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Rennes Emotion Map 2010-11
19 octobre 2011, par
Mis à jour : Juillet 2013
Langue : français
Type : Texte
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Sur d’autres sites (7233)
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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 -
Evolution #2470 : Publier le site (non public par défaut)
27 décembre 2011, par tetue -Bref (j’ai oublié une phrase), la publication du site doit résulter d’une action volontaire du webmestre et non une être surprise : « C’est super, il y a même des statistiques ! Mais... pourquoi y a-t-il des visites alors que je n’ai pas encore publié le site ? » Cf. : (...)
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flv reencode to mp4 for iphone/ipod via ffmpeg and x264 (quality issue)
3 octobre 2011, par zeroasteriskThere are a lot of questions on this topic, and I've read most of them and most of the google search results I could come up with.
When I use FFMPEG to convert a FLV to a iphone3 compatble MP4 file, it just doesn't preserver enough of the quality. Yes, I've worked the hell out of
-sameqand-band-btsettings, text just isn't readable.Next I tried to split the video out and process it directly, using these instructions :
https://sites.google.com/site/linuxencoding/x264-encoding-guideThe problem is myplayer (via ffmpeg) was not able to determine the duration of the FLV (even though the metadata was set).
(I assume) Because of that unknown duration, when I create the MP4 file, the resulting x264 file plays through super-fast while the audio plays at the normal rate.
user@server:/tmp# mplayer -nosound -benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'
MPlayer 1.0rc4-4.4.5 (C) 2000-2010 MPlayer Team
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing input.flv.
libavformat file format detected.
[flv @ 0x1202460]Estimating duration from bitrate, this may be inaccurate
[lavf] stream 0: video (vp6f), -vid 0
[lavf] stream 1: audio (nellymoser), -aid 0
VIDEO: [VP6F] 1680x992 0bpp 1000.000 fps 33.4 kbps ( 4.1 kbyte/s)
Clip info:
audiocodecid: 6
audiodatarate: 86
audiosamplerate: 44100
audiosamplesize: 16
audiosize: 6097005
canSeekToEnd: true
datasize: 8609138
duration: 567
framerate: 2
hasAudio: true
hasCuePoints: false
hasKeyframes: true
hasMetadata: true
hasVideo: true
height: 992
lasttimestamp: 567
metadatacreator: flvtool++ (Facebook, Motion project, dweatherford)
stereo: false
totalframes: 1043
videocodecid: 4
videodatarate: 33
videosize: 2316256
width: 1680
Using (default) progressive frame mode.Opening video filter: [expand w=640 h=480]
Expand: 640 x 480, -1 ; -1, osd: 0, aspect: 0.000000, round: 1
Opening video filter: [scale w=0 h=0]
Opening video filter: [dsize=640:480:0]
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
Selected video codec: [ffvp6f] vfm: ffmpeg (FFmpeg VP6 Flash)
==========================================================================
Audio: no sound
Starting playback...
Movie-Aspect is undefined - no prescaling applied.
[swscaler @ 0x7f0c738b9620]Lanczos scaler, from yuv420p to yuv420p using MMX2
VO: [yuv4mpeg] 640x480 => 641x480 Planar YV12I have also tried specifying FPS, but no change in results
user@server:/tmp# mplayer -nosound -fps 25-benchmark -sws 9 -vf dsize=640:480:0,scale=0:0,expand=640:480 -vo yuv4mpeg:file=>(x264 --demuxer y4m --fps 25 --crf 0 --preset slow --threads auto --output output.264 - 2>x264.log) 'input.flv'Can someone tell me how to either :
- fix my split A/V processing/timing/duration issues ?
- improve the
quality of the FFMPEG conversion of FLV to iphone3 compatible
format ?
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