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author | Michael Niedermayer <michaelni@gmx.at> | 2014-07-27 04:10:06 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2014-07-27 12:03:51 +0200 |
commit | c437765be7628c9dacb64aa1c1c46c0c39df1ca9 (patch) | |
tree | c8fb8ba1065857fc6ae407e797ac20531cb78139 | |
parent | ad91bf854b559f1afc42fc06f8a7dfbb75def5c8 (diff) | |
download | ffmpeg-c437765be7628c9dacb64aa1c1c46c0c39df1ca9.tar.gz |
doc/examples/muxing: Exchange tmp_frame and frame
Reduces difference to 56f98e340fca894a76d1ddbe33118b8d8c4db34a
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
-rw-r--r-- | doc/examples/muxing.c | 23 |
1 files changed, 12 insertions, 11 deletions
diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c index 4216a563a7..934698a1d7 100644 --- a/doc/examples/muxing.c +++ b/doc/examples/muxing.c @@ -235,10 +235,10 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A else nb_samples = c->frame_size; - ost->frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout, + ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout, + c->sample_rate, nb_samples); + ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout, c->sample_rate, nb_samples); - ost->tmp_frame = alloc_audio_frame(c->sample_fmt, c->channel_layout, - c->sample_rate, ost->frame->nb_samples); /* create resampler context */ ost->swr_ctx = swr_alloc(); @@ -266,15 +266,16 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A * 'nb_channels' channels. */ static AVFrame *get_audio_frame(OutputStream *ost) { + AVFrame *frame = ost->tmp_frame; int j, i, v; - int16_t *q = (int16_t*)ost->frame->data[0]; + int16_t *q = (int16_t*)frame->data[0]; /* check if we want to generate more frames */ if (av_compare_ts(ost->next_pts, ost->st->codec->time_base, STREAM_DURATION, (AVRational){ 1, 1 }) >= 0) return NULL; - for (j = 0; j < ost->frame->nb_samples; j++) { + for (j = 0; j <frame->nb_samples; j++) { v = (int)(sin(ost->t) * 10000); for (i = 0; i < ost->st->codec->channels; i++) *q++ = v; @@ -282,10 +283,10 @@ static AVFrame *get_audio_frame(OutputStream *ost) ost->tincr += ost->tincr2; } - ost->frame->pts = ost->next_pts; - ost->next_pts += ost->frame->nb_samples; + frame->pts = ost->next_pts; + ost->next_pts += frame->nb_samples; - return ost->frame; + return frame; } /* @@ -317,19 +318,19 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost) * internally; * make sure we do not overwrite it here */ - ret = av_frame_make_writable(ost->tmp_frame); + ret = av_frame_make_writable(ost->frame); if (ret < 0) exit(1); /* convert to destination format */ ret = swr_convert(ost->swr_ctx, - ost->tmp_frame->data, dst_nb_samples, + ost->frame->data, dst_nb_samples, (const uint8_t **)frame->data, frame->nb_samples); if (ret < 0) { fprintf(stderr, "Error while converting\n"); exit(1); } - frame = ost->tmp_frame; + frame = ost->frame; frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base); ost->samples_count += dst_nb_samples; |