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authorMichael Niedermayer <michaelni@gmx.at>2012-07-05 21:23:37 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-07-05 21:55:31 +0200
commit24823a761cb28eff5876d5de266a7d7103075e94 (patch)
treea476ad08e498a91e05f8a2b789bf6b4f495a145d /ffmpeg.c
parent92c7ef1e30d677f29c06caf765c527c9db486861 (diff)
parentbb58c43c69078c6cf29a9efee12e14469e2c21f8 (diff)
Merge remote-tracking branch 'qatar/master'
* qatar/master: qdm2: remove broken and disabled dump_context() debug function x86: h264_intrapred: use newly introduced SPLAT* and PSHUFLW macros x86inc: add SPLATB_LOAD, SPLATB_REG, PSHUFLW macros x86inc: modify ALIGN to not generate long nops on i586 x86: h264_intrapred: port to cpuflag macros avplay: update input filter pointer when the filtergraph is reset. avconv: fix parsing of -force_key_frames option. h264: use templates to avoid excessive inlining xtea: Make the count parameter match the documentation blowfish: Make the count parameter match the documentation mpegvideo: Don't use ff_mspel_motion() for vc1 xtea: invert branch and loop precedence blowfish: invert branch and loop precedence flvdec: optionally trust the metadata avconv: Set audio filter time base to the sample rate vp8: Add ifdef guards around the sse2 loopfilter in the sse2slow branch too Conflicts: ffmpeg.c ffplay.c libavcodec/h264.c libavcodec/mpegvideo_common.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'ffmpeg.c')
-rw-r--r--ffmpeg.c13
1 files changed, 11 insertions, 2 deletions
diff --git a/ffmpeg.c b/ffmpeg.c
index 1119fdf013..2fa8acb8df 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -1077,7 +1077,7 @@ static int configure_input_audio_filter(FilterGraph *fg, InputFilter *ifilter,
snprintf(args, sizeof(args), "time_base=%d/%d:sample_rate=%d:sample_fmt=%s"
":channel_layout=0x%"PRIx64,
- ist->st->time_base.num, ist->st->time_base.den,
+ 1, ist->st->codec->sample_rate,
ist->st->codec->sample_rate,
av_get_sample_fmt_name(ist->st->codec->sample_fmt),
ist->st->codec->channel_layout);
@@ -2451,6 +2451,10 @@ static int decode_audio(InputStream *ist, AVPacket *pkt, int *got_output)
}
}
+ if (decoded_frame->pts != AV_NOPTS_VALUE)
+ decoded_frame->pts = av_rescale_q(decoded_frame->pts,
+ ist->st->time_base,
+ (AVRational){1, ist->st->codec->sample_rate});
for (i = 0; i < ist->nb_filters; i++)
av_buffersrc_add_frame(ist->filters[i]->filter, decoded_frame, 0);
@@ -2804,12 +2808,17 @@ static void parse_forced_key_frames(char *kf, OutputStream *ost,
av_log(NULL, AV_LOG_FATAL, "Could not allocate forced key frames array.\n");
exit_program(1);
}
+
p = kf;
for (i = 0; i < n; i++) {
char *next = strchr(p, ',');
- if (next) *next++ = 0;
+
+ if (next)
+ *next++ = 0;
+
t = parse_time_or_die("force_key_frames", p, 1);
ost->forced_kf_pts[i] = av_rescale_q(t, AV_TIME_BASE_Q, avctx->time_base);
+
p = next;
}
}