Fix problems if first packet into NetEq is rejected
This CL fixes the problem described in issue 4021. In summary, of the
very first packet coming in to NetEq gets rejected because the RTP
payload type is unknown, subsequent GetAudio calls would trigger asserts
(in debug builds). The problem was that the first_packet_ variable was
reset and new_codec_ was set, even though the packet was discarded
further down the line. Now, these variables are modified after the
packet has been verified.
BUG=4021
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7724 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index f3d1a4f..7e8af3c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -458,8 +458,10 @@
bool update_sample_rate_and_channels = false;
// Reinitialize NetEq if it's needed (changed SSRC or first call).
if ((main_header.ssrc != ssrc_) || first_packet_) {
+ // Note: |first_packet_| will be cleared further down in this method, once
+ // the packet has been successfully inserted into the packet buffer.
+
rtcp_.Init(main_header.sequenceNumber);
- first_packet_ = false;
// Flush the packet buffer and DTMF buffer.
packet_buffer_->Flush();
@@ -475,13 +477,10 @@
timestamp_ = main_header.timestamp;
current_rtp_payload_type_ = main_header.payloadType;
- // Set MCU to update codec on next SignalMCU call.
- new_codec_ = true;
-
// Reset timestamp scaling.
timestamp_scaler_->Reset();
- // Triger an update of sampling rate and the number of channels.
+ // Trigger an update of sampling rate and the number of channels.
update_sample_rate_and_channels = true;
}
@@ -612,6 +611,13 @@
PacketBuffer::DeleteAllPackets(&packet_list);
return kOtherError;
}
+
+ if (first_packet_) {
+ first_packet_ = false;
+ // Update the codec on the next GetAudio call.
+ new_codec_ = true;
+ }
+
if (current_rtp_payload_type_ != 0xFF) {
const DecoderDatabase::DecoderInfo* dec_info =
decoder_database_->GetDecoderInfo(current_rtp_payload_type_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index efe122c..56ea425 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -589,8 +589,7 @@
// This test verifies that NetEq can handle the situation where the first
// incoming packet is rejected.
-// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=4021.
-TEST_F(NetEqImplTest, DISABLED_FirstPacketUnknown) {
+TEST_F(NetEqImplTest, FirstPacketUnknown) {
UseNoMocks();
CreateInstance();
