I just cann't seem to get the FFMpeg working with using the library. Everytime I try to convert asf file to wmv. I get the following issue on run time:
[wmv1 # 0x81ee000]error, slice code was 2
[wmv1 # 0x81ee000]header damaged
This my code:
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int out_size, len, in_size;
FILE *f, *outfile;
uint8_t *outbuf;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
av_init_packet(&avpkt);
printf("Audio decoding\n");
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(CODEC_ID_WMV1);
if (!codec) {
fprintf(stderr, "codec not found\n");
return;
}
c= avcodec_alloc_context2(CODEC_TYPE_AUDIO);
/* open it */
if (avcodec_open(c, codec) < 0) {
fprintf(stderr, "could not open codec\n");
return;
}
outbuf = malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "could not open %s\n", filename);
return;
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
return;
}
/* decode until eof */
avpkt.data = inbuf;
len = avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
NSLog(#"%d", avpkt.size);
while (avpkt.size > 0) {
out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
len = avcodec_decode_audio2(c, (short *)outbuf, &out_size, inbuf,len);// avpkt.size);
NSLog(#"%d", len);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
fclose(outfile);
return;
}
if (out_size > 0) {
/* if a frame has been decoded, output it */
fwrite(outbuf, 1, out_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
free(outbuf);
avcodec_close(c);
av_free(c);
}
I have try the command line utilities and it successfully convert the file. Any help would be helpfully. thanks
Make the mistake of opening the wrong file
Related
I'm trying to make the app using FreeTDS. Following are my code
/*
* Purpose: Test to see if row buffering and blobs works correctly.
* Functions: dbbind dbnextrow dbopen dbresults dbsqlexec dbgetrow
*/
#include "common.h"
#include "stdio.h"
#include<stdlib.h>
#include "readline/readline.h"
#include "readline/history.h"
#include "string.h"
int
main(int argc, char **argv)
{
char failed;
int hasil;
char ch;
char* r = malloc(30);
LOGINREC *login;
DBPROCESS *dbproc;
int i;
int p = 0;
char teststr[1024];
DBINT testint;
read_login_info(argc, argv);
fprintf(stdout, "Starting %s\n", argv[0]);
/* Fortify_EnterScope(); */
while(1)
{
char qrcode[50];
char perintahsql[100] = "select * from REF_COBA where id=";
char *alokasi="";
while(ch != '\n') // terminates if user hit enter
{
ch = getchar();
qrcode[p] = ch;
p++;
}
qrcode[p]='\0';
strcpy(r,qrcode);
alokasi = strsep(&r, ",");
strcpy(r,"");
p=0;
ch = "";
dbinit();
dberrhandle(syb_err_handler);
dbmsghandle(syb_msg_handler);
fprintf(stdout, "About to logon\n");
login = dblogin();
DBSETLPWD(login, PASSWORD);
DBSETLUSER(login, USER);
DBSETLAPP(login, "text_buffer");
//DBSETLHOST(login, "ntbox.dntis.ro");
DBSETLHOST(login, "192.168.1.9");
fprintf(stdout, "About to open\n");
dbproc = dbopen(login, SERVER);
if (strlen(DATABASE))
dbuse(dbproc, DATABASE);
dbloginfree(login);
#ifdef MICROSOFT_DBLIB
dbsetopt(dbproc, DBBUFFER, "100");
#else
dbsetopt(dbproc, DBBUFFER, "100", 0);
