virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2025-02-04 06:08:26 +02:00
Code Issues Releases Activity

Merge commit '63fd0d866c8300a8f251a15b1535e9ce40a407fb'

Browse Source 
* commit '63fd0d866c8300a8f251a15b1535e9ce40a407fb':
  output example: allocate the audio frame only once

Conflicts:
	doc/examples/muxing.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2014-06-26 23:20:54 +02:00
commit 5a55d5e445
1 changed files with 45 additions and 42 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

87
doc/examples/muxing.c
View File

@ -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. */