I am getting a 'Segmentation fault (core dumped) error'
I know that means I am accessing memory I shouldn't.
The error is coming in the outer else loop with the fwrite function. fwrite(buffer, 1, 512, img);
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
int main(int argc, char *argv[])
{
if (argc != 2)
{
printf("Usage: ./recover IMAGE\n");
return 1;
}
FILE *file = fopen(argv[1], "r");
if (file == NULL)
{
printf("File entered does not exist\n");
return 1;
}
typedef uint8_t BYTE;
BYTE buffer[512];
char filename[8];
int i = 0;
FILE *img;
while (fread(buffer, 1, 512, file) == 512)
{
// check if first three bytes are 0xff 0xd8 0xff
// check if 4th byte is 0xe0, 0xe1, 0xe2, ..., 0xef
// if those 4 bytes are found then start writing these bytes to a file ###.jpg
// once you see the 4 bytes again, stop writing to the current file and make a new file ###.jpg to write to
if (buffer[0] == 0xff && buffer[1] == 0xd8 && buffer[2] == 0xff && (buffer[3] & 0xf0) == 0xe0)
{
if (i == 0)
{
sprintf(filename, "%03i.jpg", i);
img = fopen(filename, "w");
i += 1;
}
else
{
fclose(img);
sprintf(filename, "%03i.jpg", i);
img = fopen(filename, "w");
i += 1;
}
fwrite(buffer, 1, 512, img);
}
else
{
fwrite(buffer, 1, 512, img);
}
}
fclose(img);
return 0;
}
Any ideas, hints or tips would be appreciated.
This is my first time posting in stackoverflow :D
Solved:
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
int main(int argc, char *argv[])
{
if (argc != 2)
{
printf("Usage: ./recover IMAGE\n");
return 1;
}
FILE *file = fopen(argv[1], "r");
if (file == NULL)
{
printf("File entered does not exist\n");
return 1;
}
typedef uint8_t BYTE;
BYTE buffer[512];
char filename[8];
int i = 0;
FILE *img;
while (fread(buffer, 1, 512, file) == 512)
{
// check if first three bytes are 0xff 0xd8 0xff
// check if 4th byte is 0xe0, 0xe1, 0xe2, ..., 0xef
// if those 4 bytes are found then start writing these bytes to a file ###.jpg
// once you see the 4 bytes again, stop writing to the current file and make a new file ###.jpg to write to
if (buffer[0] == 0xff && buffer[1] == 0xd8 && buffer[2] == 0xff && (buffer[3] & 0xf0) == 0xe0)
{
if (i == 0)
{
sprintf(filename, "%03i.jpg", i);
img = fopen(filename, "w");
i += 1;
}
else
{
fclose(img);
sprintf(filename, "%03i.jpg", i);
img = fopen(filename, "w");
i += 1;
}
fwrite(buffer, 1, 512, img);
}
else
{
if (i > 0)
{
fwrite(buffer, 512, 1, img);
}
}
}
fclose(img);
return 0;
}
So I'm doing the CS50 pset4 recover task (where you need to search for jpg files on a memory card and whenever you find one- you open a new file and write the jpg found to the new file). I have written the code in a slightly different manner then what is told in the course but I think(hope) my logic is right. I am able to recover all 50 images but the images are incomplete and distorted. Also for some reason the 050th image is not opening. Here is my code
#include <stdio.h>
#include <stdlib.h>
#include <stdint.h>
typedef uint8_t BYTE; //defining a byte = 8 bits or 8 0's or 1's
int main(int argc, char *argv[]) {
//assigning variables
int BLOCK_SIZE = 512;
BYTE buffer[BLOCK_SIZE];
int i = 0;
char* nf = malloc(sizeof(int)*3);
//checking to see a file name is given
if (argc != 2)
{
printf("Usage: ./recover IMAGE\n");
return 1;
}
//opening said file
FILE *rfile = fopen(argv[1], "r");
if(rfile == NULL)
{
printf("File '%s' does not exist\n", argv[1]);
return 1;
}
//opening a copy
FILE *c = fopen(argv[1], "r");
if(c == NULL)
{
printf("File '%s' does not exist\n", argv[1]);
return 1;
}
//checking 512 bytes again and again for orginal
while (fread (buffer, 1, BLOCK_SIZE, rfile) == BLOCK_SIZE)
{
fseek(c, ftell(rfile) , SEEK_SET); //fread (buffer, 1, BLOCK_SIZE, c);
sprintf(nf, "%03i.jpg", i); //000, 001, 002
//opening a file to write in
FILE *img = fopen(nf, "w");
if(img == NULL)
{
return 1;
}
//Is it a jpeg!?