#endif
dbcmd(dbproc, strcat(perintahsql,alokasi));
dbsqlexec(dbproc);
if (dbresults(dbproc) != SUCCEED)
{
failed = 1;
fprintf(stdout, "Was expecting a result set.");
exit(1);
}
fprintf(stdout, "select\n");
for (i = 1; i <= dbnumcols(dbproc); i++)
printf("col %d is %s\n", i, dbcolname(dbproc, i));
fprintf(stdout, "setelah for\n");
dbbind(dbproc, 1, INTBIND, 0, (BYTE *) & testint);
dbbind(dbproc, 2, CHARBIND, 0, (BYTE *) teststr);
for (i = 1; i < 2; i++)
{
char expected[1024];
sprintf(expected, "row %03d", i);
if (i % 100 == 0)
{
dbclrbuf(dbproc, 100);
}
if (REG_ROW != dbnextrow(dbproc))
{
failed = 1;
fprintf(stderr, "Failed. Expected a row\n");
//exit(1);
}
else
{
fprintf(stdout, "jalanin usb\n");
printf("Read a row of data -> %d %s\n", (int) testint, teststr);
hasil = system("/opt/usb1rc_linux USB1REL-172 R");
sleep(2);
hasil = system("/opt/usb1rc_linux USB1REL-172 r");
}
}
dbfreebuf(dbproc);
dbexit();
}
return 0;
}
I modified the source from FreeTDS, after compilation there are some warning
bash-4.2# make text_buffer
CC text_buffer.o
text_buffer.c: In function 'main':
text_buffer.c:54:8: warning: assignment makes integer from pointer without a cast [enabled by default]
ch = "";
^
text_buffer.c:18:7: warning: variable 'hasil' set but not used [-Wunused-but-set-variable]
int hasil;
^
text_buffer.c:17:8: warning: variable 'failed' set but not used [-Wunused-but-set-variable]
char failed;
^
CC common.o
CCLD text_buffer
I execute the program without error. This program will check input from user then run the sql command. The format of input shall be , e.g 123456,john doe then press enter. After 50-60 times of input, the double-linked list error will happen.
could anybody help why the error happen?
This is the Original code:
#include <stdio.h>
#include <string.h>
#include <assert.h>
#if defined(_WIN32) && !defined(__CYGWIN__)
#include <windows.h>
#else
#include <sys/select.h>
#endif
#include <sphinxbase/err.h>
#include <sphinxbase/ad.h>
#include "pocketsphinx.h"
static const arg_t cont_args_def[] = {
POCKETSPHINX_OPTIONS,
/* Argument file. */
{"-argfile",
ARG_STRING,
NULL,
"Argument file giving extra arguments."},
{"-adcdev",
ARG_STRING,
NULL,
"Name of audio device to use for input."},
{"-infile",
ARG_STRING,
NULL,
"Audio file to transcribe."},
{"-inmic",
ARG_BOOLEAN,
"no",
"Transcribe audio from microphone."},
{"-time",
ARG_BOOLEAN,
"no",
"Print word times in file transcription."},
CMDLN_EMPTY_OPTION
};
static ps_decoder_t *ps;
static cmd_ln_t *config;
static FILE *rawfd;
static void
print_word_times()
{
int frame_rate = cmd_ln_int32_r(config, "-frate");
ps_seg_t *iter = ps_seg_iter(ps);
while (iter != NULL) {
int32 sf, ef, pprob;
float conf;
ps_seg_frames(iter, &sf, &ef);
pprob = ps_seg_prob(iter, NULL, NULL, NULL);
conf = logmath_exp(ps_get_logmath(ps), pprob);
printf("%s %.3f %.3f %f\n", ps_seg_word(iter), ((float)sf / frame_rate),
((float) ef / frame_rate), conf);
iter = ps_seg_next(iter);
}
}
static int
check_wav_header(char *header, int expected_sr)
{
int sr;
if (header[34] != 0x10) {
