ffplay: fix indentation after last commit

Signed-off-by: Marton Balint <cus@passwd.hu>

1 files changed, 89 insertions, 93 deletions

diff --git a/ffplay.c b/ffplay.c
index 24bcae2a6b..a3b34fd966 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -2424,105 +2424,101 @@ static int audio_decode_frame(VideoState *is)
     int wanted_nb_samples;
     Frame *af;
 
-    {
-        if (is->paused)
-            return -1;
+    if (is->paused)
+        return -1;
 
-        do {
-            if (!(af = frame_queue_peek_readable(&is->sampq)))
+    do {
+        if (!(af = frame_queue_peek_readable(&is->sampq)))
+            return -1;
+        frame_queue_next(&is->sampq);
+    } while (af->serial != is->audioq.serial);
+
+    data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame),
+                                           af->frame->nb_samples,
+                                           af->frame->format, 1);
+
+    dec_channel_layout =
+        (af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
+        af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame));
+    wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
+
+    if (af->frame->format        != is->audio_src.fmt            ||
+        dec_channel_layout       != is->audio_src.channel_layout ||
+        af->frame->sample_rate   != is->audio_src.freq           ||
+        (wanted_nb_samples       != af->frame->nb_samples && !is->swr_ctx)) {
+        swr_free(&is->swr_ctx);
+        is->swr_ctx = swr_alloc_set_opts(NULL,
+                                         is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
+                                         dec_channel_layout,           af->frame->format, af->frame->sample_rate,
+                                         0, NULL);
+        if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
+            av_log(NULL, AV_LOG_ERROR,
+                   "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
+                    af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame),
+                    is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
+            swr_free(&is->swr_ctx);
+            return -1;
+        }
+        is->audio_src.channel_layout = dec_channel_layout;
+        is->audio_src.channels       = av_frame_get_channels(af->frame);
+        is->audio_src.freq = af->frame->sample_rate;
+        is->audio_src.fmt = af->frame->format;
+    }
+
+    if (is->swr_ctx) {
+        const uint8_t **in = (const uint8_t **)af->frame->extended_data;
+        uint8_t **out = &is->audio_buf1;
+        int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
+        int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
+        int len2;
+        if (out_size < 0) {
+            av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
+            return -1;
+        }
+        if (wanted_nb_samples != af->frame->nb_samples) {
+            if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
+                                        wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
+                av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
                 return -1;
-            frame_queue_next(&is->sampq);
-        } while (af->serial != is->audioq.serial);
-
-        {
-            data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame),
-                                                   af->frame->nb_samples,
-                                                   af->frame->format, 1);
-
-            dec_channel_layout =
-                (af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
-                af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame));
-            wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
-
-            if (af->frame->format        != is->audio_src.fmt            ||
-                dec_channel_layout       != is->audio_src.channel_layout ||
-                af->frame->sample_rate   != is->audio_src.freq           ||
-                (wanted_nb_samples       != af->frame->nb_samples && !is->swr_ctx)) {
-                swr_free(&is->swr_ctx);
-                is->swr_ctx = swr_alloc_set_opts(NULL,
-                                                 is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
-                                                 dec_channel_layout,           af->frame->format, af->frame->sample_rate,
-                                                 0, NULL);
-                if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
-                    av_log(NULL, AV_LOG_ERROR,
-                           "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
-                            af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame),
-                            is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
-                    swr_free(&is->swr_ctx);
-                    return -1;
-                }
-                is->audio_src.channel_layout = dec_channel_layout;
-                is->audio_src.channels       = av_frame_get_channels(af->frame);
-                is->audio_src.freq = af->frame->sample_rate;
-                is->audio_src.fmt = af->frame->format;
-            }
-
-            if (is->swr_ctx) {
-                const uint8_t **in = (const uint8_t **)af->frame->extended_data;
-                uint8_t **out = &is->audio_buf1;
-                int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
-                int out_size  = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
-                int len2;
-                if (out_size < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
-                    return -1;
-                }
-                if (wanted_nb_samples != af->frame->nb_samples) {
-                    if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
-                                                wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
-                        av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
-                        return -1;
-                    }
-                }
-                av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
-                if (!is->audio_buf1)
-                    return AVERROR(ENOMEM);
-                len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
-                if (len2 < 0) {
-                    av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
-                    return -1;
-                }
-                if (len2 == out_count) {
-                    av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
-                    if (swr_init(is->swr_ctx) < 0)
-                        swr_free(&is->swr_ctx);
-                }
-                is->audio_buf = is->audio_buf1;
-                resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
-            } else {
-                is->audio_buf = af->frame->data[0];
-                resampled_data_size = data_size;
             }
+        }
+        av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
+        if (!is->audio_buf1)
+            return AVERROR(ENOMEM);
+        len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
+        if (len2 < 0) {
+            av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
+            return -1;
+        }
+        if (len2 == out_count) {
+            av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
+            if (swr_init(is->swr_ctx) < 0)
+                swr_free(&is->swr_ctx);
+        }
+        is->audio_buf = is->audio_buf1;
+        resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
+    } else {
+        is->audio_buf = af->frame->data[0];
+        resampled_data_size = data_size;
+    }
 
-            audio_clock0 = is->audio_clock;
-            /* update the audio clock with the pts */
-            if (!isnan(af->pts))
-                is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
-            else
-                is->audio_clock = NAN;
-            is->audio_clock_serial = af->serial;
+    audio_clock0 = is->audio_clock;
+    /* update the audio clock with the pts */
+    if (!isnan(af->pts))
+        is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
+    else
+        is->audio_clock = NAN;
+    is->audio_clock_serial = af->serial;
 #ifdef DEBUG
-            {
-                static double last_clock;
-                printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
-                       is->audio_clock - last_clock,
-                       is->audio_clock, audio_clock0);
-                last_clock = is->audio_clock;
-            }
-#endif
-            return resampled_data_size;
-        }
+    {
+        static double last_clock;
+        printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
+               is->audio_clock - last_clock,
+               is->audio_clock, audio_clock0);
+        last_clock = is->audio_clock;
     }
+#endif
+    return resampled_data_size;
 }
 
 /* prepare a new audio buffer */