Delete in_order argument to RtpReceiver::IncomingRtpPacket
Bug: webrtc:7135
Change-Id: I35fbc76a5ca8d50caff918bbfd2cb13dce4cbd21
Reviewed-on: https://webrtc-review.googlesource.com/4141
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20154}diff --git a/modules/rtp_rtcp/include/rtp_receiver.h b/modules/rtp_rtcp/include/rtp_receiver.h
index 8931d78..9738116 100644
--- a/modules/rtp_rtcp/include/rtp_receiver.h
+++ b/modules/rtp_rtcp/include/rtp_receiver.h
@@ -79,8 +79,17 @@
virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
const uint8_t* payload,
size_t payload_length,
- PayloadUnion payload_specific,
- bool in_order) = 0;
+ PayloadUnion payload_specific) = 0;
+ // TODO(nisse): Deprecated version, delete as soon as downstream
+ // applications are updated.
+ bool IncomingRtpPacket(const RTPHeader& rtp_header,
+ const uint8_t* payload,
+ size_t payload_length,
+ PayloadUnion payload_specific,
+ bool in_order /* Ignored */) {
+ return IncomingRtpPacket(rtp_header, payload, payload_length,
+ payload_specific);
+ }
// Gets the RTP timestamp and the corresponding monotonic system
// time for the most recent in-order packet. Returns true on
diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index a47aa15..792e08b 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -20,6 +20,7 @@
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -27,6 +28,26 @@
namespace webrtc {
+namespace {
+bool InOrderPacket(rtc::Optional<uint16_t> latest_sequence_number,
+ uint16_t current_sequence_number) {
+ if (!latest_sequence_number)
+ return true;
+
+ // We need to distinguish between a late or retransmitted packet,
+ // and a sequence number discontinuity.
+ if (IsNewerSequenceNumber(current_sequence_number, *latest_sequence_number)) {
+ return true;
+ } else {
+ // If we have a restart of the remote side this packet is still in order.
+ return !IsNewerSequenceNumber(
+ current_sequence_number,
+ *latest_sequence_number - kDefaultMaxReorderingThreshold);
+ }
+}
+
+} // namespace
+
using RtpUtility::Payload;
// Only return the sources in the last 10 seconds.
@@ -142,12 +163,10 @@
return rtp_media_receiver_->Energy(array_of_energy);
}
-bool RtpReceiverImpl::IncomingRtpPacket(
- const RTPHeader& rtp_header,
- const uint8_t* payload,
- size_t payload_length,
- PayloadUnion payload_specific,
- bool in_order) {
+bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header,
+ const uint8_t* payload,
+ size_t payload_length,
+ PayloadUnion payload_specific) {
// Trigger our callbacks.
CheckSSRCChanged(rtp_header);
@@ -186,13 +205,18 @@
{
rtc::CritScope lock(&critical_section_rtp_receiver_);
- if (in_order) {
- if (last_received_timestamp_ != rtp_header.timestamp) {
- last_received_timestamp_ = rtp_header.timestamp;
- last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
- }
+ // TODO(nisse): Do not rely on InOrderPacket for recovered packets, when
+ // packet is passed as RtpPacketReceived and that information is available.
+ // We should ideally never record timestamps for retransmitted or recovered
+ // packets.
+ if (InOrderPacket(last_received_sequence_number_,
+ rtp_header.sequenceNumber)) {
+ last_received_sequence_number_.emplace(rtp_header.sequenceNumber);
+ last_received_timestamp_ = rtp_header.timestamp;
+ last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
}
}
+
return true;
}
@@ -237,7 +261,7 @@
bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp,
int64_t* receive_time_ms) const {
rtc::CritScope lock(&critical_section_rtp_receiver_);
- if (last_received_frame_time_ms_ < 0)
+ if (!last_received_sequence_number_)
return false;
*timestamp = last_received_timestamp_;
diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.h b/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 9257029..cd6b619 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -16,6 +16,7 @@
#include <unordered_map>
#include <vector>
+#include "api/optional.h"
#include "modules/rtp_rtcp/include/rtp_receiver.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -45,8 +46,7 @@
bool IncomingRtpPacket(const RTPHeader& rtp_header,
const uint8_t* payload,
size_t payload_length,
- PayloadUnion payload_specific,
- bool in_order) override;
+ PayloadUnion payload_specific) override;
bool GetLatestTimestamps(uint32_t* timestamp,
int64_t* receive_time_ms) const override;
@@ -95,6 +95,9 @@
uint32_t current_remote_csrc_[kRtpCsrcSize] RTC_GUARDED_BY(
critical_section_rtp_receiver_);
+ // Sequence number and timestamps for the latest in-order packet.
