[Libav-user] Closing codec properly after an audio decoding
Polochon Street
polochonstreet at gmx.fr
Sat Aug 29 17:25:03 CEST 2015
Okay thanks for your help; I changed my code with what you wrote, but I
still have a more important leak.
I also found that I have an indirect memory leak when decoding songs
with a embedded album picture tag (which appears as Stream #0:1: Video:
mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 160x160 [SAR 1:1 DAR 1:1],
90k tbr, 90k tbn, 90k tbc):
Indirect leak of 8686 byte(s) in 2 object(s) allocated from:
#0 0x7f7877199386 in __interceptor_posix_memalign
/build/gcc-multilib/src/gcc-5.2.0/libsanitizer/asan/asan_malloc_linux.cc:105
#1 0x7f78767c743f in av_malloc (/usr/lib/libavutil.so.54+0x2343f)
Maybe I should open another thread for this one? Now that we've seen
that the leak does not come from the codec closing, do you have a clue
where I could search?
The 29/08/2015 17:10, J Decker wrote :
> On Sat, Aug 29, 2015 at 5:02 AM, Polochon Street <polochonstreet at gmx.fr> wrote:
>> 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:
> somethign more like...
> avpkt.data = NULL;
> avpkt.size = 0;
> // just use your existing buffer here...
> decoded_frame = av_frame_alloc();
> do {
> avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
> } while (got_frame);
> av_frame_unref(decoded_frame);
> av_free_packet(&avpkt);
>
>> 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 */
>>>>
>
> avpkt.data = NULL;
> avpkt.size = 0;
> // just use your existing buffer here...
> decoded_frame = av_frame_alloc();
> do {
> avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
> } while (got_frame);
>
>>>>
>>>> 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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