E_ERROR("Input audio file has [%d] bits per sample instead of 16\n", header[34]);
return 0;
}
if (header[20] != 0x1) {
E_ERROR("Input audio file has compression [%d] and not required PCM\n", header[20]);
return 0;
}
if (header[22] != 0x1) {
E_ERROR("Input audio file has [%d] channels, expected single channel mono\n", header[22]);
return 0;
}
sr = ((header[24] & 0xFF) | ((header[25] & 0xFF) << 8) | ((header[26] & 0xFF) << 16) | ((header[27] & 0xFF) << 24));
if (sr != expected_sr) {
E_ERROR("Input audio file has sample rate [%d], but decoder expects [%d]\n", sr, expected_sr);
return 0;
}
return 1;
}
/*
* Continuous recognition from a file
*/
static void
recognize_from_file()
{
int16 adbuf[2048];
const char *fname;
const char *hyp;
int32 k;
uint8 utt_started, in_speech;
int32 print_times = cmd_ln_boolean_r(config, "-time");
fname = cmd_ln_str_r(config, "-infile");
if ((rawfd = fopen(fname, "rb")) == NULL) {
E_FATAL_SYSTEM("Failed to open file '%s' for reading",
fname);
}
if (strlen(fname) > 4 && strcmp(fname + strlen(fname) - 4, ".wav") == 0) {
char waveheader[44];
fread(waveheader, 1, 44, rawfd);
if (!check_wav_header(waveheader, (int)cmd_ln_float32_r(config, "-samprate")))
E_FATAL("Failed to process file '%s' due to format mismatch.\n", fname);
}
if (strlen(fname) > 4 && strcmp(fname + strlen(fname) - 4, ".mp3") == 0) {
E_FATAL("Can not decode mp3 files, convert input file to WAV 16kHz 16-bit mono before decoding.\n");
}
ps_start_utt(ps);
utt_started = FALSE;
while ((k = fread(adbuf, sizeof(int16), 2048, rawfd)) > 0) {
ps_process_raw(ps, adbuf, k, FALSE, FALSE);
in_speech = ps_get_in_speech(ps);
if (in_speech && !utt_started) {
utt_started = TRUE;
}
if (!in_speech && utt_started) {
ps_end_utt(ps);
hyp = ps_get_hyp(ps, NULL);
if (hyp != NULL)
printf("%s\n", hyp);
if (print_times)
print_word_times();
fflush(stdout);
ps_start_utt(ps);
utt_started = FALSE;
}
}
ps_end_utt(ps);
if (utt_started) {
hyp = ps_get_hyp(ps, NULL);
if (hyp != NULL) {
printf("%s\n", hyp);
if (print_times) {
print_word_times();
}
}
}
fclose(rawfd);
}
/* Sleep for specified msec */
static void
sleep_msec(int32 ms)
{
#if (defined(_WIN32) && !defined(GNUWINCE)) || defined(_WIN32_WCE)
Sleep(ms);
#else
/* ------------------- Unix ------------------ */
struct timeval tmo;
tmo.tv_sec = 0;
tmo.tv_usec = ms * 1000;
select(0, NULL, NULL, NULL, &tmo);
#endif
}
/*
* Main utterance processing loop:
* for (;;) {
* start utterance and wait for speech to process
* decoding till end-of-utterance silence will be detected
* print utterance result;
* }
*/
static void
recognize_from_microphone()
{
ad_rec_t *ad;
int16 adbuf[2048];
uint8 utt_started, in_speech;
int32 k;
char const *hyp;
if ((ad = ad_open_dev(cmd_ln_str_r(config, "-adcdev"),
(int) cmd_ln_float32_r(config,
"-samprate"))) == NULL)
E_FATAL("Failed to open audio device\n");
if (ad_start_rec(ad) < 0)
E_FATAL("Failed to start recording\n");
if (ps_start_utt(ps) < 0)
E_FATAL("Failed to start utterance\n");
utt_started = FALSE;
E_INFO("Ready....\n");
for (;;) {
if ((k = ad_read(ad, adbuf, 2048)) < 0)
E_FATAL("Failed to read audio\n");
