examples/muxing: reindent after previous commit
authorStefano Sabatini <stefasab@gmail.com>
Thu, 23 Jan 2014 00:08:24 +0000 (01:08 +0100)
committerStefano Sabatini <stefasab@gmail.com>
Thu, 23 Jan 2014 00:08:24 +0000 (01:08 +0100)
doc/examples/muxing.c

index b0c91a8ba0a5be8a49db9f7faae1fb1b08f08154..a849e0abc69634a950ef346010af1bb1370cdf71 100644 (file)
@@ -265,41 +265,41 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
     c = st->codec;
 
     if (!flush) {
-    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
-
-    /* convert samples from native format to destination codec format, using the resampler */
-    if (swr_ctx) {
-        /* compute destination number of samples */
-        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
-                                        c->sample_rate, c->sample_rate, AV_ROUND_UP);
-        if (dst_nb_samples > max_dst_nb_samples) {
-            av_free(dst_samples_data[0]);
-            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
-                                   dst_nb_samples, c->sample_fmt, 0);
-            if (ret < 0)
-                exit(1);
-            max_dst_nb_samples = dst_nb_samples;
-            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
-                                                          c->sample_fmt, 0);
-        }
+        get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+        /* convert samples from native format to destination codec format, using the resampler */
+        if (swr_ctx) {
+            /* compute destination number of samples */
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+                                            c->sample_rate, c->sample_rate, AV_ROUND_UP);
+            if (dst_nb_samples > max_dst_nb_samples) {
+                av_free(dst_samples_data[0]);
+                ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                       dst_nb_samples, c->sample_fmt, 0);
+                if (ret < 0)
+                    exit(1);
+                max_dst_nb_samples = dst_nb_samples;
+                dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                              c->sample_fmt, 0);
+            }
 
-        /* convert to destination format */
-        ret = swr_convert(swr_ctx,
-                          dst_samples_data, dst_nb_samples,
-                          (const uint8_t **)src_samples_data, src_nb_samples);
-        if (ret < 0) {
-            fprintf(stderr, "Error while converting\n");
-            exit(1);
+            /* convert to destination format */
+            ret = swr_convert(swr_ctx,
+                              dst_samples_data, dst_nb_samples,
+                              (const uint8_t **)src_samples_data, src_nb_samples);
+            if (ret < 0) {
+                fprintf(stderr, "Error while converting\n");
+                exit(1);
+            }
+        } else {
+            dst_nb_samples = src_nb_samples;
         }
-    } else {
-        dst_nb_samples = src_nb_samples;
-    }
 
-    audio_frame->nb_samples = dst_nb_samples;
-    audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-    avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
-                             dst_samples_data[0], dst_samples_size, 0);
-    samples_count += dst_nb_samples;
+        audio_frame->nb_samples = dst_nb_samples;
+        audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
+                                 dst_samples_data[0], dst_samples_size, 0);
+        samples_count += dst_nb_samples;
     }
 
     ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);