Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f.
Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260.
Original change's description:
> Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.
>
> This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time.
>
> Bug: webrtc:10668
> Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Commit-Queue: Chen Xing <chxg@google.com>
> Cr-Commit-Position: refs/heads/master@{#28434}
TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com
Bug: webrtc:10668, chromium:982260
Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28561}
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index ded54bf..517f4ac 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -11,8 +11,6 @@
#include "modules/audio_coding/neteq/neteq_impl.h"
#include <memory>
-#include <utility>
-#include <vector>
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -33,7 +31,6 @@
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/timestamp_scaler.h"
#include "rtc_base/numerics/safe_conversions.h"
-#include "system_wrappers/include/clock.h"
#include "test/audio_decoder_proxy_factory.h"
#include "test/function_audio_decoder_factory.h"
#include "test/gmock.h"
@@ -44,17 +41,14 @@
using ::testing::_;
using ::testing::AtLeast;
using ::testing::DoAll;
-using ::testing::ElementsAre;
using ::testing::InSequence;
using ::testing::Invoke;
-using ::testing::IsEmpty;
using ::testing::IsNull;
using ::testing::Pointee;
using ::testing::Return;
using ::testing::ReturnNull;
using ::testing::SetArgPointee;
using ::testing::SetArrayArgument;
-using ::testing::SizeIs;
using ::testing::WithArg;
namespace webrtc {
@@ -69,12 +63,12 @@
class NetEqImplTest : public ::testing::Test {
protected:
- NetEqImplTest() : clock_(0) { config_.sample_rate_hz = 8000; }
+ NetEqImplTest() { config_.sample_rate_hz = 8000; }
void CreateInstance(
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
ASSERT_TRUE(decoder_factory);
- NetEqImpl::Dependencies deps(config_, &clock_, decoder_factory);
+ NetEqImpl::Dependencies deps(config_, decoder_factory);
// Get a local pointer to NetEq's TickTimer object.
tick_timer_ = deps.tick_timer.get();
@@ -224,10 +218,6 @@
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- // DTMF packets are immediately consumed by |InsertPacket()| and won't be
- // returned by |GetAudio()|.
- EXPECT_THAT(output.packet_infos_, IsEmpty());
-
// Verify first 64 samples of actual output.
const std::vector<int16_t> kOutput(
{0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
@@ -243,7 +233,6 @@
std::unique_ptr<NetEqImpl> neteq_;
NetEq::Config config_;
- SimulatedClock clock_;
TickTimer* tick_timer_ = nullptr;
MockBufferLevelFilter* mock_buffer_level_filter_ = nullptr;
BufferLevelFilter* buffer_level_filter_ = nullptr;
@@ -275,9 +264,7 @@
// TODO(hlundin): Move to separate file?
TEST(NetEq, CreateAndDestroy) {
NetEq::Config config;
- SimulatedClock clock(0);
- NetEq* neteq =
- NetEq::Create(config, &clock, CreateBuiltinAudioDecoderFactory());
+ NetEq* neteq = NetEq::Create(config, CreateBuiltinAudioDecoderFactory());
delete neteq;
}
@@ -469,10 +456,6 @@
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
- rtp_header.numCSRCs = 3;
- rtp_header.arrOfCSRCs[0] = 43;
- rtp_header.arrOfCSRCs[1] = 65;
- rtp_header.arrOfCSRCs[2] = 17;
// This is a dummy decoder that produces as many output samples as the input
// has bytes. The output is an increasing series, starting at 1 for the first
@@ -516,8 +499,6 @@
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
- clock_.AdvanceTimeMilliseconds(123456);
- int64_t expected_receive_time_ms = clock_.TimeInMilliseconds();
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
@@ -531,17 +512,6 @@
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- // Verify |output.packet_infos_|.
- ASSERT_THAT(output.packet_infos_, SizeIs(1));
- {
- const auto& packet_info = output.packet_infos_[0];
- EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
- EXPECT_THAT(packet_info.csrcs(), ElementsAre(43, 65, 17));
- EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
- EXPECT_FALSE(packet_info.audio_level().has_value());
- EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms);
- }
-
// Start with a simple check that the fake decoder is behaving as expected.