+ rtc::Optional<uint16_t> last_received_sequence_number_
+ RTC_GUARDED_BY(critical_section_rtp_receiver_);
uint32_t last_received_timestamp_
RTC_GUARDED_BY(critical_section_rtp_receiver_);
int64_t last_received_frame_time_ms_
diff --git a/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
index c8c9362..3d23b50 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
@@ -34,7 +34,6 @@
const uint32_t kSsrc2 = 124;
const uint32_t kCsrc1 = 111;
const uint32_t kCsrc2 = 222;
-const bool kInOrder = true;
static uint32_t rtp_timestamp(int64_t time_ms) {
return static_cast<uint32_t>(time_ms * kTestRate / 1000);
@@ -90,7 +89,7 @@
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
auto sources = rtp_receiver_->GetSources();
// One SSRC source and two CSRC sources.
EXPECT_THAT(sources, UnorderedElementsAre(
@@ -102,7 +101,7 @@
// contributing source object with same source id and updated timestamp.
fake_clock_.AdvanceTimeMilliseconds(1);
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
now_ms = fake_clock_.TimeInMilliseconds();
EXPECT_THAT(sources, UnorderedElementsAre(
@@ -141,7 +140,7 @@
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
auto sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
@@ -153,7 +152,7 @@
header.ssrc = kSsrc2;
header.timestamp = rtp_timestamp(now_ms);
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(
RtpSource(prev_time_ms, kSsrc1, RtpSourceType::SSRC),
@@ -167,7 +166,7 @@
prev_time_ms = now_ms;
now_ms = fake_clock_.TimeInMilliseconds();
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(
RtpSource(prev_time_ms, kSsrc2, RtpSourceType::SSRC),
@@ -177,7 +176,7 @@
fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
now_ms = fake_clock_.TimeInMilliseconds();
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(
RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC)));
@@ -197,9 +196,8 @@
for (size_t i = 0; i < kSourceListSize; ++i) {
header.ssrc = i;
header.arrOfCSRCs[0] = (i + 1);
- EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload,
- sizeof(kTestPayload),
- payload_specific, !kInOrder));
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
}
RtpSource source(0, 0, RtpSourceType::SSRC);
@@ -237,7 +235,7 @@
header.ssrc = kSsrc1;
header.arrOfCSRCs[0] = kCsrc1;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
auto rtp_receiver_impl = static_cast<RtpReceiverImpl*>(rtp_receiver_.get());
auto ssrc_sources = rtp_receiver_impl->ssrc_sources_for_testing();
ASSERT_EQ(1u, ssrc_sources.size());
@@ -268,7 +266,7 @@
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
auto sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
@@ -283,7 +281,7 @@
header.extension.audioLevel = 20;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources,
UnorderedElementsAre(
@@ -300,7 +298,7 @@
header.extension.audioLevel = 30;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources,
UnorderedElementsAre(
@@ -321,7 +319,7 @@
AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
auto sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(RtpSource(
time1_ms, kSsrc1, RtpSourceType::SSRC, 10)));
@@ -334,10 +332,141 @@
header.extension.hasAudioLevel = false;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, kTestPayload, sizeof(kTestPayload), payload_specific, !kInOrder));
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
sources = rtp_receiver_->GetSources();
EXPECT_THAT(sources, UnorderedElementsAre(
RtpSource(time2_ms, kSsrc1, RtpSourceType::SSRC)));
}
+TEST_F(RtpReceiverTest, UpdatesTimestampsIfAndOnlyIfPacketArrivesInOrder) {
+ RTPHeader header;
+ int64_t time1_ms = fake_clock_.TimeInMilliseconds();
+ header.payloadType = kPcmuPayloadType;
+ header.ssrc = kSsrc1;
+ header.timestamp = rtp_timestamp(time1_ms);
+ header.extension.hasAudioLevel = true;
+ header.extension.audioLevel = 10;
+ header.sequenceNumber = 0xfff0;
+
+ const PayloadUnion payload_specific{
+ AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
+ uint32_t latest_timestamp;
+ int64_t latest_receive_time_ms;
+
+ // No packet received yet.