ps_process_raw(ps, adbuf, k, FALSE, FALSE);
in_speech = ps_get_in_speech(ps);
if (in_speech && !utt_started) {
utt_started = TRUE;
E_INFO("Listening...\n");
}
if (!in_speech && utt_started) {
/* speech -> silence transition, time to start new utterance */
ps_end_utt(ps);
hyp = ps_get_hyp(ps, NULL );
if (hyp != NULL) {
printf("%s\n", hyp);
fflush(stdout);
}
if (ps_start_utt(ps) < 0)
E_FATAL("Failed to start utterance\n");
utt_started = FALSE;
E_INFO("Ready....\n");
}
sleep_msec(100);
}
ad_close(ad);
}
int
main(int argc, char *argv[])
{
char const *cfg;
config = cmd_ln_parse_r(NULL, cont_args_def, argc, argv, TRUE);
/* Handle argument file as -argfile. */
if (config && (cfg = cmd_ln_str_r(config, "-argfile")) != NULL) {
config = cmd_ln_parse_file_r(config, cont_args_def, cfg, FALSE);
}
if (config == NULL || (cmd_ln_str_r(config, "-infile") == NULL && cmd_ln_boolean_r(config, "-inmic") == FALSE)) {
E_INFO("Specify '-infile <file.wav>' to recognize from file or '-inmic yes' to recognize from microphone.\n");
cmd_ln_free_r(config);
return 1;
}
ps_default_search_args(config);
ps = ps_init(config);
if (ps == NULL) {
cmd_ln_free_r(config);
return 1;
}
E_INFO("%s COMPILED ON: %s, AT: %s\n\n", argv[0], __DATE__, __TIME__);
if (cmd_ln_str_r(config, "-infile") != NULL) {
recognize_from_file();
} else if (cmd_ln_boolean_r(config, "-inmic")) {
recognize_from_microphone();
}
ps_free(ps);
cmd_ln_free_r(config);
return 0;
}
#if defined(_WIN32_WCE)
#pragma comment(linker,"/entry:mainWCRTStartup")
#include <windows.h>
//Windows Mobile has the Unicode main only
int
wmain(int32 argc, wchar_t * wargv[])
{
char **argv;
size_t wlen;
size_t len;
int i;
argv = malloc(argc * sizeof(char *));
for (i = 0; i < argc; i++) {
wlen = lstrlenW(wargv[i]);
len = wcstombs(NULL, wargv[i], wlen);
argv[i] = malloc(len + 1);
wcstombs(argv[i], wargv[i], wlen);
}
//assuming ASCII parameters
return main(argc, argv);
}
#endif
I can compile it by this command:
g++ -o output continuous.cpp -DMODELDIR=\"`pkg-config --variable=modeldir pocketsphinx`\" `pkg-config --cflags --libs pocketsphinx sphinxbase`
And run it by this command : output -inmic yes .
But I like to convert the code as it has no need to get inmic yes and it automatically starts the program from microphone. But I got segmentation fault(core dumped) error when I changed these parts:
static const arg_t cont_args_def= {"-inmic",
ARG_BOOLEAN,
"no",
"Transcribe audio from microphone."};
int main(int argc, char *argv[])
{
config = cmd_ln_parse_r(NULL, cont_args_def, argc, argv, TRUE);
if (cmd_ln_boolean_r(config, "-inmic")) {
recognize_from_microphone();
}
// recognize_from_microphone();
ps_free(ps);
cmd_ln_free_r(config);
return 0;
}
I searched a lot and red the documentation but couldn't understand what's the problem?
Change the last argument passed to cmd_ln_parse_r from TRUE to FALSE.
It has something to do with strict checking.
I figured this out by reading the source code for cmd_ln.c in the sphinxbase code.
I also changed the boolean value for -inmic in cont_args_def from "no" to "yes".
