Revert of Change NetEq::InsertPacket to take an RTPHeader (patchset #2 id:20001 of https://codereview.webrtc.org/2807273004/ )
Reason for revert:
Broke downstream dependencies.
Original issue's description:
> Change NetEq::InsertPacket to take an RTPHeader
>
> It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> a member. None of the other member in WebRtcRTPHeader where used in
> NetEq.
>
> This CL adapts the production code; tests and tools will be converted
> in a follow-up CL.
>
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2807273004
> Cr-Commit-Position: refs/heads/master@{#17652}
> Committed: https://chromium.googlesource.com/external/webrtc/+/4d027576a6f7420fc4ec6be7f4f991cfad34b826
TBR=ivoc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2812933002
Cr-Commit-Position: refs/heads/master@{#17657}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index f6f1786..2438bc8 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -104,8 +104,8 @@
}
} // |crit_sect_| is released.
- if (neteq_->InsertPacket(rtp_header.header, incoming_payload,
- receive_timestamp) < 0) {
+ if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) <
+ 0) {
LOG(LERROR) << "AcmReceiver::InsertPacket "
<< static_cast<int>(header->payloadType)
<< " Failed to insert packet";
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 322a86f..450318e 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -26,6 +26,7 @@
// Forward declarations.
class AudioFrame;
+struct WebRtcRTPHeader;
class AudioDecoderFactory;
struct NetEqNetworkStatistics {
@@ -140,7 +141,7 @@
// of the time when the packet was received, and should be measured with
// the same tick rate as the RTP timestamp of the current payload.
// Returns 0 on success, -1 on failure.
- virtual int InsertPacket(const RTPHeader& rtp_header,
+ virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) = 0;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index 0ccba6d..4b3c0b7 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -210,8 +210,8 @@
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) override {
// Insert packet in internal decoder.
- ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(
- rtp_header.header, payload, receive_timestamp));
+ ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload,
+ receive_timestamp));
// Insert packet in external decoder instance.
NetEqExternalDecoderUnitTest::InsertPacket(rtp_header, payload,
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 89bddec..501e567 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -131,7 +131,7 @@
NetEqImpl::~NetEqImpl() = default;
-int NetEqImpl::InsertPacket(const RTPHeader& rtp_header,
+int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) {
rtc::MsanCheckInitialized(payload);
@@ -581,7 +581,7 @@
// Methods below this line are private.
-int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
+int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) {
if (payload.empty()) {
@@ -594,24 +594,24 @@
packet_list.push_back([&rtp_header, &payload] {
// Convert to Packet.
Packet packet;
- packet.payload_type = rtp_header.payloadType;
- packet.sequence_number = rtp_header.sequenceNumber;
- packet.timestamp = rtp_header.timestamp;
+ packet.payload_type = rtp_header.header.payloadType;
+ packet.sequence_number = rtp_header.header.sequenceNumber;
+ packet.timestamp = rtp_header.header.timestamp;
packet.payload.SetData(payload.data(), payload.size());
// Waiting time will be set upon inserting the packet in the buffer.
RTC_DCHECK(!packet.waiting_time);
return packet;
}());
- bool update_sample_rate_and_channels =
- first_packet_ || (rtp_header.ssrc != ssrc_);
+ bool update_sample_rate_and_channels = first_packet_ ||
+ (rtp_header.header.ssrc != ssrc_);
if (update_sample_rate_and_channels) {
// Reset timestamp scaling.
timestamp_scaler_->Reset();
}
- if (!decoder_database_->IsRed(rtp_header.payloadType)) {
+ if (!decoder_database_->IsRed(rtp_header.header.payloadType)) {
// Scale timestamp to internal domain (only for some codecs).
timestamp_scaler_->ToInternal(&packet_list);
}
@@ -627,14 +627,14 @@
// Note: |first_packet_| will be cleared further down in this method, once
// the packet has been successfully inserted into the packet buffer.
- rtcp_.Init(rtp_header.sequenceNumber);
+ rtcp_.Init(rtp_header.header.sequenceNumber);
// Flush the packet buffer and DTMF buffer.
packet_buffer_->Flush();
dtmf_buffer_->Flush();
// Store new SSRC.