+ EXPECT_FALSE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ // Initial packet
+ const uint32_t timestamp_1 = header.timestamp;
+ const int64_t receive_time_1 = fake_clock_.TimeInMilliseconds();
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_1);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_1);
+
+ // Late packet, timestamp not recorded.
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ header.timestamp -= 900;
+ header.sequenceNumber -= 2;
+
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_1);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_1);
+
+ // New packet, still late, no wraparound.
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ header.timestamp += 1800;
+ header.sequenceNumber += 1;
+
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_1);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_1);
+
+ // New packet, new timestamp recorded
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ header.timestamp += 900;
+ header.sequenceNumber += 2;
+ const uint32_t timestamp_2 = header.timestamp;
+ const int64_t receive_time_2 = fake_clock_.TimeInMilliseconds();
+ const uint16_t seqno_2 = header.sequenceNumber;
+
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_2);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_2);
+
+ // New packet, timestamp wraps around
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ header.timestamp += 900;
+ header.sequenceNumber += 20;
+ const uint32_t timestamp_3 = header.timestamp;
+ const int64_t receive_time_3 = fake_clock_.TimeInMilliseconds();
+ EXPECT_LT(header.sequenceNumber, seqno_2); // Wrap-around
+
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_3);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_3);
+}
+
+TEST_F(RtpReceiverTest, UpdatesTimestampsWhenStreamResets) {
+ RTPHeader header;
+ int64_t time1_ms = fake_clock_.TimeInMilliseconds();
+ header.payloadType = kPcmuPayloadType;
+ header.ssrc = kSsrc1;
+ header.timestamp = rtp_timestamp(time1_ms);
+ header.extension.hasAudioLevel = true;
+ header.extension.audioLevel = 10;
+ header.sequenceNumber = 0xfff0;
+
+ const PayloadUnion payload_specific{
+ AudioPayload{SdpAudioFormat("foo", 8000, 1), 0}};
+ uint32_t latest_timestamp;
+ int64_t latest_receive_time_ms;
+
+ // No packet received yet.
+ EXPECT_FALSE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ // Initial packet
+ const uint32_t timestamp_1 = header.timestamp;
+ const int64_t receive_time_1 = fake_clock_.TimeInMilliseconds();
+ const uint16_t seqno_1 = header.sequenceNumber;
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_1);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_1);
+
+ // Packet with far in the past seqno, but unlikely to be a wrap-around.
+ // Treated as a seqno discontinuity, and timestamp is recorded.
+ fake_clock_.AdvanceTimeMilliseconds(10);
+ header.timestamp += 900;
+ header.sequenceNumber = 0x9000;
+
+ const uint32_t timestamp_2 = header.timestamp;
+ const int64_t receive_time_2 = fake_clock_.TimeInMilliseconds();
+ const uint16_t seqno_2 = header.sequenceNumber;
+ EXPECT_LT(seqno_1 - seqno_2, 0x8000); // In the past.
