Those are private fields, no reason to have them exposed in a public
header.
---
libavformat/avformat.h | 29 -----------------------------
libavformat/internal.h | 29 +++++++++++++++++++++++++++++
libavformat/mov.c | 12 ++++++------
libavformat/mp3dec.c | 8 ++++----
libavformat/swfdec.c | 2 +-
libavformat/utils.c | 26 +++++++++++++-------------
6 files changed, 53 insertions(+), 53 deletions(-)
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 977680ec83..8028b1b558 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -1100,35 +1100,6 @@ typedef struct AVStream {
*/
int skip_to_keyframe;
- /**
- * Number of samples to skip at the start of the frame decoded from the
next packet.
- */
- int skip_samples;
-
- /**
- * If not 0, the number of samples that should be skipped from the start of
- * the stream (the samples are removed from packets with pts==0, which also
- * assumes negative timestamps do not happen).
- * Intended for use with formats such as mp3 with ad-hoc gapless audio
- * support.
- */
- int64_t start_skip_samples;
-
- /**
- * If not 0, the first audio sample that should be discarded from the
stream.
- * This is broken by design (needs global sample count), but can't be
- * avoided for broken by design formats such as mp3 with ad-hoc gapless
- * audio support.
- */
- int64_t first_discard_sample;
-
- /**
- * The sample after last sample that is intended to be discarded after
- * first_discard_sample. Works on frame boundaries only. Used to prevent
- * early EOF if the gapless info is broken (considered concatenated mp3s).
- */
- int64_t last_discard_sample;
-
/**
* An opaque field for libavformat internal usage.
* Must not be accessed in any way by callers.
diff --git a/libavformat/internal.h b/libavformat/internal.h
index b1112fe463..12105aa7d0 100644
--- a/libavformat/internal.h
+++ b/libavformat/internal.h
@@ -225,6 +225,35 @@ struct AVStreamInternal {
} *info;
+ /**
+ * Number of samples to skip at the start of the frame decoded from the
next packet.
+ */
+ int skip_samples;
+
+ /**
+ * If not 0, the number of samples that should be skipped from the start of
+ * the stream (the samples are removed from packets with pts==0, which also
+ * assumes negative timestamps do not happen).
+ * Intended for use with formats such as mp3 with ad-hoc gapless audio
+ * support.
+ */
+ int64_t start_skip_samples;
+
+ /**
+ * If not 0, the first audio sample that should be discarded from the
stream.
+ * This is broken by design (needs global sample count), but can't be
+ * avoided for broken by design formats such as mp3 with ad-hoc gapless
+ * audio support.
+ */
+ int64_t first_discard_sample;
+
+ /**
+ * The sample after last sample that is intended to be discarded after
+ * first_discard_sample. Works on frame boundaries only. Used to prevent
+ * early EOF if the gapless info is broken (considered concatenated mp3s).
+ */
+ int64_t last_discard_sample;
+
/**
* Number of internally decoded frames, used internally in libavformat, do
not access
* its lifetime differs from info which is why it is not in that structure.
diff --git a/libavformat/mov.c b/libavformat/mov.c
index 82fd1d74f6..6103678cdb 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -3550,7 +3550,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st)
}
if (first_non_zero_audio_edit > 0)
- st->skip_samples = msc->start_pad = 0;
+ st->internal->skip_samples = msc->start_pad = 0;
}
// While reordering frame index according to edit list we must handle
properly
@@ -3625,7 +3625,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st)
curr_cts < edit_list_media_time && curr_cts +
frame_duration > edit_list_media_time &&
first_non_zero_audio_edit > 0) {
packet_skip_samples = edit_list_media_time - curr_cts;
- st->skip_samples += packet_skip_samples;
+ st->internal->skip_samples += packet_skip_samples;
// Shift the index entry timestamp by packet_skip_samples
to be correct.
edit_list_dts_counter -= packet_skip_samples;
@@ -3658,7 +3658,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st)
// Increment skip_samples for the first non-zero audio
edit list
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
first_non_zero_audio_edit > 0 &&
st->codecpar->codec_id != AV_CODEC_ID_VORBIS) {
- st->skip_samples += frame_duration;
+ st->internal->skip_samples += frame_duration;
}
}
}
@@ -3744,7 +3744,7 @@ static void mov_fix_index(MOVContext *mov, AVStream *st)
// Update av stream length, if it ends up shorter than the track's media
duration
st->duration = FFMIN(st->duration, edit_list_dts_entry_end - start_dts);
- msc->start_pad = st->skip_samples;
+ msc->start_pad = st->internal->skip_samples;
// Free the old index and the old CTTS structures
av_free(e_old);
@@ -7615,7 +7615,7 @@ static int mov_read_header(AVFormatContext *s)
fix_timescale(mov, sc);
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
st->codecpar->codec_id == AV_CODEC_ID_AAC) {
- st->skip_samples = sc->start_pad;
+ st->internal->skip_samples = sc->start_pad;
}
if (st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO &&
sc->nb_frames_for_fps > 0 && sc->duration_for_fps > 0)
