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Revert "Wires up WebrtcKeyValueBasedConfig in media engines."

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This reverts commit 591b2ab82ead157b5f5a85d5082bd15fe8c51809.

Reason for revert: Breaks downstream project

Original change's description:
> Wires up WebrtcKeyValueBasedConfig in media engines.
> 
> This replaces field_trial:: -based functions from system_wrappers.
> Field trials are still used as fallback, but injectable trials are now
> possible.
> 
> Bug: webrtc:11926
> Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32129}

TBR=mbonadei@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I3e169149a8b787aa6366bb357abb71794534c63a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184507
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32132}
This commit is contained in:
Artem Titov 2020-09-17 20:17:19 +00:00 committed by Commit Bot
parent 467b7c0ab2
commit 5956a17ed6
40 changed files with 210 additions and 410 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
api/BUILD.gn
View File

@ -55,7 +55,6 @@ if (!build_with_chromium) {
"audio:audio_mixer_api",
"audio_codecs:audio_codecs_api",
"task_queue:default_task_queue_factory",
"transport:field_trial_based_config",
"video_codecs:video_codecs_api",
]
}

3
api/create_peerconnection_factory.cc
View File

@ -18,7 +18,6 @@
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "media/base/media_engine.h"
#include "media/engine/webrtc_media_engine.h"
#include "modules/audio_device/include/audio_device.h"
@ -46,7 +45,6 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
dependencies.call_factory = CreateCallFactory();
dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(
dependencies.task_queue_factory.get());
dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_dependencies;
media_dependencies.task_queue_factory = dependencies.task_queue_factory.get();
@ -61,7 +59,6 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
media_dependencies.audio_mixer = std::move(audio_mixer);
media_dependencies.video_encoder_factory = std::move(video_encoder_factory);
media_dependencies.video_decoder_factory = std::move(video_decoder_factory);
media_dependencies.trials = dependencies.trials.get();
dependencies.media_engine =
cricket::CreateMediaEngine(std::move(media_dependencies));

6
call/call.cc
View File

@ -260,8 +260,6 @@ class Call final : public webrtc::Call,
Stats GetStats() const override;
const WebRtcKeyValueConfig& trials() const override;
// Implements PacketReceiver.
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
@ -1114,10 +1112,6 @@ Call::Stats Call::GetStats() const {
return stats;
}
const WebRtcKeyValueConfig& Call::trials() const {
return *config_.trials;
}
void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
RTC_DCHECK_RUN_ON(worker_thread_);
switch (media) {

2
call/call.h
View File

@ -156,8 +156,6 @@ class Call {
virtual void SetClientBitratePreferences(
const BitrateSettings& preferences) = 0;
virtual const WebRtcKeyValueConfig& trials() const = 0;
virtual ~Call() {}
};

4
call/degraded_call.cc
View File

@ -266,10 +266,6 @@ Call::Stats DegradedCall::GetStats() const {
return call_->GetStats();
}
const WebRtcKeyValueConfig& DegradedCall::trials() const {
return call_->trials();
}
void DegradedCall::SignalChannelNetworkState(MediaType media,
NetworkState state) {
call_->SignalChannelNetworkState(media, state);

2
call/degraded_call.h
View File

@ -85,8 +85,6 @@ class DegradedCall : public Call, private PacketReceiver {
Stats GetStats() const override;
const WebRtcKeyValueConfig& trials() const override;
void SignalChannelNetworkState(MediaType media, NetworkState state) override;
void OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) override;

5
media/BUILD.gn
View File

@ -93,7 +93,6 @@ rtc_library("rtc_media_base") {
"../api/crypto:frame_encryptor_interface",
"../api/crypto:options",
"../api/transport:stun_types",
"../api/transport:webrtc_key_value_config",
"../api/transport/rtp:rtp_source",
"../api/video:video_bitrate_allocation",
"../api/video:video_bitrate_allocator_factory",
@ -288,8 +287,6 @@ rtc_library("rtc_audio_video") {
"../api/audio_codecs:audio_codecs_api",
"../api/task_queue",
"../api/transport:bitrate_settings",
"../api/transport:field_trial_based_config",
"../api/transport:webrtc_key_value_config",
"../api/transport/rtp:rtp_source",
"../api/units:data_rate",
"../api/video:video_bitrate_allocation",
@ -327,6 +324,7 @@ rtc_library("rtc_audio_video") {
"../rtc_base/system:rtc_export",
"../rtc_base/third_party/base64",
"../system_wrappers",
"../system_wrappers:field_trial",
"../system_wrappers:metrics",
]
absl_deps = [
@ -458,7 +456,6 @@ if (rtc_include_tests) {
"../api:call_api",
"../api:fec_controller_api",
"../api:scoped_refptr",
"../api/transport:field_trial_based_config",
"../api/video:encoded_image",
"../api/video:video_bitrate_allocation",
"../api/video:video_frame",

13
media/base/media_engine.cc
View File

@ -161,20 +161,11 @@ webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
}
CompositeMediaEngine::CompositeMediaEngine(
std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials,
std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VoiceEngineInterface> voice_engine,
std::unique_ptr<VideoEngineInterface> video_engine)
: trials_(std::move(trials)),
voice_engine_(std::move(audio_engine)),
: voice_engine_(std::move(voice_engine)),
video_engine_(std::move(video_engine)) {}
CompositeMediaEngine::CompositeMediaEngine(
std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine)
: CompositeMediaEngine(nullptr,
std::move(audio_engine),
std::move(video_engine)) {}
CompositeMediaEngine::~CompositeMediaEngine() = default;
bool CompositeMediaEngine::Init() {

6
media/base/media_engine.h
View File

@ -19,7 +19,6 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/audio_state.h"
#include "media/base/codec.h"
@ -132,12 +131,8 @@ class MediaEngineInterface {
// CompositeMediaEngine constructs a MediaEngine from separate
// voice and video engine classes.
// Optionally owns a WebRtcKeyValueConfig trials map.
class CompositeMediaEngine : public MediaEngineInterface {
public:
CompositeMediaEngine(std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials,
std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine);
CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine);
~CompositeMediaEngine() override;
@ -149,7 +144,6 @@ class CompositeMediaEngine : public MediaEngineInterface {
const VideoEngineInterface& video() const override;
private:
const std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials_;
std::unique_ptr<VoiceEngineInterface> voice_engine_;
std::unique_ptr<VideoEngineInterface> video_engine_;
};

6
media/engine/fake_webrtc_call.h
View File

@ -24,7 +24,6 @@
#include <string>
#include <vector>
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_frame.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
@ -363,10 +362,6 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
webrtc::Call::Stats GetStats() const override;
const webrtc::WebRtcKeyValueConfig& trials() const override {
return trials_;
}
void SignalChannelNetworkState(webrtc::MediaType media,
webrtc::NetworkState state) override;
void OnAudioTransportOverheadChanged(
@ -390,7 +385,6 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
int num_created_send_streams_;
int num_created_receive_streams_;
webrtc::FieldTrialBasedConfig trials_;
};
} // namespace cricket

4
media/engine/null_webrtc_video_engine_unittest.cc
View File

@ -15,7 +15,6 @@
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "media/engine/webrtc_voice_engine.h"
#include "modules/audio_device/include/mock_audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
@ -32,12 +31,11 @@ TEST(NullWebRtcVideoEngineTest, CheckInterface) {
webrtc::CreateDefaultTaskQueueFactory();
rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
webrtc::test::MockAudioDeviceModule::CreateNice();
webrtc::FieldTrialBasedConfig trials;
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
webrtc::AudioProcessingBuilder().Create(), trials);
webrtc::AudioProcessingBuilder().Create());
CompositeMediaEngine engine(std::move(audio_engine),
std::make_unique<NullWebRtcVideoEngine>());

