Imported Upstream version 6.1

[platform/upstream/ffmpeg.git] / doc / examples / transcode_aac.c

diff --git a/doc/examples/transcode_aac.c b/doc/examples/transcode_aac.c
index 2d4f9a5..cf6edc9 100644 (file)
--- 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;