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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
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1
api/audio/BUILD.gn
View File

@ -16,6 +16,7 @@ rtc_source_set("audio_frame_api") {
]
deps = [
"..:rtp_packet_info",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]

2
api/audio/audio_frame.cc
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@ -39,6 +39,7 @@ void AudioFrame::ResetWithoutMuting() {
speech_type_ = kUndefined;
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
packet_infos_ = RtpPacketInfos();
}
void AudioFrame::UpdateFrame(uint32_t timestamp,
@ -72,6 +73,7 @@ void AudioFrame::CopyFrom(const AudioFrame& src) {
timestamp_ = src.timestamp_;
elapsed_time_ms_ = src.elapsed_time_ms_;
ntp_time_ms_ = src.ntp_time_ms_;
packet_infos_ = src.packet_infos_;
muted_ = src.muted();
samples_per_channel_ = src.samples_per_channel_;
sample_rate_hz_ = src.sample_rate_hz_;

17
api/audio/audio_frame.h
View File

@ -14,6 +14,7 @@
#include <stddef.h>
#include <stdint.h>
#include "api/rtp_packet_infos.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -115,6 +116,22 @@ class AudioFrame {
// class/struct needs an explicit out-of-line destructor" build error.
int64_t profile_timestamp_ms_ = 0;
// Information about packets used to assemble this audio frame. This is needed
// by |SourceTracker| when the frame is delivered to the RTCRtpReceiver's
// MediaStreamTrack, in order to implement getContributingSources(). See:
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getcontributingsources
//
// TODO(bugs.webrtc.org/10757):
// Note that this information might not be fully accurate since we currently
// don't have a proper way to track it across the audio sync buffer. The
// sync buffer is the small sample-holding buffer located after the audio
// decoder and before where samples are assembled into output frames.
//
// |RtpPacketInfos| may also be empty if the audio samples did not come from
// RTP packets. E.g. if the audio were locally generated by packet loss
// concealment, comfort noise generation, etc.
RtpPacketInfos packet_infos_;
private:
// A permamently zeroed out buffer to represent muted frames. This is a
// header-only class, so the only way to avoid creating a separate empty

1
audio/remix_resample.cc
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@ -27,6 +27,7 @@ void RemixAndResample(const AudioFrame& src_frame,
dst_frame->timestamp_ = src_frame.timestamp_;
dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
dst_frame->packet_infos_ = src_frame.packet_infos_;
}
void RemixAndResample(const int16_t* src_data,

4
modules/audio_coding/BUILD.gn
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@ -1017,6 +1017,7 @@ rtc_static_library("neteq") {
"..:module_api_public",
"../../api:array_view",
"../../api:rtp_headers",
"../../api:rtp_packet_info",
"../../api:scoped_refptr",
"../../api/audio:audio_frame_api",
"../../api/audio_codecs:audio_codecs_api",
@ -1029,6 +1030,7 @@ rtc_static_library("neteq") {
"../../rtc_base:safe_minmax",
"../../rtc_base:sanitizer",
"../../rtc_base/system:fallthrough",
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/memory",
@ -1066,6 +1068,7 @@ rtc_source_set("neteq_tools_minimal") {
"../../api/audio_codecs:audio_codecs_api",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
"//third_party/abseil-cpp/absl/types:optional",
@ -1591,6 +1594,7 @@ if (rtc_include_tests) {
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
"//testing/gtest",

4
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -34,7 +34,9 @@ namespace acm2 {
AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
: last_audio_buffer_(new int16_t[AudioFrame::kMaxDataSizeSamples]),
neteq_(NetEq::Create(config.neteq_config, config.decoder_factory)),
neteq_(NetEq::Create(config.neteq_config,
config.clock,
config.decoder_factory)),
clock_(config.clock),
resampled_last_output_frame_(true) {
RTC_DCHECK(clock_);

2
modules/audio_coding/neteq/include/neteq.h
View File

@ -31,6 +31,7 @@ namespace webrtc {
// Forward declarations.
class AudioFrame;
class AudioDecoderFactory;
class Clock;
struct NetEqNetworkStatistics {
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
@ -149,6 +150,7 @@ class NetEq {
// method.
static NetEq* Create(
const NetEq::Config& config,
Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
virtual ~NetEq() {}

