[FFmpeg-devel] [PATCH] ffplay: rename temporary variable pts in audio_decode_frame()

[Stefano Sabatini]

Also rename variable pts to pts_time.
---
 ffplay.c |   16 +++++++---------
 1 file changed, 7 insertions(+), 9 deletions(-)

diff --git a/ffplay.c b/ffplay.c
index f5acbca..80e3187 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -2073,7 +2073,7 @@ static int synchronize_audio(VideoState *is, int nb_samples)
  * stored in is->audio_buf, with size in bytes given by the return
  * value.
  */
-static int audio_decode_frame(VideoState *is, double *pts_ptr)
+static int audio_decode_frame(VideoState *is, double *pts_time)
 {
     AVPacket *pkt_temp = &is->audio_pkt_temp;
     AVPacket *pkt = &is->audio_pkt;
@@ -2081,7 +2081,6 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
     int len1, len2, data_size, resampled_data_size;
     int64_t dec_channel_layout;
     int got_frame;
-    double pts;
     int new_packet = 0;
     int flush_complete = 0;
     int wanted_nb_samples;
@@ -2175,16 +2174,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
             }
 
             /* if no pts, then compute it */
-            pts = is->audio_clock;
-            *pts_ptr = pts;
+            *pts_time = is->audio_clock;
             is->audio_clock += (double)data_size /
                 (is->frame->channels * is->frame->sample_rate * av_get_bytes_per_sample(is->frame->format));
 #ifdef DEBUG
             {
                 static double last_clock;
-                printf("audio: delay=%0.3f clock=%0.3f pts=%0.3f\n",
+                printf("audio: delay=%0.3f clock=%0.3f pts_time=%0.3f\n",
                        is->audio_clock - last_clock,
-                       is->audio_clock, pts);
+                       is->audio_clock, *pts_time);
                 last_clock = is->audio_clock;
             }
 #endif
@@ -2214,7 +2212,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
 
         *pkt_temp = *pkt;
 
-        /* if update the audio clock with the pts */
+        /* if update the audio clock with the pts time */
         if (pkt->pts != AV_NOPTS_VALUE) {
             is->audio_clock = av_q2d(is->audio_st->time_base)*pkt->pts;
         }
@@ -2228,13 +2226,13 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
     int audio_size, len1;
     int bytes_per_sec;
     int frame_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, 1, is->audio_tgt.fmt, 1);
-    double pts;
+    double pts_time;
 
     audio_callback_time = av_gettime();
 
     while (len > 0) {
         if (is->audio_buf_index >= is->audio_buf_size) {
-           audio_size = audio_decode_frame(is, &pts);
+           audio_size = audio_decode_frame(is, &pts_time);
            if (audio_size < 0) {
                 /* if error, just output silence */
                is->audio_buf      = is->silence_buf;
-- 
1.7.9.5