[FFmpeg-cvslog] avconv: rename InputStream.pts to last_dts.
Anton Khirnov
git at videolan.org
Wed Feb 8 06:04:55 CET 2012
ffmpeg | branch: master | Anton Khirnov <anton at khirnov.net> | Sun Feb 5 21:12:43 2012 +0100| [23576b3fbba3f5438fbd6aa6e5e73a39682b942b] | committer: Anton Khirnov
avconv: rename InputStream.pts to last_dts.
It more accurately describes what does this variable store.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=23576b3fbba3f5438fbd6aa6e5e73a39682b942b
---
avconv.c | 31 ++++++++++++++++---------------
1 files changed, 16 insertions(+), 15 deletions(-)
diff --git a/avconv.c b/avconv.c
index ab4fb7c..a4f67ff 100644
--- a/avconv.c
+++ b/avconv.c
@@ -171,7 +171,8 @@ typedef struct InputStream {
/* predicted dts of the next packet read for this stream or (when there are
* several frames in a packet) of the next frame in current packet */
int64_t next_dts;
- int64_t pts; /* current pts */
+ /* dts of the last packet read for this stream */
+ int64_t last_dts;
PtsCorrectionContext pts_ctx;
double ts_scale;
int is_start; /* is 1 at the start and after a discontinuity */
@@ -1093,7 +1094,7 @@ need_realloc:
}
if (audio_sync_method) {
- double delta = get_sync_ipts(ost, ist->pts) * enc->sample_rate - ost->sync_opts -
+ double delta = get_sync_ipts(ost, ist->last_dts) * enc->sample_rate - ost->sync_opts -
av_fifo_size(ost->fifo) / (enc->channels * osize);
int idelta = delta * dec->sample_rate / enc->sample_rate;
int byte_delta = idelta * isize * dec->channels;
@@ -1136,7 +1137,7 @@ need_realloc:
}
}
} else
- ost->sync_opts = lrintf(get_sync_ipts(ost, ist->pts) * enc->sample_rate) -
+ ost->sync_opts = lrintf(get_sync_ipts(ost, ist->last_dts) * enc->sample_rate) -
av_fifo_size(ost->fifo) / (enc->channels * osize); // FIXME wrong
if (ost->audio_resample) {
@@ -1748,7 +1749,7 @@ static int check_output_constraints(InputStream *ist, OutputStream *ost)
if (ost->source_index != ist_index)
return 0;
- if (of->start_time && ist->pts < of->start_time)
+ if (of->start_time && ist->last_dts < of->start_time)
return 0;
return 1;
@@ -1767,7 +1768,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
return;
if (of->recording_time != INT64_MAX &&
- ist->pts >= of->recording_time + of->start_time) {
+ ist->last_dts >= of->recording_time + of->start_time) {
ost->is_past_recording_time = 1;
return;
}
@@ -1786,7 +1787,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
opkt.pts = AV_NOPTS_VALUE;
if (pkt->dts == AV_NOPTS_VALUE)
- opkt.dts = av_rescale_q(ist->pts, AV_TIME_BASE_Q, ost->st->time_base);
+ opkt.dts = av_rescale_q(ist->last_dts, AV_TIME_BASE_Q, ost->st->time_base);
else
opkt.dts = av_rescale_q(pkt->dts, ist->st->time_base, ost->st->time_base);
opkt.dts -= ost_tb_start_time;
@@ -1814,7 +1815,7 @@ static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *p
static void rate_emu_sleep(InputStream *ist)
{
if (input_files[ist->file_index].rate_emu) {
- int64_t pts = av_rescale(ist->pts, 1000000, AV_TIME_BASE);
+ int64_t pts = av_rescale(ist->last_dts, 1000000, AV_TIME_BASE);
