Commit 91a28b0e authored by Justin Ruggles's avatar Justin Ruggles
Browse files

avcodec: add ff_samples_to_time_base() convenience function to internal.h

parent 41ac9bb2
......@@ -26,6 +26,7 @@
#include <stdint.h>
#include "libavutil/mathematics.h"
#include "libavutil/pixfmt.h"
#include "avcodec.h"
......@@ -127,4 +128,14 @@ int avpriv_unlock_avformat(void);
*/
int ff_alloc_packet(AVPacket *avpkt, int size);
/**
* Rescale from sample rate to AVCodecContext.time_base.
*/
static av_always_inline int64_t ff_samples_to_time_base(AVCodecContext *avctx,
int64_t samples)
{
return av_rescale_q(samples, (AVRational){ 1, avctx->sample_rate },
avctx->time_base);
}
#endif /* AVCODEC_INTERNAL_H */
......@@ -67,7 +67,6 @@
#include <speex/speex.h>
#include <speex/speex_header.h>
#include <speex/speex_stereo.h>
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
......@@ -258,9 +257,7 @@ static int encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size,
/* write output if all frames for the packet have been encoded */
if (s->pkt_frame_count == s->frames_per_packet) {
s->pkt_frame_count = 0;
avctx->coded_frame->pts =
av_rescale_q(s->next_pts, (AVRational){ 1, avctx->sample_rate },
avctx->time_base);
avctx->coded_frame->pts = ff_samples_to_time_base(avctx, s->next_pts);
s->next_pts += s->pkt_sample_count;
s->pkt_sample_count = 0;
if (buf_size > speex_bits_nbytes(&s->bits)) {
......
......@@ -29,8 +29,8 @@
#include "libavutil/opt.h"
#include "avcodec.h"
#include "bytestream.h"
#include "internal.h"
#include "vorbis.h"
#include "libavutil/mathematics.h"
#undef NDEBUG
#include <assert.h>
......@@ -216,7 +216,8 @@ static int oggvorbis_encode_frame(AVCodecContext *avccontext,
op2->packet = context->buffer + sizeof(ogg_packet);
l = op2->bytes;
avccontext->coded_frame->pts = av_rescale_q(op2->granulepos, (AVRational) { 1, avccontext->sample_rate }, avccontext->time_base);
avccontext->coded_frame->pts = ff_samples_to_time_base(avccontext,
op2->granulepos);
//FIXME we should reorder the user supplied pts and not assume that they are spaced by 1/sample_rate
if (l > buf_size) {
......
......@@ -886,9 +886,8 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
if (!ret && *got_packet_ptr) {
if (!(avctx->codec->capabilities & CODEC_CAP_DELAY)) {
avpkt->pts = frame->pts;
avpkt->duration = av_rescale_q(frame->nb_samples,
(AVRational){ 1, avctx->sample_rate },
avctx->time_base);
avpkt->duration = ff_samples_to_time_base(avctx,
frame->nb_samples);
}
avpkt->dts = avpkt->pts;
} else {
......@@ -944,9 +943,8 @@ int attribute_align_arg avcodec_encode_audio2(AVCodecContext *avctx,
once all encoders supporting CODEC_CAP_SMALL_LAST_FRAME use
encode2() */
if (fs_tmp) {
avpkt->duration = av_rescale_q(avctx->frame_size,
(AVRational){ 1, avctx->sample_rate },
avctx->time_base);
avpkt->duration = ff_samples_to_time_base(avctx,
avctx->frame_size);
}
}
avpkt->size = ret;
......@@ -1018,9 +1016,8 @@ int attribute_align_arg avcodec_encode_audio(AVCodecContext *avctx,
/* fabricate frame pts from sample count.
this is needed because the avcodec_encode_audio() API does not have
a way for the user to provide pts */
frame->pts = av_rescale_q(avctx->internal->sample_count,
(AVRational){ 1, avctx->sample_rate },
avctx->time_base);
frame->pts = ff_samples_to_time_base(avctx,
avctx->internal->sample_count);
avctx->internal->sample_count += frame->nb_samples;
} else {
frame = NULL;
......
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