51
media/engine/simulcast.cc
View File

@ -16,7 +16,6 @@
#include <algorithm>
#include <string>
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/video/video_codec_constants.h"
#include "media/base/media_constants.h"
@ -27,6 +26,7 @@
#include "rtc_base/experiments/normalize_simulcast_size_experiment.h"
#include "rtc_base/experiments/rate_control_settings.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
namespace cricket {
@ -103,9 +103,7 @@ constexpr const SimulcastFormat kSimulcastFormats[] = {
const int kMaxScreenshareSimulcastLayers = 2;
// Multiway: Number of temporal layers for each simulcast stream.
int DefaultNumberOfTemporalLayers(int simulcast_id,
bool screenshare,
const webrtc::WebRtcKeyValueConfig& trials) {
int DefaultNumberOfTemporalLayers(int simulcast_id, bool screenshare) {
RTC_CHECK_GE(simulcast_id, 0);
RTC_CHECK_LT(simulcast_id, webrtc::kMaxSimulcastStreams);
@ -116,8 +114,10 @@ int DefaultNumberOfTemporalLayers(int simulcast_id,
: kDefaultNumTemporalLayers;
const std::string group_name =
screenshare ? trials.Lookup("WebRTC-VP8ScreenshareTemporalLayers")
: trials.Lookup("WebRTC-VP8ConferenceTemporalLayers");
screenshare ? webrtc::field_trial::FindFullName(
"WebRTC-VP8ScreenshareTemporalLayers")
: webrtc::field_trial::FindFullName(
"WebRTC-VP8ConferenceTemporalLayers");
if (group_name.empty())
return default_num_temporal_layers;
@ -231,10 +231,9 @@ webrtc::DataRate GetTotalMaxBitrate(
size_t LimitSimulcastLayerCount(int width,
int height,
size_t need_layers,
size_t layer_count,
const webrtc::WebRtcKeyValueConfig& trials) {
if (!absl::StartsWith(trials.Lookup(kUseLegacySimulcastLayerLimitFieldTrial),
"Disabled")) {
size_t layer_count) {
if (!webrtc::field_trial::IsDisabled(
kUseLegacySimulcastLayerLimitFieldTrial)) {
size_t adaptive_layer_count = std::max(
need_layers,
kSimulcastFormats[FindSimulcastFormatIndex(width, height)].max_layers);
@ -255,28 +254,27 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(
double bitrate_priority,
int max_qp,
bool is_screenshare_with_conference_mode,
bool temporal_layers_supported,
const webrtc::WebRtcKeyValueConfig& trials) {
bool temporal_layers_supported) {
RTC_DCHECK_LE(min_layers, max_layers);
RTC_DCHECK(max_layers > 1 || is_screenshare_with_conference_mode);
const bool base_heavy_tl3_rate_alloc =
webrtc::RateControlSettings::ParseFromKeyValueConfig(&trials)
webrtc::RateControlSettings::ParseFromFieldTrials()
.Vp8BaseHeavyTl3RateAllocation();
if (is_screenshare_with_conference_mode) {
return GetScreenshareLayers(max_layers, width, height, bitrate_priority,
max_qp, temporal_layers_supported,
base_heavy_tl3_rate_alloc, trials);
base_heavy_tl3_rate_alloc);
} else {
// Some applications rely on the old behavior limiting the simulcast layer
// count based on the resolution automatically, which they can get through
// the WebRTC-LegacySimulcastLayerLimit field trial until they update.
max_layers =
LimitSimulcastLayerCount(width, height, min_layers, max_layers, trials);
LimitSimulcastLayerCount(width, height, min_layers, max_layers);
return GetNormalSimulcastLayers(max_layers, width, height, bitrate_priority,
max_qp, temporal_layers_supported,
base_heavy_tl3_rate_alloc, trials);
base_heavy_tl3_rate_alloc);
}
}
@ -287,8 +285,7 @@ std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
double bitrate_priority,
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials) {
bool base_heavy_tl3_rate_alloc) {
std::vector<webrtc::VideoStream> layers(layer_count);
// Format width and height has to be divisible by |2 ^ num_simulcast_layers -
@ -303,13 +300,11 @@ std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
// TODO(pbos): Fill actual temporal-layer bitrate thresholds.
layers[s].max_qp = max_qp;
layers[s].num_temporal_layers =
temporal_layers_supported
? DefaultNumberOfTemporalLayers(s, false, trials)
: 1;
temporal_layers_supported ? DefaultNumberOfTemporalLayers(s, false) : 1;
layers[s].max_bitrate_bps = FindSimulcastMaxBitrate(width, height).bps();
layers[s].target_bitrate_bps =
FindSimulcastTargetBitrate(width, height).bps();
int num_temporal_layers = DefaultNumberOfTemporalLayers(s, false, trials);
int num_temporal_layers = DefaultNumberOfTemporalLayers(s, false);
if (s == 0) {
// If alternative temporal rate allocation is selected, adjust the
// bitrate of the lowest simulcast stream so that absolute bitrate for
@ -361,8 +356,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
double bitrate_priority,
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials) {
bool base_heavy_tl3_rate_alloc) {
auto max_screenshare_layers = kMaxScreenshareSimulcastLayers;
size_t num_simulcast_layers =
std::min<int>(max_layers, max_screenshare_layers);
@ -385,8 +379,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
// restrictions. The base simulcast layer will still use legacy setup.
if (num_simulcast_layers == kMaxScreenshareSimulcastLayers) {
// Add optional upper simulcast layer.
const int num_temporal_layers =
DefaultNumberOfTemporalLayers(1, true, trials);
const int num_temporal_layers = DefaultNumberOfTemporalLayers(1, true);
int max_bitrate_bps;
bool using_boosted_bitrate = false;
if (!temporal_layers_supported) {
@ -396,7 +389,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
kScreenshareHighStreamMaxBitrate.bps() *
webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
num_temporal_layers, 0, base_heavy_tl3_rate_alloc));
} else if (DefaultNumberOfTemporalLayers(1, true, trials) != 3 ||
} else if (DefaultNumberOfTemporalLayers(1, true) != 3 ||
base_heavy_tl3_rate_alloc) {
// Experimental temporal layer mode used, use increased max bitrate.
max_bitrate_bps = kScreenshareHighStreamMaxBitrate.bps();
@ -416,9 +409,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
layers[1].max_qp = max_qp;
layers[1].max_framerate = kDefaultVideoMaxFramerate;
layers[1].num_temporal_layers =
temporal_layers_supported
? DefaultNumberOfTemporalLayers(1, true, trials)
: 1;
temporal_layers_supported ? DefaultNumberOfTemporalLayers(1, true) : 1;
layers[1].min_bitrate_bps = using_boosted_bitrate
? kScreenshareHighStreamMinBitrate.bps()
: layers[0].target_bitrate_bps * 2;

10
media/engine/simulcast.h
View File

@ -15,7 +15,6 @@
#include <vector>
#include "api/transport/webrtc_key_value_config.h"
#include "api/units/data_rate.h"
#include "api/video_codecs/video_encoder_config.h"
@ -42,8 +41,7 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(
double bitrate_priority,
int max_qp,
bool is_screenshare_with_conference_mode,
bool temporal_layers_supported,
const webrtc::WebRtcKeyValueConfig& trials);
bool temporal_layers_supported);
// Gets the simulcast config layers for a non-screensharing case.
std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
@ -53,8 +51,7 @@ std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
double bitrate_priority,
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials);
bool base_heavy_tl3_rate_alloc);
// Gets simulcast config layers for screenshare settings.
std::vector<webrtc::VideoStream> GetScreenshareLayers(
@ -64,8 +61,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
double bitrate_priority,
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials);
bool base_heavy_tl3_rate_alloc);
} // namespace cricket

