[FFmpeg-cvslog] doc/examples/muxing: Always use swr, simplifies code slightly

Michael Niedermayer git at videolan.org
Sun Jul 27 01:56:48 CEST 2014


ffmpeg | branch: master | Michael Niedermayer <michaelni at gmx.at> | Sun Jul 27 01:32:19 2014 +0200| [fbd46e2f1caeb190a601f0f600b6199e632d7292] | committer: Michael Niedermayer

doc/examples/muxing: Always use swr, simplifies code slightly

Idea-from: 56f98e340fca894a76d1ddbe33118b8d8c4db34a
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=fbd46e2f1caeb190a601f0f600b6199e632d7292
---

 doc/examples/muxing.c |    6 ------
 1 file changed, 6 deletions(-)

diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
index 66027bf..2d97fde 100644
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -241,7 +241,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
                                        c->sample_rate, ost->frame->nb_samples);
 
     /* create resampler context */
-    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
         ost->swr_ctx = swr_alloc();
         if (!ost->swr_ctx) {
             fprintf(stderr, "Could not allocate resampler context\n");
@@ -261,7 +260,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
             fprintf(stderr, "Failed to initialize the resampling context\n");
             exit(1);
         }
-    }
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -318,7 +316,6 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
 
     if (frame) {
         /* convert samples from native format to destination codec format, using the resampler */
-        if (ost->swr_ctx) {
             /* compute destination number of samples */
             dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
                                             c->sample_rate, c->sample_rate, AV_ROUND_UP);
@@ -333,9 +330,6 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
                 exit(1);
             }
             frame = ost->tmp_frame;
-        } else {
-            dst_nb_samples = frame->nb_samples;
-        }
 
         frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
         ost->samples_count += dst_nb_samples;



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