Merge remote-tracking branch 'cus/stable'

* cus/stable:
  ffplay: do not wait for the picture allocation to finish on exit
  ffplay: remove two unneeded av_free_packet calls
  ffplay: remove some unneded av_frame_unref calls
  ffplay: remove unneeded avcodec_get_frame_defaults
  ffplay: use precalculated frame size and bytes per sec values
  ffplay: precalculate audio output frame size and byte per sec

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 12 insertions, 13 deletions

diff --git a/ffplay.c b/ffplay.c
index 0b5331b72f..164872aaea 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -145,6 +145,8 @@ typedef struct AudioParams {
     int channels;
     int64_t channel_layout;
     enum AVSampleFormat fmt;
+    int frame_size;
+    int bytes_per_sec;
 } AudioParams;
 
 typedef struct Clock {
@@ -1608,7 +1610,7 @@ static int queue_picture(VideoState *is, AVFrame *src_frame, double pts, double
         }
         /* if the queue is aborted, we have to pop the pending ALLOC event or wait for the allocation to complete */
         if (is->videoq.abort_request && SDL_PeepEvents(&event, 1, SDL_GETEVENT, SDL_EVENTMASK(FF_ALLOC_EVENT)) != 1) {
-            while (!vp->allocated) {
+            while (!vp->allocated && !is->abort_request) {
                 SDL_CondWait(is->pictq_cond, is->pictq_mutex);
             }
         }
@@ -1920,7 +1922,6 @@ static int video_thread(void *arg)
         while (is->paused && !is->videoq.abort_request)
             SDL_Delay(10);
 
-        avcodec_get_frame_defaults(frame);
         av_free_packet(&pkt);
 
         ret = get_video_frame(is, frame, &pkt, &serial);
@@ -1947,7 +1948,6 @@ static int video_thread(void *arg)
                 event.type = FF_QUIT_EVENT;
                 event.user.data1 = is;
                 SDL_PushEvent(&event);
-                av_free_packet(&pkt);
                 goto the_end;
             }
             filt_in  = is->in_video_filter;
@@ -1962,9 +1962,6 @@ static int video_thread(void *arg)
         ret = av_buffersrc_add_frame(filt_in, frame);
         if (ret < 0)
             goto the_end;
-        av_frame_unref(frame);
-        avcodec_get_frame_defaults(frame);
-        av_free_packet(&pkt);
 
         while (ret >= 0) {
             is->frame_last_returned_time = av_gettime() / 1000000.0;
@@ -2241,7 +2238,6 @@ static int audio_decode_frame(VideoState *is)
 
                 if ((ret = av_buffersrc_add_frame(is->in_audio_filter, is->frame)) < 0)
                     return ret;
-                av_frame_unref(is->frame);
 #endif
             }
 #if CONFIG_AVFILTER
@@ -2378,8 +2374,6 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
 {
     VideoState *is = opaque;
     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);
 
     audio_callback_time = av_gettime();
 
@@ -2389,7 +2383,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
            if (audio_size < 0) {
                 /* if error, just output silence */
                is->audio_buf      = is->silence_buf;
-               is->audio_buf_size = sizeof(is->silence_buf) / frame_size * frame_size;
+               is->audio_buf_size = sizeof(is->silence_buf) / is->audio_tgt.frame_size * is->audio_tgt.frame_size;
            } else {
                if (is->show_mode != SHOW_MODE_VIDEO)
                    update_sample_display(is, (int16_t *)is->audio_buf, audio_size);
@@ -2405,11 +2399,10 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len)
         stream += len1;
         is->audio_buf_index += len1;
     }
-    bytes_per_sec = is->audio_tgt.freq * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
     is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index;
     /* Let's assume the audio driver that is used by SDL has two periods. */
     if (!isnan(is->audio_clock)) {
-        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
+        set_clock_at(&is->audclk, is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / is->audio_tgt.bytes_per_sec, is->audio_clock_serial, audio_callback_time / 1000000.0);
         sync_clock_to_slave(&is->extclk, &is->audclk);
     }
 }
@@ -2468,6 +2461,12 @@ static int audio_open(void *opaque, int64_t wanted_channel_layout, int wanted_nb
     audio_hw_params->freq = spec.freq;
     audio_hw_params->channel_layout = wanted_channel_layout;
     audio_hw_params->channels =  spec.channels;
+    audio_hw_params->frame_size = av_samples_get_buffer_size(NULL, audio_hw_params->channels, 1, audio_hw_params->fmt, 1);
+    audio_hw_params->bytes_per_sec = av_samples_get_buffer_size(NULL, audio_hw_params->channels, audio_hw_params->freq, audio_hw_params->fmt, 1);
+    if (audio_hw_params->bytes_per_sec <= 0 || audio_hw_params->frame_size <= 0) {
+        av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size failed\n");
+        return -1;
+    }
     return spec.size;
 }
 
@@ -2572,7 +2571,7 @@ static int stream_component_open(VideoState *is, int stream_index)
         is->audio_diff_avg_count = 0;
         /* since we do not have a precise anough audio fifo fullness,
            we correct audio sync only if larger than this threshold */
-        is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / av_samples_get_buffer_size(NULL, is->audio_tgt.channels, is->audio_tgt.freq, is->audio_tgt.fmt, 1);
+        is->audio_diff_threshold = 2.0 * is->audio_hw_buf_size / is->audio_tgt.bytes_per_sec;
 
         memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
         memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));