if (buffer[0]==0xff && buffer[1]==0xd8 && buffer[2]==0xff && (buffer[3] & 0xf0) == 0xe0)
{
//writing the img in a file called img
fwrite(buffer, 512, 1, img);
fwrite(c, BLOCK_SIZE, 2048, img);
i++; //making sure a new file is opened next time
}
fclose(img);
}
fclose(c);
fclose(rfile);
free(nf); }
So I am currently attempting recover.c from the cs50 pset3 and I have recovered all 49 jpeg files. However, all these jpeg files are empty (with a grey and white grid). Could someone please explain where my code went wrong? I tried check50 to see if my code was correct but it said the recovered images do not match.
I changed my "w" to "wb" in my fopen function too but that didn't seem to work either.
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <stdint.h>
#include <stdbool.h>
int main(int argc, char *argv[])
{
//a data type that can store a byte
typedef uint8_t BYTE;
//Checking to see if there is only one command line argument
if (argc != 2)
{
fprintf(stderr, "Usage: ./recover image\n" );
return 1;
}
//Opening the file to see if its correct
char *infile = argv[1];
FILE *memory = fopen(infile, "r");
if (memory == NULL)
{
fprintf(stderr, "Could not open %s.\n", infile);
return 2;
}
//Creation of a buffer
BYTE buffer[512] = {0};
//Whether or not we have found a JPEG or not
bool jpegfound = false;
//the number of JPEG files found
int numJPEGfile = 0;
//declaring the new to be JPEG file so that it has a scope for the
whole while loop
FILE *img = NULL;
//declaring the new JPEG filename
char filename[8];
//Repeating until the end of card
while(fread(buffer, 512, 1, memory) == 1)
{
//Start of a new JPEG?
if (buffer[0] == 0xff && buffer[1] == 0xd8 && buffer[2] == 0xff && (buffer[3] & 0xf0) == 0xe0)
{
jpegfound = true;
sprintf(filename, "%03i.jpg", numJPEGfile);
numJPEGfile += 1;
img = fopen(filename, "wb");
fwrite(buffer, 512, 1, img);
}
//Have we already found a JPEG?
if(jpegfound)
{
jpegfound = false;
fclose(img);
}
}
//Close any remaining files
fclose(memory);
return 0;
}
If you run the below command in the terminal, how large are the jpg-files you recovered? I think this will give you a hint for solving this pset.
ls -l
My question is regarding the Recover assignment as part of CS50.
The code is running (finally) and it produces 50 JPEG files, and most of them are the correct images, except the first file is not an image, therefore it doesn't pass check50.
I have spent a long time trying to figure out what the problem is but I cannot pinpoint it so I am hoping someone might be able to help me out so I can move on.