85
media/engine/simulcast_unittest.cc
View File

@ -10,7 +10,6 @@
#include "media/engine/simulcast.h"
#include "api/transport/field_trial_based_config.h"
#include "media/base/media_constants.h"
#include "media/engine/constants.h"
#include "test/field_trial.h"
@ -81,13 +80,12 @@ TEST(SimulcastTest, BandwidthAboveTotalMaxBitrateGivenToHighestStream) {
TEST(SimulcastTest, GetConfig) {
const std::vector<VideoStream> kExpected = GetSimulcastBitrates720p();
const FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 1280, 720, kBitratePriority, kQpMax,
!kScreenshare, true, trials);
!kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(320u, streams[0].width);
@ -115,7 +113,6 @@ TEST(SimulcastTest, GetConfig) {
TEST(SimulcastTest, GetConfigWithBaseHeavyVP8TL3RateAllocation) {
test::ScopedFieldTrials field_trials(
"WebRTC-UseBaseHeavyVP8TL3RateAllocation/Enabled/");
FieldTrialBasedConfig trials;
const std::vector<VideoStream> kExpected = GetSimulcastBitrates720p();
@ -123,7 +120,7 @@ TEST(SimulcastTest, GetConfigWithBaseHeavyVP8TL3RateAllocation) {
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 1280, 720, kBitratePriority, kQpMax,
!kScreenshare, true, trials);
!kScreenshare, true);
EXPECT_EQ(kExpected[0].min_bitrate_bps, streams[0].min_bitrate_bps);
EXPECT_EQ(static_cast<int>(0.4 * kExpected[0].target_bitrate_bps / 0.6),
@ -140,10 +137,9 @@ TEST(SimulcastTest, GetConfigWithBaseHeavyVP8TL3RateAllocation) {
TEST(SimulcastTest, GetConfigWithLimitedMaxLayers) {
const size_t kMinLayers = 1;
const size_t kMaxLayers = 2;
FieldTrialBasedConfig trials;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 1280, 720, kBitratePriority, kQpMax,
!kScreenshare, true, trials);
!kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(640u, streams[0].width);
@ -155,12 +151,11 @@ TEST(SimulcastTest, GetConfigWithLimitedMaxLayers) {
TEST(SimulcastTest, GetConfigWithLimitedMaxLayersForResolution) {
test::ScopedFieldTrials field_trials(
"WebRTC-LegacySimulcastLayerLimit/Enabled/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 800, 600, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(2u, streams.size());
EXPECT_EQ(400u, streams[0].width);
@ -172,12 +167,11 @@ TEST(SimulcastTest, GetConfigWithLimitedMaxLayersForResolution) {
TEST(SimulcastTest, GetConfigWithLowResolutionScreenshare) {
test::ScopedFieldTrials field_trials(
"WebRTC-LegacySimulcastLayerLimit/Enabled/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 100, 100, kBitratePriority, kQpMax, kScreenshare,
true, trials);
std::vector<VideoStream> streams =
cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 100, 100,
kBitratePriority, kQpMax, kScreenshare, true);
// Simulcast streams number is never decreased for screenshare,
// even for very low resolution.
@ -187,12 +181,11 @@ TEST(SimulcastTest, GetConfigWithLowResolutionScreenshare) {
TEST(SimulcastTest, GetConfigWithNotLimitedMaxLayersForResolution) {
test::ScopedFieldTrials field_trials(
"WebRTC-LegacySimulcastLayerLimit/Disabled/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 800, 600, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(200u, streams[0].width);
@ -204,12 +197,11 @@ TEST(SimulcastTest, GetConfigWithNotLimitedMaxLayersForResolution) {
}
TEST(SimulcastTest, GetConfigWithNormalizedResolution) {
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 2;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 640 + 1, 360 + 1, kBitratePriority, kQpMax,
!kScreenshare, true, trials);
!kScreenshare, true);
// Must be divisible by |2 ^ (num_layers - 1)|.
EXPECT_EQ(kMaxLayers, streams.size());
@ -222,13 +214,12 @@ TEST(SimulcastTest, GetConfigWithNormalizedResolution) {
TEST(SimulcastTest, GetConfigWithNormalizedResolutionDivisibleBy4) {
test::ScopedFieldTrials field_trials(
"WebRTC-NormalizeSimulcastResolution/Enabled-2/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 2;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 709, 501, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
// Must be divisible by |2 ^ 2|.
EXPECT_EQ(kMaxLayers, streams.size());
@ -241,13 +232,12 @@ TEST(SimulcastTest, GetConfigWithNormalizedResolutionDivisibleBy4) {
TEST(SimulcastTest, GetConfigWithNormalizedResolutionDivisibleBy8) {
test::ScopedFieldTrials field_trials(
"WebRTC-NormalizeSimulcastResolution/Enabled-3/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 2;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 709, 501, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
// Must be divisible by |2 ^ 3|.
EXPECT_EQ(kMaxLayers, streams.size());
@ -260,56 +250,53 @@ TEST(SimulcastTest, GetConfigWithNormalizedResolutionDivisibleBy8) {
TEST(SimulcastTest, GetConfigForLegacyLayerLimit) {
test::ScopedFieldTrials field_trials(
"WebRTC-LegacySimulcastLayerLimit/Enabled/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const int kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 320, 180, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(1u, streams.size());
streams = cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 640, 360,
kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(2u, streams.size());
streams = cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 1920, 1080,
kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(3u, streams.size());
}
TEST(SimulcastTest, GetConfigForLegacyLayerLimitWithRequiredHD) {
test::ScopedFieldTrials field_trials(
"WebRTC-LegacySimulcastLayerLimit/Enabled/");
FieldTrialBasedConfig trials;
const size_t kMinLayers = 3; // "HD" layer must be present!
const int kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 320, 180, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(3u, streams.size());
streams = cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 640, 360,
kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(3u, streams.size());
streams = cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 1920, 1080,
kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(3u, streams.size());
}
TEST(SimulcastTest, GetConfigForScreenshareSimulcast) {
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 1400, 800, kBitratePriority, kQpMax, kScreenshare,
true, trials);
std::vector<VideoStream> streams =
cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 1400, 800,
kBitratePriority, kQpMax, kScreenshare, true);
EXPECT_GT(streams.size(), 1u);
for (size_t i = 0; i < streams.size(); ++i) {
@ -326,12 +313,11 @@ TEST(SimulcastTest, GetConfigForScreenshareSimulcast) {
}
TEST(SimulcastTest, GetConfigForScreenshareSimulcastWithLimitedMaxLayers) {
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 1;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 1400, 800, kBitratePriority, kQpMax, kScreenshare,
true, trials);
std::vector<VideoStream> streams =
cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 1400, 800,
kBitratePriority, kQpMax, kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
}
@ -340,23 +326,22 @@ TEST(SimulcastTest, SimulcastScreenshareMaxBitrateAdjustedForResolution) {
constexpr int kScreenshareHighStreamMinBitrateBps = 600000;
constexpr int kScreenshareHighStreamMaxBitrateBps = 1250000;
constexpr int kMaxBitrate960_540 = 1200000;
FieldTrialBasedConfig trials;
// Normal case, max bitrate not limited by resolution.
const size_t kMinLayers = 1;
const size_t kMaxLayers = 2;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 1920, 1080, kBitratePriority, kQpMax,
kScreenshare, true, trials);
std::vector<VideoStream> streams =
cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 1920, 1080,
kBitratePriority, kQpMax, kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(streams[1].max_bitrate_bps, kScreenshareHighStreamMaxBitrateBps);
EXPECT_EQ(streams[1].min_bitrate_bps, kScreenshareHighStreamMinBitrateBps);
EXPECT_GE(streams[1].max_bitrate_bps, streams[1].min_bitrate_bps);
// At 960x540, the max bitrate is limited to 900kbps.
streams = cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 960, 540,
kBitratePriority, kQpMax, kScreenshare,
true, trials);
streams =
cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 960, 540,
kBitratePriority, kQpMax, kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(streams[1].max_bitrate_bps, kMaxBitrate960_540);
EXPECT_EQ(streams[1].min_bitrate_bps, kScreenshareHighStreamMinBitrateBps);
@ -364,9 +349,9 @@ TEST(SimulcastTest, SimulcastScreenshareMaxBitrateAdjustedForResolution) {
// At 480x270, the max bitrate is limited to 450kbps. This is lower than
// the min bitrate, so use that as a lower bound.
streams = cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 480, 270,
kBitratePriority, kQpMax, kScreenshare,
true, trials);
streams =
cricket::GetSimulcastConfig(kMinLayers, kMaxLayers, 480, 270,
kBitratePriority, kQpMax, kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(streams[1].max_bitrate_bps, kScreenshareHighStreamMinBitrateBps);
EXPECT_EQ(streams[1].min_bitrate_bps, kScreenshareHighStreamMinBitrateBps);
@ -374,12 +359,11 @@ TEST(SimulcastTest, SimulcastScreenshareMaxBitrateAdjustedForResolution) {
}
TEST(SimulcastTest, AveragesBitratesForNonStandardResolution) {
FieldTrialBasedConfig trials;
const size_t kMinLayers = 1;
const size_t kMaxLayers = 3;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, 900, 800, kBitratePriority, kQpMax, !kScreenshare,
true, trials);
true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(900u, streams[2].width);
@ -396,11 +380,10 @@ TEST(SimulcastTest, BitratesForCloseToStandardResolution) {
const size_t kWidth = 1280;
const size_t kHeight = 716;
const std::vector<VideoStream> kExpectedNear = GetSimulcastBitrates720p();
FieldTrialBasedConfig trials;
std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
kMinLayers, kMaxLayers, kWidth, kHeight, kBitratePriority, kQpMax,
!kScreenshare, true, trials);
!kScreenshare, true);
EXPECT_EQ(kMaxLayers, streams.size());
EXPECT_EQ(kWidth, streams[2].width);

21
media/engine/webrtc_media_engine.cc
View File

@ -14,8 +14,8 @@
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "media/engine/webrtc_voice_engine.h"
#include "system_wrappers/include/field_trial.h"
#ifdef HAVE_WEBRTC_VIDEO
#include "media/engine/webrtc_video_engine.h"
@ -27,27 +27,20 @@ namespace cricket {
std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
MediaEngineDependencies dependencies) {
// TODO(sprang): Make populating |dependencies.trials| mandatory and remove
// these fallbacks.
std::unique_ptr<webrtc::WebRtcKeyValueConfig> fallback_trials(
dependencies.trials ? nullptr : new webrtc::FieldTrialBasedConfig());
const webrtc::WebRtcKeyValueConfig& trials =
dependencies.trials ? *dependencies.trials : *fallback_trials;
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
dependencies.task_queue_factory, std::move(dependencies.adm),
std::move(dependencies.audio_encoder_factory),
std::move(dependencies.audio_decoder_factory),
std::move(dependencies.audio_mixer),
std::move(dependencies.audio_processing), trials);
std::move(dependencies.audio_processing));
#ifdef HAVE_WEBRTC_VIDEO
auto video_engine = std::make_unique<WebRtcVideoEngine>(
std::move(dependencies.video_encoder_factory),
std::move(dependencies.video_decoder_factory), trials);
std::move(dependencies.video_decoder_factory));
#else
auto video_engine = std::make_unique<NullWebRtcVideoEngine>();
#endif
return std::make_unique<CompositeMediaEngine>(std::move(fallback_trials),
std::move(audio_engine),
return std::make_unique<CompositeMediaEngine>(std::move(audio_engine),
std::move(video_engine));
}
@ -94,8 +87,7 @@ bool ValidateRtpExtensions(
std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions,
bool (*supported)(absl::string_view),
bool filter_redundant_extensions,
const webrtc::WebRtcKeyValueConfig& trials) {
bool filter_redundant_extensions) {
RTC_DCHECK(ValidateRtpExtensions(extensions));
RTC_DCHECK(supported);
std::vector<webrtc::RtpExtension> result;
@ -129,8 +121,7 @@ std::vector<webrtc::RtpExtension> FilterRtpExtensions(
result.erase(it, result.end());
// Keep just the highest priority extension of any in the following lists.
if (absl::StartsWith(trials.Lookup("WebRTC-FilterAbsSendTimeExtension"),
"Enabled")) {
if (webrtc::field_trial::IsEnabled("WebRTC-FilterAbsSendTimeExtension")) {
static const char* const kBweExtensionPriorities[] = {
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kAbsSendTimeUri,

