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ffmpeg.wasm-core
Commit b9bfd888 authored by Michael Niedermayer's avatar Michael Niedermayer
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Merge commit 'edd5f957' 

* commit 'edd5f957':
  output example: use OutputStream for audio streams as well

Conflicts:
	doc/examples/muxing.c
Merged-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parents 11991a7d edd5f957
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Showing with 40 additions and 46 deletions
doc/examples/muxing.c
View file @ b9bfd888
......@@ -56,6 +56,9 @@ typedef struct OutputStream {
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
int audio_input_frame_size;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
......@@ -81,9 +84,9 @@ static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AV
}
/* Add an output stream. */
static AVStream *add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
......@@ -147,27 +150,21 @@ static AVStream *add_stream(OutputStream *ost, AVFormatContext *oc,
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return ost->st;
}
/**************************************************************/
/* audio output */
static float t, tincr, tincr2;
static int src_nb_samples;
int samples_count;
struct SwrContext *swr_ctx = NULL;
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
{
AVCodecContext *c;
int ret;
c = st->codec;
c = ost->st->codec;
/* open it */
ret = avcodec_open2(c, codec, NULL);
......@@ -177,15 +174,15 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, 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)
src_nb_samples = 10000;
ost->audio_input_frame_size = 10000;
else
src_nb_samples = c->frame_size;
ost->audio_input_frame_size = c->frame_size;
/* create resampler context */
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
......@@ -213,7 +210,7 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
/* 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];
......@@ -227,15 +224,15 @@ 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, int flush)
static void write_audio_frame(AVFormatContext *oc, OutputStream *ost, int flush)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
......@@ -244,11 +241,11 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
int dst_nb_samples;
av_init_packet(&pkt);
c = st->codec;
c = ost->st->codec;
if (!flush) {
frame->sample_rate = c->sample_rate;
frame->nb_samples = src_nb_samples;
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);
......@@ -257,14 +254,14 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
exit(1);
}
get_audio_frame(frame, c->channels);
get_audio_frame(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) + src_nb_samples,
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + ost->audio_input_frame_size,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
tmp_frame->sample_rate = c->sample_rate;
tmp_frame->nb_samples = dst_nb_samples;
......@@ -279,7 +276,7 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
/* convert to destination format */
ret = swr_convert(swr_ctx,
tmp_frame->data, dst_nb_samples,
(const uint8_t **)frame->data, src_nb_samples);
(const uint8_t **)frame->data, ost->audio_input_frame_size);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
......@@ -287,7 +284,7 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
av_frame_free(&frame);
frame = tmp_frame;
} else {
dst_nb_samples = src_nb_samples;
dst_nb_samples = ost->audio_input_frame_size;
}
frame->nb_samples = dst_nb_samples;
......@@ -307,7 +304,7 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
return;
}
ret = write_frame(oc, &c->time_base, st, &pkt);
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
......@@ -315,9 +312,9 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st, int flush)
}
}
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);
}
/**************************************************************/
......@@ -492,15 +489,14 @@ 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;
AVCodec *audio_codec, *video_codec;
double audio_time, video_time;
int flush, ret;
int have_video = 0;
int have_video = 0, have_audio = 0;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
......@@ -531,15 +527,13 @@ 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_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
OutputStream dummy;
audio_st = add_stream(&dummy, oc, &audio_codec, fmt->audio_codec);
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
......@@ -547,8 +541,8 @@ int main(int argc, char **argv)
if (have_video)
open_video(oc, video_codec, &video_st);
if (audio_st)
open_audio(oc, audio_codec, audio_st);
if (have_audio)
open_audio(oc, audio_codec, &audio_st);
av_dump_format(oc, 0, filename, 1);
......@@ -571,20 +565,20 @@ int main(int argc, char **argv)
}
flush = 0;
while ((have_video && !video_is_eof) || (audio_st && !audio_is_eof)) {
while ((have_video && !video_is_eof) || (have_audio && !audio_is_eof)) {
/* Compute current audio and video time. */
audio_time = (audio_st && !audio_is_eof) ? audio_st->pts.val * av_q2d(audio_st->time_base) : INFINITY;
audio_time = (have_audio && !audio_is_eof) ? audio_st.st->pts.val * av_q2d(audio_st.st->time_base) : INFINITY;
video_time = (have_video && !video_is_eof) ? video_st.st->pts.val * av_q2d(video_st.st->time_base) : INFINITY;
if (!flush &&
(!audio_st || audio_time >= STREAM_DURATION) &&
(!have_audio || audio_time >= STREAM_DURATION) &&
(!have_video || video_time >= STREAM_DURATION)) {
flush = 1;
}
/* write interleaved audio and video frames */
if (audio_st && !audio_is_eof && audio_time <= video_time) {
write_audio_frame(oc, audio_st, flush);
if (have_audio && !audio_is_eof && audio_time <= video_time) {
write_audio_frame(oc, &audio_st, flush);
} else if (have_video && !video_is_eof && video_time < audio_time) {
write_video_frame(oc, &video_st, flush);
}
......@@ -599,8 +593,8 @@ 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);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
......
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