Thanks in advance! Here is my code:
int main(int argc, char *argv[])
{
// ensure proper usage
if (argc != 2)
{
fprintf(stderr, "Usage: Name of Memory Card File\n");
return 1;
}
char *readfile = argv[1];
// open memory card file
FILE *card_ptr = fopen(readfile, "r");
if (card_ptr == NULL)
{
fprintf(stderr, "Could not open %s.\n", readfile);
return 2;
}
//Declare a buffer to read into
unsigned char *buffer = malloc(512);
//to check if we have already found a file
bool (jpgAlreadyNew) = false;
//declare counter for the number of files found and a file pointer
int filenumber = 0;
FILE *new_jpg_ptr = NULL;
char filename[8];
//read in bytes until reach EOF
while (fread(buffer, 1, 512, card_ptr) != 0x00)
{
//if we reach the header pattern of bytes
if (buffer [0] == 0xff && buffer [1] == 0xd8 && buffer [2] == 0xff && (buffer [3] & 0xf0) == 0xe0)
{
//if there is not already a JPEG file found
if (!jpgAlreadyNew)
{
//change the bool value
(jpgAlreadyNew) = true;
//open new file
sprintf(filename, "%03i.jpg", filenumber);
new_jpg_ptr = fopen(filename, "w");
if (new_jpg_ptr == NULL)
{
return 3;
}
//add to counter of files found
filenumber++;
//write files from buffer into new img file
fwrite(buffer, 1, 512, new_jpg_ptr);
}
//if there is already a JPEG file found
if (jpgAlreadyNew)
{
//close the previous file which would now be complete
fclose(new_jpg_ptr);
//open new file
sprintf(filename, "%03i.jpg", filenumber);
new_jpg_ptr = fopen(filename, "w");
if (new_jpg_ptr == NULL)
{
return 4;
}
//add to counter of files found
filenumber++;
//write files from buffer into new img file
fwrite(buffer, 1, 512, new_jpg_ptr);
}
}
// else if we do not see pattern of header bytes
else
{
//if already found a jpg file which is open then write the bytes to that file
if (jpgAlreadyNew)
{
fwrite(buffer, 1, 512, new_jpg_ptr);
}
//if no file found yet, discard and move on
if (!jpgAlreadyNew)
{
continue;
}
}
}
//free memory
free (buffer);
//close pointers and end program successfully
fclose(card_ptr);
fclose(new_jpg_ptr);
return 0;
}
Let's walk through the program starting with finding the first jpeg signature:
This if (!jpgAlreadyNew) evaluates to true, so it enters the if block; the first thing it does is (jpgAlreadyNew) = true;. When it is done creating the file and writing the first block, what happens next? This if (jpgAlreadyNew). Which also evaluates to true. So it closes 000.jpg and moves along.
Since jpgAlreadyNew is a boolean, an if {} else {} construct would suffice.
I'm using the FFmpeg library to generate MP4 files containing audio from various files, such as MP3, WAV, OGG, but I'm having some troubles (I'm also putting video in there, but for simplicity's sake I'm omitting that for this question, since I've got that working). My current code opens an audio file, decodes the content and converts it into the MP4 container and finally writes it into the destination file as interleaved frames.
It works perfectly for most MP3 files, but when inputting WAV or OGG, the audio in the resulting MP4 is slightly distorted and often plays at the wrong speed (up to many times faster or slower).
I've looked at countless of examples of using the converting functions (swr_convert), but I can't seem to get rid of the noise in the exported audio.
Here's how I add an audio stream to the MP4 (outContext is the AVFormatContext for the output file):
audioCodec = avcodec_find_encoder(outContext->oformat->audio_codec);
if (!audioCodec)
die("Could not find audio encoder!");
// Start stream
audioStream = avformat_new_stream(outContext, audioCodec);
if (!audioStream)
die("Could not allocate audio stream!");
audioCodecContext = audioStream->codec;
audioStream->id = 1;
// Setup
audioCodecContext->sample_fmt = AV_SAMPLE_FMT_S16;
audioCodecContext->bit_rate = 128000;
audioCodecContext->sample_rate = 44100;
audioCodecContext->channels = 2;
audioCodecContext->channel_layout = AV_CH_LAYOUT_STEREO;
// Open the codec
if (avcodec_open2(audioCodecContext, audioCodec, NULL) < 0)
die("Could not open audio codec");
And to open a sound file from MP3/WAV/OGG (from the filename variable)...