6
media/engine/webrtc_media_engine.h
View File

@ -21,7 +21,6 @@
#include "api/rtp_parameters.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "media/base/codec.h"
@ -49,8 +48,6 @@ struct MediaEngineDependencies {
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory;
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory;
const webrtc::WebRtcKeyValueConfig* trials = nullptr;
};
// CreateMediaEngine may be called on any thread, though the engine is
@ -69,8 +66,7 @@ bool ValidateRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions,
bool (*supported)(absl::string_view),
bool filter_redundant_extensions,
const webrtc::WebRtcKeyValueConfig& trials);
bool filter_redundant_extensions);
webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec);

45
media/engine/webrtc_media_engine_unittest.cc
View File

@ -13,7 +13,6 @@
#include <memory>
#include <utility>
#include "api/transport/field_trial_based_config.h"
#include "media/engine/webrtc_media_engine_defaults.h"
#include "test/field_trial.h"
#include "test/gtest.h"
@ -102,17 +101,15 @@ TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_OverlappingIds_EndOfSet) {
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_EmptyList) {
std::vector<RtpExtension> extensions;
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions1, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions1, true);
EXPECT_EQ(0u, filtered.size());
}
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_IncludeOnlySupported) {
std::vector<RtpExtension> extensions = MakeUniqueExtensions();
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions1, false, trials);
FilterRtpExtensions(extensions, SupportedExtensions1, false);
EXPECT_EQ(2u, filtered.size());
EXPECT_EQ("c", filtered[0].uri);
EXPECT_EQ("i", filtered[1].uri);
@ -120,27 +117,24 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_IncludeOnlySupported) {
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_SortedByName_1) {
std::vector<RtpExtension> extensions = MakeUniqueExtensions();
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, false, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, false);
EXPECT_EQ(12u, filtered.size());
EXPECT_TRUE(IsSorted(filtered));
}
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_SortedByName_2) {
std::vector<RtpExtension> extensions = MakeUniqueExtensions();
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(12u, filtered.size());
EXPECT_TRUE(IsSorted(filtered));
}
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_DontRemoveRedundant) {
std::vector<RtpExtension> extensions = MakeRedundantExtensions();
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, false, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, false);
EXPECT_EQ(12u, filtered.size());
EXPECT_TRUE(IsSorted(filtered));
EXPECT_EQ(filtered[0].uri, filtered[1].uri);
@ -148,9 +142,8 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_DontRemoveRedundant) {
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundant) {
std::vector<RtpExtension> extensions = MakeRedundantExtensions();
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(6u, filtered.size());
EXPECT_TRUE(IsSorted(filtered));
EXPECT_NE(filtered[0].uri, filtered[1].uri);
@ -162,9 +155,8 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantEncrypted_1) {
extensions.push_back(webrtc::RtpExtension("b", 2, true));
extensions.push_back(webrtc::RtpExtension("c", 3));
extensions.push_back(webrtc::RtpExtension("b", 4));
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(3u, filtered.size());
EXPECT_TRUE(IsSorted(filtered));
EXPECT_EQ(filtered[0].uri, filtered[1].uri);
@ -179,9 +171,8 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantEncrypted_2) {
extensions.push_back(webrtc::RtpExtension("b", 2));
extensions.push_back(webrtc::RtpExtension("c", 3));
extensions.push_back(webrtc::RtpExtension("b", 4));
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(3u, filtered.size());
EXPECT_TRUE(IsSorted(filtered));
EXPECT_EQ(filtered[0].uri, filtered[1].uri);
@ -193,7 +184,6 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantEncrypted_2) {
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_1) {
webrtc::test::ScopedFieldTrials override_field_trials_(
"WebRTC-FilterAbsSendTimeExtension/Enabled/");
webrtc::FieldTrialBasedConfig trials;
std::vector<RtpExtension> extensions;
extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 3));
@ -203,7 +193,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_1) {
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 1));
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(1u, filtered.size());
EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[0].uri);
}
@ -218,9 +208,8 @@ TEST(WebRtcMediaEngineTest,
extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 1));
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(2u, filtered.size());
EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[0].uri);
EXPECT_EQ(RtpExtension::kAbsSendTimeUri, filtered[1].uri);
@ -229,7 +218,6 @@ TEST(WebRtcMediaEngineTest,
TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBweEncrypted_1) {
webrtc::test::ScopedFieldTrials override_field_trials_(
"WebRTC-FilterAbsSendTimeExtension/Enabled/");
webrtc::FieldTrialBasedConfig trials;
std::vector<RtpExtension> extensions;
extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 3));
@ -243,7 +231,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBweEncrypted_1) {
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 2, true));
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(2u, filtered.size());
EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[0].uri);
EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[1].uri);
@ -264,9 +252,8 @@ TEST(WebRtcMediaEngineTest,
extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri, 2, true));
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(3u, filtered.size());
EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[0].uri);
EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[1].uri);
@ -279,9 +266,8 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_2) {
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 1));
extensions.push_back(RtpExtension(RtpExtension::kAbsSendTimeUri, 14));
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 7));
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(1u, filtered.size());
EXPECT_EQ(RtpExtension::kAbsSendTimeUri, filtered[0].uri);
}
@ -290,9 +276,8 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_3) {
std::vector<RtpExtension> extensions;
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 2));
extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
webrtc::FieldTrialBasedConfig trials;
std::vector<webrtc::RtpExtension> filtered =
FilterRtpExtensions(extensions, SupportedExtensions2, true, trials);
FilterRtpExtensions(extensions, SupportedExtensions2, true);
EXPECT_EQ(1u, filtered.size());
EXPECT_EQ(RtpExtension::kTimestampOffsetUri, filtered[0].uri);
}
@ -300,8 +285,6 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_3) {
TEST(WebRtcMediaEngineTest, Create) {
MediaEngineDependencies deps;
webrtc::SetMediaEngineDefaults(&deps);
webrtc::FieldTrialBasedConfig trials;
deps.trials = &trials;
std::unique_ptr<MediaEngineInterface> engine =
CreateMediaEngine(std::move(deps));

