zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Delete redundant logic for setting is_first_packet_in_frame

Browse Source 
A value for this flag was derived in RtpReceiverImpl::IncomingRtpPacket.
For audio, it was never used, and for video, it was overridden by
the result from RtpDepacketizer::ParsedPayload.

Bug: webrtc:7135
Change-Id: I597a57ca77d13b9a9145a9ee5e6624c1986777b9
Reviewed-on: https://webrtc-review.googlesource.com/3660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19985}
This commit is contained in:
Niels Möller
2017-09-26 14:05:05 +02:00
committed by Commit Bot
parent 4d5030f8e6
commit bbf389c7af
7 changed files with 8 additions and 40 deletions
Download Patch File Download Diff File Expand all files Collapse all files

26
modules/rtp_rtcp/source/rtp_receiver_impl.cc
View File

@ -65,14 +65,11 @@ RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
rtp_payload_registry_(rtp_payload_registry),
rtp_media_receiver_(rtp_media_receiver),
cb_rtp_feedback_(incoming_messages_callback),
last_receive_time_(0),
last_received_payload_length_(0),
ssrc_(0),
num_csrcs_(0),
current_remote_csrc_(),
last_received_timestamp_(0),
last_received_frame_time_ms_(-1),
last_received_sequence_number_(0) {
last_received_frame_time_ms_(-1) {
assert(incoming_messages_callback);
memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
@ -169,23 +166,9 @@ bool RtpReceiverImpl::IncomingRtpPacket(
: rtc::Optional<uint8_t>();
UpdateSources(audio_level);
size_t payload_data_length = payload_length - rtp_header.paddingLength;
bool is_first_packet_in_frame = false;
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
if (HaveReceivedFrame()) {
is_first_packet_in_frame =
last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
last_received_timestamp_ != rtp_header.timestamp;
} else {
is_first_packet_in_frame = true;
}
}
int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
&webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
clock_->TimeInMilliseconds(), is_first_packet_in_frame);
clock_->TimeInMilliseconds());
if (ret_val < 0) {
return false;
@ -194,15 +177,11 @@ bool RtpReceiverImpl::IncomingRtpPacket(
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
last_receive_time_ = clock_->TimeInMilliseconds();
last_received_payload_length_ = payload_data_length;
if (in_order) {
if (last_received_timestamp_ != rtp_header.timestamp) {
last_received_timestamp_ = rtp_header.timestamp;
last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
}
last_received_sequence_number_ = rtp_header.sequenceNumber;
}
}
return true;
@ -285,7 +264,6 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
new_ssrc = true;
last_received_timestamp_ = 0;
last_received_sequence_number_ = 0;
last_received_frame_time_ms_ = -1;
// Do we have a SSRC? Then the stream is restarted.