// Create contex
formatContext = avformat_alloc_context();
if (avformat_open_input(&formatContext, filename, NULL, NULL)<0)
die("Could not open file");
// Find info
if (avformat_find_stream_info(formatContext, 0)<0)
die("Could not find file info");
av_dump_format(formatContext, 0, filename, false);
// Find audio stream
streamId = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, NULL, 0);
if (streamId < 0)
die("Could not find Audio Stream");
codecContext = formatContext->streams[streamId]->codec;
// Find decoder
codec = avcodec_find_decoder(codecContext->codec_id);
if (codec == NULL)
die("cannot find codec!");
// Open codec
if (avcodec_open2(codecContext, codec, 0)<0)
die("Codec cannot be found");
// Set up resample context
swrContext = swr_alloc();
if (!swrContext)
die("Failed to alloc swr context");
av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
av_opt_set_int(swrContext, "in_channel_layout", codecContext->channel_layout, 0);
av_opt_set_int(swrContext, "in_sample_rate", codecContext->sample_rate, 0);
av_opt_set_sample_fmt(swrContext, "in_sample_fmt", codecContext->sample_fmt, 0);
av_opt_set_int(swrContext, "out_channel_count", audioCodecContext->channels, 0);
av_opt_set_int(swrContext, "out_channel_layout", audioCodecContext->channel_layout, 0);
av_opt_set_int(swrContext, "out_sample_rate", audioCodecContext->sample_rate, 0);
av_opt_set_sample_fmt(swrContext, "out_sample_fmt", audioCodecContext->sample_fmt, 0);
if (swr_init(swrContext))
die("Failed to init swr context");
Finally, to decode+convert+encode...
// Allocate and init re-usable frames
audioFrameDecoded = av_frame_alloc();
if (!audioFrameDecoded)
die("Could not allocate audio frame");
audioFrameDecoded->format = fileCodecContext->sample_fmt;
audioFrameDecoded->channel_layout = fileCodecContext->channel_layout;
audioFrameDecoded->channels = fileCodecContext->channels;
audioFrameDecoded->sample_rate = fileCodecContext->sample_rate;
audioFrameConverted = av_frame_alloc();
if (!audioFrameConverted)
die("Could not allocate audio frame");
audioFrameConverted->nb_samples = audioCodecContext->frame_size;
audioFrameConverted->format = audioCodecContext->sample_fmt;
audioFrameConverted->channel_layout = audioCodecContext->channel_layout;
audioFrameConverted->channels = audioCodecContext->channels;
audioFrameConverted->sample_rate = audioCodecContext->sample_rate;
AVPacket inPacket;
av_init_packet(&inPacket);
inPacket.data = NULL;
inPacket.size = 0;
int frameFinished = 0;
while (av_read_frame(formatContext, &inPacket) >= 0) {
if (inPacket.stream_index == streamId) {
int len = avcodec_decode_audio4(fileCodecContext, audioFrameDecoded, &frameFinished, &inPacket);
if (frameFinished) {
// Convert
uint8_t *convertedData=NULL;
if (av_samples_alloc(&convertedData,
NULL,
audioCodecContext->channels,
audioFrameConverted->nb_samples,
audioCodecContext->sample_fmt, 0) < 0)
die("Could not allocate samples");
int outSamples = swr_convert(swrContext,
&convertedData,
audioFrameConverted->nb_samples,
(const uint8_t **)audioFrameDecoded->data,
audioFrameDecoded->nb_samples);
if (outSamples < 0)
die("Could not convert");
size_t buffer_size = av_samples_get_buffer_size(NULL,
audioCodecContext->channels,
audioFrameConverted->nb_samples,
audioCodecContext->sample_fmt,
0);
if (buffer_size < 0)
die("Invalid buffer size");
if (avcodec_fill_audio_frame(audioFrameConverted,
audioCodecContext->channels,
audioCodecContext->sample_fmt,
convertedData,
buffer_size,
0) < 0)
die("Could not fill frame");
AVPacket outPacket;
av_init_packet(&outPacket);
outPacket.data = NULL;
outPacket.size = 0;
if (avcodec_encode_audio2(audioCodecContext, &outPacket, audioFrameConverted, &frameFinished) < 0)
die("Error encoding audio frame");
if (frameFinished) {
outPacket.stream_index = audioStream->index;
if (av_interleaved_write_frame(outContext, &outPacket) != 0)
die("Error while writing audio frame");
av_free_packet(&outPacket);
}
}
}
}
av_frame_free(&audioFrameConverted);
av_frame_free(&audioFrameDecoded);
av_free_packet(&inPacket);
I have also tried setting appropriate pts values for outgoing frames, but that doesn't seem to affect the sound quality at all.
