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

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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}
This commit is contained in:
Chen Xing
2019-07-01 17:16:32 +02:00
committed by Commit Bot
parent 62eb89d221
commit 3e8ef940fe
23 changed files with 195 additions and 26 deletions
Download Patch File Download Diff File Expand all files Collapse all files

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

@ -9,6 +9,8 @@
*/
#include <memory>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
@ -30,6 +32,7 @@
#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"
@ -40,14 +43,17 @@
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 {
@ -62,12 +68,12 @@ int DeletePacketsAndReturnOk(PacketList* packet_list) {
class NetEqImplTest : public ::testing::Test {
protected:
NetEqImplTest() { config_.sample_rate_hz = 8000; }
NetEqImplTest() : clock_(0) { config_.sample_rate_hz = 8000; }
void CreateInstance(
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
ASSERT_TRUE(decoder_factory);
NetEqImpl::Dependencies deps(config_, decoder_factory);
NetEqImpl::Dependencies deps(config_, &clock_, decoder_factory);
// Get a local pointer to NetEq's TickTimer object.
tick_timer_ = deps.tick_timer.get();
@ -217,6 +223,10 @@ 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, -1578, -2816, -3460, -3403, -2709, -1594,
@ -231,6 +241,7 @@ 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;
@ -263,7 +274,9 @@ class NetEqImplTest : public ::testing::Test {
// TODO(hlundin): Move to separate file?
TEST(NetEq, CreateAndDestroy) {
NetEq::Config config;
NetEq* neteq = NetEq::Create(config, CreateBuiltinAudioDecoderFactory());
SimulatedClock clock(0);
NetEq* neteq =
NetEq::Create(config, &clock, CreateBuiltinAudioDecoderFactory());
delete neteq;
}
@ -458,6 +471,10 @@ 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
@ -501,6 +518,8 @@ 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));
@ -514,6 +533,17 @@ 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));
@ -561,6 +591,8 @@ 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())
@ -583,6 +615,8 @@ 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));
@ -595,16 +629,32 @@ 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));
@ -627,6 +677,17 @@ 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());
}
@ -663,6 +724,7 @@ 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,
@ -685,6 +747,7 @@ 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));
}
}
@ -722,6 +785,7 @@ 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.
@ -741,6 +805,7 @@ 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();
@ -975,12 +1040,14 @@ 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.
@ -1082,6 +1149,7 @@ 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());
}
@ -1178,6 +1246,7 @@ 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));
@ -1191,12 +1260,14 @@ 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());
}
@ -1625,4 +1696,4 @@ TEST_F(NetEqImplTest120ms, Accelerate) {
EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
}
}// namespace webrtc
} // namespace webrtc