3
私はアンドロイドでビデオチャットをしています。私はffmpegをストリームrtspまたはrtmpに移植したいと思いますが、今はRTSPで試しています。 どうやら問題は今av_write_frameまたはav_interleaved_write_frameがうまく動作しないかクラッシュするだけです。 はたぶん... audiorecordのサンプルフォーマットは 枠を設定FFMPEGに等しいされていないので、コード... AudioRecorder http://pastebin.com/iWtB3Jhy パッケージをcom.curtis.broadcaster.PublisherAndroid AudioRecord to FFMPEGエンコードネイティブAAC
に等しいされていない受信。
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("[email protected] JNI work fine @-");
}
void Java_com_curtis_broadcaster_Publisher_Publisher_jniInitFFMpeg(JNIEnv* env,
jobject this) {
LOGI("[email protected] 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("[email protected] Finish Init Encorder @-");
}
void Java_com_curtis_broadcaster_Publisher_Publisher_jniSetAudioSample(
JNIEnv* env, jobject this, unsigned char *inSample[]) {
if (audio_st) {
LOGI("[email protected] Start setAudioSample @-");
samples = (int16_t *) inSample;
write_audio_frame(avForCtx, audio_st);
LOGI("[email protected] Finish setAudioSample @-");
}
}
void Java_com_curtis_broadcaster_Publisher_Publisher_jniStopAll(JNIEnv* env,
jobject this) {
LOGI("[email protected] Stopping All @-");
//close_audio(avForCtx, audio_st);
//close_video(avForCtx, video_st);
LOGI("[email protected] Stopped All @-");
}
/*/////////////////////////////END JNI Bridge////////////////////////////////////// */
/* New Added Coding */
AVFormatContext *createAVFormatContext() {
LOGI("[email protected] - [email protected]");
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("[email protected]_alloc_output_context2 [email protected]");
}*/
// 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("[email protected]_alloc_output_context2 [email protected]");
}
LOGI("[email protected] - [email protected]");
return ctx;
}
/**************************************************************/
/* audio output */
/*
* add an audio output stream
*/
AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id) {
LOGI("[email protected] - [email protected]");
AVCodecContext *c;
AVStream *st = avformat_new_stream(oc, avcodec_find_encoder(codec_id));
if (!st) {
LOGI("[email protected]_audio_stream - Could not alloc [email protected]");
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("[email protected] - [email protected]");
return st;
}
void open_audio(AVFormatContext *oc, AVStream *st) {
LOGI("@- open_audio [email protected]");
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 [email protected]");
exit(1);
}
/* open it */
if (avcodec_open(c, codec) < 0) {
LOGI("%d",avcodec_open(c, codec));
LOGI("@- open_audio E:could not open [email protected]");
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 [email protected]");
}
/* 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 [email protected]");
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 [email protected]");
}
LOGI("@- CLOSE get_audio_frame [email protected]");
}
void write_audio_frame(AVFormatContext *oc, AVStream *st) {
LOGI("@- write_audio_frame [email protected]");
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 [email protected]");
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 [email protected]");
// 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 [email protected]");
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のマニフェストは、設定され、すべてを初期化されています。 、それはあなたや、将来的に同じ質問時につまずく誰か他の人を助けている場合、私はそのおそらくこの質問に答えるためには遅すぎる知っているが、念のhttp://pastebin.com/uPD5LyH2にあなたに...私に