Remove the unused `receive_timestamp` arg to NetEq::InsertPacket
The implementation just ignores the provided timestamp, and gets the
time from the current clock instead.
Bug: webrtc:11028
Change-Id: I7a1fee36bef862c68d8f15fd19ee53b2bbb25892
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156164
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29434}
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 2f152c9..8862905 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -192,7 +192,6 @@
void TestDtmfPacket(int sample_rate_hz) {
const size_t kPayloadLength = 4;
const uint8_t kPayloadType = 110;
- const uint32_t kReceiveTime = 17;
const int kSampleRateHz = 16000;
config_.sample_rate_hz = kSampleRateHz;
UseNoMocks();
@@ -209,8 +208,7 @@
kPayloadType, SdpAudioFormat("telephone-event", sample_rate_hz, 1)));
// Insert first packet.
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Pull audio once.
const size_t kMaxOutputSize =
@@ -312,7 +310,6 @@
const uint16_t kFirstSequenceNumber = 0x1234;
const uint32_t kFirstTimestamp = 0x12345678;
const uint32_t kSsrc = 0x87654321;
- const uint32_t kFirstReceiveTime = 17;
uint8_t payload[kPayloadLength] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
@@ -383,12 +380,12 @@
}
// Insert first packet.
- neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime);
+ neteq_->InsertPacket(rtp_header, payload);
// Insert second packet.
rtp_header.timestamp += 160;
rtp_header.sequenceNumber += 1;
- neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155);
+ neteq_->InsertPacket(rtp_header, payload);
}
TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
@@ -398,7 +395,6 @@
const int 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};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
@@ -411,8 +407,7 @@
// Insert packets. The buffer should not flush.
for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
rtp_header.timestamp += kPayloadLengthSamples;
rtp_header.sequenceNumber += 1;
EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
@@ -420,8 +415,7 @@
// Insert one more packet and make sure the buffer got flushed. That is, it
// should only hold one single packet.
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
const Packet* test_packet = packet_buffer_->PeekNextPacket();
EXPECT_EQ(rtp_header.timestamp, test_packet->timestamp);
@@ -448,7 +442,6 @@
// through to the sync buffer and to the playout timestamp.
TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
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.
@@ -508,8 +501,7 @@
// Insert one packet.
clock_.AdvanceTimeMilliseconds(123456);
int64_t expected_receive_time_ms = clock_.TimeInMilliseconds();
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -568,7 +560,6 @@
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
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.
@@ -603,8 +594,7 @@
// Insert one packet.
clock_.AdvanceTimeMilliseconds(123456);
int64_t expected_receive_time_ms = clock_.TimeInMilliseconds();
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -633,16 +623,14 @@
rtp_header.extension.audioLevel = 1;
payload[0] = 1;
clock_.AdvanceTimeMilliseconds(1000);
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
rtp_header.extension.audioLevel = 2;
payload[0] = 2;
clock_.AdvanceTimeMilliseconds(2000);
expected_receive_time_ms = clock_.TimeInMilliseconds();
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Expect only the second packet to be decoded (the one with "2" as the first
// payload byte).
@@ -684,7 +672,6 @@
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.
@@ -698,8 +685,7 @@
// Insert one packet. Note that we have not registered any payload type, so
// this packet will be rejected.
- EXPECT_EQ(NetEq::kFail,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_header, payload));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -720,8 +706,7 @@
for (size_t i = 0; i < 10; ++i) {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
@@ -745,7 +730,6 @@
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.
@@ -778,8 +762,7 @@
for (size_t i = 0; i < 10; ++i) {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
@@ -808,7 +791,6 @@
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
const uint8_t kPayloadType = 17; // Just an arbitrary number.
- const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateKhz = 48;
const size_t kPayloadLengthSamples =
static_cast<size_t>(20 * kSampleRateKhz); // 20 ms.
@@ -867,15 +849,13 @@
SdpAudioFormat("opus", 48000, 2)));
// Insert one packet (decoder will return speech).
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Insert second packet (decoder will return CNG).
payload[0] = 1;
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
AudioFrame output;
@@ -925,8 +905,7 @@
payload[0] = 2;
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
for (size_t i = 6; i < 8; ++i) {
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
@@ -953,7 +932,6 @@
static const size_t kChannels = 2;
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 =
@@ -1001,8 +979,7 @@
// Insert one packet.
payload[0] = kFirstPayloadValue; // This will make Decode() fail.
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
// Insert another packet.
payload[0] = kSecondPayloadValue; // This will make Decode() successful.
@@ -1010,8 +987,7 @@
// The second timestamp needs to be at least 30 ms after the first to make
// the second packet get decoded.
rtp_header.timestamp += 3 * kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
AudioFrame output;
bool muted;
@@ -1048,7 +1024,6 @@
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};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
@@ -1062,8 +1037,7 @@
// 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, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
rtp_header.timestamp += rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
++rtp_header.sequenceNumber;
}
@@ -1083,7 +1057,6 @@
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
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.
@@ -1116,8 +1089,7 @@
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
@@ -1144,7 +1116,6 @@
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
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 int kDecoderErrorCode = -97; // Any negative number.
@@ -1210,8 +1181,7 @@
for (int i = 0; i < 6; ++i) {
rtp_header.sequenceNumber += 1;
rtp_header.timestamp += kFrameLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
}
// Pull audio.
@@ -1258,7 +1228,6 @@
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
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 int kDecoderErrorCode = -97; // Any negative number.
@@ -1321,8 +1290,7 @@
for (int i = 0; i < 2; ++i) {
rtp_header.sequenceNumber += 1;
rtp_header.timestamp += kFrameLengthSamples;
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
}
// Pull audio.
@@ -1438,7 +1406,6 @@
const int 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;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
@@ -1448,8 +1415,7 @@
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
@@ -1459,8 +1425,7 @@
rtp_header.timestamp -= kPayloadLengthSamples;
EXPECT_CALL(*mock_delay_manager_,
Update(rtp_header.sequenceNumber, rtp_header.timestamp, _));
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
}
class Decoder120ms : public AudioDecoder {
@@ -1537,7 +1502,7 @@
rtp_header.ssrc = 15;
const size_t kPayloadLengthBytes = 1; // This can be arbitrary.
uint8_t payload[kPayloadLengthBytes] = {0};
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload, 10));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
sequence_number_++;
}