int64_t now = av_gettime() - ist->start;
if (pts > now)
usleep(pts - now);
@@ -1941,7 +1942,7 @@ static int transcode_video(InputStream *ist, AVPacket *pkt, int *got_output, int
avcodec_get_frame_defaults(ist->decoded_frame);
decoded_frame = ist->decoded_frame;
pkt->pts = *pkt_pts;
- pkt->dts = ist->pts;
+ pkt->dts = ist->last_dts;
*pkt_pts = AV_NOPTS_VALUE;
ret = avcodec_decode_video2(ist->st->codec,
@@ -2091,7 +2092,7 @@ static int output_packet(InputStream *ist,
AVPacket avpkt;
if (ist->next_dts == AV_NOPTS_VALUE)
- ist->next_dts = ist->pts;
+ ist->next_dts = ist->last_dts;
if (pkt == NULL) {
/* EOF handling */
@@ -2104,7 +2105,7 @@ static int output_packet(InputStream *ist,
}
if (pkt->dts != AV_NOPTS_VALUE)
- ist->next_dts = ist->pts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q);
+ ist->next_dts = ist->last_dts = av_rescale_q(pkt->dts, ist->st->time_base, AV_TIME_BASE_Q);
if (pkt->pts != AV_NOPTS_VALUE)
pkt_pts = av_rescale_q(pkt->pts, ist->st->time_base, AV_TIME_BASE_Q);
@@ -2113,7 +2114,7 @@ static int output_packet(InputStream *ist,
int ret = 0;
handle_eof:
- ist->pts = ist->next_dts;
+ ist->last_dts = ist->next_dts;
if (avpkt.size && avpkt.size != pkt->size) {
av_log(NULL, ist->showed_multi_packet_warning ? AV_LOG_VERBOSE : AV_LOG_WARNING,
@@ -2150,7 +2151,7 @@ static int output_packet(InputStream *ist,
/* handle stream copy */
if (!ist->decoding_needed) {
rate_emu_sleep(ist);
- ist->pts = ist->next_dts;
+ ist->last_dts = ist->next_dts;
switch (ist->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
ist->next_dts += ((int64_t)AV_TIME_BASE * ist->st->codec->frame_size) /
@@ -2235,7 +2236,7 @@ static int init_input_stream(int ist_index, OutputStream *output_streams, int nb
assert_avoptions(ist->opts);
}
- ist->pts = ist->st->avg_frame_rate.num ? - ist->st->codec->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0;
+ ist->last_dts = ist->st->avg_frame_rate.num ? - ist->st->codec->has_b_frames * AV_TIME_BASE / av_q2d(ist->st->avg_frame_rate) : 0;
ist->next_dts = AV_NOPTS_VALUE;
init_pts_correction(&ist->pts_ctx);
ist->is_start = 1;
@@ -2685,7 +2686,7 @@ static int transcode(OutputFile *output_files,
(os->pb && avio_tell(os->pb) >= of->limit_filesize))
continue;
opts = ost->st->pts.val * av_q2d(ost->st->time_base);
- ipts = ist->pts;
+ ipts = ist->last_dts;
if (!input_files[ist->file_index].eof_reached) {
if (ipts < ipts_min) {
ipts_min = ipts;
@@ -2765,7 +2766,7 @@ static int transcode(OutputFile *output_files,
&& (is->iformat->flags & AVFMT_TS_DISCONT)) {
int64_t pkt_dts = av_rescale_q(pkt.dts, ist->st->time_base, AV_TIME_BASE_Q);
int64_t delta = pkt_dts - ist->next_dts;
- if ((FFABS(delta) > 1LL * dts_delta_threshold * AV_TIME_BASE || pkt_dts + 1 < ist->pts) && !copy_ts) {
+ if ((FFABS(delta) > 1LL * dts_delta_threshold * AV_TIME_BASE || pkt_dts + 1 < ist->last_dts) && !copy_ts) {
input_files[ist->file_index].ts_offset -= delta;
av_log(NULL, AV_LOG_DEBUG,
"timestamp discontinuity %"PRId64", new offset= %"PRId64"\n",
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