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Calculate max payload size for an rtp packet to fit full video frame

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instead of sometimes incorrectly guessing it

Bug: webrtc:9868
Change-Id: I8b15ecca4c660d83ea129dc9df6ec174ad83b4c6
Reviewed-on: https://webrtc-review.googlesource.com/c/106281
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25213}
This commit is contained in:
Danil Chapovalov
2018-10-16 16:22:33 +02:00
committed by Commit Bot
parent f5e767dbbc
commit 6026f05ef1
1 changed files with 17 additions and 26 deletions
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43
modules/rtp_rtcp/source/rtp_sender_video.cc
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@ -387,20 +387,19 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
int packet_capacity = rtp_sender_->MaxRtpPacketSize() - fec_packet_overhead -
(rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
auto create_packet = [&] {
std::unique_ptr<RtpPacketToSend> rtp_packet = rtp_sender_->AllocatePacket();
RTC_DCHECK_LE(packet_capacity, rtp_packet->capacity());
std::unique_ptr<RtpPacketToSend> single_packet =
rtp_sender_->AllocatePacket();
RTC_DCHECK_LE(packet_capacity, single_packet->capacity());
single_packet->SetPayloadType(payload_type);
single_packet->SetTimestamp(rtp_timestamp);
single_packet->set_capture_time_ms(capture_time_ms);
rtp_packet->SetPayloadType(payload_type);
rtp_packet->SetTimestamp(rtp_timestamp);
rtp_packet->set_capture_time_ms(capture_time_ms);
return rtp_packet;
};
auto first_packet = create_packet();
auto middle_packet = absl::make_unique<RtpPacketToSend>(*first_packet);
auto last_packet = absl::make_unique<RtpPacketToSend>(*first_packet);
auto first_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
auto middle_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
auto last_packet = absl::make_unique<RtpPacketToSend>(*single_packet);
// Simplest way to estimate how much extensions would occupy is to set them.
AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
/*first=*/true, /*last=*/true, single_packet.get());
AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
/*first=*/true, /*last=*/false, first_packet.get());
AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
@ -408,12 +407,17 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
/*first=*/false, /*last=*/true, last_packet.get());
RTC_DCHECK_GT(packet_capacity, single_packet->headers_size());
RTC_DCHECK_GT(packet_capacity, first_packet->headers_size());
RTC_DCHECK_GT(packet_capacity, middle_packet->headers_size());
RTC_DCHECK_GT(packet_capacity, last_packet->headers_size());
RtpPacketizer::PayloadSizeLimits limits;
limits.max_payload_len = packet_capacity - middle_packet->headers_size();
RTC_DCHECK_GE(single_packet->headers_size(), middle_packet->headers_size());
limits.single_packet_reduction_len =
single_packet->headers_size() - middle_packet->headers_size();
RTC_DCHECK_GE(first_packet->headers_size(), middle_packet->headers_size());
limits.first_packet_reduction_len =
first_packet->headers_size() - middle_packet->headers_size();
@ -422,16 +426,6 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
limits.last_packet_reduction_len =
last_packet->headers_size() - middle_packet->headers_size();
// TODO(bugs.webrtc.org/9868, bugs.webrtc.org/7990): Calculate correctly:
// Single packet might need more space for extensions than sum of first and
// last packet reductions when two byte header extension is used. It also may
// need more even for one-byte header extension because of padding.
// e.g. if middle_packet uses 5 bytes for extensions, it is padded to 8.
// if first_packet and last_packet use 2 bytes for extensions each, reduction
// would be 0 for both of them, but single packet would need 4 extra bytes.
limits.single_packet_reduction_len =
limits.first_packet_reduction_len + limits.last_packet_reduction_len;
RTPVideoHeader minimized_video_header;
const RTPVideoHeader* packetize_video_header = video_header;
if (first_packet->HasExtension<RtpGenericFrameDescriptorExtension>() &&
@ -457,10 +451,7 @@ bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
int expected_payload_capacity;
// Choose right packet template:
if (num_packets == 1) {
// No prepared template, create a new packet.
packet = create_packet();
AddRtpHeaderExtensions(*video_header, frame_type, set_video_rotation,
/*first=*/true, /*last=*/true, packet.get());
packet = std::move(single_packet);
expected_payload_capacity =
limits.max_payload_len - limits.single_packet_reduction_len;
} else if (i == 0) {