I'm also unsure how/if I should be allocating the converted data, can av_samples_alloc be used for this? What about avcodec_fill_audio_frame? Am I on the right track?
Any input is appreciated (I can also send the exported MP4s if necessary, if you want to hear the distortion).
if (avcodec_encode_audio2(audioCodecContext, &outPacket, audioFrameConverted, &frameFinished) < 0)
die("Error encoding audio frame");
You seem to be assuming that the encoder will eat all submitted samples - it doesn't. It also doesn't cache them internally. It will eat a specific number of samples (AVCodecContext.frame_size), and the rest should be resubmitted in the next call to avcodec_encode_audio2().
[edit]
ok, so your edited code is better, but not there yet. You're still assuming the decoder will output at least frame_size samples for each call to avcodec_decode_audioN() (after resampling), which may not be the case. If that happens (and it does, for ogg), your avcodec_encode_audioN() call will encode an incomplete input buffer (because you say it's got frame_size samples, but it doesn't). Likewise, your code also doesn't deal with cases where the decoder outputs a number significantly bigger than frame_size (like 10*frame_size) expected by the encoder, in which case you'll get overruns - basically your 1:1 decode/encode mapping is the main source of your problem.
As a solution, consider the swrContext a FIFO, where you input all decoder samples, and loop over it until it's got less than frame_size samples left. I'll leave it up to you to learn how to deal with end-of-stream, because you'll need to flush cached samples out of the decoder (by calling avcodec_decode_audioN() with AVPacket where .data = NULL and .size = 0), flush the swrContext (by calling swr_context() until it returns 0) as well as flush the encoder (by feeding it NULL AVFrames until it returns AVPacket with .size = 0). Right now you'll probably get an output file where the end is slightly truncated. That shouldn't be hard to figure out.
