diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
index b6958a5e0b00357f760201e5bf92518bcee875ed..df3e3149515eeef9a698c7cd639d073a60303245 100644
--- a/doc/examples/muxing.c
+++ b/doc/examples/muxing.c
@@ -34,6 +34,7 @@
#include <string.h>
#include <math.h>
+#include <libavutil/avassert.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
@@ -58,7 +59,6 @@ typedef struct OutputStream {
AVFrame *tmp_frame;
float t, tincr, tincr2;
- int audio_input_frame_size;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
@@ -179,10 +179,27 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+ ost->frame = av_frame_alloc();
+ if (!ost->frame)
+ exit(1);
+
+ ost->frame->sample_rate = c->sample_rate;
+ ost->frame->format = AV_SAMPLE_FMT_S16;
+ ost->frame->channel_layout = c->channel_layout;
+
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
- ost->audio_input_frame_size = 10000;
+ ost->frame->nb_samples = 10000;
else
- ost->audio_input_frame_size = c->frame_size;
+ ost->frame->nb_samples = c->frame_size;
+
+ ost->tmp_frame = av_frame_alloc();
+ if (!ost->frame)
+ exit(1);
+
+ ost->tmp_frame->sample_rate = c->sample_rate;
+ ost->tmp_frame->format = c->sample_fmt;
+ ost->tmp_frame->channel_layout = c->channel_layout;
+ ost->tmp_frame->nb_samples = ost->frame->nb_samples;
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
@@ -206,6 +223,17 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
exit(1);
}
}
+
+ ret = av_frame_get_buffer(ost->frame, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate an audio frame.\n");
+ exit(1);
+ }
+ ret = av_frame_get_buffer(ost->tmp_frame, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate an audio frame.\n");
+ exit(1);
+ }
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -236,58 +264,37 @@ static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
- AVFrame *frame = av_frame_alloc();
int got_packet, ret;
int dst_nb_samples;
+ AVFrame *frame;
av_init_packet(&pkt);
c = ost->st->codec;
if (!flush) {
- frame->sample_rate = c->sample_rate;
- frame->nb_samples = ost->audio_input_frame_size;
- frame->format = AV_SAMPLE_FMT_S16;
- frame->channel_layout = c->channel_layout;
- ret = av_frame_get_buffer(frame, 0);
- if (ret < 0) {
- fprintf(stderr, "Could not allocate an audio frame.\n");
- exit(1);
- }
-
- get_audio_frame(ost, frame, c->channels);
+ get_audio_frame(ost, ost->frame, c->channels);
/* convert samples from native format to destination codec format, using the resampler */
if (swr_ctx) {
- AVFrame *tmp_frame = av_frame_alloc();
-
/* compute destination number of samples */
- dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + ost->audio_input_frame_size,
+ dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + ost->frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
- tmp_frame->sample_rate = c->sample_rate;
- tmp_frame->nb_samples = dst_nb_samples;
- tmp_frame->format = c->sample_fmt;
- tmp_frame->channel_layout = c->channel_layout;
- ret = av_frame_get_buffer(tmp_frame, 0);
- if (ret < 0) {
- fprintf(stderr, "Could not allocate an audio frame.\n");
- exit(1);
- }
+ av_assert0(dst_nb_samples == ost->frame->nb_samples);
/* convert to destination format */
ret = swr_convert(swr_ctx,
- tmp_frame->data, dst_nb_samples,
- (const uint8_t **)frame->data, ost->audio_input_frame_size);
+ ost->tmp_frame->data, dst_nb_samples,
+ (const uint8_t **)ost->frame->data, ost->frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
- av_frame_free(&frame);
- frame = tmp_frame;
+ frame = ost->tmp_frame;
} else {
- dst_nb_samples = ost->audio_input_frame_size;
+ dst_nb_samples = ost->frame->nb_samples;
+ frame = ost->frame;
}
- frame->nb_samples = dst_nb_samples;
frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
samples_count += dst_nb_samples;
}
@@ -301,6 +308,7 @@ static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush)
if (!got_packet) {
if (flush)
audio_is_eof = 1;
+
return;
}
@@ -312,11 +320,6 @@ static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush)
}
}
-static void close_audio(AVFormatContext *oc, OutputStream *ost)
-{
- avcodec_close(ost->st->codec);
-}
-
/**************************************************************/
/* video output */
@@ -477,7 +480,7 @@ static void write_video_frame(AVFormatContext *oc, OutputStream *ost, int flush)
frame_count++;
}
-static void close_video(AVFormatContext *oc, OutputStream *ost)
+static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_close(ost->st->codec);
av_frame_free(&ost->frame);
@@ -489,7 +492,7 @@ static void close_video(AVFormatContext *oc, OutputStream *ost)
int main(int argc, char **argv)
{
- OutputStream video_st, audio_st;
+ OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
@@ -592,9 +595,9 @@ int main(int argc, char **argv)
/* Close each codec. */
if (have_video)
- close_video(oc, &video_st);
+ close_stream(oc, &video_st);
if (have_audio)
- close_audio(oc, &audio_st);
+ close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */