Propagate arrival time inside NetEq
Bug: webrtc:341266986
Change-Id: I0fdd14e3fc5b09cbc9369497501f399464964211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352920
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42414}
diff --git a/api/neteq/BUILD.gn b/api/neteq/BUILD.gn
index f73085a..c29660d 100644
--- a/api/neteq/BUILD.gn
+++ b/api/neteq/BUILD.gn
@@ -23,6 +23,7 @@
"../../rtc_base:stringutils",
"../../system_wrappers:system_wrappers",
"../audio_codecs:audio_codecs_api",
+ "../units:timestamp",
"//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index 927bd62..d16f7ba 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -23,6 +23,7 @@
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
+#include "api/units/timestamp.h"
namespace webrtc {
@@ -183,10 +184,20 @@
virtual ~NetEq() {}
+ virtual int InsertPacket(const RTPHeader& rtp_header,
+ rtc::ArrayView<const uint8_t> payload) {
+ // TODO: webrtc:343501093 - removed unused method.
+ return InsertPacket(rtp_header, payload,
+ /*receive_time=*/Timestamp::MinusInfinity());
+ }
// Inserts a new packet into NetEq.
// Returns 0 on success, -1 on failure.
virtual int InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload) = 0;
+ rtc::ArrayView<const uint8_t> payload,
+ Timestamp receive_time) {
+ // TODO: webrtc:343501093 - Make this method pure virtual.
+ return InsertPacket(rtp_header, payload);
+ }
// Lets NetEq know that a packet arrived with an empty payload. This typically
// happens when empty packets are used for probing the network channel, and
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 1fa396e..b2fa0bf 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -196,8 +196,8 @@
private:
void ReceivePacket(const uint8_t* packet,
size_t packet_length,
- const RTPHeader& header)
- RTC_RUN_ON(worker_thread_checker_);
+ const RTPHeader& header,
+ Timestamp receive_time) RTC_RUN_ON(worker_thread_checker_);
int ResendPackets(const uint16_t* sequence_numbers, int length);
void UpdatePlayoutTimestamp(bool rtcp, int64_t now_ms)
RTC_RUN_ON(worker_thread_checker_);
@@ -205,7 +205,8 @@
int GetRtpTimestampRateHz() const;
void OnReceivedPayloadData(rtc::ArrayView<const uint8_t> payload,
- const RTPHeader& rtpHeader)
+ const RTPHeader& rtpHeader,
+ Timestamp receive_time)
RTC_RUN_ON(worker_thread_checker_);
void InitFrameTransformerDelegate(
@@ -254,7 +255,6 @@
AudioSinkInterface* audio_sink_ = nullptr;
AudioLevel _outputAudioLevel;
- Clock* const clock_;
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
// Timestamp of the audio pulled from NetEq.
@@ -320,7 +320,8 @@
void ChannelReceive::OnReceivedPayloadData(
rtc::ArrayView<const uint8_t> payload,
- const RTPHeader& rtpHeader) {
+ const RTPHeader& rtpHeader,
+ Timestamp receive_time) {
if (!playing_) {
// Avoid inserting into NetEQ when we are not playing. Count the
// packet as discarded.
@@ -335,7 +336,7 @@
// the SourceTracker from updating RtpSource information.
if (source_tracker_) {
RtpPacketInfos::vector_type packet_vector = {
- RtpPacketInfo(rtpHeader, clock_->CurrentTime())};
+ RtpPacketInfo(rtpHeader, receive_time)};
source_tracker_->OnFrameDelivered(RtpPacketInfos(packet_vector));
}
@@ -343,7 +344,7 @@
}
// Push the incoming payload (parsed and ready for decoding) into the ACM
- if (acm_receiver_.InsertPacket(rtpHeader, payload) != 0) {
+ if (acm_receiver_.InsertPacket(rtpHeader, payload, receive_time) != 0) {
RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to "
"push data to the ACM";
return;
@@ -372,7 +373,9 @@
receive_audio_callback = [this](rtc::ArrayView<const uint8_t> packet,
const RTPHeader& header) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
- OnReceivedPayloadData(packet, header);
+ // TODO(lionelk): Get the receive time.