- ssrc_ = rtp_header.ssrc;
+ ssrc_ = rtp_header.header.ssrc;
// Update audio buffer timestamp.
sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_);
@@ -644,19 +644,19 @@
}
// Update RTCP statistics, only for regular packets.
- rtcp_.Update(rtp_header, receive_timestamp);
+ rtcp_.Update(rtp_header.header, receive_timestamp);
if (nack_enabled_) {
RTC_DCHECK(nack_);
if (update_sample_rate_and_channels) {
nack_->Reset();
}
- nack_->UpdateLastReceivedPacket(rtp_header.sequenceNumber,
- rtp_header.timestamp);
+ nack_->UpdateLastReceivedPacket(rtp_header.header.sequenceNumber,
+ rtp_header.header.timestamp);
}
// Check for RED payload type, and separate payloads into several packets.
- if (decoder_database_->IsRed(rtp_header.payloadType)) {
+ if (decoder_database_->IsRed(rtp_header.header.payloadType)) {
if (!red_payload_splitter_->SplitRed(&packet_list)) {
return kRedundancySplitError;
}
@@ -675,7 +675,7 @@
// Update main_timestamp, if new packets appear in the list
// after RED splitting.
- if (decoder_database_->IsRed(rtp_header.payloadType)) {
+ if (decoder_database_->IsRed(rtp_header.header.payloadType)) {
timestamp_scaler_->ToInternal(&packet_list);
main_timestamp = packet_list.front().timestamp;
main_payload_type = packet_list.front().payload_type;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 863bfbb..88c0308 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -105,7 +105,7 @@
// of the time when the packet was received, and should be measured with
// the same tick rate as the RTP timestamp of the current payload.
// Returns 0 on success, -1 on failure.
- int InsertPacket(const RTPHeader& rtp_header,
+ int InsertPacket(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) override;
@@ -222,7 +222,7 @@
// Inserts a new packet into NetEq. This is used by the InsertPacket method
// above. Returns 0 on success, otherwise an error code.
// TODO(hlundin): Merge this with InsertPacket above?
- int InsertPacketInternal(const RTPHeader& rtp_header,
+ int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 9897f69..47c2847 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -193,7 +193,7 @@
// Insert first packet.
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize =
@@ -384,12 +384,12 @@
}
// Insert first packet.
- neteq_->InsertPacket(rtp_header.header, payload, kFirstReceiveTime);
+ neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime);
// Insert second packet.
rtp_header.header.timestamp += 160;
rtp_header.header.sequenceNumber += 1;
- neteq_->InsertPacket(rtp_header.header, payload, kFirstReceiveTime + 155);
+ neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155);
}
TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
@@ -413,7 +413,7 @@
// 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.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.header.timestamp += kPayloadLengthSamples;
rtp_header.header.sequenceNumber += 1;
EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
@@ -422,7 +422,7 @@
// 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.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
const Packet* test_packet = packet_buffer_->PeekNextPacket();
EXPECT_EQ(rtp_header.header.timestamp, test_packet->timestamp);
@@ -502,7 +502,7 @@
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -583,7 +583,7 @@
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
@@ -600,12 +600,12 @@
rtp_header.header.timestamp -= kPayloadLengthSamples;
payload[0] = 1;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.header.sequenceNumber += 2;
rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
payload[0] = 2;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Expect only the second packet to be decoded (the one with "2" as the first
// payload byte).
@@ -651,7 +651,7 @@
// 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.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
// Pull audio once.
@@ -673,7 +673,7 @@
rtp_header.header.sequenceNumber++;
rtp_header.header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
@@ -760,14 +760,14 @@
// Insert one packet (decoder will return speech).
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Insert second packet (decoder will return CNG).
payload[0] = 1;
rtp_header.header.sequenceNumber++;
rtp_header.header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
AudioFrame output;
@@ -818,7 +818,7 @@
rtp_header.header.sequenceNumber += 2;
rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
for (size_t i = 6; i < 8; ++i) {
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
@@ -896,7 +896,7 @@
// Insert one packet.
payload[0] = kFirstPayloadValue; // This will make Decode() fail.
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Insert another packet.
payload[0] = kSecondPayloadValue; // This will make Decode() successful.