147
media/engine/webrtc_video_engine.cc
View File

@ -40,6 +40,7 @@
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
namespace cricket {
@ -60,13 +61,23 @@ const char* StreamTypeToString(
return nullptr;
}
bool IsEnabled(const webrtc::WebRtcKeyValueConfig& trials,
absl::string_view name) {
return absl::StartsWith(trials.Lookup(name), "Enabled");
// If this field trial is enabled, we will enable sending FlexFEC and disable
// sending ULPFEC whenever the former has been negotiated in the SDPs.
bool IsFlexfecFieldTrialEnabled() {
return webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03");
}
void AddDefaultFeedbackParams(VideoCodec* codec,
const webrtc::WebRtcKeyValueConfig& trials) {
// If this field trial is enabled, the "flexfec-03" codec will be advertised
// as being supported. This means that "flexfec-03" will appear in the default
// SDP offer, and we therefore need to be ready to receive FlexFEC packets from
// the remote. It also means that FlexFEC SSRCs will be generated by
// MediaSession and added as "a=ssrc:" and "a=ssrc-group:" lines in the local
// SDP.
bool IsFlexfecAdvertisedFieldTrialEnabled() {
return webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03-Advertised");
}
void AddDefaultFeedbackParams(VideoCodec* codec) {
// Don't add any feedback params for RED and ULPFEC.
if (codec->name == kRedCodecName || codec->name == kUlpfecCodecName)
return;
@ -80,7 +91,7 @@ void AddDefaultFeedbackParams(VideoCodec* codec,
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
if (codec->name == kVp8CodecName &&
IsEnabled(trials, "WebRTC-RtcpLossNotification")) {
webrtc::field_trial::IsEnabled("WebRTC-RtcpLossNotification")) {
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamLntf, kParamValueEmpty));
}
}
@ -90,8 +101,7 @@ void AddDefaultFeedbackParams(VideoCodec* codec,
// codecs for recognized codecs (VP8, VP9, H264, and RED). It will also add
// default feedback params to the codecs.
std::vector<VideoCodec> AssignPayloadTypesAndDefaultCodecs(
std::vector<webrtc::SdpVideoFormat> input_formats,
const webrtc::WebRtcKeyValueConfig& trials) {
std::vector<webrtc::SdpVideoFormat> input_formats) {
if (input_formats.empty())
return std::vector<VideoCodec>();
static const int kFirstDynamicPayloadType = 96;
@ -101,7 +111,7 @@ std::vector<VideoCodec> AssignPayloadTypesAndDefaultCodecs(
input_formats.push_back(webrtc::SdpVideoFormat(kRedCodecName));
input_formats.push_back(webrtc::SdpVideoFormat(kUlpfecCodecName));
if (IsEnabled(trials, "WebRTC-FlexFEC-03-Advertised")) {
if (IsFlexfecAdvertisedFieldTrialEnabled()) {
webrtc::SdpVideoFormat flexfec_format(kFlexfecCodecName);
// This value is currently arbitrarily set to 10 seconds. (The unit
// is microseconds.) This parameter MUST be present in the SDP, but
@ -115,7 +125,7 @@ std::vector<VideoCodec> AssignPayloadTypesAndDefaultCodecs(
for (const webrtc::SdpVideoFormat& format : input_formats) {
VideoCodec codec(format);
codec.id = payload_type;
AddDefaultFeedbackParams(&codec, trials);
AddDefaultFeedbackParams(&codec);
output_codecs.push_back(codec);
// Increment payload type.
@ -149,8 +159,7 @@ std::vector<VideoCodec> AssignPayloadTypesAndDefaultCodecs(
template <class T>
std::vector<VideoCodec> GetPayloadTypesAndDefaultCodecs(
const T* factory,
bool is_decoder_factory,
const webrtc::WebRtcKeyValueConfig& trials) {
bool is_decoder_factory) {
if (!factory) {
return {};
}
@ -161,8 +170,7 @@ std::vector<VideoCodec> GetPayloadTypesAndDefaultCodecs(
AddH264ConstrainedBaselineProfileToSupportedFormats(&supported_formats);
}
return AssignPayloadTypesAndDefaultCodecs(std::move(supported_formats),
trials);
return AssignPayloadTypesAndDefaultCodecs(std::move(supported_formats));
}
bool IsTemporalLayersSupported(const std::string& codec_name) {
@ -237,9 +245,8 @@ static bool ValidateStreamParams(const StreamParams& sp) {
}
// Returns true if the given codec is disallowed from doing simulcast.
bool IsCodecDisabledForSimulcast(const std::string& codec_name,
const webrtc::WebRtcKeyValueConfig& trials) {
return !absl::StartsWith(trials.Lookup("WebRTC-H264Simulcast"), "Disabled")
bool IsCodecDisabledForSimulcast(const std::string& codec_name) {
return !webrtc::field_trial::IsDisabled("WebRTC-H264Simulcast")
? absl::EqualsIgnoreCase(codec_name, kVp9CodecName)
: absl::EqualsIgnoreCase(codec_name, kH264CodecName) ||
absl::EqualsIgnoreCase(codec_name, kVp9CodecName);
@ -265,11 +272,9 @@ static int GetMaxDefaultVideoBitrateKbps(int width,
return max_bitrate;
}
bool GetVp9LayersFromFieldTrialGroup(
size_t* num_spatial_layers,
size_t* num_temporal_layers,
const webrtc::WebRtcKeyValueConfig& trials) {
std::string group = trials.Lookup("WebRTC-SupportVP9SVC");
bool GetVp9LayersFromFieldTrialGroup(size_t* num_spatial_layers,
size_t* num_temporal_layers) {
std::string group = webrtc::field_trial::FindFullName("WebRTC-SupportVP9SVC");
if (group.empty())
return false;
@ -288,21 +293,19 @@ bool GetVp9LayersFromFieldTrialGroup(
return true;
}
absl::optional<size_t> GetVp9SpatialLayersFromFieldTrial(
const webrtc::WebRtcKeyValueConfig& trials) {
absl::optional<size_t> GetVp9SpatialLayersFromFieldTrial() {
size_t num_sl;
size_t num_tl;
if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl, trials)) {
if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
return num_sl;
}
return absl::nullopt;
}
absl::optional<size_t> GetVp9TemporalLayersFromFieldTrial(
const webrtc::WebRtcKeyValueConfig& trials) {
absl::optional<size_t> GetVp9TemporalLayersFromFieldTrial() {
size_t num_sl;
size_t num_tl;
if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl, trials)) {
if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
return num_tl;
}
return absl::nullopt;
@ -464,14 +467,14 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
const size_t default_num_spatial_layers =
parameters_.config.rtp.ssrcs.size();
const size_t num_spatial_layers =
GetVp9SpatialLayersFromFieldTrial(call_->trials())
.value_or(default_num_spatial_layers);
GetVp9SpatialLayersFromFieldTrial().value_or(
default_num_spatial_layers);
const size_t default_num_temporal_layers =
num_spatial_layers > 1 ? kConferenceDefaultNumTemporalLayers : 1;
const size_t num_temporal_layers =
GetVp9TemporalLayersFromFieldTrial(call_->trials())
.value_or(default_num_temporal_layers);
GetVp9TemporalLayersFromFieldTrial().value_or(
default_num_temporal_layers);
vp9_settings.numberOfSpatialLayers = std::min<unsigned char>(
num_spatial_layers, kConferenceMaxNumSpatialLayers);
@ -493,7 +496,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
{"onkeypic", webrtc::InterLayerPredMode::kOnKeyPic}});
webrtc::ParseFieldTrial(
{&interlayer_pred_experiment_enabled, &inter_layer_pred_mode},
call_->trials().Lookup("WebRTC-Vp9InterLayerPred"));
webrtc::field_trial::FindFullName("WebRTC-Vp9InterLayerPred"));
if (interlayer_pred_experiment_enabled) {
vp9_settings.interLayerPred = inter_layer_pred_mode;
} else {
@ -564,11 +567,9 @@ void DefaultUnsignalledSsrcHandler::SetDefaultSink(
WebRtcVideoEngine::WebRtcVideoEngine(
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
const webrtc::WebRtcKeyValueConfig& trials)
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory)
: decoder_factory_(std::move(video_decoder_factory)),
encoder_factory_(std::move(video_encoder_factory)),
trials_(trials) {
encoder_factory_(std::move(video_encoder_factory)) {
RTC_LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()";
}
@ -589,12 +590,12 @@ VideoMediaChannel* WebRtcVideoEngine::CreateMediaChannel(
}
std::vector<VideoCodec> WebRtcVideoEngine::send_codecs() const {
return GetPayloadTypesAndDefaultCodecs(encoder_factory_.get(),
/*is_decoder_factory=*/false, trials_);
/*is_decoder_factory=*/false);
}
std::vector<VideoCodec> WebRtcVideoEngine::recv_codecs() const {
return GetPayloadTypesAndDefaultCodecs(decoder_factory_.get(),
/*is_decoder_factory=*/true, trials_);
/*is_decoder_factory=*/true);
}
std::vector<webrtc::RtpHeaderExtensionCapability>
@ -613,10 +614,11 @@ WebRtcVideoEngine::GetRtpHeaderExtensions() const {
webrtc::RtpExtension::kRidUri, webrtc::RtpExtension::kRepairedRidUri}) {
result.emplace_back(uri, id++, webrtc::RtpTransceiverDirection::kSendRecv);
}
result.emplace_back(webrtc::RtpExtension::kGenericFrameDescriptorUri00, id,
IsEnabled(trials_, "WebRTC-GenericDescriptorAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
result.emplace_back(
webrtc::RtpExtension::kGenericFrameDescriptorUri00, id,
webrtc::field_trial::IsEnabled("WebRTC-GenericDescriptorAdvertised")
? webrtc::RtpTransceiverDirection::kSendRecv
: webrtc::RtpTransceiverDirection::kStopped);
return result;
}
@ -638,13 +640,12 @@ WebRtcVideoChannel::WebRtcVideoChannel(
bitrate_allocator_factory_(bitrate_allocator_factory),
default_send_options_(options),
last_stats_log_ms_(-1),
discard_unknown_ssrc_packets_(
IsEnabled(call_->trials(),
"WebRTC-Video-DiscardPacketsWithUnknownSsrc")),
discard_unknown_ssrc_packets_(webrtc::field_trial::IsEnabled(
"WebRTC-Video-DiscardPacketsWithUnknownSsrc")),
crypto_options_(crypto_options),
unknown_ssrc_packet_buffer_(
IsEnabled(call_->trials(),
"WebRTC-Video-BufferPacketsWithUnknownSsrc")
webrtc::field_trial::IsEnabled(
"WebRTC-Video-BufferPacketsWithUnknownSsrc")
? new UnhandledPacketsBuffer()
: nullptr) {
RTC_DCHECK(thread_checker_.IsCurrent());
@ -652,7 +653,7 @@ WebRtcVideoChannel::WebRtcVideoChannel(
rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
sending_ = false;
recv_codecs_ = MapCodecs(GetPayloadTypesAndDefaultCodecs(
decoder_factory_, /*is_decoder_factory=*/true, call_->trials()));
decoder_factory_, /*is_decoder_factory=*/true));
recv_flexfec_payload_type_ =
recv_codecs_.empty() ? 0 : recv_codecs_.front().flexfec_payload_type;
}
@ -746,7 +747,7 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
}
// Never enable sending FlexFEC, unless we are in the experiment.
if (!IsEnabled(call_->trials(), "WebRTC-FlexFEC-03")) {
if (!IsFlexfecFieldTrialEnabled()) {
RTC_LOG(LS_INFO) << "WebRTC-FlexFEC-03 field trial is not enabled.";
for (VideoCodecSettings& codec : negotiated_codecs)
codec.flexfec_payload_type = -1;
@ -766,8 +767,7 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
changed_params->extmap_allow_mixed = params.extmap_allow_mixed;
}
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true,
call_->trials());
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
changed_params->rtp_header_extensions =
absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
@ -1119,8 +1119,7 @@ bool WebRtcVideoChannel::GetChangedRecvParameters(
if (params.is_stream_active) {
const std::vector<VideoCodec> local_supported_codecs =
GetPayloadTypesAndDefaultCodecs(decoder_factory_,
/*is_decoder_factory=*/true,
call_->trials());
/*is_decoder_factory=*/true);
for (const VideoCodecSettings& mapped_codec : mapped_codecs) {
if (!FindMatchingCodec(local_supported_codecs, mapped_codec.codec)) {
RTC_LOG(LS_ERROR)
@ -1138,8 +1137,7 @@ bool WebRtcVideoChannel::GetChangedRecvParameters(
// Handle RTP header extensions.
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false,
call_->trials());
params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
if (filtered_extensions != recv_rtp_extensions_) {
changed_params->rtp_header_extensions =
absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
@ -1468,7 +1466,7 @@ void WebRtcVideoChannel::ConfigureReceiverRtp(
// TODO(brandtr): Generalize when we add support for multistream protection.
flexfec_config->payload_type = recv_flexfec_payload_type_;
if (IsEnabled(call_->trials(), "WebRTC-FlexFEC-03-Advertised") &&
if (IsFlexfecAdvertisedFieldTrialEnabled() &&
sp.GetFecFrSsrc(ssrc, &flexfec_config->remote_ssrc)) {
flexfec_config->protected_media_ssrcs = {ssrc};
flexfec_config->local_ssrc = config->rtp.local_ssrc;
@ -1701,7 +1699,7 @@ void WebRtcVideoChannel::BackfillBufferedPackets(
int delivery_packet_error_cnt = 0;
webrtc::PacketReceiver* receiver = this->call_->Receiver();
unknown_ssrc_packet_buffer_->BackfillPackets(
ssrcs, [&](uint32_t /*ssrc*/, int64_t packet_time_us,
ssrcs, [&](uint32_t ssrc, int64_t packet_time_us,
rtc::CopyOnWriteBuffer packet) {
switch (receiver->DeliverPacket(webrtc::MediaType::VIDEO, packet,
packet_time_us)) {
@ -1762,7 +1760,7 @@ void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
// which case that is used as UDP recevie buffer size. All other values shall
// result in the default value being used.
const std::string group_name_recv_buf_size =
call_->trials().Lookup("WebRTC-IncreasedReceivebuffers");
webrtc::field_trial::FindFullName("WebRTC-IncreasedReceivebuffers");
int recv_buffer_size = kVideoRtpRecvBufferSize;
if (!group_name_recv_buf_size.empty() &&
(sscanf(group_name_recv_buf_size.c_str(), "%d", &recv_buffer_size) != 1 ||
@ -1780,7 +1778,7 @@ void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
// due to lack of socket buffer space, although it's not yet clear what the
// ideal value should be.
const std::string group_name_send_buf_size =
call_->trials().Lookup("WebRTC-SendBufferSizeBytes");
webrtc::field_trial::FindFullName("WebRTC-SendBufferSizeBytes");
int send_buffer_size = kVideoRtpSendBufferSize;
if (!group_name_send_buf_size.empty() &&
(sscanf(group_name_send_buf_size.c_str(), "%d", &send_buffer_size) != 1 ||
@ -1963,8 +1961,8 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
sending_(false),
disable_automatic_resize_(
IsEnabled(call->trials(), "WebRTC-Video-DisableAutomaticResize")) {
disable_automatic_resize_(webrtc::field_trial::IsEnabled(
"WebRTC-Video-DisableAutomaticResize")) {
// Maximum packet size may come in RtpConfig from external transport, for
// example from QuicTransportInterface implementation, so do not exceed
// given max_packet_size.
@ -1985,7 +1983,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
// FlexFEC SSRCs.
// TODO(brandtr): This code needs to be generalized when we add support for
// multistream protection.
if (IsEnabled(call_->trials(), "WebRTC-FlexFEC-03")) {
if (IsFlexfecFieldTrialEnabled()) {
uint32_t flexfec_ssrc;
bool flexfec_enabled = false;
for (uint32_t primary_ssrc : parameters_.config.rtp.ssrcs) {
@ -2090,7 +2088,8 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const {
webrtc::VideoTrackInterface::ContentHint::kText) {
degradation_preference =
webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
} else if (IsEnabled(call_->trials(), "WebRTC-Video-BalancedDegradation")) {
} else if (webrtc::field_trial::IsEnabled(
"WebRTC-Video-BalancedDegradation")) {
// Standard wants balanced by default, but it needs to be tuned first.
degradation_preference = webrtc::DegradationPreference::BALANCED;
} else {
@ -2341,7 +2340,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
// or a screencast (and not in simulcast screenshare experiment), only
// configure a single stream.
encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();
if (IsCodecDisabledForSimulcast(codec.name, call_->trials())) {
if (IsCodecDisabledForSimulcast(codec.name)) {
encoder_config.number_of_streams = 1;
}
@ -2936,7 +2935,8 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::RecreateWebRtcVideoStream() {
MaybeAssociateFlexfecWithVideo();
stream_->Start();
if (IsEnabled(call_->trials(), "WebRTC-Video-BufferPacketsWithUnknownSsrc")) {
if (webrtc::field_trial::IsEnabled(
"WebRTC-Video-BufferPacketsWithUnknownSsrc")) {
channel_->BackfillBufferedPackets(stream_params_.ssrcs);
}
}
@ -3387,18 +3387,15 @@ void WebRtcVideoChannel::SetDepacketizerToDecoderFrameTransformer(
// TODO(bugs.webrtc.org/8785): Consider removing max_qp as member of
// EncoderStreamFactory and instead set this value individually for each stream
// in the VideoEncoderConfig.simulcast_layers.
EncoderStreamFactory::EncoderStreamFactory(
std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode,
const webrtc::WebRtcKeyValueConfig* trials)
EncoderStreamFactory::EncoderStreamFactory(std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode)
: codec_name_(codec_name),
max_qp_(max_qp),
is_screenshare_(is_screenshare),
conference_mode_(conference_mode),
trials_(trials ? *trials : fallback_trials_) {}
conference_mode_(conference_mode) {}
std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams(
int width,
@ -3520,7 +3517,7 @@ EncoderStreamFactory::CreateSimulcastOrConferenceModeScreenshareStreams(
encoder_config.number_of_streams, width, height,
encoder_config.bitrate_priority, max_qp_,
is_screenshare_ && conference_mode_,
temporal_layers_supported, trials_);
temporal_layers_supported);
// Allow an experiment to override the minimum bitrate for the lowest
// spatial layer. The experiment's configuration has the lowest priority.
if (experimental_min_bitrate) {

