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Unified Diff: webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

Issue 1307893004: NetEq: Fixing a bug that caused rtc::checked_cast to trigger (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Review comments Created 5 years, 4 months ago
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Index: webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 006a5ad542397cb839b2fe70bd5e59c093fbdbc8..1020a45f468aac88297512ab832701b7dc08f680 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -13,6 +13,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/safe_conversions.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
@@ -907,4 +908,43 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
EXPECT_EQ(kChannels, num_channels);
}
+// This test inserts packets until the buffer is flushed. After that, it asks
+// NetEq for the network statistics. The purpose of the test is to make sure
+// that even though the buffer size increment is negative (which it becomes when
+// the packet causing a flush is inserted), the packet length stored in the
+// decision logic remains valid.
+TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
+ UseNoMocks();
+ CreateInstance();
+
+ const size_t kPayloadLengthSamples = 80;
+ const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit.
+ const uint8_t kPayloadType = 17; // Just an arbitrary number.
+ const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
+ uint8_t payload[kPayloadLengthBytes] = {0};
+ WebRtcRTPHeader rtp_header;
+ rtp_header.header.payloadType = kPayloadType;
+ rtp_header.header.sequenceNumber = 0x1234;
+ rtp_header.header.timestamp = 0x12345678;
+ rtp_header.header.ssrc = 0x87654321;
+
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
+
+ // Insert packets until the buffer flushes.
+ for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
+ EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
+ kReceiveTime));
+ rtp_header.header.timestamp +=
+ rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
+ ++rtp_header.header.sequenceNumber;
+ }
+ EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+
+ // Ask for network statistics. This should not crash.
+ NetEqNetworkStatistics stats;
+ EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
+}
} // namespace webrtc
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