+
+ EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
+ header, kTestPayload, sizeof(kTestPayload), payload_specific));
+ EXPECT_TRUE(rtp_receiver_->GetLatestTimestamps(&latest_timestamp,
+ &latest_receive_time_ms));
+ EXPECT_EQ(latest_timestamp, timestamp_2);
+ EXPECT_EQ(latest_receive_time_ms, receive_time_2);
+}
+
} // namespace webrtc
diff --git a/modules/rtp_rtcp/test/testAPI/test_api.cc b/modules/rtp_rtcp/test/testAPI/test_api.cc
index c247fc7..e2afa42 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api.cc
@@ -58,7 +58,7 @@
const size_t payload_length = len - header.headerLength;
receive_statistics_->IncomingPacket(header, len, false);
return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
- pl->typeSpecific, true);
+ pl->typeSpecific);
}
bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) {
diff --git a/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 26964fd..63b7851 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -171,7 +171,7 @@
const uint8_t* payload = padding_packet + header.headerLength;
const size_t payload_length = packet_size - header.headerLength;
EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
- header, payload, payload_length, pl->typeSpecific, true));
+ header, payload, payload_length, pl->typeSpecific));
EXPECT_EQ(0u, receiver_->payload_size());
EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
}
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 2b1b05e..aecdfe3 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -280,8 +280,7 @@
RTPHeader header;
packet.GetHeader(&header);
- bool in_order = IsPacketInOrder(header);
- ReceivePacket(rtp_packet, rtp_packet_length, header, in_order);
+ ReceivePacket(rtp_packet, rtp_packet_length, header);
}
// TODO(pbos): Remove as soon as audio can handle a changing payload type
@@ -339,7 +338,7 @@
// TODO(nisse): Why isn't this done for recovered packets?
rtp_payload_registry_.SetIncomingPayloadType(header);
}
- ReceivePacket(packet.data(), packet.size(), header, in_order);
+ ReceivePacket(packet.data(), packet.size(), header);
// Update receive statistics after ReceivePacket.
// Receive statistics will be reset if the payload type changes (make sure
// that the first packet is included in the stats).
@@ -443,8 +442,7 @@
void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet,
size_t packet_length,
- const RTPHeader& header,
- bool in_order) {
+ const RTPHeader& header) {
if (rtp_payload_registry_.IsRed(header)) {
ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
return;
@@ -456,7 +454,7 @@
rtp_payload_registry_.PayloadTypeToPayload(header.payloadType);
if (pl) {
rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
- pl->typeSpecific, in_order);
+ pl->typeSpecific);
}
}
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 25ea486..90e8200 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -154,8 +154,7 @@
bool AddReceiveCodec(const VideoCodec& video_codec);
void ReceivePacket(const uint8_t* packet,
size_t packet_length,
- const RTPHeader& header,
- bool in_order);
+ const RTPHeader& header);
// Parses and handles for instance RTX and RED headers.
// This function assumes that it's being called from only one thread.
void ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index c3c06a0..19873f1 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -595,7 +595,8 @@
rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
if (header.payload_type_frequency < 0)
return false;
- return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
+ // TODO(nisse): Pass RtpPacketReceived with |recovered()| true.
+ return ReceivePacket(rtp_packet, rtp_packet_length, header);
}
AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
@@ -1148,14 +1149,13 @@
header, packet.size(), IsPacketRetransmitted(header, in_order));
rtp_payload_registry_->SetIncomingPayloadType(header);
- ReceivePacket(packet.data(), packet.size(), header, in_order);
+ ReceivePacket(packet.data(), packet.size(), header);
}
}
bool Channel::ReceivePacket(const uint8_t* packet,
size_t packet_length,
- const RTPHeader& header,
- bool in_order) {
+ const RTPHeader& header) {
const uint8_t* payload = packet + header.headerLength;
assert(packet_length >= header.headerLength);
size_t payload_length = packet_length - header.headerLength;
@@ -1165,7 +1165,7 @@
return false;
}
return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
- pl->typeSpecific, in_order);
+ pl->typeSpecific);
}
bool Channel::IsPacketInOrder(const RTPHeader& header) const {
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 9221115..9a79b2b 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -341,8 +341,7 @@
bool ReceivePacket(const uint8_t* packet,
size_t packet_length,
- const RTPHeader& header,
- bool in_order);
+ const RTPHeader& header);
bool IsPacketInOrder(const RTPHeader& header) const;
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
int ResendPackets(const uint16_t* sequence_numbers, int length);