I have to encode .wav file and write it into same file,or other file using
ffmpeg library,here is my code for encoding
-(void)audioencode:(const char *)fileName
{
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
const char *format_name = "wav",
const char *file_url = "/Users/xxxx/Downloads/simple-drum-beat.wav";
avcodec_register_all();
av_register_all();
AVOutputFormat *format = NULL;
for (AVOutputFormat *formatIter = av_oformat_next(NULL); formatIter != NULL; formatIter = av_oformat_next(formatIter)
{
int hasEncoder = NULL != avcodec_find_encoder(formatIter->audio_codec);
if (0 == strcmp(format_name, formatIter->name)) {
format = formatIter;
break;
}
}
AVCodec *codec = avcodec_find_encoder(format->audio_codec);
NSLog(#"tet test tststs");
AVCodecContext *c;
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
c->bit_rate = 64000;//705600;
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->frame_size = av_get_audio_frame_duration(c, 16);
int bits_per_sample = av_get_bits_per_sample(c->codec_id);
int frameSize = av_get_audio_frame_duration(c,16);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(fileName, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", fileName);
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = frameSize/*c->frame_size*/;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
buffer_size = av_samples_get_buffer_size(NULL, c->channels,frameSize /*c->frame_size*/,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
float t, tincr;
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<800;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < frameSize/*c->frame_size*/; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
}
but after encoded file size is zero,
Please suggest what m doing wrong,any help will be appreciate, thanks in advance
How to decode a .wav file and write it into another file using ffmpeg?
I got decoded data by this piece of code:
-(void)audioDecode:(const char *)outfilename inFileName:(const char *)filename
{
const char* input_filename=filename;
//avcodec_register_all();
av_register_all();
avcodec_register_all();
// av_register_all();
//av_ini
// AVFormatContext* container=avformat_alloc_context();
// if(avformat_open_input(&container,filename,NULL,NULL)<0){
// NSLog(#"Could not open file");
// }
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
avcodec_register_all();
AVFrame* frame = av_frame_alloc();
if (!frame)
{
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
AVFormatContext* formatContext = NULL;
/* Opening the file, and check if it has opened */
if (avformat_open_input(&formatContext, filename, NULL, NULL) != 0)
{
av_frame_free(&frame);
NSLog(#"Could not open file");
}
if (avformat_find_stream_info(formatContext, NULL) < 0)
{
av_frame_free(&frame);
avformat_close_input(&formatContext);
NSLog(#"Error finding the stream info");
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* Find the audio Stream, if no audio stream are found, clean and exit */
AVCodec* cdc = NULL;
int streamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &cdc, 0);
if (streamIndex < 0)
{
av_frame_free(&frame);
avformat_close_input(&formatContext);
NSLog(#"Could not find any audio stream in the file");
exit(1);
}
/* Open the audio stream to read data in audioStream */
AVStream* audioStream = formatContext->streams[streamIndex];
/* Initialize the codec context */
AVCodecContext* codecContext = audioStream->codec;
codecContext->codec = cdc;
/* Open the codec, and verify if it has opened */
if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
{
av_frame_free(&frame);
avformat_close_input(&formatContext);
NSLog(#"Couldn't open the context with the decoder");
exit(1);
}
/* Initialize buffer to store compressed packets */
AVPacket readingPacket;
av_init_packet(&readingPacket);
int lenght = 1;
long long int chunk = ((formatContext->bit_rate)*lenght/8);
int j=0;
int count = 0;
//audioChunk output;
while(av_read_frame(formatContext, &readingPacket)==0){
//if((count+readingPacket.size)>start){
if(readingPacket.stream_index == audioStream->index){
AVPacket decodingPacket = readingPacket;
// Audio packets can have multiple audio frames in a single packet
while (decodingPacket.size > 0){
// Try to decode the packet into a frame
// Some frames rely on multiple packets, so we have to make sure the frame is finished before
// we can use it
int gotFrame = 0;
int result = avcodec_decode_audio4(codecContext, frame, &gotFrame, &decodingPacket);
count += result;
if (result >= 0 && gotFrame)
{
decodingPacket.size -= result;
decodingPacket.data += result;
int a;
for(int i=0;i<result-1;i++){
fwrite(frame->data[0], 1, decodingPacket.size, outfile);
// *(output.data+j)=frame->data[0][i];
//
j++;
if(j>=chunk) break;
}
// We now have a fully decoded audio frame
}
else
{
decodingPacket.size = 0;
decodingPacket.data = NULL;
}
// if(j>=chunk) break;
}
}
// }else count+=readingPacket.size;
//
// // To prevent memory leak, must free packet.