3
modules/audio_coding/neteq/neteq.cc
View File

@ -39,9 +39,10 @@ std::string NetEq::Config::ToString() const {
// Return the new object.
NetEq* NetEq::Create(
const NetEq::Config& config,
Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
return new NetEqImpl(config,
NetEqImpl::Dependencies(config, decoder_factory));
NetEqImpl::Dependencies(config, clock, decoder_factory));
}
} // namespace webrtc

38
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -15,6 +15,7 @@
#include <cstdint>
#include <cstring>
#include <list>
#include <map>
#include <utility>
#include <vector>
@ -52,13 +53,16 @@
#include "rtc_base/sanitizer.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
NetEqImpl::Dependencies::Dependencies(
const NetEq::Config& config,
Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
: tick_timer(new TickTimer),
: clock(clock),
tick_timer(new TickTimer),
stats(new StatisticsCalculator),
buffer_level_filter(new BufferLevelFilter),
decoder_database(
@ -86,7 +90,8 @@ NetEqImpl::Dependencies::~Dependencies() = default;
NetEqImpl::NetEqImpl(const NetEq::Config& config,
Dependencies&& deps,
bool create_components)
: tick_timer_(std::move(deps.tick_timer)),
: clock_(deps.clock),
tick_timer_(std::move(deps.tick_timer)),
buffer_level_filter_(std::move(deps.buffer_level_filter)),
decoder_database_(std::move(deps.decoder_database)),
delay_manager_(std::move(deps.delay_manager)),
@ -468,17 +473,20 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
RTC_LOG_F(LS_ERROR) << "payload is empty";
return kInvalidPointer;
}
int64_t receive_time_ms = clock_->TimeInMilliseconds();
stats_->ReceivedPacket();
PacketList packet_list;
// Insert packet in a packet list.
packet_list.push_back([&rtp_header, &payload] {
packet_list.push_back([&rtp_header, &payload, &receive_time_ms] {
// Convert to Packet.
Packet packet;
packet.payload_type = rtp_header.payloadType;
packet.sequence_number = rtp_header.sequenceNumber;
packet.timestamp = rtp_header.timestamp;
packet.payload.SetData(payload.data(), payload.size());
packet.packet_info = RtpPacketInfo(rtp_header, receive_time_ms);
// Waiting time will be set upon inserting the packet in the buffer.
RTC_DCHECK(!packet.waiting_time);
return packet;
@ -611,6 +619,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
const auto sequence_number = packet.sequence_number;
const auto payload_type = packet.payload_type;
const Packet::Priority original_priority = packet.priority;
const auto& packet_info = packet.packet_info;
auto packet_from_result = [&](AudioDecoder::ParseResult& result) {
Packet new_packet;
new_packet.sequence_number = sequence_number;
@ -618,6 +627,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
new_packet.timestamp = result.timestamp;
new_packet.priority.codec_level = result.priority;
new_packet.priority.red_level = original_priority.red_level;
new_packet.packet_info = packet_info;
new_packet.frame = std::move(result.frame);
return new_packet;
};
@ -879,7 +889,16 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
comfort_noise_->Reset();
}
// Copy from |algorithm_buffer| to |sync_buffer_|.
// We treat it as if all packets referenced to by |last_decoded_packet_infos_|
// were mashed together when creating the samples in |algorithm_buffer_|.
RtpPacketInfos packet_infos(std::move(last_decoded_packet_infos_));
last_decoded_packet_infos_.clear();
// Copy samples from |algorithm_buffer_| to |sync_buffer_|.
//
// TODO(bugs.webrtc.org/10757):
// We would in the future also like to pass |packet_infos| so that we can do
// sample-perfect tracking of that information across |sync_buffer_|.
sync_buffer_->PushBack(*algorithm_buffer_);
// Extract data from |sync_buffer_| to |output|.
@ -897,6 +916,13 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
audio_frame);
audio_frame->sample_rate_hz_ = fs_hz_;
// TODO(bugs.webrtc.org/10757):
// We don't have the ability to properly track individual packets once their
// audio samples have entered |sync_buffer_|. So for now, treat it as if
// |packet_infos| from packets decoded by the current |GetAudioInternal()|
// call were all consumed assembling the current audio frame and the current
// audio frame only.
audio_frame->packet_infos_ = std::move(packet_infos);
if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
// The sync buffer should always contain |overlap_length| samples, but now
// too many samples have been extracted. Reinstall the |overlap_length|
@ -1392,6 +1418,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
int* decoded_length,
AudioDecoder::SpeechType* speech_type) {
RTC_DCHECK(last_decoded_timestamps_.empty());
RTC_DCHECK(last_decoded_packet_infos_.empty());
// Do decoding.
while (!packet_list->empty() && !decoder_database_->IsComfortNoise(
@ -1409,6 +1436,8 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
decoded_buffer_length_ - *decoded_length));
last_decoded_timestamps_.push_back(packet_list->front().timestamp);
last_decoded_packet_infos_.push_back(
std::move(packet_list->front().packet_info));
packet_list->pop_front();
if (opt_result) {
const auto& result = *opt_result;
@ -1424,6 +1453,7 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list,
// TODO(ossu): What to put here?
RTC_LOG(LS_WARNING) << "Decode error";
*decoded_length = -1;
last_decoded_packet_infos_.clear();
packet_list->clear();
break;
}

