecord is PCM - 16 bit encoded, and you don't seem to have specified the format of the incoming audio stream FFmpeg.

The command issued to ffmpeg could be as follows:

ffmpeg -f u16le -acodec pcm_s16le -i - -acodec <output-file-codec> <rtsp-stream-address>

The audio data received from the audio source was written to the input stream of the FFmpeg process.

The audio data could also be streamed to ffmpeg through a pipe. ParcelFileDescriptor.createPipe() could be used to create pipes on the Android Platform and the command line replacement for -i - would be -i pipe:<fd> where fd is the file descriptor of the read side of the created pipe.

I'd rather suggest to access ffmpeg through command line interface rather than using JNI, as it is well documented and could also be beneficial in detecting the problem using the debug log level.