+ OnReceivedPayloadData(packet, header,
+ /*receive_time=*/Timestamp::MinusInfinity());
};
frame_transformer_delegate_ =
rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
@@ -545,7 +548,6 @@
jitter_buffer_fast_playout,
jitter_buffer_min_delay_ms)),
_outputAudioLevel(),
- clock_(clock),
ntp_estimator_(clock),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
@@ -663,12 +665,14 @@
rtc::saturated_cast<uint32_t>(packet_copy.payload_type_frequency()),
header.extension.absolute_capture_time);
- ReceivePacket(packet_copy.data(), packet_copy.size(), header);
+ ReceivePacket(packet_copy.data(), packet_copy.size(), header,
+ packet.arrival_time());
}
void ChannelReceive::ReceivePacket(const uint8_t* packet,
size_t packet_length,
- const RTPHeader& header) {
+ const RTPHeader& header,
+ Timestamp receive_time) {
const uint8_t* payload = packet + header.headerLength;
RTC_DCHECK_GE(packet_length, header.headerLength);
size_t payload_length = packet_length - header.headerLength;
@@ -688,7 +692,8 @@
const FrameDecryptorInterface::Result decrypt_result =
frame_decryptor_->Decrypt(
cricket::MEDIA_TYPE_AUDIO, csrcs,
- /*additional_data=*/nullptr,
+ /*additional_data=*/
+ nullptr,
rtc::ArrayView<const uint8_t>(payload, payload_data_length),
decrypted_audio_payload);
@@ -720,7 +725,7 @@
frame_transformer_delegate_->Transform(payload_data, header, remote_ssrc_,
mime_type.str());
} else {
- OnReceivedPayloadData(payload_data, header);
+ OnReceivedPayloadData(payload_data, header, receive_time);
}
}
diff --git a/audio/voip/audio_ingress.cc b/audio/voip/audio_ingress.cc
index 80f2115..29363c1 100644
--- a/audio/voip/audio_ingress.cc
+++ b/audio/voip/audio_ingress.cc
@@ -11,6 +11,7 @@
#include "audio/voip/audio_ingress.h"
#include <algorithm>
+#include <ctime>
#include <utility>
#include <vector>
@@ -50,7 +51,8 @@
rtp_receive_statistics_(receive_statistics),
rtp_rtcp_(rtp_rtcp),
acm_receiver_(CreateAcmConfig(decoder_factory)),
- ntp_estimator_(clock) {}
+ ntp_estimator_(clock),
+ clock_(clock) {}
AudioIngress::~AudioIngress() = default;
@@ -184,7 +186,8 @@
auto data_view = rtc::ArrayView<const uint8_t>(payload, payload_data_length);
// Push the incoming payload (parsed and ready for decoding) into the ACM.
- if (acm_receiver_.InsertPacket(header, data_view) != 0) {
+ if (acm_receiver_.InsertPacket(header, data_view, clock_->CurrentTime()) !=
+ 0) {
RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
"push data to the ACM";
}
diff --git a/audio/voip/audio_ingress.h b/audio/voip/audio_ingress.h
index 11bde7c..36876de 100644
--- a/audio/voip/audio_ingress.h
+++ b/audio/voip/audio_ingress.h
@@ -133,6 +133,8 @@
RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(lock_);
+ Clock* clock_;
+
// For receiving RTP statistics, this tracks the sampling rate value
// per payload type set when caller set via SetReceiveCodecs.
std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 0ae4e34..655a934 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -47,6 +47,7 @@
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
"../../api/neteq:neteq_api",
+ "../../api/units:timestamp",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:audio_format_to_string",
@@ -712,6 +713,7 @@
"../../api/neteq:neteq_api",
"../../api/neteq:neteq_controller_api",
"../../api/neteq:tick_timer",
+ "../../api/units:timestamp",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:audio_format_to_string",
@@ -784,6 +786,7 @@
"../../api/neteq:custom_neteq_factory",
"../../api/neteq:default_neteq_controller_factory",
"../../api/neteq:neteq_api",
+ "../../api/units:timestamp",
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:copy_on_write_buffer",
@@ -938,6 +941,7 @@
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../api/neteq:neteq_api",
+ "../../api/units:timestamp",
"../../rtc_base:checks",
"../../rtc_base:digest",
"../../rtc_base:stringutils",
@@ -1067,6 +1071,7 @@
"../../api/audio_codecs/ilbc:audio_encoder_ilbc",
"../../api/audio_codecs/opus:audio_decoder_opus",
"../../api/audio_codecs/opus:audio_encoder_opus",
+ "../../api/units:timestamp",
"../../common_audio",
"../../rtc_base:checks",
"../../rtc_base:logging",
@@ -1356,6 +1361,7 @@
":neteq_tools_minimal",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../api/neteq:neteq_api",
+ "../../api/units:timestamp",
"../../rtc_base:checks",
"../../rtc_base:stringutils",
"../../system_wrappers",
@@ -1684,6 +1690,7 @@
"../../api/neteq:tick_timer",
"../../api/neteq:tick_timer_unittest",
"../../api/rtc_event_log",
+ "../../api/units:timestamp",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../common_audio:mock_common_audio",
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 66f6255..20d0a2f 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/modules/audio_coding/acm2/acm_receive_test.cc
@@ -122,7 +122,8 @@
EXPECT_EQ(0, acm_receiver_->InsertPacket(
packet->header(),
rtc::ArrayView<const uint8_t>(
- packet->payload(), packet->payload_length_bytes())))
+ packet->payload(), packet->payload_length_bytes()),
+ clock_.CurrentTime()))
<< "Failure when inserting packet:" << std::endl
<< " PT = " << static_cast<int>(packet->header().payloadType)
<< std::endl
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index ab11358..c0ac681c 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -20,6 +20,7 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/neteq/neteq.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
@@ -102,7 +103,8 @@
}
int AcmReceiver::InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> incoming_payload) {
+ rtc::ArrayView<const uint8_t> incoming_payload,
+ Timestamp receive_time) {
if (incoming_payload.empty()) {
neteq_->InsertEmptyPacket(rtp_header);
return 0;
@@ -137,7 +139,7 @@
}
} // `mutex_` is released.
- if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) {
+ if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_time) < 0) {
RTC_LOG(LS_ERROR) << "AcmReceiver::InsertPacket "
<< static_cast<int>(rtp_header.payloadType)
<< " Failed to insert packet";
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 21bee16..d631403 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -28,6 +28,7 @@
#include "api/audio_codecs/audio_format.h"
#include "api/neteq/neteq.h"
#include "api/neteq/neteq_factory.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/acm2/acm_resampler.h"
#include "modules/audio_coding/acm2/call_statistics.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
@@ -70,13 +71,15 @@
// information about payload type, sequence number,
// timestamp, SSRC and marker bit.
// - incoming_payload : Incoming audio payload.
- // - length_payload : Length of incoming audio payload in bytes.
+ // - receive_time : Timestamp when the packet has been seen on the
+ // network card.
//
// Return value : 0 if OK.
// <0 if NetEq returned an error.
//
int InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> incoming_payload);
+ rtc::ArrayView<const uint8_t> incoming_payload,
+ Timestamp receive_time = Timestamp::MinusInfinity());
//
// Asks NetEq for 10 milliseconds of decoded audio.
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 8b35f4a..e86c718 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -16,6 +16,7 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
@@ -119,7 +120,8 @@
int ret_val = receiver_->InsertPacket(
rtp_header_,
- rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes));
+ rtc::ArrayView<const uint8_t>(payload_data, payload_len_bytes),
+ Timestamp::MinusInfinity());
if (ret_val < 0) {
RTC_DCHECK_NOTREACHED();
return -1;
@@ -373,7 +375,8 @@
for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
- ASSERT_EQ(0, receiver_->InsertPacket(rtp_header, kPayload));
+ ASSERT_EQ(0, receiver_->InsertPacket(rtp_header, kPayload,
+ Timestamp::MinusInfinity()));
++rtp_header.sequenceNumber;
rtp_header.timestamp += kFrameSizeSamples;
ASSERT_EQ(0, receiver_->GetAudio(-1, &audio_frame, &muted));
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 75e38bc..7674538 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
@@ -25,6 +25,7 @@
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/acm2/acm_receive_test.h"
#include "modules/audio_coding/acm2/acm_send_test.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@@ -209,9 +210,10 @@
virtual void InsertPacket() {
const uint8_t kPayload[kPayloadSizeBytes] = {0};
- ASSERT_EQ(0, acm_receiver_->InsertPacket(rtp_header_,
- rtc::ArrayView<const uint8_t>(
- kPayload, kPayloadSizeBytes)));
+ ASSERT_EQ(0, acm_receiver_->InsertPacket(
+ rtp_header_,
+ rtc::ArrayView<const uint8_t>(kPayload, kPayloadSizeBytes),
+ /*receive_time=*/Timestamp::MinusInfinity()));
rtp_utility_->Forward(&rtp_header_);
}
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 1ee4dc4..dcdac98 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -193,11 +193,12 @@
NetEqImpl::~NetEqImpl() = default;
int NetEqImpl::InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload) {
+ rtc::ArrayView<const uint8_t> payload,
+ Timestamp receive_time) {
rtc::MsanCheckInitialized(payload);
TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
MutexLock lock(&mutex_);
- if (InsertPacketInternal(rtp_header, payload) != 0) {
+ if (InsertPacketInternal(rtp_header, payload, receive_time) != 0) {
return kFail;
}
return kOK;
@@ -464,13 +465,12 @@
// Methods below this line are private.
int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload) {
+ rtc::ArrayView<const uint8_t> payload,
+ Timestamp receive_time) {
if (payload.empty()) {
RTC_LOG_F(LS_ERROR) << "payload is empty";
return kInvalidPointer;
}
-
- Timestamp receive_time = clock_->CurrentTime();
stats_->ReceivedPacket();
PacketList packet_list;
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index f808c9e..8730d9a 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -13,6 +13,7 @@
#include <map>
#include <memory>
+#include <optional>
#include <string>
#include <utility>
#include <vector>
@@ -24,6 +25,7 @@
#include "api/neteq/neteq_controller_factory.h"
#include "api/neteq/tick_timer.h"
#include "api/rtp_packet_info.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/expand_uma_logger.h"
#include "modules/audio_coding/neteq/packet.h"
@@ -127,7 +129,8 @@
// Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
int InsertPacket(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload) override;
+ rtc::ArrayView<const uint8_t> payload,
+ Timestamp receive_time) override;
void InsertEmptyPacket(const RTPHeader& rtp_header) override;
@@ -204,7 +207,8 @@
// above. Returns 0 on success, otherwise an error code.
// TODO(hlundin): Merge this with InsertPacket above?
int InsertPacketInternal(const RTPHeader& rtp_header,
- rtc::ArrayView<const uint8_t> payload)
+ rtc::ArrayView<const uint8_t> payload,
+ Timestamp receive_time)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
// Returns true if the payload type changed (this should be followed by
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 07c924c..9018d78 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -194,7 +194,9 @@
kPayloadType, SdpAudioFormat("telephone-event", sample_rate_hz, 1)));
// Insert first packet.
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
// Pull audio once.
const size_t kMaxOutputSize =
@@ -373,12 +375,14 @@
}
// Insert first packet.
- neteq_->InsertPacket(rtp_header, payload);
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime());
// Insert second packet.
rtp_header.timestamp += 160;
rtp_header.sequenceNumber += 1;
- neteq_->InsertPacket(rtp_header, payload);
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime());
}
TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
@@ -400,7 +404,9 @@
// 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));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
rtp_header.timestamp += kPayloadLengthSamples;
rtp_header.sequenceNumber += 1;
EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
@@ -408,7 +414,9 @@
// 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));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
const Packet* test_packet = packet_buffer_->PeekNextPacket();
EXPECT_EQ(rtp_header.timestamp, test_packet->timestamp);
@@ -494,7 +502,8 @@
// Insert one packet.
clock_.AdvanceTimeMilliseconds(123456);
Timestamp expected_receive_time = clock_.CurrentTime();
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload, expected_receive_time));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -588,7 +597,9 @@
// Insert one packet.
clock_.AdvanceTimeMilliseconds(123456);
Timestamp expected_receive_time = clock_.CurrentTime();
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -618,14 +629,18 @@
rtp_header.extension.set_audio_level(AudioLevel(/*voice_activity=*/false, 1));
payload[0] = 1;
clock_.AdvanceTimeMilliseconds(1000);
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
rtp_header.extension.set_audio_level(AudioLevel(/*voice_activity=*/false, 2));
payload[0] = 2;
clock_.AdvanceTimeMilliseconds(2000);
expected_receive_time = clock_.CurrentTime();
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
// Expect only the second packet to be decoded (the one with "2" as the first
// payload byte).
@@ -684,7 +699,9 @@
// 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));
+ EXPECT_EQ(NetEq::kFail,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -705,7 +722,9 @@
for (size_t i = 0; i < 10; ++i) {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
@@ -777,7 +796,8 @@
header.extension.set_audio_level(AudioLevel(/*voice_activity=*/false, 12));
header.numCSRCs = 1;
header.arrOfCSRCs[0] = 123;
- neteq_->InsertPacket(header, payload);
+ neteq_->InsertPacket(header, payload,
+ /*receive_time=*/clock_.CurrentTime());
AudioFrame frame;
bool muted;
neteq_->GetAudio(&frame, &muted);
@@ -875,7 +895,9 @@
auto insert_packet = [&]() {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
};
// Insert 10 packets.
for (size_t i = 0; i < 10; ++i) {
@@ -959,7 +981,9 @@
auto insert_packet = [&]() {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
};
// Insert 10 packets.
for (size_t i = 0; i < 10; ++i) {
@@ -1033,7 +1057,9 @@
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
if (!lost)
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
};
// Insert and decode 10 packets.
@@ -1122,7 +1148,9 @@
rtp_header.sequenceNumber += packets[i].sequence_number_delta;
rtp_header.timestamp += packets[i].timestamp_delta;
payload[0] = i;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
// Pointee(x) verifies that first byte of the payload equals x, this makes
// it possible to verify that the correct payload is fed to Decode().
@@ -1220,7 +1248,9 @@
// Insert one packet.
payload[0] = kFirstPayloadValue; // This will make Decode() fail.
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
// Insert another packet.
payload[0] = kSecondPayloadValue; // This will make Decode() successful.
@@ -1228,7 +1258,9 @@
// 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));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
AudioFrame output;
bool muted;
@@ -1281,7 +1313,9 @@
// 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));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
rtp_header.timestamp += rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
++rtp_header.sequenceNumber;
}
@@ -1333,7 +1367,9 @@
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
@@ -1425,7 +1461,9 @@
for (int i = 0; i < 20; ++i) {
rtp_header.sequenceNumber += 1;
rtp_header.timestamp += kFrameLengthSamples;
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
}
// Pull audio.
@@ -1550,7 +1588,9 @@
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
@@ -1572,7 +1612,9 @@
Field(&NetEqController::PacketArrivedInfo::main_timestamp,
rtp_header.timestamp))));
- EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
}
// When using a codec with 1000 channels, there should be no crashes.
@@ -1626,7 +1668,8 @@
}));
// Insert first packet.
- neteq_->InsertPacket(rtp_header, payload);
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime());
AudioFrame audio_frame;
bool muted;
@@ -1655,12 +1698,16 @@
header.payloadType = kCnPayloadType;
uint8_t payload[320] = {0};
- EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK);
+ EXPECT_EQ(neteq_->InsertPacket(header, payload,
+ /*receive_time=*/clock_.CurrentTime()),
+ NetEq::kOK);
EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 0u);
header.payloadType = kSpeechPayloadType;
header.timestamp += 160;
- EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK);
+ EXPECT_EQ(neteq_->InsertPacket(header, payload,
+ /*receive_time=*/clock_.CurrentTime()),
+ NetEq::kOK);
// CN packet should be discarded, since it does not match the
// new speech sample rate.
EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 1u);
@@ -1688,12 +1735,16 @@
header.timestamp = 1234;
uint8_t payload[160] = {0};
- EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK);
+ EXPECT_EQ(neteq_->InsertPacket(header, payload,
+ /*receive_time=*/clock_.CurrentTime()),
+ NetEq::kOK);
EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 0u);
header.payloadType = kPcmaPayloadType;
header.timestamp += 80;
- EXPECT_EQ(neteq_->InsertPacket(header, payload), NetEq::kOK);
+ EXPECT_EQ(neteq_->InsertPacket(header, payload,
+ /*receive_time=*/clock_.CurrentTime()),
+ NetEq::kOK);
// The previous packet should be discarded since the codec changed.
EXPECT_EQ(neteq_->GetLifetimeStatistics().packets_discarded, 1u);
@@ -1780,7 +1831,9 @@
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));
+ EXPECT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload,
+ /*receive_time=*/clock_.CurrentTime()));
sequence_number_++;
}
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index da51698..69538d2 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -238,7 +238,9 @@
kPayloadType, frame_size_samples_, &rtp_header_);
if (!Lost(next_send_time)) {
static const uint8_t payload[kPayloadSizeByte] = {0};
- ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header_, payload));
+ ASSERT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header_, payload,
+ Timestamp::Millis(next_send_time)));
}
}
bool muted = true;
diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 6fa56fd..d7bd794 100644
--- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -18,6 +18,7 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/neteq/neteq.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
@@ -109,7 +110,7 @@
if (frame_size_samples_ * 2 != payload_size_bytes_) {
return -1;
}
- int next_send_time = rtp_generator_mono_.GetRtpHeader(
+ int next_send_time_ms = rtp_generator_mono_.GetRtpHeader(
kPayloadTypeMono, frame_size_samples_, &rtp_header_mono_);
MakeMultiChannelInput();
multi_payload_size_bytes_ = WebRtcPcm16b_Encode(
@@ -120,7 +121,7 @@
}
rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
&rtp_header_);
- return next_send_time;
+ return next_send_time_ms;
}
virtual void MakeMultiChannelInput() {
@@ -151,32 +152,35 @@
void RunTest(int num_loops) {
// Get next input packets (mono and multi-channel).
