[FFmpeg-cvslog] output example: use OutputStream for audio streams as well
Anton Khirnov
git at videolan.org
Thu Jun 26 23:07:35 CEST 2014
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Tue Jun 24 08:56:27 2014 +0200| [edd5f957646dcbf1bb55718bc7bf1e5481c25bcb] | committer: Anton Khirnov
output example: use OutputStream for audio streams as well
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=edd5f957646dcbf1bb55718bc7bf1e5481c25bcb
---
doc/examples/output.c | 81 ++++++++++++++++++++++++-------------------------
1 file changed, 39 insertions(+), 42 deletions(-)
diff --git a/doc/examples/output.c b/doc/examples/output.c
index 1f8ff96..768f1b6 100644
--- a/doc/examples/output.c
+++ b/doc/examples/output.c
@@ -53,21 +53,21 @@ typedef struct OutputStream {
AVFrame *frame;
AVFrame *tmp_frame;
+
+ float t, tincr, tincr2;
+ int audio_input_frame_size;
} OutputStream;
/**************************************************************/
/* audio output */
-static float t, tincr, tincr2;
-static int audio_input_frame_size;
-
/*
* add an audio output stream
*/
-static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
+static void add_audio_stream(OutputStream *ost, AVFormatContext *oc,
+ enum AVCodecID codec_id)
{
AVCodecContext *c;
- AVStream *st;
AVCodec *codec;
/* find the audio encoder */
@@ -77,13 +77,13 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
exit(1);
}
- st = avformat_new_stream(oc, codec);
- if (!st) {
+ ost->st = avformat_new_stream(oc, codec);
+ if (!ost->st) {
fprintf(stderr, "Could not alloc stream\n");
exit(1);
}
- c = st->codec;
+ c = ost->st->codec;
/* put sample parameters */
c->sample_fmt = AV_SAMPLE_FMT_S16;
@@ -95,15 +95,13 @@ static AVStream *add_audio_stream(AVFormatContext *oc, enum AVCodecID codec_id)
// some formats want stream headers to be separate
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
- return st;
}
-static void open_audio(AVFormatContext *oc, AVStream *st)
+static void open_audio(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
- c = st->codec;
+ c = ost->st->codec;
/* open it */
if (avcodec_open2(c, NULL, NULL) < 0) {
@@ -112,20 +110,20 @@ static void open_audio(AVFormatContext *oc, AVStream *st)
}
/* init signal generator */
- t = 0;
- tincr = 2 * M_PI * 110.0 / c->sample_rate;
+ ost->t = 0;
+ ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
- 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;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
- audio_input_frame_size = 10000;
+ ost->audio_input_frame_size = 10000;
else
- audio_input_frame_size = c->frame_size;
+ ost->audio_input_frame_size = c->frame_size;
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
-static void get_audio_frame(AVFrame *frame, int nb_channels)
+static void get_audio_frame(OutputStream *ost, AVFrame *frame, int nb_channels)
{
int j, i, v, ret;
int16_t *q = (int16_t*)frame->data[0];
@@ -139,15 +137,15 @@ static void get_audio_frame(AVFrame *frame, int nb_channels)
exit(1);
for (j = 0; j < frame->nb_samples; j++) {
- v = (int)(sin(t) * 10000);
+ v = (int)(sin(ost->t) * 10000);
for (i = 0; i < nb_channels; i++)
*q++ = v;
- t += tincr;
- tincr += tincr2;
+ ost->t += ost->tincr;
+ ost->tincr += ost->tincr2;
}
}
-static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+static void write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
@@ -155,10 +153,10 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
int got_packet, ret;
av_init_packet(&pkt);
- c = st->codec;
+ c = ost->st->codec;
frame->sample_rate = c->sample_rate;
- frame->nb_samples = audio_input_frame_size;
+ 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);
@@ -167,13 +165,13 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
exit(1);
}
- get_audio_frame(frame, c->channels);
+ get_audio_frame(ost, frame, c->channels);
avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (!got_packet)
return;
- pkt.stream_index = st->index;
+ pkt.stream_index = ost->st->index;
/* Write the compressed frame to the media file. */
if (av_interleaved_write_frame(oc, &pkt) != 0) {
@@ -183,9 +181,9 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
av_frame_free(&frame);
}
-static void close_audio(AVFormatContext *oc, AVStream *st)
+static void close_audio(AVFormatContext *oc, OutputStream *ost)
{
- avcodec_close(st->codec);
+ avcodec_close(ost->st->codec);
}
/**************************************************************/
@@ -413,13 +411,12 @@ static void close_video(AVFormatContext *oc, OutputStream *ost)
int main(int argc, char **argv)
{
- OutputStream video_st;
+ OutputStream video_st, audio_st;
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
- AVStream *audio_st;
double audio_pts, video_pts;
- int have_video = 0;
+ int have_video = 0, have_audio = 0;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
@@ -458,21 +455,21 @@ int main(int argc, char **argv)
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
- audio_st = NULL;
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_video_stream(&video_st, oc, fmt->video_codec);
have_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
- audio_st = add_audio_stream(oc, fmt->audio_codec);
+ add_audio_stream(&audio_st, oc, fmt->audio_codec);
+ have_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, &video_st);
- if (audio_st)
- open_audio(oc, audio_st);
+ if (have_audio)
+ open_audio(oc, &audio_st);
av_dump_format(oc, 0, filename, 1);
@@ -489,8 +486,8 @@ int main(int argc, char **argv)
for (;;) {
/* Compute current audio and video time. */
- if (audio_st)
- audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
+ if (have_audio)
+ audio_pts = (double)audio_st.st->pts.val * audio_st.st->time_base.num / audio_st.st->time_base.den;
else
audio_pts = 0.0;
@@ -500,13 +497,13 @@ int main(int argc, char **argv)
else
video_pts = 0.0;
- if ((!audio_st || audio_pts >= STREAM_DURATION) &&
+ if ((!have_audio || audio_pts >= STREAM_DURATION) &&
(!have_video || video_pts >= STREAM_DURATION))
break;
/* write interleaved audio and video frames */
- if (!have_video || (have_video && audio_st && audio_pts < video_pts)) {
- write_audio_frame(oc, audio_st);
+ if (!have_video || (have_video && have_audio && audio_pts < video_pts)) {
+ write_audio_frame(oc, &audio_st);
} else {
write_video_frame(oc, &video_st);
}
@@ -521,8 +518,8 @@ int main(int argc, char **argv)
/* Close each codec. */
if (have_video)
close_video(oc, &video_st);
- if (audio_st)
- close_audio(oc, audio_st);
+ if (have_audio)
+ close_audio(oc, &audio_st);
/* Free the streams. */
for (i = 0; i < oc->nb_streams; i++) {
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