[Libav-user] Code crashes and cant seem to find out why, even code in demo crashes at the same point
JULIAN GARDNER
joolzg at btinternet.com
Tue Apr 12 20:45:28 CEST 2016
I am in the middle of trying to add a VU Meter to my application but when I try and convert the audio buffer from FLTP to S16 I get a crash and I cannot see what im doing wrong. Now this code also fails in the demo applications.
outputSettings->audio_channels = 2;outputSettings->audio_format = AV_SAMPLE_FMT_S16;outputSettings->audio_sample_rate = 32000;
This is the source file and its outputffprobe version N-78956-g920f592 Copyright (c) 2007-2016 the FFmpeg developers
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'mv/ABC - The Look Of Love (1982).mp4':
Metadata:
major_brand : isom
minor_version : 1
compatible_brands: isomavc1
creation_time : 2011-08-31 10:04:04
Duration: 00:03:33.21, start: 0.000000, bitrate: 1114 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 919 kb/s, 29.97 fps, 29.97 tbr, 120k tbn, 59.94 tbc (default)
Metadata:
creation_time : 2011-08-31 10:04:04
Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, stereo, fltp, 192 kb/s (default)
Metadata:
creation_time : 2011-08-31 10:04:08
Side data:
audio service type: main
So I want the audio to be converted to S16 so I can use this as the output to the new stream and also processed as a VuMeter
if( !inputSource->audio_scale_ctx) {
/* create resampler context */
inputSource->audio_scale_ctx = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(outputSettings->audio_channels),
outputSettings->audio_format,
outputSettings->audio_sample_rate,
av_get_default_channel_layout(inputSource->fmt_ctx->streams[inputSource->audio_stream_idx]->codec->channels),
inputSource->fmt_ctx->streams[inputSource->audio_stream_idx]->codec->sample_fmt,
inputSource->fmt_ctx->streams[inputSource->audio_stream_idx]->codec->sample_rate,
0, NULL);
/* initialize the resampling context */
if ( swr_init(inputSource->audio_scale_ctx)<0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
printf( "Tile audio resampling from %d:%d:%s to %d:%d:%s\n",
inputSource->fmt_ctx->streams[inputSource->audio_stream_idx]->codec->channels,
inputSource->fmt_ctx->streams[inputSource->audio_stream_idx]->codec->sample_rate,
audioFormats[inputSource->fmt_ctx->streams[inputSource->audio_stream_idx]->codec->sample_fmt].name,
outputSettings->audio_channels,
outputSettings->audio_sample_rate,
audioFormats[outputSettings->audio_format].name);
}
dst_nb_samples = av_rescale_rnd(swr_get_delay(inputSource->audio_scale_ctx, inputSource->audio_dec_ctx->sample_rate) + frame->nb_samples,
outputSettings->audio_sample_rate, inputSource->audio_dec_ctx->sample_rate, AV_ROUND_UP);
if( dst_nb_samples>inputSource->audio_frame_size) {
if( inputSource->audio_frame)
av_frame_free(&inputSource->audio_frame);
inputSource->audio_frame = alloc_audio_frame(outputSettings->audio_format, outputSettings->audio_channels, outputSettings->audio_sample_rate, dst_nb_samples);
// inputSource->audio_frame_size = dst_nb_samples;
printf( "Tile audio resampling frame size changed from %d to %d\r\n", inputSource->audio_frame_size, dst_nb_samples);
}
else if( !inputSource->audio_frame_size) {
inputSource->audio_frame = alloc_audio_frame(outputSettings->audio_format, outputSettings->audio_channels, outputSettings->audio_sample_rate, dst_nb_samples);
// inputSource->audio_frame_size = dst_nb_samples;
printf( "Tile audio resampling frame size changed from %d to %d\r\n", inputSource->audio_frame_size, dst_nb_samples);
}
if( av_frame_make_writable(inputSource->audio_frame)<0) {
exit(1);
}
/* convert to destination format */
printf( "%p %p %d %p %d\n",
inputSource->audio_scale_ctx, inputSource->audio_frame->data, inputSource->audio_frame->nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
/* convert to destination format */
ret = swr_convert(inputSource->audio_scale_ctx, inputSource->audio_frame->data, inputSource->audio_frame->nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
output from the above with debugTile audio resampling from 2:48000:FLTP to 2:32000:S16
Tile audio resampling frame size changed from 0 to 1024
0x7f24b40008c0 0x7f24b4045bc0 1024 0x7f24f46b49c0 1536*** Error in `./generator': double free or corruption (!prev): 0x00007f24b4045e00 ***
Now in Valgrind i get the following stack trace==13145== Thread 23:
==13145== Invalid write of size 8
==13145== at 0xD4EDBF: ff_pack_2ch_float_to_int16_a_sse2 (audio_convert.asm:667)
==13145== by 0xD49279: swri_audio_convert (audioconvert.c:207)
==13145== by 0xD45881: swr_convert_internal.part.3 (swresample.c:686)
==13145== by 0xD4652B: swr_convert_internal (swresample.c:783)
==13145== by 0xD4652B: swr_convert (swresample.c:747)
==13145== by 0x47551E: inputThreadAudio (generator.c:2106)
==13145== by 0x7900181: start_thread (pthread_create.c:312)
==13145== by 0x7F7747C: clone (clone.S:111)
==13145== Address 0x1ee85080 is 0 bytes after a block of size 2,048 alloc'd
==13145== at 0x4C2D110: memalign (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so)
==13145== by 0x4C2D227: posix_memalign (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so)
==13145== by 0xDCAA39: av_malloc (mem.c:97)
==13145== by 0xDBA917: av_buffer_alloc (buffer.c:71)
==13145== by 0xDC4401: get_audio_buffer (frame.c:251)
==13145== by 0xDC4401: av_frame_get_buffer (frame.c:278)
==13145== by 0x475DF3: alloc_audio_frame (generator.c:3023)
==13145== by 0x4753DF: inputThreadAudio (generator.c:2086)
==13145== by 0x7900181: start_thread (pthread_create.c:312)
==13145== by 0x7F7747C: clone (clone.S:111)
Anybody see my silly mistake as I cant. Joolz Joolz
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