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diff --git a/doc/examples/muxing.c b/doc/examples/muxing.c
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+/*
+ * Copyright (c) 2003 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat API example.
+ *
+ * Output a media file in any supported libavformat format.
+ * The default codecs are used.
+ * @example doc/examples/muxing.c
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+#include <libavutil/opt.h>
+#include <libavutil/mathematics.h>
+#include <libavformat/avformat.h>
+#include <libswscale/swscale.h>
+#include <libswresample/swresample.h>
+
+/* 5 seconds stream duration */
+#define STREAM_DURATION 200.0
+#define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
+#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
+
+static int sws_flags = SWS_BICUBIC;
+
+/* Add an output stream. */
+static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
+ enum AVCodecID codec_id)
+{
+ AVCodecContext *c;
+ AVStream *st;
+
+ /* find the encoder */
+ *codec = avcodec_find_encoder(codec_id);
+ if (!(*codec)) {
+ fprintf(stderr, "Could not find encoder for '%s'\n",
+ avcodec_get_name(codec_id));
+ exit(1);
+ }
+
+ st = avformat_new_stream(oc, *codec);
+ if (!st) {
+ fprintf(stderr, "Could not allocate stream\n");
+ exit(1);
+ }
+ st->id = oc->nb_streams-1;
+ c = st->codec;
+
+ switch ((*codec)->type) {
+ case AVMEDIA_TYPE_AUDIO:
+ c->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ c->bit_rate = 64000;
+ c->sample_rate = 44100;
+ c->channels = 2;
+ break;
+
+ case AVMEDIA_TYPE_VIDEO:
+ c->codec_id = codec_id;
+
+ c->bit_rate = 400000;
+ /* Resolution must be a multiple of two. */
+ c->width = 352;
+ c->height = 288;
+ /* timebase: This is the fundamental unit of time (in seconds) in terms
+ * of which frame timestamps are represented. For fixed-fps content,
+ * timebase should be 1/framerate and timestamp increments should be
+ * identical to 1. */
+ c->time_base.den = STREAM_FRAME_RATE;
+ c->time_base.num = 1;
+ c->gop_size = 12; /* emit one intra frame every twelve frames at most */
+ c->pix_fmt = STREAM_PIX_FMT;
+ if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
+ /* just for testing, we also add B frames */
+ c->max_b_frames = 2;
+ }
+ if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
+ /* Needed to avoid using macroblocks in which some coeffs overflow.
+ * This does not happen with normal video, it just happens here as
+ * the motion of the chroma plane does not match the luma plane. */
+ c->mb_decision = 2;
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ /* Some formats want stream headers to be separate. */
+ if (oc->oformat->flags & AVFMT_GLOBALHEADER)
+ c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+ return st;
+}
+
+/**************************************************************/
+/* audio output */
+
+static float t, tincr, tincr2;
+
+static uint8_t **src_samples_data;
+static int src_samples_linesize;
+static int src_nb_samples;
+
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int dst_samples_linesize;
+int dst_samples_size;
+
+struct SwrContext *swr_ctx = NULL;
+
+static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+{
+ AVCodecContext *c;
+ int ret;
+
+ c = st->codec;
+
+ /* open it */
+ ret = avcodec_open2(c, codec, NULL);
+ if (ret < 0) {
+ fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
+ exit(1);
+ }
+
+ /* init signal generator */
+ t = 0;
+ 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;
+
+ src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+ 10000 : c->frame_size;
+
+ ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+ src_nb_samples, AV_SAMPLE_FMT_S16, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate source samples\n");
+ exit(1);
+ }
+
+ /* compute the number of converted samples: buffering is avoided
+ * ensuring that the output buffer will contain at least all the
+ * converted input samples */
+ max_dst_nb_samples = src_nb_samples;
+
+ /* create resampler context */
+ if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+ swr_ctx = swr_alloc();
+ if (!swr_ctx) {
+ fprintf(stderr, "Could not allocate resampler context\n");
+ exit(1);
+ }
+
+ /* set options */
+ av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
+ av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
+ av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
+ av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
+ av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
+ av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
+
+ /* initialize the resampling context */
+ if ((ret = swr_init(swr_ctx)) < 0) {
+ fprintf(stderr, "Failed to initialize the resampling context\n");
+ exit(1);
+ }
+
+ ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+ max_dst_nb_samples, c->sample_fmt, 0);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate destination samples\n");
+ exit(1);
+ }
+ } else {
+ dst_samples_data = src_samples_data;