EXPECT_EQ(kPayloadLengthSamples,
static_cast<size_t>(decoder_.next_value() - 1));
@@ -589,8 +559,6 @@
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
- rtp_header.extension.hasAudioLevel = true;
- rtp_header.extension.audioLevel = 42;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, SampleRateHz())
@@ -613,8 +581,6 @@
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
- clock_.AdvanceTimeMilliseconds(123456);
- int64_t expected_receive_time_ms = clock_.TimeInMilliseconds();
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
@@ -627,32 +593,16 @@
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- // Verify |output.packet_infos_|.
- ASSERT_THAT(output.packet_infos_, SizeIs(1));
- {
- const auto& packet_info = output.packet_infos_[0];
- EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
- EXPECT_THAT(packet_info.csrcs(), IsEmpty());
- EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
- EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
- EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms);
- }
-
// Insert two more packets. The first one is out of order, and is already too
// old, the second one is the expected next packet.
rtp_header.sequenceNumber -= 1;
rtp_header.timestamp -= kPayloadLengthSamples;
- rtp_header.extension.audioLevel = 1;
payload[0] = 1;
- clock_.AdvanceTimeMilliseconds(1000);
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
- rtp_header.extension.audioLevel = 2;
payload[0] = 2;
- clock_.AdvanceTimeMilliseconds(2000);
- expected_receive_time_ms = clock_.TimeInMilliseconds();
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
@@ -675,17 +625,6 @@
// out-of-order packet should have been discarded.
EXPECT_TRUE(packet_buffer_->Empty());
- // Verify |output.packet_infos_|. Expect to only see the second packet.
- ASSERT_THAT(output.packet_infos_, SizeIs(1));
- {
- const auto& packet_info = output.packet_infos_[0];
- EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
- EXPECT_THAT(packet_info.csrcs(), IsEmpty());
- EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
- EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
- EXPECT_EQ(packet_info.receive_time_ms(), expected_receive_time_ms);
- }
-
EXPECT_CALL(mock_decoder, Die());
}
@@ -722,7 +661,6 @@
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
- EXPECT_THAT(output.packet_infos_, IsEmpty());
// Register the payload type.
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
@@ -745,7 +683,6 @@
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
<< "NetEq did not decode the packets as expected.";
- EXPECT_THAT(output.packet_infos_, SizeIs(1));
}
}
@@ -783,7 +720,6 @@
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
- EXPECT_THAT(output.packet_infos_, IsEmpty());
}
// Insert 10 packets.
@@ -803,7 +739,6 @@
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
<< "NetEq did not decode the packets as expected.";
- EXPECT_THAT(output.packet_infos_, SizeIs(1));
}
auto lifetime_stats = neteq_->GetLifetimeStatistics();
@@ -1036,14 +971,12 @@
const size_t kExpectedOutputSize = 10 * (kSampleRateHz / 1000) * kChannels;
EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
EXPECT_EQ(kChannels, output.num_channels_);
- EXPECT_THAT(output.packet_infos_, IsEmpty());
// Second call to GetAudio will decode the packet that is ok. No errors are
// expected.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
EXPECT_EQ(kChannels, output.num_channels_);
- EXPECT_THAT(output.packet_infos_, SizeIs(1));
// Die isn't called through NiceMock (since it's called by the
// MockAudioDecoder constructor), so it needs to be mocked explicitly.
@@ -1145,7 +1078,6 @@
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- EXPECT_THAT(output.packet_infos_, SizeIs(1));
EXPECT_CALL(mock_decoder, Die());
}
@@ -1240,7 +1172,6 @@
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- EXPECT_THAT(output.packet_infos_, SizeIs(2)); // 5 ms packets vs 10 ms output
// Pull audio again. Decoder fails.
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
@@ -1254,14 +1185,12 @@
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
- EXPECT_THAT(output.packet_infos_, IsEmpty());
// Pull audio again, should behave normal.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
- EXPECT_THAT(output.packet_infos_, SizeIs(2)); // 5 ms packets vs 10 ms output
EXPECT_CALL(mock_decoder, Die());
}
@@ -1689,4 +1618,4 @@
EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
}
-} // namespace webrtc
+}// namespace webrtc