- int next_send_time;
- int next_arrival_time;
+ int next_send_time_ms;
+ int next_arrival_time_ms;
do {
- next_send_time = GetNewPackets();
- ASSERT_NE(-1, next_send_time);
- next_arrival_time = GetArrivalTime(next_send_time);
+ next_send_time_ms = GetNewPackets();
+ ASSERT_NE(-1, next_send_time_ms);
+ next_arrival_time_ms = GetArrivalTime(next_send_time_ms);
} while (Lost()); // If lost, immediately read the next packet.
- int time_now = 0;
+ int time_now_ms = 0;
for (int k = 0; k < num_loops; ++k) {
- while (time_now >= next_arrival_time) {
+ while (time_now_ms >= next_arrival_time_ms) {
// Insert packet in mono instance.
ASSERT_EQ(NetEq::kOK,
- neteq_mono_->InsertPacket(
- rtp_header_mono_, rtc::ArrayView<const uint8_t>(
- encoded_, payload_size_bytes_)));
+ neteq_mono_->InsertPacket(rtp_header_mono_,
+ rtc::ArrayView<const uint8_t>(
+ encoded_, payload_size_bytes_),
+ Timestamp::Millis(time_now_ms)));
// Insert packet in multi-channel instance.
- ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
- rtp_header_, rtc::ArrayView<const uint8_t>(
- encoded_multi_channel_,
- multi_payload_size_bytes_)));
+ ASSERT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(
+ rtp_header_,
+ rtc::ArrayView<const uint8_t>(encoded_multi_channel_,
+ multi_payload_size_bytes_),
+ Timestamp::Millis(time_now_ms)));
// Get next input packets (mono and multi-channel).
do {
- next_send_time = GetNewPackets();
- ASSERT_NE(-1, next_send_time);
- next_arrival_time = GetArrivalTime(next_send_time);
+ next_send_time_ms = GetNewPackets();
+ ASSERT_NE(-1, next_send_time_ms);
+ next_arrival_time_ms = GetArrivalTime(next_send_time_ms);
} while (Lost()); // If lost, immediately read the next packet.
}
// Get audio from mono instance.
@@ -197,7 +201,7 @@
// Compare mono and multi-channel.
ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
- time_now += kTimeStepMs;
+ time_now_ms += kTimeStepMs;
clock_.AdvanceTimeMilliseconds(kTimeStepMs);
}
}
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 7104b7a..8ebed6f 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -132,7 +132,7 @@
rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
@@ -237,7 +237,8 @@
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.payloadType = 1; // Not registered as a decoder.
- EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload));
+ EXPECT_EQ(NetEq::kFail,
+ neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
@@ -252,7 +253,7 @@
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.payloadType = 103; // iSAC, but the payload is invalid.
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
// Set all of `out_data_` to 1, and verify that it was set to 0 by the call
// to GetAudio.
int16_t* out_frame_data = out_frame_.mutable_data();
@@ -340,8 +341,10 @@
WebRtcPcm16b_Encode(block.data(), block.size(), payload);
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
- payload, enc_len_bytes)));
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtp_info,
+ rtc::ArrayView<const uint8_t>(payload, enc_len_bytes),
+ clock_.CurrentTime()));
output.Reset();
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
ASSERT_EQ(1u, output.num_channels_);
@@ -445,7 +448,7 @@
bool muted;
for (int i = 0; i < 3; ++i) {
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
++seq_no;
timestamp += kSamples;
@@ -462,8 +465,10 @@
size_t payload_len;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
// This is the first time this CNG packet is inserted.
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
- payload, payload_len)));
+ ASSERT_EQ(0,
+ neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len),
+ clock_.CurrentTime()));
// Pull audio once and make sure CNG is played.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
@@ -476,8 +481,10 @@
// Insert the same CNG packet again. Note that at this point it is old, since
// we have already decoded the first copy of it.
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
- payload, payload_len)));
+ ASSERT_EQ(0,
+ neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len),
+ clock_.CurrentTime()));
// Pull audio until we have played `kCngPeriodMs` of CNG. Start at 10 ms since
// we have already pulled out CNG once.
