我在android中进行视频聊天，我想将ffmpeg移植到流rtsp或rtmp，但现在我首先尝试使用RTSP。 不知何故，问题现在是av_write_frame或av_interleaved_write_frame无法正常工作或崩溃。 也许... AudioRecord样本格式不等于FFMPEG设置 帧接收不等于
所以代码...... AudioRecorder http://pastebin.com/iWtB3Jhy 包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已设置并初始化所有内容。 请给我一些想法.. 一些日志消息发送给您http://pastebin.com/uPD5LyH2
答案 0 :(得分：0)
我知道回答这个问题可能为时已晚，但如果它可以帮助你或其他人在将来偶然发现同一个问题，这是一个解决方法。
我一直在研究类似的项目，但不同之处在于，我选择了一个已编译的本机二进制文件，并使用Java Process API与二进制文件进行通信，而不是使用JNI和编译的Native FFmpeg共享库。 p>
需要提示FFmpeg输入数据的性质。
由AudioRecord创建的音频帧是PCM-16位编码，您似乎没有指定传入音频流FFmpeg的格式。
发给ffmpeg的命令可能如下：
ffmpeg -f u16le -acodec pcm_s16le -i - -acodec <output-file-codec> <rtsp-stream-address>
从音频源接收的音频数据被写入FFmpeg进程的输入流。
音频数据也可以通过管道流式传输到ffmpeg。
ParcelFileDescriptor.createPipe()可用于在Android平台上创建管道，-i -的命令行替换为-i pipe:<fd>，其中fd是创建的读取端的文件描述符管。
我宁愿建议通过命令行界面访问ffmpeg而不是使用JNI，因为它已被充分记录，并且在使用调试日志级别检测问题时也是有益的。