// av_free_packet(&readingPacket);
// if(j>=chunk) break;
}
fclose(outfile);
But the file is created with zero bytes. I don't know what's wrong with this code. And I got one more error: "Format adp detected only with low score of 25, misdetection possible!"
I have been trying to produce a "flv" video file in the following sequence:
av_register_all();
// Open video file
if (avformat_open_input(&pFormatCtx, "6.mp4", NULL, NULL) != 0)
return -1; // Couldn't open file
// Retrieve stream information
if (avformat_find_stream_info(pFormatCtx, NULL) < 0)
return -1; // Couldn't find stream information
// Dump information about file onto standard error
av_dump_format(pFormatCtx, 0, "input_file.mp4", 0);
// Find the first video stream
videoStream = -1;
for (i = 0; i < pFormatCtx->nb_streams; i++)
if (pFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
videoStream = i;
break;
}
if (videoStream == -1)
return -1; // Didn't find a video stream
// Get a pointer to the codec context for the video stream
pCodecCtx = pFormatCtx->streams[videoStream]->codec;
// Find the decoder for the video stream
pCodec = avcodec_find_decoder(pCodecCtx->codec_id);
if (pCodec == NULL) {
fprintf(stderr, "Unsupported codec!\n");
return -1; // Codec not found
}
// Open codec
if (avcodec_open2(pCodecCtx, pCodec, NULL) < 0)
return -1; // Could not open codec
// Allocate video frame
pFrame = avcodec_alloc_frame();
// Allocate video frame
pFrame = avcodec_alloc_frame();
// Allocate an AVFrame structure
pFrameYUV420 = avcodec_alloc_frame();
if (pFrameYUV420 == NULL)
return -1;
// Determine required buffer size and allocate buffer
numBytes = avpicture_get_size(pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height);
buffer = (uint8_t *) av_malloc(numBytes * sizeof(uint8_t));
// Assign appropriate parts of buffer to image planes in pFrameYUV420
// Note that pFrameYUV420 is an AVFrame, but AVFrame is a superset of AVPicture
avpicture_fill((AVPicture *) pFrameRGB, buffer, pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height);
// Setup scaler
img_convert_ctx = sws_getContext(pCodecCtx->width, pCodecCtx->height, pCodecCtx->pix_fmt, pCodecCtx->width, pCodecCtx->height, pCodecCtx->pix_fmt, SWS_BILINEAR, 0, 0, 0);
if (img_convert_ctx == NULL) {
fprintf(stderr, "Cannot initialize the conversion context!\n");
exit(1);
}
// Setup encoder/muxing now
filename = "output_file.flv";
fmt = av_guess_format("flv", filename, NULL);
if (fmt == NULL) {
printf("Could not guess format.\n");
return -1;
}
/* allocate the output media context */
oc = avformat_alloc_context();
if (oc == NULL) {
printf("could not allocate context.\n");
return -1;
}
oc->oformat = fmt;
snprintf(oc->filename, sizeof(oc->filename), "%s", filename);
video_st = NULL;
if (fmt->video_codec != AV_CODEC_ID_NONE) {
video_st = add_stream(oc, &video_codec, fmt->video_codec);
}
// Let's see some information about our format
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename, av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n", av_err2str(ret));
return 1;
}
// Setup x264 params
x264_param_t param;
x264_param_default_preset(¶m, "veryfast", "zerolatency");
param.i_threads = 1;
param.i_width = video_st->codec->width;
param.i_height = video_st->codec->height;
param.i_fps_num = STREAM_FRAME_RATE; // 30 fps, same as video
param.i_fps_den = 1;
// Intra refres:
param.i_keyint_max = STREAM_FRAME_RATE;
param.b_intra_refresh = 1;
// Rate control:
param.rc.i_rc_method = X264_RC_CRF;
param.rc.f_rf_constant = 25;
