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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
Autres articles (92)
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L’utiliser, en parler, le critiquer
10 avril 2011La première attitude à adopter est d’en parler, soit directement avec les personnes impliquées dans son développement, soit autour de vous pour convaincre de nouvelles personnes à l’utiliser.
Plus la communauté sera nombreuse et plus les évolutions seront rapides ...
Une liste de discussion est disponible pour tout échange entre utilisateurs. -
Websites made with MediaSPIP
2 mai 2011, parThis page lists some websites based on MediaSPIP.
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Use, discuss, criticize
13 avril 2011, parTalk to people directly involved in MediaSPIP’s development, or to people around you who could use MediaSPIP to share, enhance or develop their creative projects.
The bigger the community, the more MediaSPIP’s potential will be explored and the faster the software will evolve.
A discussion list is available for all exchanges between users.
Sur d’autres sites (10879)
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The New Samples Regime
1er décembre 2011, par Multimedia Mike — GeneralA little while ago, I got a big head over the fact that I owned and controlled the feared and revered MPlayer samples archive. This is the repository that retains more than a decade of multimedia samples.
Conflict
Where once there was one multimedia project (FFmpeg), there are now 2 (also Libav). There were various political and technical snafus regarding the previous infrastructure. I volunteered to take over hosting the vast samples archive (53 GB at the time) at samples.mplayerhq.hu (s.mphq for this post).However, a brand new server is online at samples.libav.org (s.libav for this post).
Policies
The server at s.libav will be the authoritative samples repository going forward. Why does s.libav receive the honor ? Mostly by virtue of having more advanced features. My simple (yet bandwidth-rich) web hosting plan does not provide for rsync or anonymous FTP services, both of which have traditionally been essential for the samples server. In the course of hosting s.mphq for the past few months, a few more discrepancies have come to light– apparently, the symlinks weren’t properly replicated. And perhaps most unusual is that if a directory contains aREADMEfile, it won’t be displayed in the directory listing (which frustrated me greatly when I couldn’t find this README file that I carefully and lovingly crafted years ago).The s.mphq archive will continue to exist — nay, must exist — going forward since there are years’ worth of web links pointing into it. I’ll likely set up a mirroring script that periodically (daily) rsyncs from s.libav to my local machine and then uses lftp (the best facility I have available) to mirror the files up to s.mphq.
Also, since we’re starting fresh with a new upload directory, I think we need to be far more ruthless about policing its content. This means making sure that anything that is uploaded has an accompanying file which explains why it’s there and ideally links the sample to a bug report somewhere. No explanation = sample terminated.
RSS
I think it would be nifty to have an RSS feed that shows the latest samples to appear in the repository. I figure that I can use the Unix ‘find’ command on my local repository in concert with something like PyRSS2Gen to accomplish this goal.Monetization
In the few months that I have been managing the repository, I have had numerous requests for permission to leech the entire collection in one recursive web-suck. These requests often from commercial organizations who wish to test their multimedia product on a large corpus of diverse samples. Personally, I believe the archive makes a rather poor corpus for such an endeavor, but so be it. Go ahead ; hosting this archive barely makes a dent in my fairly low-end web hosting plan. However, at least one person indicated that it might be easier to mail a hard drive to me, have me copy it, and send it back.This got me thinking about monetization opportunities. Perhaps, I should provide a service to send HDs filled with samples for the cost of the HD, shipping, and a small donation to the multimedia projects. I immediately realized that that is precisely the point at which the vast multimedia samples archive — with all of its media of questionable fair use status — would officially run afoul of copyright laws.
Which brings me to…
Clean Up
I think we need to clean up some samples, starting with the ones that were marked not-readable in the old repository. Apparently, some ‘samples’ were, e.g., full anime videos and were responsible for a large bandwidth burden when linked from various sources.We multimedia nerds are a hoarding lot, never willing to throw anything away. This will probably the most challenging proposal to implement.
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parse rtmp stream to detect a particular frame
1er juin 2012, par sccI need to analyse an rtmp stream and when a given frame is streamed (predefined) I need to detect it and later on be able to create several recordings all split by that given frame. For example, stream emits predefined frame, program 1 starts, runs and ends, stream emits predefined frame again to signal program has ended. Now I need to create a video with program 1 and send it to youtube.
Is there any library that can detect the given frame (step 1) (if python or nodejs even better) ? For splitting, ffmpeg should be enough, yes ?
Any pointers very welcome :)
Thanks :)
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Android AudioRecord to FFMPEG encode native AAC
8 mars 2013, par Curtis KiuI 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 equalsSo 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
#include <android></android>log.h>
#include <android></android>bitmap.h>
#include
#include
#include
#include
#include <sys></sys>time.h>
#include "libavformat/rtsp.h"
#include <libavutil></libavutil>mathematics.h>
#include <libavformat></libavformat>avformat.h>
#include <libavcodec></libavcodec>avcodec.h>
#include <libswscale></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
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