doc/examples/muxing: add alloc_audio_frame() and use it to simplify code.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Anton Khirnov 2014-07-27 01:12:25 +02:00 committed by Michael Niedermayer
parent a98cadef7f
commit 22e9fe06eb

View File

@ -178,9 +178,38 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
/**************************************************************/ /**************************************************************/
/* audio output */ /* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg) static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{ {
AVCodecContext *c; AVCodecContext *c;
int nb_samples;
int ret; int ret;
AVDictionary *opt = NULL; AVDictionary *opt = NULL;
@ -201,27 +230,15 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
/* increment frequency by 110 Hz per second */ /* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate; 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) if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
ost->frame->nb_samples = 10000; nb_samples = 10000;
else else
ost->frame->nb_samples = c->frame_size; nb_samples = c->frame_size;
ost->tmp_frame = av_frame_alloc(); ost->frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
if (!ost->frame) c->sample_rate, nb_samples);
exit(1); ost->tmp_frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, ost->frame->nb_samples);
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 */ /* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) { if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
@ -245,17 +262,6 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, A
exit(1); 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 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and