This code works for me for m4a/ogg/mp3 to m4a/aac conversion:
#include "libswresample/swresample.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "libavutil/opt.h"
#include <stdio.h>
#include <stdlib.h>
static void die(char *str) {
fprintf(stderr, "%s\n", str);
exit(1);
}
static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
{
AVCodecContext *c;
AVCodec *encoder = avcodec_find_encoder(codec_id);
AVStream *st = avformat_new_stream(oc, encoder);
if (!st) die("av_new_stream");
c = st->codec;
c->codec_id = codec_id;
c->codec_type = AVMEDIA_TYPE_AUDIO;
/* put sample parameters */
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
c->sample_fmt = encoder->sample_fmts[0];
c->channel_layout = AV_CH_LAYOUT_STEREO;
// some formats want stream headers to be separate
if(oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
static void open_audio(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c = st->codec;
AVCodec *codec;
/* find the audio encoder */
codec = avcodec_find_encoder(c->codec_id);
if (!codec) die("avcodec_find_encoder");
/* open it */
AVDictionary *dict = NULL;
av_dict_set(&dict, "strict", "+experimental", 0);
int res = avcodec_open2(c, codec, &dict);
if (res < 0) die("avcodec_open");
}
int main(int argc, char *argv[]) {
av_register_all();
if (argc != 3) {
fprintf(stderr, "%s <in> <out>\n", argv[0]);
exit(1);
}
// Allocate and init re-usable frames
AVCodecContext *fileCodecContext, *audioCodecContext;
AVFormatContext *formatContext, *outContext;
AVStream *audioStream;
SwrContext *swrContext;
int streamId;
// input file
const char *file = argv[1];
int res = avformat_open_input(&formatContext, file, NULL, NULL);
if (res != 0) die("avformat_open_input");
res = avformat_find_stream_info(formatContext, NULL);
if (res < 0) die("avformat_find_stream_info");
AVCodec *codec;
res = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
if (res < 0) die("av_find_best_stream");
streamId = res;
fileCodecContext = avcodec_alloc_context3(codec);
avcodec_copy_context(fileCodecContext, formatContext->streams[streamId]->codec);
res = avcodec_open2(fileCodecContext, codec, NULL);
if (res < 0) die("avcodec_open2");
// output file
const char *outfile = argv[2];
AVOutputFormat *fmt = fmt = av_guess_format(NULL, outfile, NULL);
if (!fmt) die("av_guess_format");
outContext = avformat_alloc_context();
outContext->oformat = fmt;
audioStream = add_audio_stream(outContext, fmt->audio_codec);
open_audio(outContext, audioStream);
res = avio_open2(&outContext->pb, outfile, AVIO_FLAG_WRITE, NULL, NULL);
if (res < 0) die("url_fopen");
avformat_write_header(outContext, NULL);
audioCodecContext = audioStream->codec;
// resampling
swrContext = swr_alloc();
av_opt_set_channel_layout(swrContext, "in_channel_layout", fileCodecContext->channel_layout, 0);
av_opt_set_channel_layout(swrContext, "out_channel_layout", audioCodecContext->channel_layout, 0);
av_opt_set_int(swrContext, "in_sample_rate", fileCodecContext->sample_rate, 0);
av_opt_set_int(swrContext, "out_sample_rate", audioCodecContext->sample_rate, 0);
av_opt_set_sample_fmt(swrContext, "in_sample_fmt", fileCodecContext->sample_fmt, 0);
av_opt_set_sample_fmt(swrContext, "out_sample_fmt", audioCodecContext->sample_fmt, 0);
res = swr_init(swrContext);
if (res < 0) die("swr_init");
AVFrame *audioFrameDecoded = av_frame_alloc();
if (!audioFrameDecoded)
die("Could not allocate audio frame");
audioFrameDecoded->format = fileCodecContext->sample_fmt;
audioFrameDecoded->channel_layout = fileCodecContext->channel_layout;
audioFrameDecoded->channels = fileCodecContext->channels;
audioFrameDecoded->sample_rate = fileCodecContext->sample_rate;
AVFrame *audioFrameConverted = av_frame_alloc();
if (!audioFrameConverted) die("Could not allocate audio frame");
audioFrameConverted->nb_samples = audioCodecContext->frame_size;
audioFrameConverted->format = audioCodecContext->sample_fmt;
audioFrameConverted->channel_layout = audioCodecContext->channel_layout;
audioFrameConverted->channels = audioCodecContext->channels;