13
modules/audio_coding/neteq/neteq_impl.h
View File

@ -11,11 +11,15 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
#include <map>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
#include "api/rtp_packet_info.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h" // Modes, Operations
#include "modules/audio_coding/neteq/expand_uma_logger.h"
@ -34,6 +38,7 @@ namespace webrtc {
class Accelerate;
class BackgroundNoise;
class BufferLevelFilter;
class Clock;
class ComfortNoise;
class DecisionLogic;
class DecoderDatabase;
@ -93,11 +98,13 @@ class NetEqImpl : public webrtc::NetEq {
// before sending the struct to the NetEqImpl constructor. However, there
// are dependencies between some of the classes inside the struct, so
// swapping out one may make it necessary to re-create another one.
explicit Dependencies(
Dependencies(
const NetEq::Config& config,
Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
~Dependencies();
Clock* const clock;
std::unique_ptr<TickTimer> tick_timer;
std::unique_ptr<StatisticsCalculator> stats;
std::unique_ptr<BufferLevelFilter> buffer_level_filter;
@ -338,6 +345,8 @@ class NetEqImpl : public webrtc::NetEq {
// Creates DecisionLogic object with the mode given by |playout_mode_|.
virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
Clock* const clock_;
rtc::CriticalSection crit_sect_;
const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
@ -403,6 +412,8 @@ class NetEqImpl : public webrtc::NetEq {
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
RTC_GUARDED_BY(crit_sect_);
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
std::vector<RtpPacketInfo> last_decoded_packet_infos_
RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_); // Only used for test.

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

4
modules/audio_coding/neteq/neteq_network_stats_unittest.cc
View File

@ -17,6 +17,7 @@
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/ref_counted_object.h"
#include "system_wrappers/include/clock.h"
#include "test/audio_decoder_proxy_factory.h"
#include "test/gmock.h"
@ -163,7 +164,8 @@ class NetEqNetworkStatsTest {
packet_loss_interval_(0xffffffff) {
NetEq::Config config;
config.sample_rate_hz = format.clockrate_hz;
neteq_ = absl::WrapUnique(NetEq::Create(config, decoder_factory_));
neteq_ = absl::WrapUnique(
NetEq::Create(config, Clock::GetRealTimeClock(), decoder_factory_));
neteq_->RegisterPayloadType(kPayloadType, format);
}

8
modules/audio_coding/neteq/neteq_stereo_unittest.cc
View File

@ -22,6 +22,7 @@
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -57,6 +58,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
frame_size_samples_(
static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
output_size_samples_(10 * samples_per_ms_),
clock_(0),
rtp_generator_mono_(samples_per_ms_),
rtp_generator_(samples_per_ms_),
payload_size_bytes_(0),
@ -67,8 +69,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
config.sample_rate_hz = sample_rate_hz_;
rtc::scoped_refptr<AudioDecoderFactory> factory =
CreateBuiltinAudioDecoderFactory();
neteq_mono_ = NetEq::Create(config, factory);
neteq_ = NetEq::Create(config, factory);
neteq_mono_ = NetEq::Create(config, &clock_, factory);
neteq_ = NetEq::Create(config, &clock_, factory);
input_ = new int16_t[frame_size_samples_];
encoded_ = new uint8_t[2 * frame_size_samples_];
input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
@ -196,6 +198,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
time_now += kTimeStepMs;
clock_.AdvanceTimeMilliseconds(kTimeStepMs);
}
}
@ -205,6 +208,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
const int frame_size_ms_;
const size_t frame_size_samples_;
const size_t output_size_samples_;
SimulatedClock clock_;
NetEq* neteq_mono_;
NetEq* neteq_;
test::RtpGenerator rtp_generator_mono_;

