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Fix problems if first packet into NetEq is rejected

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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
This commit is contained in:
henrik.lundin@webrtc.org 2014-11-20 14:14:49 +00:00
parent ed91068bf1
commit 6ff3ac1db8
2 changed files with 12 additions and 7 deletions
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16
webrtc/modules/audio_coding/neteq/neteq_impl.cc
View File

@ -458,8 +458,10 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
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 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
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 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
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_);

3
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -589,8 +589,7 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
// 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();