@@ -497,7 +504,7 @@
// Insert speech again.
for (int i = 0; i < 4; ++i) {
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
++seq_no;
timestamp += kSamples;
}
@@ -529,7 +536,8 @@
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(NetEq::kOK,
neteq_->InsertPacket(
- rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len)));
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len),
+ clock_.CurrentTime()));
++seq_no;
timestamp += kCngPeriodSamples;
@@ -545,7 +553,7 @@
do {
ASSERT_LT(timeout_counter++, 20) << "Test timed out";
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
++seq_no;
timestamp += kSamples;
@@ -571,7 +579,7 @@
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
}
void InsertCngPacket(uint32_t rtp_timestamp) {
@@ -580,8 +588,9 @@
size_t payload_len;
PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
- payload, payload_len)));
+ neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len),
+ clock_.CurrentTime()));
}
bool GetAudioReturnMuted() {
@@ -791,8 +800,8 @@
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
- EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
+ EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
AudioFrame out_frame1, out_frame2;
bool muted;
@@ -815,8 +824,9 @@
// packet.
for (int i = 0; i < 5; ++i) {
PopulateRtpInfo(0, kSamples * 1000 + kSamples * i, &rtp_info);
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
- EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
+ EXPECT_EQ(0,
+ neteq2_->InsertPacket(rtp_info, payload, clock_.CurrentTime()));
}
int counter = 0;
@@ -854,7 +864,7 @@
for (int j = 0; j < 10; j++) {
rtp_info.sequenceNumber = seq_no++;
rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
- neteq_->InsertPacket(rtp_info, payload);
+ neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime());
neteq_->GetAudio(&out_frame_, &muted);
}
@@ -894,7 +904,7 @@
if (packets_sent < kNumPackets) {
rtp_info.sequenceNumber = packets_sent++;
rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
- neteq_->InsertPacket(rtp_info, payload);
+ neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime());
}
// Get packet.
@@ -950,17 +960,17 @@
const uint8_t payload[kPayloadBytes] = {0};
int expected_target_delay = neteq_->TargetDelayMs() * kSamples;
- neteq_->InsertPacket(rtp_info, payload);
+ neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime());
bool muted;
neteq_->GetAudio(&out_frame_, &muted);
rtp_info.sequenceNumber += 1;
rtp_info.timestamp += kSamples;
- neteq_->InsertPacket(rtp_info, payload);
+ neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime());
rtp_info.sequenceNumber += 1;
rtp_info.timestamp += kSamples;
- neteq_->InsertPacket(rtp_info, payload);
+ neteq_->InsertPacket(rtp_info, payload, clock_.CurrentTime());
expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples;
// We have two packets in the buffer and kAccelerate operation will
diff --git a/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
index e626d09..1767aa6 100644
--- a/modules/audio_coding/neteq/test/neteq_decoding_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -13,6 +13,7 @@
#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtp_headers.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/neteq/test/result_sink.h"
#include "rtc_base/strings/string_builder.h"
@@ -106,11 +107,11 @@
// Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
if (packet_->header().payloadType != 104)
#endif
- ASSERT_EQ(
- 0, neteq_->InsertPacket(
- packet_->header(),
- rtc::ArrayView<const uint8_t>(
- packet_->payload(), packet_->payload_length_bytes())));
+ ASSERT_EQ(0, neteq_->InsertPacket(packet_->header(),
+ rtc::ArrayView<const uint8_t>(
+ packet_->payload(),
+ packet_->payload_length_bytes()),
+ clock_.CurrentTime()));
}
// Get next packet.
packet_ = rtp_source_->NextPacket();
@@ -239,7 +240,8 @@
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
// This sequence number was not in the set to drop. Insert it.
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload,
+ Timestamp::Millis(t_ms)));
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
@@ -298,7 +300,8 @@
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(
+ 0, neteq_->InsertPacket(rtp_info, payload, Timestamp::Millis(t_ms)));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
@@ -325,8 +328,10 @@
size_t payload_len;
RTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
- payload, payload_len)));
+ ASSERT_EQ(
+ 0, neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len),
+ Timestamp::Millis(t_ms)));
++seq_no;
timestamp += kCngPeriodSamples;
next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
@@ -367,8 +372,10 @@
size_t payload_len;
RTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
- payload, payload_len)));
+ ASSERT_EQ(
+ 0, neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len),
+ Timestamp::Millis(t_ms)));
++seq_no;
timestamp += kCngPeriodSamples;
next_input_time_ms += kCngPeriodMs * drift_factor;
@@ -384,7 +391,8 @@
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+ ASSERT_EQ(
+ 0, neteq_->InsertPacket(rtp_info, payload, Timestamp::Millis(t_ms)));
++seq_no;
timestamp += kSamples;
next_input_time_ms += kFrameSizeMs * drift_factor;
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index ccaa87b..a560b0d 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -87,7 +87,8 @@
}
if (!lost) {
// Insert packet.