18
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -36,6 +36,7 @@
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/arch.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -288,11 +289,11 @@ class NetEqDecodingTest : public ::testing::Test {
void DuplicateCng();
SimulatedClock clock_;
NetEq* neteq_;
NetEq::Config config_;
std::unique_ptr<test::RtpFileSource> rtp_source_;
std::unique_ptr<test::Packet> packet_;
unsigned int sim_clock_;
AudioFrame out_frame_;
int output_sample_rate_;
int algorithmic_delay_ms_;
@ -306,16 +307,16 @@ const size_t NetEqDecodingTest::kBlockSize32kHz;
const int NetEqDecodingTest::kInitSampleRateHz;
NetEqDecodingTest::NetEqDecodingTest()
: neteq_(NULL),
: clock_(0),
neteq_(NULL),
config_(),
sim_clock_(0),
output_sample_rate_(kInitSampleRateHz),
algorithmic_delay_ms_(0) {
config_.sample_rate_hz = kInitSampleRateHz;
}
void NetEqDecodingTest::SetUp() {
neteq_ = NetEq::Create(config_, CreateBuiltinAudioDecoderFactory());
neteq_ = NetEq::Create(config_, &clock_, CreateBuiltinAudioDecoderFactory());
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;
@ -333,7 +334,7 @@ void NetEqDecodingTest::OpenInputFile(const std::string& rtp_file) {
void NetEqDecodingTest::Process() {
// Check if time to receive.
while (packet_ && sim_clock_ >= packet_->time_ms()) {
while (packet_ && clock_.TimeInMilliseconds() >= packet_->time_ms()) {
if (packet_->payload_length_bytes() > 0) {
#ifndef WEBRTC_CODEC_ISAC
// Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
@ -363,7 +364,7 @@ void NetEqDecodingTest::Process() {
EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
// Increase time.
sim_clock_ += kTimeStepMs;
clock_.AdvanceTimeMilliseconds(kTimeStepMs);
}
void NetEqDecodingTest::DecodeAndCompare(
@ -394,7 +395,7 @@ void NetEqDecodingTest::DecodeAndCompare(
output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
if (clock_.TimeInMilliseconds() % 1000 == 0) {
// Process NetworkStatistics.
NetEqNetworkStatistics current_network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
@ -1435,7 +1436,8 @@ class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
}
void CreateSecondInstance() {
neteq2_.reset(NetEq::Create(config2_, CreateBuiltinAudioDecoderFactory()));
neteq2_.reset(
NetEq::Create(config2_, &clock_, CreateBuiltinAudioDecoderFactory()));
ASSERT_TRUE(neteq2_);
LoadDecoders(neteq2_.get());
}

1
modules/audio_coding/neteq/packet.cc
View File

@ -28,6 +28,7 @@ Packet Packet::Clone() const {
clone.payload_type = payload_type;
clone.payload.SetData(payload.data(), payload.size());
clone.priority = priority;
clone.packet_info = packet_info;
return clone;
}

2
modules/audio_coding/neteq/packet.h
View File

@ -16,6 +16,7 @@
#include <memory>
#include "api/audio_codecs/audio_decoder.h"
#include "api/rtp_packet_info.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/buffer.h"
#include "rtc_base/checks.h"
@ -72,6 +73,7 @@ struct Packet {
// Datagram excluding RTP header and header extension.
rtc::Buffer payload;
Priority priority;
RtpPacketInfo packet_info;
std::unique_ptr<TickTimer::Stopwatch> waiting_time;
std::unique_ptr<AudioDecoder::EncodedAudioFrame> frame;

