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authorStefano Sabatini <stefasab@gmail.com>2014-01-23 01:08:24 +0100
committerStefano Sabatini <stefasab@gmail.com>2014-01-23 01:08:24 +0100
commit35fe88bb51692612858cb78b3d2f11274adf554e (patch)
tree0e42a370e51ba2423b116ecd655fb7bdc6ac49a7 /doc
parentc92d2f98db68a9201b805445f126a0c51b10844d (diff)
downloadffmpeg-35fe88bb51692612858cb78b3d2f11274adf554e.tar.gz
examples/muxing: reindent after previous commit
Diffstat (limited to 'doc')
-rw-r--r--doc/examples/muxing.c64
1 files changed, 32 insertions, 32 deletions
diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
index b0c91a8ba0..a849e0abc6 100644
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -265,41 +265,41 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
c = st->codec;
if (!flush) {
- get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
-
- /* convert samples from native format to destination codec format, using the resampler */
- if (swr_ctx) {
- /* compute destination number of samples */
- dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
- c->sample_rate, c->sample_rate, AV_ROUND_UP);
- if (dst_nb_samples > max_dst_nb_samples) {
- av_free(dst_samples_data[0]);
- ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
- dst_nb_samples, c->sample_fmt, 0);
- if (ret < 0)
- exit(1);
- max_dst_nb_samples = dst_nb_samples;
- dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
- c->sample_fmt, 0);
- }
+ get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+ /* convert samples from native format to destination codec format, using the resampler */
+ if (swr_ctx) {
+ /* compute destination number of samples */
+ dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+ c->sample_rate, c->sample_rate, AV_ROUND_UP);
+ if (dst_nb_samples > max_dst_nb_samples) {
+ av_free(dst_samples_data[0]);
+ ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+ dst_nb_samples, c->sample_fmt, 0);
+ if (ret < 0)
+ exit(1);
+ max_dst_nb_samples = dst_nb_samples;
+ dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+ c->sample_fmt, 0);
+ }
- /* convert to destination format */
- ret = swr_convert(swr_ctx,
- dst_samples_data, dst_nb_samples,
- (const uint8_t **)src_samples_data, src_nb_samples);
- if (ret < 0) {
- fprintf(stderr, "Error while converting\n");
- exit(1);
+ /* convert to destination format */
+ ret = swr_convert(swr_ctx,
+ dst_samples_data, dst_nb_samples,
+ (const uint8_t **)src_samples_data, src_nb_samples);
+ if (ret < 0) {
+ fprintf(stderr, "Error while converting\n");
+ exit(1);
+ }
+ } else {
+ dst_nb_samples = src_nb_samples;
}
- } else {
- dst_nb_samples = src_nb_samples;
- }
- audio_frame->nb_samples = dst_nb_samples;
- audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
- avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
- dst_samples_data[0], dst_samples_size, 0);
- samples_count += dst_nb_samples;
+ audio_frame->nb_samples = dst_nb_samples;
+ audio_frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+ avcodec_fill_audio_frame(audio_frame, c->channels, c->sample_fmt,
+ dst_samples_data[0], dst_samples_size, 0);
+ samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, flush ? NULL : audio_frame, &got_packet);