@@ -905,7 +905,7 @@
// the second packet get decoded.
rtp_header.header.timestamp += 3 * kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
AudioFrame output;
bool muted;
@@ -957,7 +957,7 @@
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.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.header.timestamp +=
rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
++rtp_header.header.sequenceNumber;
@@ -1013,7 +1013,7 @@
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
@@ -1109,7 +1109,7 @@
rtp_header.header.sequenceNumber += 1;
rtp_header.header.timestamp += kFrameLengthSamples;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
}
// Pull audio.
@@ -1221,7 +1221,7 @@
rtp_header.header.sequenceNumber += 1;
rtp_header.header.timestamp += kFrameLengthSamples;
EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(rtp_header.header, payload, kReceiveTime));
+ neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
}
// Pull audio.
@@ -1341,7 +1341,7 @@
rtp_header.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.header, payload, 10));
+ EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload, 10));
sequence_number_++;
}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 73c25a4..0d15f88 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -197,17 +197,16 @@
while (time_now >= next_arrival_time) {
// Insert packet in mono instance.
ASSERT_EQ(NetEq::kOK,
- neteq_mono_->InsertPacket(rtp_header_mono_.header,
+ neteq_mono_->InsertPacket(rtp_header_mono_,
rtc::ArrayView<const uint8_t>(
encoded_, payload_size_bytes_),
next_arrival_time));
// Insert packet in multi-channel instance.
- ASSERT_EQ(NetEq::kOK,
- neteq_->InsertPacket(
- rtp_header_.header,
- rtc::ArrayView<const uint8_t>(encoded_multi_channel_,
- multi_payload_size_bytes_),
- next_arrival_time));
+ ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
+ rtp_header_, rtc::ArrayView<const uint8_t>(
+ encoded_multi_channel_,
+ multi_payload_size_bytes_),
+ next_arrival_time));
// Get next input packets (mono and multi-channel).
do {
next_send_time = GetNewPackets();
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 1a54c54..33b4005 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -324,13 +324,12 @@
// Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
if (rtp_header.header.payloadType != 104)
#endif
- ASSERT_EQ(0,
- neteq_->InsertPacket(
- rtp_header.header,
- rtc::ArrayView<const uint8_t>(
- packet_->payload(), packet_->payload_length_bytes()),
- static_cast<uint32_t>(packet_->time_ms() *
- (output_sample_rate_ / 1000))));
+ ASSERT_EQ(0, neteq_->InsertPacket(
+ rtp_header,
+ rtc::ArrayView<const uint8_t>(
+ packet_->payload(), packet_->payload_length_bytes()),
+ static_cast<uint32_t>(packet_->time_ms() *
+ (output_sample_rate_ / 1000))));
}
// Get next packet.
packet_ = rtp_source_->NextPacket();
@@ -527,7 +526,7 @@
rtp_info.header.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.header.payloadType = 94; // PCM16b WB codec.
rtp_info.header.markerBit = 0;
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
@@ -568,7 +567,7 @@
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++frame_index;
}
@@ -596,7 +595,7 @@
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++frame_index;
}
@@ -634,7 +633,7 @@
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
@@ -662,7 +661,7 @@
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(0, neteq_->InsertPacket(
- rtp_info.header,
+ rtp_info,
rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
++seq_no;
timestamp += kCngPeriodSamples;
@@ -705,7 +704,7 @@
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(0, neteq_->InsertPacket(
- rtp_info.header,
+ rtp_info,
rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
++seq_no;
timestamp += kCngPeriodSamples;
@@ -722,7 +721,7 @@
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += kFrameSizeMs * drift_factor;
@@ -834,7 +833,7 @@
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = 1; // Not registered as a decoder.
- EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0));
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}
@@ -850,7 +849,7 @@
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = 103; // iSAC, but the payload is invalid.