- int error = neteq->InsertPacket(rtp_header, input_payload);
+ int error = neteq->InsertPacket(rtp_header, input_payload,
+ Timestamp::Millis(time_now_ms));
if (error != NetEq::kOK)
return -1;
}
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 18e6091..77a0542 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -16,6 +16,7 @@
#include "absl/flags/flag.h"
#include "absl/strings/string_view.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
#include "modules/audio_coding/neteq/tools/output_audio_file.h"
@@ -415,7 +416,8 @@
if (!PacketLost()) {
int ret = neteq_->InsertPacket(
rtp_header_,
- rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_));
+ rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_),
+ Timestamp::Millis(packet_input_time_ms));
if (ret != NetEq::kOK)
return -1;
Log() << "was sent.";
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index d47671f..7755b4e 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -13,6 +13,7 @@
#include <iomanip>
#include <iostream>
+#include "api/units/timestamp.h"
#include "modules/audio_coding/neteq/default_neteq_factory.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "system_wrappers/include/clock.h"
@@ -114,7 +115,8 @@
if (payload_data_length != 0) {
int error = neteq_->InsertPacket(
packet_data->header,
- rtc::ArrayView<const uint8_t>(packet_data->payload));
+ rtc::ArrayView<const uint8_t>(packet_data->payload),
+ Timestamp::Millis(time_now_ms));
if (error != NetEq::kOK && callbacks_.error_callback) {
callbacks_.error_callback->OnInsertPacketError(*packet_data);
}
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 8f634db..f09b357 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -83,7 +83,8 @@
}
status = _receiverACM->InsertPacket(
- rtp_header, rtc::ArrayView<const uint8_t>(_payloadData, payloadDataSize));
+ rtp_header, rtc::ArrayView<const uint8_t>(_payloadData, payloadDataSize),
+ /*receive_time=*/Timestamp::MinusInfinity());
return status;
}
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index cab3645..ef997f1 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -170,8 +170,10 @@
}
EXPECT_EQ(0, _acm_receiver->InsertPacket(
- _rtpHeader, rtc::ArrayView<const uint8_t>(
- _incomingPayload, _realPayloadSizeBytes)));
+ _rtpHeader,
+ rtc::ArrayView<const uint8_t>(_incomingPayload,
+ _realPayloadSizeBytes),
+ /*receive_time=*/Timestamp::Millis(_nextTime)));
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
_payloadSizeBytes, &_nextTime);
if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index c4f6656..6b52000 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -14,6 +14,7 @@
#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/units/timestamp.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@@ -58,9 +59,10 @@
}
if (!PacketLost()) {
- _acm_receiver->InsertPacket(
- _rtpHeader, rtc::ArrayView<const uint8_t>(_incomingPayload,
- _realPayloadSizeBytes));
+ _acm_receiver->InsertPacket(_rtpHeader,
+ rtc::ArrayView<const uint8_t>(
+ _incomingPayload, _realPayloadSizeBytes),
+ Timestamp::Millis(_nextTime));
}
packet_counter_++;
_realPayloadSizeBytes = _rtpStream->Read(&_rtpHeader, _incomingPayload,
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index dd51760..1a56fbc 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -84,7 +84,8 @@
memcpy(payload_data_, payload_data, payload_size);
status = receiver_acm_->InsertPacket(
- rtp_header, rtc::ArrayView<const uint8_t>(payload_data_, payload_size));
+ rtp_header, rtc::ArrayView<const uint8_t>(payload_data_, payload_size),
+ /*receive_time=*/Timestamp::MinusInfinity());
payload_size_ = payload_size;
timestamp_diff_ = timestamp - last_in_timestamp_;
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 1e65e4a..4573a7e 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -61,7 +61,8 @@
if (lost_packet_ == false) {
status = receiver_acm_->InsertPacket(
- rtp_header, rtc::ArrayView<const uint8_t>(payload_data, payload_size));
+ rtp_header, rtc::ArrayView<const uint8_t>(payload_data, payload_size),
+ /*receive_time=*/Timestamp::MinusInfinity());
if (frame_type != AudioFrameType::kAudioFrameCN) {
payload_size_ = static_cast<int>(payload_size);
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 2a71627..cd8fb34 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -13,6 +13,7 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/rtp_headers.h"
+#include "api/units/timestamp.h"
#include "modules/audio_coding/acm2/acm_receiver.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/include/audio_coding_module.h"
@@ -85,7 +86,8 @@
rtp_header_.sequenceNumber++;
ASSERT_EQ(0, receiver_.InsertPacket(rtp_header_,
rtc::ArrayView<const uint8_t>(
- payload_, kFrameSizeSamples * 2)));
+ payload_, kFrameSizeSamples * 2),
+ Timestamp::MinusInfinity()));
}
// Pull audio equivalent to the amount of audio in one RTP packet.