+ }
+ dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+ c->sample_fmt, 0);
+}
+
+/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
+ * 'nb_channels' channels. */
+static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
+{
+ int j, i, v;
+ int16_t *q;
+
+ q = samples;
+ for (j = 0; j < frame_size; j++) {
+ v = (int)(sin(t) * 10000);
+ for (i = 0; i < nb_channels; i++)
+ *q++ = v;
+ t += tincr;
+ tincr += tincr2;
+ }
+}
+
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
+{
+ AVCodecContext *c;
+ AVPacket pkt = { 0 }; // data and size must be 0;
+ AVFrame *frame = av_frame_alloc();
+ int got_packet, ret, dst_nb_samples;
+
+ av_init_packet(&pkt);
+ c = st->codec;
+
+ get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
+
+ /* convert samples from native format to destination codec format, using the resampler */
+ if (swr_ctx) {
+ /* compute destination number of samples */
+ dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
+ c->sample_rate, c->sample_rate, AV_ROUND_UP);
+ if (dst_nb_samples > max_dst_nb_samples) {
+ av_free(dst_samples_data[0]);
+ ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+ dst_nb_samples, c->sample_fmt, 0);
+ if (ret < 0)
+ exit(1);
+ max_dst_nb_samples = dst_nb_samples;
+ dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+ c->sample_fmt, 0);
+ }
+
+ /* convert to destination format */
+ ret = swr_convert(swr_ctx,
+ dst_samples_data, dst_nb_samples,
+ (const uint8_t **)src_samples_data, src_nb_samples);
+ if (ret < 0) {
+ fprintf(stderr, "Error while converting\n");
+ exit(1);
+ }
+ } else {
+ dst_nb_samples = src_nb_samples;
+ }
+
+ frame->nb_samples = dst_nb_samples;
+ avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+ dst_samples_data[0], dst_samples_size, 0);
+
+ ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
+ if (ret < 0) {
+ fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
+ exit(1);
+ }
+
+ if (!got_packet)
+ goto freeframe;
+
+ pkt.stream_index = st->index;
+
+ /* Write the compressed frame to the media file. */
+ ret = av_interleaved_write_frame(oc, &pkt);
+ if (ret != 0) {
+ fprintf(stderr, "Error while writing audio frame: %s\n",
+ av_err2str(ret));
+ exit(1);
+ }
+freeframe:
+ av_frame_free(&frame);
+}
+
+static void close_audio(AVFormatContext *oc, AVStream *st)
+{
+ avcodec_close(st->codec);
+ if (dst_samples_data != src_samples_data) {
+ av_free(dst_samples_data[0]);
+ av_free(dst_samples_data);
+ }
+ av_free(src_samples_data[0]);
+ av_free(src_samples_data);
+}
+
+/**************************************************************/
+/* video output */
+
+static AVFrame *frame;
+static AVPicture src_picture, dst_picture;
+static int frame_count;
+
+static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
+{
+ int ret;
+ AVCodecContext *c = st->codec;
+
+ /* open the codec */
+ ret = avcodec_open2(c, codec, NULL);
+ if (ret < 0) {
+ fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
+ exit(1);
+ }
+
+ /* allocate and init a re-usable frame */
+ frame = av_frame_alloc();
+ if (!frame) {
+ fprintf(stderr, "Could not allocate video frame\n");
+ exit(1);
+ }
+
+ /* Allocate the encoded raw picture. */
+ ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
+ exit(1);
+ }
+
+ /* If the output format is not YUV420P, then a temporary YUV420P
+ * picture is needed too. It is then converted to the required
+ * output format. */
+ if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+ ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
+ if (ret < 0) {
+ fprintf(stderr, "Could not allocate temporary picture: %s\n",
+ av_err2str(ret));
+ exit(1);
+ }
+ }
+
+ /* copy data and linesize picture pointers to frame */
+ *((AVPicture *)frame) = dst_picture;
+}
+
+/* Prepare a dummy image. */
+static void fill_yuv_image(AVPicture *pict, int frame_index,
+ int width, int height)
+{
+ int x, y, i;
+
+ i = frame_index;
+
+ /* Y */
+ for (y = 0; y < height; y++)
+ for (x = 0; x < width; x++)
+ pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
+
+ /* Cb and Cr */
+ for (y = 0; y < height / 2; y++) {
+ for (x = 0; x < width / 2; x++) {
+ pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
+ pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
+ }
+ }
+}
+
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
+{
+ int ret;
+ static struct SwsContext *sws_ctx;
+ AVCodecContext *c = st->codec;
+
+ if (frame_count >= STREAM_NB_FRAMES) {
+ /* No more frames to compress. The codec has a latency of a few
+ * frames if using B-frames, so we get the last frames by
+ * passing the same picture again. */
+ } else {
+ if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
+ /* as we only generate a YUV420P picture, we must convert it
+ * to the codec pixel format if needed */
+ if (!sws_ctx) {
+ sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
+ c->width, c->height, c->pix_fmt,
+ sws_flags, NULL, NULL, NULL);
+ if (!sws_ctx) {
+ fprintf(stderr,
