X-Git-Url: http://review.tizen.org/git/?a=blobdiff_plain;f=doc%2Fexamples%2Ftranscode_aac.c;h=cf6edc989077b78d577ad5ab6dd5216841700511;hb=2b01b7918beebe7b392ebf255f887e396a59e4c6;hp=2d4f9a59d37a7d2bd4720c056097e2acc08dd825;hpb=55e5af3c03898ffbac352fe4af83208fa4129c71;p=platform%2Fupstream%2Fffmpeg.git diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c index 2d4f9a5..cf6edc9 100644 --- a/doc/examples/transcode_aac.c +++ b/doc/examples/transcode_aac.c @@ -19,12 +19,11 @@ */ /** - * @file - * Simple audio converter - * + * @file audio transcoding to MPEG/AAC API usage example * @example transcode_aac.c - * Convert an input audio file to AAC in an MP4 container using FFmpeg. - * Formats other than MP4 are supported based on the output file extension. + * + * Convert an input audio file to AAC in an MP4 container. Formats other than + * MP4 are supported based on the output file extension. * @author Andreas Unterweger (dustsigns@gmail.com) */ @@ -448,26 +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; @@ -599,10 +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;