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Hi!<br>
I use the following code (see below) in order to decode an audio
file into an array, and I'm having a memory leak of 24kb:<br>
Direct leak of 24 byte(s) in 1 object(s) allocated from:<br>
#0 0x7f80c449e386 in __interceptor_posix_memalign
/build/gcc-multilib/src/gcc-5.2.0/libsanitizer/asan/asan_malloc_linux.cc:105<br>
#1 0x7f80c3acc43f in av_malloc
(/usr/lib/libavutil.so.54+0x2343f)<br>
<br>
So I'm thinking that it's due to some libav-specific things that I
didn't close properly, and so here's my question: is
avcodec_close(context); sufficient to free a codec context <b>and</b>
a codec? This example
(<a class="moz-txt-link-freetext" href="http://ffmpeg.org/doxygen/trunk/decoding_encoding_8c-example.html">http://ffmpeg.org/doxygen/trunk/decoding_encoding_8c-example.html</a>)
does an av_free(context), but my program crashes when I try to do
it...<br>
<br>
Thanks by advance!<br>
Polochon_street<br>
<br>
<p>#define INBUF_SIZE 4096<br>
<br>
</p>
<p>#define AUDIO_INBUF_SIZE 20480</p>
<p>#define AUDIO_REFILL_THRESH 4096</p>
<p><br>
#include "analyze.h"</p>
<p><br>
int audio_decode(const char *filename, struct song *song) { //
decode the track</p>
<p> AVCodec *codec = NULL;</p>
<p> AVCodecContext *c = NULL;</p>
<p> AVFormatContext *pFormatCtx;</p>
<p> </p>
<p> int i, d, e;</p>
<p> int len;</p>
<p> int planar;</p>
<p> AVPacket avpkt;</p>
<p> AVFrame *decoded_frame = NULL;</p>
<p> int8_t *beginning;</p>
<p> int got_frame;</p>
<p> int audioStream;</p>
<p> size_t index;</p>
<p><br>
av_register_all();</p>
<p> av_init_packet(&avpkt);</p>
<p><br>
pFormatCtx = avformat_alloc_context();</p>
<p><br>
if(avformat_open_input(&pFormatCtx, filename, NULL, NULL)
< 0) {</p>
<p> printf("Couldn't open file: %s, %d\n", filename, errno);</p>
<p> song->nSamples = 0;</p>
<p> return 1;</p>
<p> }</p>
<p><br>
if(avformat_find_stream_info(pFormatCtx, NULL) < 0) {</p>
<p> printf("Couldn't find stream information\n");</p>
<p> song->nSamples = 0;</p>
<p> return 1;</p>
<p> } </p>
<p><br>
audioStream = av_find_best_stream(pFormatCtx,
AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);</p>
<p> c = pFormatCtx->streams[audioStream]->codec;</p>
<p> </p>
<p> if (!codec) {</p>
<p> printf("Codec not found!\n");</p>
<p> song->nSamples = 0;</p>
<p> return 1;</p>
<p> }</p>
<p><br>
if(avcodec_open2(c, codec, NULL) < 0) {</p>
<p> printf("Could not open codec\n");</p>
<p> song->nSamples = 0;</p>
<p> return 1;</p>
<p> }</p>
<p> </p>
<p> song->sample_rate = c->sample_rate;</p>
<p> song->duration = pFormatCtx->duration/AV_TIME_BASE;</p>
<p> size =
(((uint64_t)(pFormatCtx->duration)*(uint64_t)song->sample_rate)/(uint64_t)AV_TIME_BASE)*c->channels*av_get_bytes_per_sample(c->sample_fmt);</p>
<p> song->nSamples =
(((uint64_t)(pFormatCtx->duration)*(uint64_t)song->sample_rate)/(uint64_t)AV_TIME_BASE)*c->channels;</p>
<p> song->sample_array = malloc(size);</p>
<p><br>
for(i = 0; i < size; ++i)</p>
<p> song->sample_array[i] = 0;</p>
<p><br>
beginning = song->sample_array;</p>
<p> index = 0;</p>
<p><br>
planar = av_sample_fmt_is_planar(c->sample_fmt);</p>
<p> song->nb_bytes_per_sample =
av_get_bytes_per_sample(c->sample_fmt);</p>
<p><br>
song->channels = c->channels;</p>
<p> </p>
<p>/* End of codec init */</p>
<p> while(av_read_frame(pFormatCtx, &avpkt) >= 0) {</p>
<p> if(avpkt.stream_index == audioStream) {</p>
<p> got_frame = 0; </p>
<p> </p>
<p> if(!decoded_frame) {</p>
<p> if(!(decoded_frame = av_frame_alloc())) {</p>
<p> printf("Could not allocate audio frame\n");</p>
<p> exit(1);</p>
<p> }</p>
<p> }</p>
<p> else </p>
<p> av_frame_unref(decoded_frame);</p>
<p><br>
len = avcodec_decode_audio4(c, decoded_frame,
&got_frame, &avpkt);</p>
<p> </p>
<p> if(len < 0)</p>
<p> avpkt.size = 0;</p>
<p><br>
av_free_packet(&avpkt);</p>
<p><br>
/* interesting part: copying decoded data into a huge
array */</p>
<p> /* flac has a different behaviour from mp3, hence the
planar condition */</p>
<p> if(got_frame) {</p>
<p> size_t data_size =
av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples, c->sample_fmt, 1); </p>
<p><br>
if(index*song->nb_bytes_per_sample + data_size
> size) {</p>
<p> beginning = realloc(beginning, (size +=
data_size));</p>
<p> song->nSamples +=
data_size/song->nb_bytes_per_sample;</p>
<p> }</p>
<p> int8_t *p =
beginning+index*song->nb_bytes_per_sample;</p>
<p> if(planar == 1) {</p>
<p> for(i = 0; i <
decoded_frame->nb_samples*song->nb_bytes_per_sample; i +=
song->nb_bytes_per_sample) { </p>
<p> for(e = 0; e < c->channels; ++e)</p>
<p> for(d = 0; d <
song->nb_bytes_per_sample; ++d) </p>
<p> *(p++) =
((int8_t*)(decoded_frame->extended_data[e]))[i+d];</p>
<p> }</p>
<p> index +=
data_size/song->nb_bytes_per_sample;</p>
<p> }</p>
<p> else if(planar == 0) {</p>
<p> memcpy(index*song->nb_bytes_per_sample +
beginning, decoded_frame->extended_data[0], data_size);</p>
<p> index +=
data_size/song->nb_bytes_per_sample; </p>
<p> }</p>
<p> }</p>
<p> }</p>
<p> }</p>
<p> song->sample_array = beginning;</p>
<p><br>
/* cleaning memory */</p>
<p> </p>
<p> avcodec_close(c);</p>
<p> av_frame_unref(decoded_frame);</p>
<p> av_frame_free(&decoded_frame);</p>
<p> av_free_packet(&avpkt);</p>
<p> avformat_close_input(&pFormatCtx);</p>
<p><br>
return 0;</p>
<p>}<br>
</p>
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