Avoid NetEq triggering a Framelength change when receiving an FEC packet.

Internally in NetEq, an FEC packet looks very similar to a split packet, which caused NetEq to miscalculate the frame length of FEC packets. This incorrect framelength calculation was incorrectly handled as a framelength change by NetEq.

Bug: webrtc:8410
Change-Id: Icaea961d055e49d7726b87811881db0b9149805b
Reviewed-on: https://webrtc-review.googlesource.com/12420
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20373}
This commit is contained in:
Ivo Creusen
2017-10-20 12:35:04 +02:00
committed by Commit Bot
parent 9434240f7e
commit fd7c0a566a
3 changed files with 17 additions and 22 deletions

View File

@ -730,9 +730,12 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
}
}
// Calculate the number of primary (non-FEC/RED) packets.
const int number_of_primary_packets = std::count_if(
parsed_packet_list.begin(), parsed_packet_list.end(),
[](const Packet& in) { return in.priority.codec_level == 0; });
// Insert packets in buffer.
const size_t buffer_length_before_insert =
packet_buffer_->NumPacketsInBuffer();
const int ret = packet_buffer_->InsertPacketList(
&parsed_packet_list, *decoder_database_, &current_rtp_payload_type_,
&current_cng_rtp_payload_type_, &stats_);
@ -795,13 +798,9 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
dec_info->IsDtmf());
if (delay_manager_->last_pack_cng_or_dtmf() == 0) {
// Calculate the total speech length carried in each packet.
const size_t buffer_length_after_insert =
packet_buffer_->NumPacketsInBuffer();
if (buffer_length_after_insert > buffer_length_before_insert) {
if (number_of_primary_packets > 0) {
const size_t packet_length_samples =
(buffer_length_after_insert - buffer_length_before_insert) *
decoder_frame_length_;
number_of_primary_packets * decoder_frame_length_;
if (packet_length_samples != decision_logic_->packet_length_samples()) {
decision_logic_->set_packet_length_samples(packet_length_samples);
delay_manager_->SetPacketAudioLength(

View File

@ -342,10 +342,6 @@ TEST_F(NetEqImplTest, InsertPacket) {
.WillRepeatedly(Return(&info));
// Expectations for packet buffer.
EXPECT_CALL(*mock_packet_buffer_, NumPacketsInBuffer())
.WillOnce(Return(0)) // First packet.
.WillOnce(Return(1)) // Second packet.
.WillOnce(Return(2)); // Second packet, checking after it was inserted.
EXPECT_CALL(*mock_packet_buffer_, Empty())
.WillOnce(Return(false)); // Called once after first packet is inserted.
EXPECT_CALL(*mock_packet_buffer_, Flush())

View File

@ -500,18 +500,18 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string output_checksum = PlatformChecksum(
"721e1e0c6effe4b2401536a4eef11512c9fb709c",
"2e3c3e451532967e981fbc39b8cfb55e1df1ff7f",
"f403940a1936bff040d1d158624f69bdccbc3423",
"721e1e0c6effe4b2401536a4eef11512c9fb709c",
"721e1e0c6effe4b2401536a4eef11512c9fb709c");
"7ea28d7edf9395f4ac8e8d8dd3a9e5c620b1bf48",
"5b1e691ab1c4465c742d6d944bc71e3b1c0e4c0e",
"b096114dd8c233eaf2b0ce9802ac95af13933772",
"7ea28d7edf9395f4ac8e8d8dd3a9e5c620b1bf48",
"7ea28d7edf9395f4ac8e8d8dd3a9e5c620b1bf48");
const std::string network_stats_checksum =
PlatformChecksum("4e749c46e2611877120ac7a20cbbe555cfbd70ea",
"1edee6d07e0005327c32a77f9b3c0c1f03780e9f",
"ff806c574f82a089dec4c37ea1224b1eb0822d23",
"4e749c46e2611877120ac7a20cbbe555cfbd70ea",
"4e749c46e2611877120ac7a20cbbe555cfbd70ea");
PlatformChecksum("9e72233c78baf685e500dd6c94212b30a4c5f27d",
"9a37270e4242fbd31e80bb47dc5e7ab82cf2d557",
"4f1e9734bc80a290faaf9d611efcb8d7802dbc4f",
"9e72233c78baf685e500dd6c94212b30a4c5f27d",
"9e72233c78baf685e500dd6c94212b30a4c5f27d");
const std::string rtcp_stats_checksum = PlatformChecksum(
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",