6
modules/audio_coding/neteq/red_payload_splitter.cc
View File

@ -117,6 +117,12 @@ bool RedPayloadSplitter::SplitRed(PacketList* packet_list) {
new_packet.priority.red_level =
rtc::dchecked_cast<int>((new_headers.size() - 1) - i);
new_packet.payload.SetData(payload_ptr, payload_length);
new_packet.packet_info = RtpPacketInfo(
/*ssrc=*/red_packet.packet_info.ssrc(),
/*csrcs=*/std::vector<uint32_t>(),
/*rtp_timestamp=*/new_packet.timestamp,
/*audio_level=*/absl::nullopt,
/*receive_time_ms=*/red_packet.packet_info.receive_time_ms());
new_packets.push_front(std::move(new_packet));
payload_ptr += payload_length;
}

5
modules/audio_coding/neteq/tools/neteq_performance_test.cc
View File

@ -39,7 +39,9 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
// Initialize NetEq instance.
NetEq::Config config;
config.sample_rate_hz = kSampRateHz;
NetEq* neteq = NetEq::Create(config, CreateBuiltinAudioDecoderFactory());
webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
NetEq* neteq =
NetEq::Create(config, clock, CreateBuiltinAudioDecoderFactory());
// Register decoder in |neteq|.
if (!neteq->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", kSampRateHz, 1)))
@ -72,7 +74,6 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
RTC_CHECK_EQ(sizeof(input_payload), payload_len);
// Main loop.
webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
int64_t start_time_ms = clock->TimeInMilliseconds();
AudioFrame out_frame;
while (time_now_ms < runtime_ms) {

4
modules/audio_coding/neteq/tools/neteq_quality_test.cc
View File

@ -16,6 +16,7 @@
#include "modules/audio_coding/neteq/tools/output_wav_file.h"
#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include "rtc_base/checks.h"
#include "system_wrappers/include/clock.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
@ -213,7 +214,8 @@ NetEqQualityTest::NetEqQualityTest(
NetEq::Config config;
config.sample_rate_hz = out_sampling_khz_ * 1000;
neteq_.reset(NetEq::Create(config, decoder_factory));
neteq_.reset(
NetEq::Create(config, Clock::GetRealTimeClock(), decoder_factory));
max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
in_data_.reset(new int16_t[in_size_samples_ * channels_]);
}

1
modules/audio_coding/neteq/tools/neteq_quality_test.h
View File

@ -20,6 +20,7 @@
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/flags.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
namespace webrtc {

5
modules/audio_coding/neteq/tools/neteq_test.cc
View File

@ -14,6 +14,7 @@
#include <iostream>
#include "modules/rtp_rtcp/source/byte_io.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace test {
@ -57,7 +58,8 @@ NetEqTest::NetEqTest(const NetEq::Config& config,
std::unique_ptr<NetEqInput> input,
std::unique_ptr<AudioSink> output,
Callbacks callbacks)
: neteq_(NetEq::Create(config, decoder_factory)),
: clock_(0),
neteq_(NetEq::Create(config, &clock_, decoder_factory)),
input_(std::move(input)),
output_(std::move(output)),
callbacks_(callbacks),
@ -92,6 +94,7 @@ NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
while (!input_->ended()) {
// Advance time to next event.
RTC_DCHECK(input_->NextEventTime());
clock_.AdvanceTimeMilliseconds(*input_->NextEventTime() - time_now_ms);
time_now_ms = *input_->NextEventTime();
// Check if it is time to insert packet.
if (input_->NextPacketTime() && time_now_ms >= *input_->NextPacketTime()) {

2
modules/audio_coding/neteq/tools/neteq_test.h
View File

@ -23,6 +23,7 @@
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace test {
@ -106,6 +107,7 @@ class NetEqTest : public NetEqSimulator {
private:
void RegisterDecoders(const DecoderMap& codecs);
SimulatedClock clock_;
absl::optional<Action> next_action_;
absl::optional<int> last_packet_time_ms_;
std::unique_ptr<NetEq> neteq_;

1
modules/audio_mixer/frame_combiner.cc
View File

@ -57,6 +57,7 @@ void SetAudioFrameFields(const std::vector<AudioFrame*>& mix_list,
audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
audio_frame_for_mixing->ntp_time_ms_ = mix_list[0]->ntp_time_ms_;
audio_frame_for_mixing->packet_infos_ = mix_list[0]->packet_infos_;
}
}