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

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This reverts commit fab3460a821abe336ab610c6d6dfc0d392dac263.

Reason for revert: fix downstream instead

Original change's description:
> Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
> 
> This reverts commit 9973933d2e606d64fcdc753acb9ba3afd6e30569.
> 
> Reason for revert: breaking downstream projects and not reviewed by direct owners
> 
> Original change's description:
> > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
> > 
> > This reverts commit 24192c267a40eb7d6b1850489ccdbf7a84f8ff0f.
> > 
> > Reason for revert: Analyzed the performance regression in more detail.
> > 
> > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit 1796a820f60cb9429bf4bcf13a40a41794ac8fb0 has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac.
> > 
> > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall.
> > 
> > Original change's description:
> > > 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}
> > 
> > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
> > 
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> > 
> > Bug: webrtc:10668, chromium:982260
> > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Chen Xing <chxg@google.com>
> > Cr-Commit-Position: refs/heads/master@{#28664}
> 
> TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com
> 
> Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10668, chromium:982260
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28671}

TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com

Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10668, chromium:982260
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28672}
This commit is contained in:
Alessio Bazzica
2019-07-24 16:47:02 +00:00
committed by Commit Bot
parent fab3460a82
commit 8f319a3472
23 changed files with 196 additions and 26 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
api/audio/BUILD.gn
View File

@ -18,6 +18,7 @@ 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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@ -40,6 +40,7 @@ void AudioFrame::ResetWithoutMuting() {
speech_type_ = kUndefined;
vad_activity_ = kVadUnknown;
profile_timestamp_ms_ = 0;
packet_infos_ = RtpPacketInfos();
}
void AudioFrame::UpdateFrame(uint32_t timestamp,
@ -77,6 +78,7 @@ 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

@ -15,6 +15,7 @@
#include <stdint.h>
#include "api/audio/channel_layout.h"
#include "api/rtp_packet_infos.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -123,6 +124,22 @@ 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 permanently 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",
@ -1030,6 +1031,7 @@ rtc_static_library("neteq") {
"../../rtc_base:sanitizer",
"../../rtc_base/experiments:field_trial_parser",
"../../rtc_base/system:fallthrough",
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"//third_party/abseil-cpp/absl/memory",
@ -1067,6 +1069,7 @@ 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",
@ -1593,6 +1596,7 @@ if (rtc_include_tests) {
":neteq_test_tools",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../rtc_base:checks",
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
"//testing/gtest",

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

@ -35,7 +35,9 @@ 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

39
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -16,6 +16,7 @@
#include <cstdint>
#include <cstring>
#include <list>
#include <map>
#include <utility>
#include <vector>
@ -53,13 +54,16 @@
#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(
@ -87,7 +91,8 @@ 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)),
@ -469,17 +474,20 @@ 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;
@ -612,6 +620,7 @@ 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;
@ -619,6 +628,7 @@ 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;
};
@ -755,6 +765,7 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
bool play_dtmf;
*muted = false;
last_decoded_timestamps_.clear();
last_decoded_packet_infos_.clear();
tick_timer_->Increment();
stats_->IncreaseCounter(output_size_samples_, fs_hz_);
const auto lifetime_stats = stats_->GetLifetimeStatistics();
@ -880,7 +891,16 @@ 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|.
@ -898,6 +918,13 @@ 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|
@ -1393,6 +1420,7 @@ 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(
@ -1410,6 +1438,8 @@ 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;
@ -1425,6 +1455,7 @@ 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;
@ -87,11 +92,13 @@ 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;
@ -332,6 +339,8 @@ 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_
@ -397,6 +406,8 @@ 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

@ -11,6 +11,8 @@
#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"
@ -31,6 +33,7 @@
#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"
@ -41,14 +44,17 @@
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 {
@ -63,12 +69,12 @@ 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();
@ -218,6 +224,10 @@ 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,
@ -233,6 +243,7 @@ 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;
@ -264,7 +275,9 @@ 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;
}
@ -456,6 +469,10 @@ 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
@ -499,6 +516,8 @@ 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));
@ -512,6 +531,17 @@ 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));
@ -559,6 +589,8 @@ 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())
@ -581,6 +613,8 @@ 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));
@ -593,16 +627,32 @@ 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));
@ -625,6 +675,17 @@ 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());
}
@ -661,6 +722,7 @@ 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,
@ -683,6 +745,7 @@ 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));
}
}
@ -720,6 +783,7 @@ 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.
@ -739,6 +803,7 @@ 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();
@ -971,12 +1036,14 @@ 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.
@ -1078,6 +1145,7 @@ 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());
}
@ -1172,6 +1240,7 @@ 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));
@ -1185,12 +1254,14 @@ 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());
}
@ -1618,4 +1689,4 @@ 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

@ -16,6 +16,7 @@
#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"
@ -162,7 +163,8 @@ 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

@ -37,6 +37,7 @@
#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"
@ -286,11 +287,11 @@ 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_;
@ -304,16 +305,16 @@ 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;
@ -331,7 +332,7 @@ 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.
@ -361,7 +362,7 @@ 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(
@ -392,7 +393,7 @@ 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));
@ -1433,7 +1434,8 @@ 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

@ -17,6 +17,7 @@
#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"
@ -73,6 +74,7 @@ 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

@ -118,6 +118,12 @@ 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

@ -20,6 +20,7 @@
#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"
const std::string& DefaultInFilename() {
@ -227,7 +228,8 @@ 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

@ -19,6 +19,7 @@
#include "modules/audio_coding/neteq/tools/audio_sink.h"
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.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_;
}
}