+ "Could not initialize the conversion context\n");
+ exit(1);
+ }
+ }
+ fill_yuv_image(&src_picture, frame_count, c->width, c->height);
+ sws_scale(sws_ctx,
+ (const uint8_t * const *)src_picture.data, src_picture.linesize,
+ 0, c->height, dst_picture.data, dst_picture.linesize);
+ } else {
+ fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
+ }
+ }
+
+ if (oc->oformat->flags & AVFMT_RAWPICTURE) {
+ /* Raw video case - directly store the picture in the packet */
+ AVPacket pkt;
+ av_init_packet(&pkt);
+
+ pkt.flags |= AV_PKT_FLAG_KEY;
+ pkt.stream_index = st->index;
+ pkt.data = dst_picture.data[0];
+ pkt.size = sizeof(AVPicture);
+
+ ret = av_interleaved_write_frame(oc, &pkt);
+ } else {
+ AVPacket pkt = { 0 };
+ int got_packet;
+ av_init_packet(&pkt);
+
+ /* encode the image */
+ ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
+ if (ret < 0) {
+ fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
+ exit(1);
+ }
+ /* If size is zero, it means the image was buffered. */
+
+ if (!ret && got_packet && pkt.size) {
+ pkt.stream_index = st->index;
+
+ /* Write the compressed frame to the media file. */
+ ret = av_interleaved_write_frame(oc, &pkt);
+ } else {
+ ret = 0;
+ }
+ }
+ if (ret != 0) {
+ fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
+ exit(1);
+ }
+ frame_count++;
+}
+
+static void close_video(AVFormatContext *oc, AVStream *st)
+{
+ avcodec_close(st->codec);
+ av_free(src_picture.data[0]);
+ av_free(dst_picture.data[0]);
+ av_free(frame);
+}
+
+/**************************************************************/
+/* media file output */
+
+int main(int argc, char **argv)
+{
+ const char *filename;
+ AVOutputFormat *fmt;
+ AVFormatContext *oc;
+ AVStream *audio_st, *video_st;
+ AVCodec *audio_codec, *video_codec;
+ double audio_time, video_time;
+ int ret;
+
+ /* Initialize libavcodec, and register all codecs and formats. */
+ av_register_all();
+
+ if (argc != 2) {
+ printf("usage: %s output_file\n"
+ "API example program to output a media file with libavformat.\n"
+ "This program generates a synthetic audio and video stream, encodes and\n"
+ "muxes them into a file named output_file.\n"
+ "The output format is automatically guessed according to the file extension.\n"
+ "Raw images can also be output by using '%%d' in the filename.\n"
+ "\n", argv[0]);
+ return 1;
+ }
+
+ filename = argv[1];
+
+ /* allocate the output media context */
+ avformat_alloc_output_context2(&oc, NULL, NULL, filename);
+ if (!oc) {
+ printf("Could not deduce output format from file extension: using MPEG.\n");
+ avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
+ }
+ if (!oc) {
+ return 1;
+ }
+ fmt = oc->oformat;
+
+ /* Add the audio and video streams using the default format codecs
+ * and initialize the codecs. */
+ video_st = NULL;
+ audio_st = NULL;
+
+ if (fmt->video_codec != AV_CODEC_ID_NONE) {
+ video_st = add_stream(oc, &video_codec, fmt->video_codec);
+ }
+ if (fmt->audio_codec != AV_CODEC_ID_NONE) {
+ audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
+ }
+
+ /* Now that all the parameters are set, we can open the audio and
+ * video codecs and allocate the necessary encode buffers. */
+ if (video_st)
+ open_video(oc, video_codec, video_st);
+ if (audio_st)
+ open_audio(oc, audio_codec, audio_st);
+
+ av_dump_format(oc, 0, filename, 1);
+
+ /* open the output file, if needed */
+ if (!(fmt->flags & AVFMT_NOFILE)) {
+ ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
+ if (ret < 0) {
+ fprintf(stderr, "Could not open '%s': %s\n", filename,
+ av_err2str(ret));
+ return 1;
+ }
+ }
+
+ /* Write the stream header, if any. */
+ ret = avformat_write_header(oc, NULL);
+ if (ret < 0) {
+ fprintf(stderr, "Error occurred when opening output file: %s\n",
+ av_err2str(ret));
+ return 1;
+ }
+
+ if (frame)
+ frame->pts = 0;
+ for (;;) {
+ /* Compute current audio and video time. */
+ audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+ video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+
+ if ((!audio_st || audio_time >= STREAM_DURATION) &&
+ (!video_st || video_time >= STREAM_DURATION))
+ break;
+
+ /* write interleaved audio and video frames */
+ if (!video_st || (video_st && audio_st && audio_time < video_time)) {
+ write_audio_frame(oc, audio_st);
+ } else {
+ write_video_frame(oc, video_st);
+ frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
+ }
+ }
+
+ /* Write the trailer, if any. The trailer must be written before you
+ * close the CodecContexts open when you wrote the header; otherwise
+ * av_write_trailer() may try to use memory that was freed on
+ * av_codec_close(). */
+ av_write_trailer(oc);
+
+ /* Close each codec. */
+ if (video_st)
+ close_video(oc, video_st);
+ if (audio_st)
+ close_audio(oc, audio_st);
+
+ if (!(fmt->flags & AVFMT_NOFILE))
+ /* Close the output file. */
+ avio_close(oc->pb);
+
+ /* free the stream */
+ avformat_free_context(oc);
+
+ return 0;
+}
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