audioFrameConverted->sample_rate = audioCodecContext->sample_rate;
AVPacket inPacket;
av_init_packet(&inPacket);
inPacket.data = NULL;
inPacket.size = 0;
int frameFinished = 0;
while (av_read_frame(formatContext, &inPacket) >= 0) {
if (inPacket.stream_index == streamId) {
int len = avcodec_decode_audio4(fileCodecContext, audioFrameDecoded, &frameFinished, &inPacket);
if (frameFinished) {
// Convert
uint8_t *convertedData=NULL;
if (av_samples_alloc(&convertedData,
NULL,
audioCodecContext->channels,
audioFrameConverted->nb_samples,
audioCodecContext->sample_fmt, 0) < 0)
die("Could not allocate samples");
int outSamples = swr_convert(swrContext, NULL, 0,
//&convertedData,
//audioFrameConverted->nb_samples,
(const uint8_t **)audioFrameDecoded->data,
audioFrameDecoded->nb_samples);
if (outSamples < 0) die("Could not convert");
for (;;) {
outSamples = swr_get_out_samples(swrContext, 0);
if (outSamples < audioCodecContext->frame_size * audioCodecContext->channels) break; // see comments, thanks to #dajuric for fixing this
outSamples = swr_convert(swrContext,
&convertedData,
audioFrameConverted->nb_samples, NULL, 0);
size_t buffer_size = av_samples_get_buffer_size(NULL,
audioCodecContext->channels,
audioFrameConverted->nb_samples,
audioCodecContext->sample_fmt,
0);
if (buffer_size < 0) die("Invalid buffer size");
if (avcodec_fill_audio_frame(audioFrameConverted,
audioCodecContext->channels,
audioCodecContext->sample_fmt,
convertedData,
buffer_size,
0) < 0)
die("Could not fill frame");
AVPacket outPacket;
av_init_packet(&outPacket);
outPacket.data = NULL;
outPacket.size = 0;
if (avcodec_encode_audio2(audioCodecContext, &outPacket, audioFrameConverted, &frameFinished) < 0)
die("Error encoding audio frame");
if (frameFinished) {
outPacket.stream_index = audioStream->index;
if (av_interleaved_write_frame(outContext, &outPacket) != 0)
die("Error while writing audio frame");
av_free_packet(&outPacket);
}
}
}
}
}
swr_close(swrContext);
swr_free(&swrContext);
av_frame_free(&audioFrameConverted);
av_frame_free(&audioFrameDecoded);
av_free_packet(&inPacket);
av_write_trailer(outContext);
avio_close(outContext->pb);
avcodec_close(fileCodecContext);
avcodec_free_context(&fileCodecContext);
avformat_close_input(&formatContext);
return 0;
}
I wanted to include a couple things I found when I was working with the above code.
I had one file get stuck in an infinite loop. The reason is the file had a sample rate of 48000 and the code changes it to a 44100. This caused it to always have extra outSamples. swr_convert & would not grab them. So I ended up changing add_audio_stream to match the input streams sample rate.
c->sample_rate = fileCodecContext->sample_rate;
Also I had to produce wav files as my output. And it had a framesize of 0. so I just chose a number after a few tests I went with 32. I noticed if I went too big (ex 128) I would get audio glitches.
if (audioFrameConverted->nb_samples <= 0) audioFrameConverted->nb_samples = 32; //wav files have a 0
Changed the if statement that breaks out of the loop to check nb_samples if frame_size is 0.
if ((outSamples < audioCodecContext->frame_size * audioCodecContext->channels) || audioCodecContext->frame_size==0 && (outSamples < audioFrameConverted->nb_samples * audioCodecContext->channels)) break; // see comments, thanks to #dajuric for fixing this
There was also a glitch when I was testing outputting to ogg files where the timestamp data was missing so the file wouldn't play correctly in vlc. There were a few lines I added that helped with that.
out_audioStream->time_base = in_audioStream->time_base; // entered before avio_open.
outPacket.dts = audioFrameDecoded->pkt_dts;//rest after avcodec_encode_audio2
outPacket.pts = audioFrameDecoded->pkt_pts;
av_packet_rescale_ts(&outPacket, in_audioStream->time_base, out_audioStream->time_base);
Variables might be a little different I converted the code to c#. Thought this might help someone.
Actually swr_convert won't work for that, try to use swr_convert_frame instead.