libavcodec: Implementation of 32 bit fixed point FFT

[ffmpeg] / ffplay.c

diff --git a/ffplay.c b/ffplay.c
index f99b657cdedfc42fdd23703ec9a7fd871db26a16..d462eee43a9abd2fa4a11ba25439312288ead542 100644 (file)
--- a/ffplay.c
+++ b/ffplay.c
@@ -216,6 +216,7 @@ typedef struct VideoState {
     int frame_drops_early;
     int frame_drops_late;
     AVFrame *frame;
+    int64_t audio_frame_next_pts;
 
     enum ShowMode {
         SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -974,7 +975,8 @@ static void video_audio_display(VideoState *s)
                 }
                 av_rdft_calc(s->rdft, data[ch]);
             }
-            // least efficient way to do this, we should of course directly access it but its more than fast enough
+            /* Least efficient way to do this, we should of course
+             * directly access it but it is more than fast enough. */
             for (y = 0; y < s->height; y++) {
                 double w = 1 / sqrt(nb_freq);
                 int a = sqrt(w * sqrt(data[0][2 * y + 0] * data[0][2 * y + 0] + data[0][2 * y + 1] * data[0][2 * y + 1]));
@@ -2128,8 +2130,6 @@ static int audio_decode_frame(VideoState *is)
     int64_t dec_channel_layout;
     int got_frame;
     av_unused double audio_clock0;
-    int new_packet = 0;
-    int flush_complete = 0;
     int wanted_nb_samples;
     AVRational tb;
     int ret;
@@ -2137,7 +2137,7 @@ static int audio_decode_frame(VideoState *is)
 
     for (;;) {
         /* NOTE: the audio packet can contain several frames */
-        while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet) || is->audio_buf_frames_pending) {
+        while (pkt_temp->stream_index != -1 || is->audio_buf_frames_pending) {
             if (!is->frame) {
                 if (!(is->frame = avcodec_alloc_frame()))
                     return AVERROR(ENOMEM);
@@ -2153,9 +2153,6 @@ static int audio_decode_frame(VideoState *is)
                 return -1;
 
             if (!is->audio_buf_frames_pending) {
-                if (flush_complete)
-                    break;
-                new_packet = 0;
                 len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
                 if (len1 < 0) {
                     /* if error, we skip the frame */
@@ -2163,23 +2160,30 @@ static int audio_decode_frame(VideoState *is)
                     break;
                 }
 
+                pkt_temp->dts =
+                pkt_temp->pts = AV_NOPTS_VALUE;
                 pkt_temp->data += len1;
                 pkt_temp->size -= len1;
+                if (pkt_temp->data && pkt_temp->size <= 0 || !pkt_temp->data && !got_frame)
+                    pkt_temp->stream_index = -1;
 
-                if (!got_frame) {
-                    /* stop sending empty packets if the decoder is finished */
-                    if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
-                        flush_complete = 1;
+                if (!got_frame)
                     continue;
-                }
 
                 tb = (AVRational){1, is->frame->sample_rate};
                 if (is->frame->pts != AV_NOPTS_VALUE)
                     is->frame->pts = av_rescale_q(is->frame->pts, dec->time_base, tb);
                 else if (is->frame->pkt_pts != AV_NOPTS_VALUE)
                     is->frame->pts = av_rescale_q(is->frame->pkt_pts, is->audio_st->time_base, tb);
-                if (pkt_temp->pts != AV_NOPTS_VALUE)
-                    pkt_temp->pts += (double) is->frame->nb_samples / is->frame->sample_rate / av_q2d(is->audio_st->time_base);
+                else if (is->audio_frame_next_pts != AV_NOPTS_VALUE)
+#if CONFIG_AVFILTER
+                    is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_filter_src.freq}, tb);
+#else
+                    is->frame->pts = av_rescale_q(is->audio_frame_next_pts, (AVRational){1, is->audio_src.freq}, tb);
+#endif
+
+                if (is->frame->pts != AV_NOPTS_VALUE)
+                    is->audio_frame_next_pts = is->frame->pts + is->frame->nb_samples;
 
 #if CONFIG_AVFILTER
                 dec_channel_layout = get_valid_channel_layout(is->frame->channel_layout, av_frame_get_channels(is->frame));
@@ -2317,6 +2321,7 @@ static int audio_decode_frame(VideoState *is)
         if (pkt->data)
             av_free_packet(pkt);
         memset(pkt_temp, 0, sizeof(*pkt_temp));
+        pkt_temp->stream_index = -1;
 
         if (is->audioq.abort_request) {
             return -1;
@@ -2326,13 +2331,15 @@ static int audio_decode_frame(VideoState *is)
             SDL_CondSignal(is->continue_read_thread);
 
         /* read next packet */
-        if ((new_packet = packet_queue_get(&is->audioq, pkt, 1, &is->audio_pkt_temp_serial)) < 0)
+        if ((packet_queue_get(&is->audioq, pkt, 1, &is->audio_pkt_temp_serial)) < 0)
             return -1;
 
         if (pkt->data == flush_pkt.data) {
             avcodec_flush_buffers(dec);
-            flush_complete = 0;
             is->audio_buf_frames_pending = 0;
+            is->audio_frame_next_pts = AV_NOPTS_VALUE;
+            if ((is->ic->iformat->flags & (AVFMT_NOBINSEARCH | AVFMT_NOGENSEARCH | AVFMT_NO_BYTE_SEEK)) && !is->ic->iformat->read_seek)
+                is->audio_frame_next_pts = is->audio_st->start_time;
         }
 
         *pkt_temp = *pkt;
@@ -2541,6 +2548,7 @@ static int stream_component_open(VideoState *is, int stream_index)
 
         memset(&is->audio_pkt, 0, sizeof(is->audio_pkt));
         memset(&is->audio_pkt_temp, 0, sizeof(is->audio_pkt_temp));
+        is->audio_pkt_temp.stream_index = -1;
 
         is->audio_stream = stream_index;
         is->audio_st = ic->streams[stream_index];
@@ -2799,7 +2807,8 @@ static int read_thread(void *arg)
     }
 
     if (is->video_stream < 0 && is->audio_stream < 0) {
-        av_log(NULL, AV_LOG_FATAL, "%s: could not open codecs\n", is->filename);
+        av_log(NULL, AV_LOG_FATAL, "Failed to open file '%s' or configure filtergraph\n",
+               is->filename);
         ret = -1;
         goto fail;
     }