param.rc.f_rf_constant_max = 35;
// For streaming:
param.b_repeat_headers = 1;
param.b_annexb = 1;
x264_param_apply_profile(¶m, "baseline");
x264_t* encoder = x264_encoder_open(¶m);
x264_picture_t pic_in, pic_out;
x264_picture_alloc(&pic_in, X264_CSP_I420, video_st->codec->width, video_st->codec->height);
x264_nal_t* nals;
int i_nals;
// The loop:
// 1. Read frames
// 2. Decode the frame
// 3. Attempt to re-encode using x264
// 4. Write the x264 encoded frame using av_interleaved_write_frame
while (av_read_frame(pFormatCtx, &packet) >= 0) {
// Is this a packet from the video stream?
if (packet.stream_index == videoStream) {
// Decode video frame
avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
// Did we get a video frame?
if (frameFinished) {
sws_scale(img_convert_ctx, pFrame->data, pFrame->linesize, 0, pCodecCtx->height, pic_in.img.plane, pic_in.img.i_stride);
int frame_size = x264_encoder_encode(encoder, &nals, &i_nals, &pic_in, &pic_out);
if (frame_size >= 0) {
if (i_nals < 0)
printf("invalid frame size: %d\n", i_nals);
// write out NALs
for (i = 0; i < i_nals; i++) {
// initalize a packet
AVPacket p;
av_init_packet(&p);
p.data = nals[i].p_payload;
p.size = nals[i].i_payload;
p.stream_index = video_st->index;
p.flags = AV_PKT_FLAG_KEY;
p.pts = AV_NOPTS_VALUE;
p.dts = AV_NOPTS_VALUE;
ret = av_interleaved_write_frame(oc, &p);
}
}
printf("encoded frame #%d\n", frame_count);
frame_count++;
}
}
// Free the packet that was allocated by av_read_frame
av_free_packet(&packet);
}
// Now we free up resources used/close codecs, and finally close our program.
Here is the implementation for the add_stream() function:
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec, enum AVCodecID codec_id) {
AVCodecContext *c;
AVStream *st;
int r;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
st = avformat_new_stream(oc, *codec);
if (!st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
st->id = oc->nb_streams - 1;
c = st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
st->id = 1;
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
break;
case AVMEDIA_TYPE_VIDEO:
avcodec_get_context_defaults3(c, *codec);
c->codec_id = codec_id;
c->bit_rate = 500*1000;
//c->rc_min_rate = 500*1000;
//c->rc_max_rate = 500*1000;
//c->rc_buffer_size = 500*1000;
/* Resolution must be a multiple of two. */
c->width = 1280;
c->height = 720;
/* timebase: 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
* identical to 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 == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_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;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
After the encoding is complete, I check the output file output_file.flv. I notice it's size is very large: 101MB and it does not play. If I use ffmpeg to decode/encode the input file, then I get an output file about 83MB in size (which is about the same size as the original .mp4 file used as input). Also, the 83MB output from just using ffmpeg C api, as opposed to using x264 for the encoding step, plays just fine. Does anyone know where I am going wrong? I have tried researching this for a few days now but with no luck :(. I feel that I am close to making it work, however, I just cannot figure out what I am doing wrong. Thank you!
To produce the correct AVPacket, you should write all nals into the same packet, as it is done in http://ffmpeg.org/doxygen/trunk/libx264_8c_source.html (see encode_nals and X264_frame functions)