22
media/engine/webrtc_video_engine.h
View File

@ -19,7 +19,6 @@
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"
@ -99,8 +98,7 @@ class WebRtcVideoEngine : public VideoEngineInterface {
// and external hardware codecs.
WebRtcVideoEngine(
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
const webrtc::WebRtcKeyValueConfig& trials);
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory);
~WebRtcVideoEngine() override;
@ -122,7 +120,6 @@ class WebRtcVideoEngine : public VideoEngineInterface {
const std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory_;
const std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
bitrate_allocator_factory_;
const webrtc::WebRtcKeyValueConfig& trials_;
};
class WebRtcVideoChannel : public VideoMediaChannel,
@ -569,7 +566,7 @@ class WebRtcVideoChannel : public VideoMediaChannel,
void FillSendAndReceiveCodecStats(VideoMediaInfo* video_media_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(thread_checker_);
rtc::Thread* const worker_thread_;
rtc::Thread* worker_thread_;
rtc::ThreadChecker thread_checker_;
uint32_t rtcp_receiver_report_ssrc_ RTC_GUARDED_BY(thread_checker_);
@ -654,18 +651,7 @@ class EncoderStreamFactory
EncoderStreamFactory(std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode)
: EncoderStreamFactory(codec_name,
max_qp,
is_screenshare,
conference_mode,
nullptr) {}
EncoderStreamFactory(std::string codec_name,
int max_qp,
bool is_screenshare,
bool conference_mode,
const webrtc::WebRtcKeyValueConfig* trials);
bool conference_mode);
private:
std::vector<webrtc::VideoStream> CreateEncoderStreams(
@ -692,8 +678,6 @@ class EncoderStreamFactory
// Allows a screenshare specific configuration, which enables temporal
// layering and various settings.
const bool conference_mode_;
const webrtc::FieldTrialBasedConfig fallback_trials_;
const webrtc::WebRtcKeyValueConfig& trials_;
};
} // namespace cricket

