[Libav-user] Closing codec properly after an audio decoding
Polochon Street
polochonstreet at gmx.fr
Sat Aug 29 14:02:20 CEST 2015
So, what should I do? Set avpkt.data to NULL and avpkt.size to 0, and
then close the codec?
I tried to add at the end of the loop:
do {
avpkt.data = NULL;
avpkt.size = 0;
decoded_frame = av_frame_alloc();
avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
av_frame_unref(decoded_frame);
av_free_packet(&avpkt);
} while (got_frame);
But now I'm leaking more memory...
The 29/08/2015 13:10, J Decker wrote :
> You didn't pass a null packet to flush the codec at the end and get
> the remaining packets from the stream.
>
> On Thu, Aug 27, 2015 at 1:21 PM, Polochon Street <polochonstreet at gmx.fr> wrote:
>> Hi!
>> 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:
>> Direct leak of 24 byte(s) in 1 object(s) allocated from:
>> #0 0x7f80c449e386 in __interceptor_posix_memalign
>> /build/gcc-multilib/src/gcc-5.2.0/libsanitizer/asan/asan_malloc_linux.cc:105
>> #1 0x7f80c3acc43f in av_malloc (/usr/lib/libavutil.so.54+0x2343f)
>>
>> 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 and a codec? This example
>> (http://ffmpeg.org/doxygen/trunk/decoding_encoding_8c-example.html) does an
>> av_free(context), but my program crashes when I try to do it...
>>
>> Thanks by advance!
>> Polochon_street
>>
>> #define INBUF_SIZE 4096
>>
>> #define AUDIO_INBUF_SIZE 20480
>>
>> #define AUDIO_REFILL_THRESH 4096
>>
>>
>> #include "analyze.h"
>>
>>
>> int audio_decode(const char *filename, struct song *song) { // decode the
>> track
>>
>> AVCodec *codec = NULL;
>>
>> AVCodecContext *c = NULL;
>>
>> AVFormatContext *pFormatCtx;
>>
>>
>>
>> int i, d, e;
>>
>> int len;
>>
>> int planar;
>>
>> AVPacket avpkt;
>>
>> AVFrame *decoded_frame = NULL;
>>
>> int8_t *beginning;
>>
>> int got_frame;
>>
>> int audioStream;
>>
>> size_t index;
>>
>>
>> av_register_all();
>>
>> av_init_packet(&avpkt);
>>
>>
>> pFormatCtx = avformat_alloc_context();
>>
>>
>> if(avformat_open_input(&pFormatCtx, filename, NULL, NULL) < 0) {
>>
>> printf("Couldn't open file: %s, %d\n", filename, errno);
>>
>> song->nSamples = 0;
>>
>> return 1;
>>
>> }
>>
>>
>> if(avformat_find_stream_info(pFormatCtx, NULL) < 0) {
>>
>> printf("Couldn't find stream information\n");
>>
>> song->nSamples = 0;
>>
>> return 1;
>>
>> }
>>
>>
>> audioStream = av_find_best_stream(pFormatCtx, AVMEDIA_TYPE_AUDIO, -1,
>> -1, &codec, 0);
>>
>> c = pFormatCtx->streams[audioStream]->codec;
>>
>>
>>
>> if (!codec) {
>>
>> printf("Codec not found!\n");
>>
>> song->nSamples = 0;
>>
>> return 1;
>>
>> }
>>
>>
>> if(avcodec_open2(c, codec, NULL) < 0) {
>>
>> printf("Could not open codec\n");
>>
>> song->nSamples = 0;
>>
>> return 1;
>>
>> }
>>
>>
>>
>> song->sample_rate = c->sample_rate;
>>
>> song->duration = pFormatCtx->duration/AV_TIME_BASE;
>>
>> 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);
>>
>> song->nSamples =
>> (((uint64_t)(pFormatCtx->duration)*(uint64_t)song->sample_rate)/(uint64_t)AV_TIME_BASE)*c->channels;
>>
>> song->sample_array = malloc(size);
>>
>>
>> for(i = 0; i < size; ++i)
>>
>> song->sample_array[i] = 0;
>>
>>
>> beginning = song->sample_array;
>>
>> index = 0;
>>
>>
>> planar = av_sample_fmt_is_planar(c->sample_fmt);
>>
>> song->nb_bytes_per_sample = av_get_bytes_per_sample(c->sample_fmt);
>>
>>
>> song->channels = c->channels;
>>
>>
>>
>> /* End of codec init */
>>
>> while(av_read_frame(pFormatCtx, &avpkt) >= 0) {
>>
>> if(avpkt.stream_index == audioStream) {
>>
>> got_frame = 0;
>>
>>
>>
>> if(!decoded_frame) {
>>
>> if(!(decoded_frame = av_frame_alloc())) {
>>
>> printf("Could not allocate audio frame\n");
>>
>> exit(1);
>>
>> }
>>
>> }
>>
>> else
>>
>> av_frame_unref(decoded_frame);
>>
>>
>> len = avcodec_decode_audio4(c, decoded_frame, &got_frame,
>> &avpkt);
>>
>>
>>
>> if(len < 0)
>>
>> avpkt.size = 0;
>>
>>
>> av_free_packet(&avpkt);
>>
>>
>> /* interesting part: copying decoded data into a huge array */
>>
>> /* flac has a different behaviour from mp3, hence the planar
>> condition */
>>
>> if(got_frame) {
>>
>> size_t data_size = av_samples_get_buffer_size(NULL,
>> c->channels, decoded_frame->nb_samples, c->sample_fmt, 1);
>>
>>
>> if(index*song->nb_bytes_per_sample + data_size > size) {
>>
>> beginning = realloc(beginning, (size += data_size));
>>
>> song->nSamples += data_size/song->nb_bytes_per_sample;
>>
>> }
>>
>> int8_t *p = beginning+index*song->nb_bytes_per_sample;
>>
>> if(planar == 1) {
>>
>> for(i = 0; i <
>> decoded_frame->nb_samples*song->nb_bytes_per_sample; i +=
>> song->nb_bytes_per_sample) {
>>
>> for(e = 0; e < c->channels; ++e)
>>
>> for(d = 0; d < song->nb_bytes_per_sample; ++d)
>>
>> *(p++) =
>> ((int8_t*)(decoded_frame->extended_data[e]))[i+d];
>>
>> }
>>
>> index += data_size/song->nb_bytes_per_sample;
>>
>> }
>>
>> else if(planar == 0) {
>>
>> memcpy(index*song->nb_bytes_per_sample + beginning,
>> decoded_frame->extended_data[0], data_size);
>>
>> index += data_size/song->nb_bytes_per_sample;
>>
>> }
>>
>> }
>>
>> }
>>
>> }
>>
>> song->sample_array = beginning;
>>
>>
>> /* cleaning memory */
>>
>>
>>
>> avcodec_close(c);
>>
>> av_frame_unref(decoded_frame);
>>
>> av_frame_free(&decoded_frame);
>>
>> av_free_packet(&avpkt);
>>
>> avformat_close_input(&pFormatCtx);
>>
>>
>> return 0;
>>
>> }
>>
>>
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>>
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