av_reduce(&st->avg_frame_rate.num, &st->avg_frame_rate.den,
@@ -8104,7 +8104,7 @@ static int mov_read_seek(AVFormatContext *s, int
stream_index, int64_t sample_ti
int64_t timestamp;
MOVStreamContext *sc = s->streams[i]->priv_data;
st = s->streams[i];
- st->skip_samples = (sample_time <= 0) ? sc->start_pad : 0;
+ st->internal->skip_samples = (sample_time <= 0) ? sc->start_pad :
0;
if (stream_index == i)
continue;
diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index b044679c02..5e7f273c6a 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -255,13 +255,13 @@ static void mp3_parse_info_tag(AVFormatContext *s,
AVStream *st,
mp3->start_pad = v>>12;
mp3-> end_pad = v&4095;
- st->start_skip_samples = mp3->start_pad + 528 + 1;
+ st->internal->start_skip_samples = mp3->start_pad + 528 + 1;
if (mp3->frames) {
- st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames *
(int64_t)spf;
- st->last_discard_sample = mp3->frames * (int64_t)spf;
+ st->internal->first_discard_sample = -mp3->end_pad + 528 + 1 +
mp3->frames * (int64_t)spf;
+ st->internal->last_discard_sample = mp3->frames * (int64_t)spf;
}
if (!st->start_time)
- st->start_time = av_rescale_q(st->start_skip_samples,
+ st->start_time = av_rescale_q(st->internal->start_skip_samples,
(AVRational){1, c->sample_rate},
st->time_base);
av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3-> end_pad);
diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c
index 2769a768de..fa11c050cd 100644
--- a/libavformat/swfdec.c
+++ b/libavformat/swfdec.c
@@ -292,7 +292,7 @@ static int swf_read_packet(AVFormatContext *s, AVPacket
*pkt)
return AVERROR(ENOMEM);
ast->duration = avio_rl32(pb); // number of samples
if (((v>>4) & 15) == 2) { // MP3 sound data record
- ast->skip_samples = avio_rl16(pb);
+ ast->internal->skip_samples = avio_rl16(pb);
len -= 2;
}
len -= 7;
diff --git a/libavformat/utils.c b/libavformat/utils.c
index ae7e91171e..9fd79ee540 100644
--- a/libavformat/utils.c
+++ b/libavformat/utils.c
@@ -1123,7 +1123,7 @@ static void update_initial_timestamps(AVFormatContext *s,
int stream_index,
if (st->start_time == AV_NOPTS_VALUE && pktl_it->pkt.pts !=
AV_NOPTS_VALUE) {
st->start_time = pktl_it->pkt.pts;
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
st->codecpar->sample_rate)
- st->start_time = av_sat_add64(st->start_time,
av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate},
st->time_base));
+ st->start_time = av_sat_add64(st->start_time,
av_rescale_q(st->internal->skip_samples, (AVRational){1,
st->codecpar->sample_rate}, st->time_base));
}
}
@@ -1136,7 +1136,7 @@ static void update_initial_timestamps(AVFormatContext *s,
int stream_index,
st->start_time = pts;
}
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO &&
st->codecpar->sample_rate)
- st->start_time = av_sat_add64(st->start_time,
av_rescale_q(st->skip_samples, (AVRational){1, st->codecpar->sample_rate},
st->time_base));
+ st->start_time = av_sat_add64(st->start_time,
av_rescale_q(st->internal->skip_samples, (AVRational){1,
st->codecpar->sample_rate}, st->time_base));
}
}
@@ -1638,25 +1638,25 @@ FF_ENABLE_DEPRECATION_WARNINGS
if (ret >= 0) {
AVStream *st = s->streams[pkt->stream_index];
int discard_padding = 0;
- if (st->first_discard_sample && pkt->pts != AV_NOPTS_VALUE) {
+ if (st->internal->first_discard_sample && pkt->pts != AV_NOPTS_VALUE) {
int64_t pts = pkt->pts - (is_relative(pkt->pts) ? RELATIVE_TS_BASE
: 0);
int64_t sample = ts_to_samples(st, pts);
int duration = ts_to_samples(st, pkt->duration);
int64_t end_sample = sample + duration;
- if (duration > 0 && end_sample >= st->first_discard_sample &&
- sample < st->last_discard_sample)
- discard_padding = FFMIN(end_sample - st->first_discard_sample,
duration);
+ if (duration > 0 && end_sample >=
st->internal->first_discard_sample &&
+ sample < st->internal->last_discard_sample)
+ discard_padding = FFMIN(end_sample -
st->internal->first_discard_sample, duration);
}
- if (st->start_skip_samples && (pkt->pts == 0 || pkt->pts ==
RELATIVE_TS_BASE))
- st->skip_samples = st->start_skip_samples;
- if (st->skip_samples || discard_padding) {
+ if (st->internal->start_skip_samples && (pkt->pts == 0 || pkt->pts ==
RELATIVE_TS_BASE))
+ st->internal->skip_samples = st->internal->start_skip_samples;
+ if (st->internal->skip_samples || discard_padding) {
uint8_t *p = av_packet_new_side_data(pkt,
AV_PKT_DATA_SKIP_SAMPLES, 10);
if (p) {
- AV_WL32(p, st->skip_samples);
+ AV_WL32(p, st->internal->skip_samples);
AV_WL32(p + 4, discard_padding);
- av_log(s, AV_LOG_DEBUG, "demuxer injecting skip %d / discard
%d\n", st->skip_samples, discard_padding);
+ av_log(s, AV_LOG_DEBUG, "demuxer injecting skip %d / discard
%d\n", st->internal->skip_samples, discard_padding);
}
- st->skip_samples = 0;
+ st->internal->skip_samples = 0;
}
if (st->internal->inject_global_side_data) {
@@ -1890,7 +1890,7 @@ void ff_read_frame_flush(AVFormatContext *s)
if (s->internal->inject_global_side_data)
st->internal->inject_global_side_data = 1;
- st->skip_samples = 0;
+ st->internal->skip_samples = 0;
}
}
--
2.28.0
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