27
media/engine/webrtc_video_engine_unittest.cc
View File

@ -257,8 +257,7 @@ class WebRtcVideoEngineTest : public ::testing::Test {
engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
encoder_factory_),
std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
decoder_factory_),
field_trials_) {
decoder_factory_)) {
// Ensure fake clock doesn't return 0, which will cause some initializations
// fail inside RTP senders.
fake_clock_.AdvanceTime(webrtc::TimeDelta::Micros(1));
@ -1044,9 +1043,8 @@ TEST_F(WebRtcVideoEngineTest, GetSourcesWithNonExistingSsrc) {
TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullFactories) {
std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory;
std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory;
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(std::move(encoder_factory),
std::move(decoder_factory), trials);
std::move(decoder_factory));
EXPECT_EQ(0u, engine.send_codecs().size());
EXPECT_EQ(0u, engine.recv_codecs().size());
}
@ -1057,10 +1055,9 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, EmptyFactories) {
new webrtc::MockVideoEncoderFactory();
webrtc::MockVideoDecoderFactory* decoder_factory =
new webrtc::MockVideoDecoderFactory();
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials);
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
// TODO(kron): Change to Times(1) once send and receive codecs are changed
// to be treated independently.
EXPECT_CALL(*encoder_factory, GetSupportedFormats()).Times(1);
@ -1088,10 +1085,9 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
webrtc::kVideoCodecVP8)))
.WillOnce(
[] { return std::make_unique<webrtc::MockVideoBitrateAllocator>(); });
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials);
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
const webrtc::SdpVideoFormat vp8_format("VP8");
const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
EXPECT_CALL(*encoder_factory, GetSupportedFormats())
@ -1208,10 +1204,9 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
rate_allocator_factory =
std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>();
webrtc::FieldTrialBasedConfig trials;
WebRtcVideoEngine engine(
(std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)),
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials);
(std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)));
const webrtc::SdpVideoFormat vp8_format("VP8");
const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format};
EXPECT_CALL(*encoder_factory, GetSupportedFormats())
@ -1329,8 +1324,7 @@ class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
webrtc::CreateBuiltinVideoEncoderFactory(),
std::make_unique<webrtc::test::FunctionVideoDecoderFactory>(
[]() { return std::make_unique<webrtc::test::FakeDecoder>(); },
kSdpVideoFormats),
field_trials_),
kSdpVideoFormats)),
channel_(absl::WrapUnique(static_cast<cricket::WebRtcVideoChannel*>(
engine_.CreateMediaChannel(
call_.get(),
@ -1462,8 +1456,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(),
field_trials_) {}
webrtc::CreateBuiltinVideoDecoderFactory()) {}
virtual void SetUp() {
// One testcase calls SetUp in a loop, only create call_ once.
@ -8162,8 +8155,7 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
encoder_factory_),
std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
decoder_factory_),
field_trials_),
decoder_factory_)),
last_ssrc_(0) {}
void SetUp() override {
@ -8223,7 +8215,7 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
expected_streams = GetSimulcastConfig(
/*min_layers=*/1, num_configured_streams, capture_width,
capture_height, webrtc::kDefaultBitratePriority, kDefaultQpMax,
screenshare && conference_mode, true, field_trials_);
screenshare && conference_mode, true);
if (screenshare && conference_mode) {
for (const webrtc::VideoStream& stream : expected_streams) {
// Never scale screen content.
@ -8324,7 +8316,6 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory>
mock_rate_allocator_factory_;
webrtc::FieldTrialBasedConfig field_trials_;
WebRtcVideoEngine engine_;
std::unique_ptr<VideoMediaChannel> channel_;
uint32_t last_ssrc_;

50
media/engine/webrtc_voice_engine.cc
View File

@ -20,7 +20,6 @@
#include "absl/strings/match.h"
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/call/audio_sink.h"
#include "api/transport/webrtc_key_value_config.h"
#include "media/base/audio_source.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
@ -46,6 +45,7 @@
#include "rtc_base/strings/string_format.h"
#include "rtc_base/third_party/base64/base64.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#if WEBRTC_ENABLE_PROTOBUF
@ -111,6 +111,12 @@ std::string ToString(const AudioCodec& codec) {
return ss.Release();
}
// If this field trial is enabled, we will negotiate and use RFC 2198
// redundancy for opus audio.
bool IsAudioRedForOpusFieldTrialEnabled() {
return webrtc::field_trial::IsEnabled("WebRTC-Audio-Red-For-Opus");
}
bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return absl::EqualsIgnoreCase(codec.name, ref_name);
}
@ -197,11 +203,6 @@ absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
}
}
bool IsEnabled(const webrtc::WebRtcKeyValueConfig& config,
absl::string_view trial) {
return absl::StartsWith(config.Lookup(trial), "Enabled");
}
struct AdaptivePtimeConfig {
bool enabled = false;
webrtc::DataRate min_payload_bitrate = webrtc::DataRate::KilobitsPerSec(16);
@ -218,8 +219,9 @@ struct AdaptivePtimeConfig {
"use_slow_adaptation", &use_slow_adaptation);
}
explicit AdaptivePtimeConfig(const webrtc::WebRtcKeyValueConfig& trials) {
Parser()->Parse(trials.Lookup("WebRTC-Audio-AdaptivePtime"));
AdaptivePtimeConfig() {
Parser()->Parse(
webrtc::field_trial::FindFullName("WebRTC-Audio-AdaptivePtime"));
#if WEBRTC_ENABLE_PROTOBUF
webrtc::audio_network_adaptor::config::ControllerManager config;
auto* frame_length_controller =
@ -241,18 +243,13 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
const webrtc::WebRtcKeyValueConfig& trials)
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing)
: task_queue_factory_(task_queue_factory),
adm_(adm),
encoder_factory_(encoder_factory),
decoder_factory_(decoder_factory),
audio_mixer_(audio_mixer),
apm_(audio_processing),
audio_red_for_opus_trial_enabled_(
IsEnabled(trials, "WebRTC-Audio-Red-For-Opus")),
minimized_remsampling_on_mobile_trial_enabled_(
IsEnabled(trials, "WebRTC-Audio-MinimizeResamplingOnMobile")) {
apm_(audio_processing) {
// This may be called from any thread, so detach thread checkers.
worker_thread_checker_.Detach();
signal_thread_checker_.Detach();
@ -418,7 +415,8 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
// performed inside the audio processing module on mobile platforms by
// whenever possible turning off the fixed AGC mode and the high-pass filter.
// (https://bugs.chromium.org/p/webrtc/issues/detail?id=6181).
if (minimized_remsampling_on_mobile_trial_enabled_) {
if (webrtc::field_trial::IsEnabled(
"WebRTC-Audio-MinimizeResamplingOnMobile")) {
options.auto_gain_control = false;
RTC_LOG(LS_INFO) << "Disable AGC according to field trial.";
if (!(options.noise_suppression.value_or(false) ||
@ -724,7 +722,8 @@ std::vector<AudioCodec> WebRtcVoiceEngine::CollectCodecs(
out.push_back(codec);
if (codec.name == kOpusCodecName && audio_red_for_opus_trial_enabled_) {
if (codec.name == kOpusCodecName &&
IsAudioRedForOpusFieldTrialEnabled()) {
map_format({kRedCodecName, 48000, 2}, &out);
}
}
@ -768,8 +767,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
const absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<webrtc::FrameEncryptorInterface> frame_encryptor,
const webrtc::CryptoOptions& crypto_options)
: adaptive_ptime_config_(call->trials()),
call_(call),
: call_(call),
config_(send_transport),
max_send_bitrate_bps_(max_send_bitrate_bps),
rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
@ -1370,9 +1368,7 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(
engine_(engine),
call_(call),
audio_config_(config.audio),
crypto_options_(crypto_options),
audio_red_for_opus_trial_enabled_(
IsEnabled(call->trials(), "WebRTC-Audio-Red-For-Opus")) {
crypto_options_(crypto_options) {
RTC_LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
RTC_DCHECK(call);
engine->RegisterChannel(this);
@ -1419,8 +1415,7 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
}
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForAudio, true,
call_->trials());
params.extensions, webrtc::RtpExtension::IsSupportedForAudio, true);
if (send_rtp_extensions_ != filtered_extensions) {
send_rtp_extensions_.swap(filtered_extensions);
for (auto& it : send_streams_) {
@ -1457,8 +1452,7 @@ bool WebRtcVoiceMediaChannel::SetRecvParameters(
return false;
}
std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
params.extensions, webrtc::RtpExtension::IsSupportedForAudio, false,
call_->trials());
params.extensions, webrtc::RtpExtension::IsSupportedForAudio, false);
if (recv_rtp_extensions_ != filtered_extensions) {
recv_rtp_extensions_.swap(filtered_extensions);
for (auto& it : recv_streams_) {
@ -1634,7 +1628,7 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
}
auto format = AudioCodecToSdpAudioFormat(codec);
if (!IsCodec(codec, kCnCodecName) && !IsCodec(codec, kDtmfCodecName) &&
(!audio_red_for_opus_trial_enabled_ ||
(!IsAudioRedForOpusFieldTrialEnabled() ||
!IsCodec(codec, kRedCodecName)) &&
!engine()->decoder_factory_->IsSupportedDecoder(format)) {
RTC_LOG(LS_ERROR) << "Unsupported codec: " << rtc::ToString(format);
@ -1788,7 +1782,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
}
}
if (audio_red_for_opus_trial_enabled_) {
if (IsAudioRedForOpusFieldTrialEnabled()) {
// Loop through the codecs to find the RED codec that matches opus
// with respect to clockrate and number of channels.
size_t red_codec_position = 0;

