FFMpeg写MP4文件例子分析

    这段时间看了FFMpeg提供的例子muxing.c，我略微修改了下源代码，使其生成一个MP4文件，音频使用AAC编码，视频使用H.264编码。代码很简单，我就不做说明了，代码如下。

    以后我们继续写如何将DirectShow中采集的音视频数据编码并生成MP4文件。

/* 5 seconds stream duration */
#define STREAM_DURATION   5.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT PIX_FMT_YUV420P /* default pix_fmt */

static int sws_flags = SWS_BICUBIC;

/**************************************************************/
/* audio output */

static float t, tincr, tincr2;
static int16_t *samples;
static uint8_t *audio_outbuf;
static int audio_outbuf_size;
static int audio_input_frame_size;

/*
* add an audio output stream
*/
static AVStream *add_audio_stream(AVFormatContext *oc, enum CodecID codec_id)
{
	AVCodecContext *c;
	AVStream *st;

	st = avformat_new_stream(oc, NULL);
	if (!st) {
		fprintf(stderr, "Could not alloc stream\n");
		exit(1);
	}
	st->id = 1;

	c = st->codec;
	c->codec_id = codec_id;
	c->codec_type = AVMEDIA_TYPE_AUDIO;

	/* put sample parameters */
	c->sample_fmt = AV_SAMPLE_FMT_S16;
	c->bit_rate = 64000;
	c->sample_rate = 44100;
	c->channels = 2;

	// 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)
{
	AVCodecContext *c;
	AVCodec *codec;

	c = st->codec;

	/* find the audio encoder */
	codec = avcodec_find_encoder(c->codec_id);
	if (!codec) {
		fprintf(stderr, "codec not found\n");
		exit(1);
	}

	/* open it */
	if (avcodec_open(c, codec) < 0) {
		fprintf(stderr, "could not open codec\n");
		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;

	audio_outbuf_size = 10000;
	audio_outbuf = (uint8_t *)av_malloc(audio_outbuf_size);

	/* ugly hack for PCM codecs (will be removed ASAP with new PCM
	support to compute the input frame size in samples */
	if (c->frame_size <= 1) {
		audio_input_frame_size = audio_outbuf_size / c->channels;
		switch(st->codec->codec_id) {
		case CODEC_ID_PCM_S16LE:
		case CODEC_ID_PCM_S16BE:
		case CODEC_ID_PCM_U16LE:
		case CODEC_ID_PCM_U16BE:
			audio_input_frame_size >>= 1;
			break;
		default:
			break;
		}
	} else {
		audio_input_frame_size = c->frame_size;
	}
	samples = (int16_t *)av_malloc(audio_input_frame_size * 2 * c->channels);
}

/* 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++) {