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lavf/output-example: use new audio encoding API correctly.

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This commit is contained in:
Anton Khirnov
2012-02-24 08:59:38 +01:00
parent 6e9ed7c7ae
commit 5ff42e3138
1 changed files with 15 additions and 29 deletions
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44
libavformat/output-example.c
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@@ -53,8 +53,6 @@ static int sws_flags = SWS_BICUBIC;
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;
/*
@@ -112,27 +110,12 @@ static void open_audio(AVFormatContext *oc, AVStream *st)
/* 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 = 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 {
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
audio_input_frame_size = 10000;
else
audio_input_frame_size = c->frame_size;
}
samples = av_malloc(audio_input_frame_size * 2 * c->channels);
samples = av_malloc(audio_input_frame_size * av_get_bytes_per_sample(c->sample_fmt)
* c->channels);
}
/* prepare a 16 bit dummy audio frame of 'frame_size' samples and
@@ -156,19 +139,23 @@ static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt;
av_init_packet(&pkt);
AVFrame *frame = avcodec_alloc_frame();
int got_packet;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame(samples, audio_input_frame_size, c->channels);
frame->nb_samples = audio_input_frame_size;
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt, (uint8_t *)samples,
audio_input_frame_size * av_get_bytes_per_sample(c->sample_fmt)
* c->channels, 1);
pkt.size = avcodec_encode_audio2(c, audio_outbuf, audio_outbuf_size, samples);
avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (!got_packet)
return;
if (c->coded_frame && c->coded_frame->pts != AV_NOPTS_VALUE)
pkt.pts= av_rescale_q(c->coded_frame->pts, c->time_base, st->time_base);
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index= st->index;
pkt.data= audio_outbuf;
/* write the compressed frame in the media file */
if (av_interleaved_write_frame(oc, &pkt) != 0) {
@@ -182,7 +169,6 @@ static void close_audio(AVFormatContext *oc, AVStream *st)
avcodec_close(st->codec);
av_free(samples);
av_free(audio_outbuf);
}
/**************************************************************/