zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."

Browse Source 
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}
This commit is contained in:
Ivo Creusen
2019-07-12 17:00:25 +02:00
committed by Commit Bot
parent e95b57cdfc
commit 24192c267a
23 changed files with 26 additions and 195 deletions
Download Patch File Download Diff File Expand all files Collapse all files

79
modules/audio_coding/neteq/neteq_impl_unittest.cc
View File

@ -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 @@ int DeletePacketsAndReturnOk(PacketList* packet_list) {
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 @@ class NetEqImplTest : public ::testing::Test {
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 @@ class NetEqImplTest : public ::testing::Test {
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 @@ class NetEqImplTest : public ::testing::Test {
// 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 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
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 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
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 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
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 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
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 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
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 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
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 @@ TEST_F(NetEqImplTest, ReorderedPacket) {
// 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 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
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 @@ TEST_F(NetEqImplTest, FirstPacketUnknown) {
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 @@ TEST_F(NetEqImplTest, NoAudioInterruptionLoggedBeforeFirstDecode) {
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 @@ TEST_F(NetEqImplTest, NoAudioInterruptionLoggedBeforeFirstDecode) {
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 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
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 @@ TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
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 @@ TEST_F(NetEqImplTest, DecodingError) {
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 @@ TEST_F(NetEqImplTest, DecodingError) {
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 @@ TEST_F(NetEqImplTest120ms, Accelerate) {
EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
}
} // namespace webrtc
}// namespace webrtc