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author | James Almer <jamrial@gmail.com> | 2023-05-05 15:26:19 -0300 |
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committer | James Almer <jamrial@gmail.com> | 2023-05-05 15:28:28 -0300 |
commit | ed3691338b25108517d354dda7f6c930fec5471c (patch) | |
tree | 9e51ee50d2ff59a55d69778bbefb2bcdca2d695b /doc | |
parent | 09b2ca93b4aa7cb24eeebdc829be61077466f021 (diff) | |
download | ffmpeg-ed3691338b25108517d354dda7f6c930fec5471c.tar.gz |
doc/examples/transcode_aac: use av_samples_alloc_array_and_samples to allocate the input samples buffer and pointers
Fixes -Wuse-after-free warnings and simplifies code.
Signed-off-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'doc')
-rw-r--r-- | doc/examples/transcode_aac.c | 20 |
1 files changed, 5 insertions, 15 deletions
diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index bb5681a7c0..cf6edc9890 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -447,26 +447,17 @@ static int init_converted_samples(uint8_t ***converted_input_samples, int error; /* Allocate as many pointers as there are audio channels. - * Each pointer will later point to the audio samples of the corresponding + * Each pointer will point to the audio samples of the corresponding * channels (although it may be NULL for interleaved formats). - */ - if (!(*converted_input_samples = calloc(output_codec_context->ch_layout.nb_channels, - sizeof(**converted_input_samples)))) { - fprintf(stderr, "Could not allocate converted input sample pointers\n"); - return AVERROR(ENOMEM); - } - - /* Allocate memory for the samples of all channels in one consecutive + * Allocate memory for the samples of all channels in one consecutive * block for convenience. */ - if ((error = av_samples_alloc(*converted_input_samples, NULL, + if ((error = av_samples_alloc_array_and_samples(converted_input_samples, NULL, output_codec_context->ch_layout.nb_channels, frame_size, output_codec_context->sample_fmt, 0)) < 0) { fprintf(stderr, "Could not allocate converted input samples (error '%s')\n", av_err2str(error)); - av_freep(&(*converted_input_samples)[0]); - free(*converted_input_samples); return error; } return 0; @@ -598,10 +589,9 @@ static int read_decode_convert_and_store(AVAudioFifo *fifo, ret = 0; cleanup: - if (converted_input_samples) { + if (converted_input_samples) av_freep(&converted_input_samples[0]); - free(converted_input_samples); - } + av_freep(&converted_input_samples); av_frame_free(&input_frame); return ret; |