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doc/examples/muxing: Exchange tmp_frame and frame

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Reduces difference to 56f98e340fca894a76d1ddbe33118b8d8c4db34a

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2014-07-27 04:10:06 +02:00
parent ad91bf854b
commit c437765be7
1 changed files with 12 additions and 11 deletions
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23
doc/examples/muxing.c
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@ -235,10 +235,10 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
else else
nb_samples = c->frame_size; 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); 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 */ /* create resampler context */
ost->swr_ctx = swr_alloc(); ost->swr_ctx = swr_alloc();
@ -266,15 +266,16 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
* 'nb_channels' channels. */ * 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost) static AVFrame *get_audio_frame(OutputStream *ost)
{ {
AVFrame *frame = ost->tmp_frame;
int j, i, v; 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 */ /* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->st->codec->time_base, if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0) STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL; 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); v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->st->codec->channels; i++) for (i = 0; i < ost->st->codec->channels; i++)
*q++ = v; *q++ = v;
@ -282,10 +283,10 @@ static AVFrame *get_audio_frame(OutputStream *ost)
ost->tincr += ost->tincr2; ost->tincr += ost->tincr2;
} }
ost->frame->pts = ost->next_pts; frame->pts = ost->next_pts;
ost->next_pts += ost->frame->nb_samples; 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; * internally;
* make sure we do not overwrite it here * 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) if (ret < 0)
exit(1); exit(1);
/* convert to destination format */ /* convert to destination format */
ret = swr_convert(ost->swr_ctx, 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); (const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) { if (ret < 0) {
fprintf(stderr, "Error while converting\n"); fprintf(stderr, "Error while converting\n");
exit(1); 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); frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples; ost->samples_count += dst_nb_samples;