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
@@ -957,10 +956,9 @@
WebRtcPcm16b_Encode(block.data(), block.size(), payload);
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
- ASSERT_EQ(0, neteq_->InsertPacket(
- rtp_info.header,
- rtc::ArrayView<const uint8_t>(payload, enc_len_bytes),
- receive_timestamp));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+ payload, enc_len_bytes),
+ receive_timestamp));
output.Reset();
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
ASSERT_EQ(1u, output.num_channels_);
@@ -1094,8 +1092,8 @@
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.header, payload,
- receive_timestamp));
+ ASSERT_EQ(0,
+ neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
++packets_inserted;
}
NetEqNetworkStatistics network_stats;
@@ -1183,7 +1181,7 @@
bool muted;
for (int i = 0; i < 3; ++i) {
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++seq_no;
timestamp += kSamples;
@@ -1200,9 +1198,9 @@
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.header,
- rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+ ASSERT_EQ(
+ 0, neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
// Pull audio once and make sure CNG is played.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
@@ -1214,9 +1212,9 @@
// 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.header,
- rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+ ASSERT_EQ(
+ 0, neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
// Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
// we have already pulled out CNG once.
@@ -1233,7 +1231,7 @@
++seq_no;
timestamp += kCngPeriodSamples;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull audio once and verify that the output is speech again.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
@@ -1266,10 +1264,10 @@
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
- ASSERT_EQ(NetEq::kOK,
- neteq_->InsertPacket(
- rtp_info.header,
- rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+ ASSERT_EQ(
+ NetEq::kOK,
+ neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
++seq_no;
timestamp += kCngPeriodSamples;
@@ -1285,7 +1283,7 @@
do {
ASSERT_LT(timeout_counter++, 20) << "Test timed out";
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
++seq_no;
timestamp += kSamples;
@@ -1311,7 +1309,7 @@
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
}
void InsertCngPacket(uint32_t rtp_timestamp) {
@@ -1319,10 +1317,10 @@
WebRtcRTPHeader rtp_info;
size_t payload_len;
PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
- EXPECT_EQ(NetEq::kOK,
- neteq_->InsertPacket(
- rtp_info.header,
- rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
+ EXPECT_EQ(
+ NetEq::kOK,
+ neteq_->InsertPacket(
+ rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
}
bool GetAudioReturnMuted() {
@@ -1547,8 +1545,8 @@
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
- EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info.header, payload, 0));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+ EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
AudioFrame out_frame1, out_frame2;
bool muted;
@@ -1570,8 +1568,8 @@
// Insert new data. Timestamp is corrected for the time elapsed since the last
// packet.
PopulateRtpInfo(0, kSamples * 1000, &rtp_info);
- EXPECT_EQ(0, neteq_->InsertPacket(rtp_info.header, payload, 0));
- EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info.header, payload, 0));
+ EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
+ EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
int counter = 0;
while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 95fdb04..ed51279 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -40,8 +40,8 @@
WebRtcRTPHeader rtp_header,
rtc::ArrayView<const uint8_t> payload,
uint32_t receive_timestamp) {
- ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header.header, payload,
- receive_timestamp));
+ ASSERT_EQ(NetEq::kOK,
+ neteq_->InsertPacket(rtp_header, payload, receive_timestamp));
}
void NetEqExternalDecoderTest::GetOutputAudio(AudioFrame* output) {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index df14081..4349a70 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -88,7 +88,7 @@
if (!lost) {
// Insert packet.
int error =
- neteq->InsertPacket(rtp_header.header, input_payload,
+ neteq->InsertPacket(rtp_header, input_payload,
packet_input_time_ms * kSampRateHz / 1000);
if (error != NetEq::kOK)
return -1;
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 732c7b8..7b3a35b 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -380,7 +380,7 @@
if (payload_size_bytes_ > 0) {
if (!PacketLost()) {
int ret = neteq_->InsertPacket(
- rtp_header_.header,
+ rtp_header_,
rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_),
packet_input_time_ms * in_sampling_khz_);
if (ret != NetEq::kOK)
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
index 13b11e8c..cc88b38 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -70,7 +70,7 @@
std::unique_ptr<NetEqInput::PacketData> packet_data = input_->PopPacket();
RTC_CHECK(packet_data);
int error = neteq_->InsertPacket(
- packet_data->header.header,
+ packet_data->header,
rtc::ArrayView<const uint8_t>(packet_data->payload),
static_cast<uint32_t>(packet_data->time_ms * sample_rate_hz_ / 1000));
if (error != NetEq::kOK && error_callback_) {
