[FFmpeg-cvslog] rtpenc: Don' t set max_frames_per_packet based on the packet frame size or frame rate
Martin Storsjö
git at videolan.org
Sun Mar 1 01:42:03 CET 2015
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Thu Feb 26 00:00:39 2015 +0200| [4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f] | committer: Martin Storsjö
rtpenc: Don't set max_frames_per_packet based on the packet frame size or frame rate
Instead check the timestamps while muxing, to avoid buffering a
too long timestamp range into one single packet.
This makes the AMR and AAC packetization slightly less efficient,
since we set a possibly unnecessarily high max_frames_per_packet.
(These packetizers end up doing a memmove of the TOC bytes if
sending a packet before max_frames_per_packet is achieved, and
we end up setting max_frames_per_packet to a value that should
be high enough for most uses.)
All packetizers that use max_frames_per_packet now set it either
to a default value, or to a value calculated based on other
parameters, so none of them rely on the previous default setting.
For iLBC, copy one frame at a time, to allow checking the timestamp
range for each of them - basically doing potentially multiple
loops to simplify the code instead of trying to calculate the
number of frames to buffer while honoring s1->max_delay.
This is in preparation for reducing the coupling between libavformat
and libavcodec, by not having the muxers use the encoder field
frame_size (which may not be available during e.g. stream copy).
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4f6cd883f06f7893a2b60a41e7a4f8ae633dac2f
---
libavformat/rtpenc.c | 61 ++++++++++++---------------------------------
libavformat/rtpenc_aac.c | 5 +++-
libavformat/rtpenc_amr.c | 5 +++-
libavformat/rtpenc_xiph.c | 7 ++++--
4 files changed, 29 insertions(+), 49 deletions(-)
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 5f1fc84..d94f254 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -149,33 +149,6 @@ static int rtp_write_header(AVFormatContext *s1)
}
s->max_payload_size = s1->packet_size - 12;
- s->max_frames_per_packet = 0;
- if (s1->max_delay > 0) {
- if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- int frame_size = av_get_audio_frame_duration(st->codec, 0);
- if (!frame_size)
- frame_size = st->codec->frame_size;
- if (frame_size == 0) {
- av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
- } else {
- s->max_frames_per_packet =
- av_rescale_q_rnd(s1->max_delay,
- AV_TIME_BASE_Q,
- (AVRational){ frame_size, st->codec->sample_rate },
- AV_ROUND_DOWN);
- }
- }
- if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
- /* FIXME: We should round down here... */
- if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
- s->max_frames_per_packet = av_rescale_q(s1->max_delay,
- (AVRational){1, 1000000},
- av_inv_q(st->avg_frame_rate));
- } else
- s->max_frames_per_packet = 1;
- }
- }
-
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
} else {
@@ -225,9 +198,7 @@ static int rtp_write_header(AVFormatContext *s1)
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
- if (!s->max_frames_per_packet)
- s->max_frames_per_packet = 15;
- s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
+ s->max_frames_per_packet = 15;
break;
case AV_CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
@@ -249,15 +220,11 @@ static int rtp_write_header(AVFormatContext *s1)
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
- if (!s->max_frames_per_packet)
- s->max_frames_per_packet = 1;
- s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
- s->max_payload_size / st->codec->block_align);
+ s->max_frames_per_packet = s->max_payload_size / st->codec->block_align;
break;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
- if (!s->max_frames_per_packet)
- s->max_frames_per_packet = 12;
+ s->max_frames_per_packet = 50;
if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
@@ -273,8 +240,7 @@ static int rtp_write_header(AVFormatContext *s1)
}
break;
case AV_CODEC_ID_AAC:
- if (!s->max_frames_per_packet)
- s->max_frames_per_packet = 5;
+ s->max_frames_per_packet = 50;
break;
default:
break;
@@ -493,18 +459,23 @@ static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
int frames = size / frame_size;
while (frames > 0) {
- int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
+ if (s->num_frames > 0 &&
+ av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
+ s1->max_delay, AV_TIME_BASE_Q) >= 0) {
+ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
+ s->num_frames = 0;
+ }
if (!s->num_frames) {
s->buf_ptr = s->buf;
s->timestamp = s->cur_timestamp;
}
- memcpy(s->buf_ptr, buf, n * frame_size);
- frames -= n;
- s->num_frames += n;
- s->buf_ptr += n * frame_size;
- buf += n * frame_size;
- s->cur_timestamp += n * frame_duration;
+ memcpy(s->buf_ptr, buf, frame_size);
+ frames--;
+ s->num_frames++;
+ s->buf_ptr += frame_size;
+ buf += frame_size;
+ s->cur_timestamp += frame_duration;
if (s->num_frames == s->max_frames_per_packet) {
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
diff --git a/libavformat/rtpenc_aac.c b/libavformat/rtpenc_aac.c
index 7805ab9..d0b4ca0 100644
--- a/libavformat/rtpenc_aac.c
+++ b/libavformat/rtpenc_aac.c
@@ -27,6 +27,7 @@
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
{
RTPMuxContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
const int max_au_headers_size = 2 + 2 * s->max_frames_per_packet;
int len, max_packet_size = s->max_payload_size - max_au_headers_size;
uint8_t *p;
@@ -41,7 +42,9 @@ void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
len = (s->buf_ptr - s->buf);
if (s->num_frames &&
(s->num_frames == s->max_frames_per_packet ||
- (len + size) > s->max_payload_size)) {
+ (len + size) > s->max_payload_size ||
+ av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
+ s1->max_delay, AV_TIME_BASE_Q) >= 0)) {
int au_size = s->num_frames * 2;
p = s->buf + max_au_headers_size - au_size - 2;
diff --git a/libavformat/rtpenc_amr.c b/libavformat/rtpenc_amr.c
index 6da5f0f..0adbf49 100644
--- a/libavformat/rtpenc_amr.c
+++ b/libavformat/rtpenc_amr.c
@@ -30,6 +30,7 @@
void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size)
{
RTPMuxContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
int max_header_toc_size = 1 + s->max_frames_per_packet;
uint8_t *p;
int len;
@@ -38,7 +39,9 @@ void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size)
len = s->buf_ptr - s->buf;
if (s->num_frames &&
(s->num_frames == s->max_frames_per_packet ||
- len + size - 1 > s->max_payload_size)) {
+ len + size - 1 > s->max_payload_size ||
+ av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
+ s1->max_delay, AV_TIME_BASE_Q) >= 0)) {
int header_size = s->num_frames + 1;
p = s->buf + max_header_toc_size - header_size;
if (p != s->buf)
diff --git a/libavformat/rtpenc_xiph.c b/libavformat/rtpenc_xiph.c
index 2167bdc..5b171c3 100644
--- a/libavformat/rtpenc_xiph.c
+++ b/libavformat/rtpenc_xiph.c
@@ -32,6 +32,7 @@
void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
{
RTPMuxContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
int max_pkt_size, xdt, frag;
uint8_t *q;
@@ -77,8 +78,10 @@ void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size)
assert(s->num_frames <= s->max_frames_per_packet);
if (s->num_frames > 0 &&
(remaining < 0 ||
- s->num_frames == s->max_frames_per_packet)) {
- // send previous packets now; no room for new data
+ s->num_frames == s->max_frames_per_packet ||
+ av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base,
+ s1->max_delay, AV_TIME_BASE_Q) >= 0)) {
+ // send previous packets now; no room for new data, or too much delay
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->num_frames = 0;
}
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