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NetEq: Fixing a corner case with depleted sync buffer

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In some cases, the number of samples (per channel) in NetEq's sync
buffer could fall below the allowed minimum (5 samples for narrowband,
scaling for other rates). If the number of samples extracted from the
buffer was smaller than the desired number, an error is
returned. However, if the decoder returns fewer samples than expected,
it could happen that the sync buffer level falls under the minimum,
but enough samples are extracted. This triggered an assert. With this
change, the minimum level of the sync buffer is always enforced.

A test is implemented to trigger the problem. It made the assert fire
without this fix, but it now passes.

BUG=webrtc:4840
R=minyue@webrtc.org

Review URL: https://codereview.webrtc.org/1324453002 .

Cr-Commit-Position: refs/heads/master@{#9828}
This commit is contained in:
Henrik Lundin
2015-09-01 11:51:58 +02:00
parent 521875a9a4
commit 05f71fcb61
4 changed files with 78 additions and 1 deletions
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63
webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
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@ -942,4 +942,67 @@ TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
NetEqNetworkStatistics stats;
EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
}
TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
UseNoMocks();
CreateInstance();
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;
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;
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.WillRepeatedly(Return(kPayloadLengthSamples));
int16_t dummy_output[kPayloadLengthSamples] = {0};
// The below expectation will make the mock decoder write
// |kPayloadLengthSamples| - 5 zeros to the output array, and mark it as
// speech. That is, the decoded length is 5 samples shorter than the expected.
EXPECT_CALL(mock_decoder,
Decode(_, kPayloadLengthBytes, kSampleRateHz, _, _, _))
.WillOnce(
DoAll(SetArrayArgument<4>(dummy_output,
dummy_output + kPayloadLengthSamples - 5),
SetArgPointee<5>(AudioDecoder::kSpeech),
Return(kPayloadLengthSamples - 5)));
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterExternalDecoder(&mock_decoder, kDecoderPCM16B,
kPayloadType, kSampleRateHz));
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
kReceiveTime));
EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
int16_t output[kMaxOutputSize];
size_t samples_per_channel;
int num_channels;
NetEqOutputType type;
EXPECT_EQ(NetEq::kOK,
neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
&num_channels, &type));
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(1, num_channels);
EXPECT_EQ(kOutputNormal, type);
EXPECT_CALL(mock_decoder, Die());
}
} // namespace webrtc