11
media/engine/webrtc_voice_engine.h
View File

@ -20,7 +20,6 @@
#include "api/scoped_refptr.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/transport/webrtc_key_value_config.h"
#include "call/audio_state.h"
#include "call/call.h"
#include "media/base/media_engine.h"
@ -50,8 +49,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
const webrtc::WebRtcKeyValueConfig& trials);
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
~WebRtcVoiceEngine() override;
// Does initialization that needs to occur on the worker thread.
@ -129,11 +127,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
int audio_jitter_buffer_min_delay_ms_ = 0;
bool audio_jitter_buffer_enable_rtx_handling_ = false;
// If this field trial is enabled, we will negotiate and use RFC 2198
// redundancy for opus audio.
const bool audio_red_for_opus_trial_enabled_;
const bool minimized_remsampling_on_mobile_trial_enabled_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceEngine);
};
@ -338,8 +331,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
unsignaled_frame_decryptor_;
const bool audio_red_for_opus_trial_enabled_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
};
} // namespace cricket

37
media/engine/webrtc_voice_engine_unittest.cc
View File

@ -158,12 +158,10 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
EXPECT_CALL(*apm, DetachAecDump());
}
{
webrtc::FieldTrialBasedConfig trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
trials);
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
}
}
@ -212,7 +210,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
engine_.reset(new cricket::WebRtcVoiceEngine(
task_queue_factory_.get(), adm_, encoder_factory, decoder_factory,
nullptr, apm_, trials_config_));
nullptr, apm_));
engine_->Init();
send_parameters_.codecs.push_back(kPcmuCodec);
recv_parameters_.codecs.push_back(kPcmuCodec);
@ -803,7 +801,6 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
private:
webrtc::test::ScopedFieldTrials override_field_trials_;
webrtc::FieldTrialBasedConfig trials_config_;
};
INSTANTIATE_TEST_SUITE_P(TestBothWithAndWithoutNullApm,
@ -3631,15 +3628,14 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
webrtc::test::MockAudioDeviceModule::CreateNice();
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
field_trials);
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3663,15 +3659,14 @@ TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
{
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
field_trials);
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3697,12 +3692,10 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
webrtc::test::MockAudioDeviceModule::CreateNice();
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
field_trials);
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
for (const cricket::AudioCodec& codec : engine.send_codecs()) {
auto is_codec = [&codec](const char* name, int clockrate = 0) {
@ -3750,15 +3743,14 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
webrtc::test::MockAudioDeviceModule::CreateNice();
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
field_trials);
webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3799,14 +3791,14 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
webrtc::test::MockAudioDeviceModule::CreateNice();
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm,
webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, field_trials);
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm);
engine.Init();
webrtc::RtcEventLogNull event_log;
webrtc::Call::Config call_config(&event_log);
webrtc::FieldTrialBasedConfig field_trials;
call_config.trials = &field_trials;
call_config.task_queue_factory = task_queue_factory.get();
auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
@ -3851,10 +3843,9 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
rtc::scoped_refptr<webrtc::AudioProcessing> apm =
use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
webrtc::FieldTrialBasedConfig field_trials;
cricket::WebRtcVoiceEngine engine(
task_queue_factory.get(), adm, unused_encoder_factory,
mock_decoder_factory, nullptr, apm, field_trials);
cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), adm,
unused_encoder_factory,
mock_decoder_factory, nullptr, apm);
engine.Init();
auto codecs = engine.recv_codecs();
EXPECT_EQ(11u, codecs.size());

1
modules/video_coding/BUILD.gn
View File

@ -759,7 +759,6 @@ if (rtc_include_tests) {
"../../api:array_view",
"../../api:videocodec_test_fixture_api",
"../../api/test/video:function_video_factory",
"../../api/transport:field_trial_based_config",
"../../api/video:video_bitrate_allocation",
"../../api/video_codecs:video_codecs_api",
"../../call:video_stream_api",

4
modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
View File

@ -22,7 +22,6 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
@ -62,11 +61,10 @@ const int kMaxFramerateFps = 30;
const int kMaxQp = 56;
void ConfigureSimulcast(VideoCodec* codec_settings) {
FieldTrialBasedConfig trials;
const std::vector<webrtc::VideoStream> streams = cricket::GetSimulcastConfig(
/*min_layer=*/1, codec_settings->numberOfSimulcastStreams,
codec_settings->width, codec_settings->height, kBitratePriority, kMaxQp,
/* is_screenshare = */ false, true, trials);
/* is_screenshare = */ false, true);
for (size_t i = 0; i < streams.size(); ++i) {
SpatialLayer* ss = &codec_settings->simulcastStream[i];

1
pc/BUILD.gn
View File

@ -584,7 +584,6 @@ if (rtc_include_tests) {
"../api/rtc_event_log",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue:default_task_queue_factory",
"../api/transport:field_trial_based_config",
"../api/transport/rtp:rtp_source",
"../api/units:time_delta",
"../api/video:builtin_video_bitrate_allocator_factory",

10
pc/peer_connection.cc
View File

@ -58,6 +58,7 @@
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
@ -4343,9 +4344,7 @@ bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
bool PeerConnection::StartRtcEventLog(
std::unique_ptr<RtcEventLogOutput> output) {
int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
if (absl::StartsWith(
call_ptr_->trials().Lookup("WebRTC-RtcEventLogNewFormat"),
"Enabled")) {
if (field_trial::IsEnabled("WebRTC-RtcEventLogNewFormat")) {
output_period_ms = 5000;
}
return StartRtcEventLog(std::move(output), output_period_ms);
@ -5788,8 +5787,9 @@ PeerConnection::InitializePortAllocator_n(
// by experiment.
if (configuration.disable_ipv6) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
} else if (absl::StartsWith(call_ptr_->trials().Lookup("WebRTC-IPv6Default"),
"Disabled")) {
} else if (absl::StartsWith(
webrtc::field_trial::FindFullName("WebRTC-IPv6Default"),
"Disabled")) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}

2
pc/peer_connection_factory.h
View File

@ -87,8 +87,6 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
const Options& options() const { return options_; }
const WebRtcKeyValueConfig& trials() const { return *trials_.get(); }
protected:
// This structure allows simple management of all new dependencies being added
// to the PeerConnectionFactory.

4
pc/peer_connection_integrationtest.cc
View File

@ -28,7 +28,6 @@
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/rtp_receiver_interface.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/uma_metrics.h"
#include "api/video_codecs/sdp_video_format.h"
#include "call/call.h"
@ -634,7 +633,6 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
pc_factory_dependencies.signaling_thread = signaling_thread;
pc_factory_dependencies.task_queue_factory =
webrtc::CreateDefaultTaskQueueFactory();
pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory =
pc_factory_dependencies.task_queue_factory.get();
@ -654,8 +652,6 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
media_deps.audio_processing = AudioProcessingBuilderForTesting().Create();
}
media_deps.trials = pc_factory_dependencies.trials.get();
pc_factory_dependencies.media_engine =
cricket::CreateMediaEngine(std::move(media_deps));
pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();

3
pc/peer_connection_interface_unittest.cc
View File

@ -43,7 +43,6 @@
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
@ -647,14 +646,12 @@ class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
dependencies.network_thread = rtc::Thread::Current();
dependencies.signaling_thread = rtc::Thread::Current();
dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = dependencies.task_queue_factory.get();
// Use fake audio device module since we're only testing the interface
// level, and using a real one could make tests flaky when run in parallel.
media_deps.adm = FakeAudioCaptureModule::Create();
SetMediaEngineDefaults(&media_deps);
media_deps.trials = dependencies.trials.get();
dependencies.media_engine =
cricket::CreateMediaEngine(std::move(media_deps));
dependencies.call_factory = webrtc::CreateCallFactory();

3
pc/peer_connection_jsep_unittest.cc
View File

@ -11,7 +11,6 @@
#include <memory>
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/engine/webrtc_media_engine_defaults.h"
#include "pc/media_session.h"
@ -48,11 +47,9 @@ PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() {
dependencies.network_thread = rtc::Thread::Current();
dependencies.signaling_thread = rtc::Thread::Current();
dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = dependencies.task_queue_factory.get();
media_deps.adm = FakeAudioCaptureModule::Create();
media_deps.trials = dependencies.trials.get();
SetMediaEngineDefaults(&media_deps);
dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
dependencies.call_factory = CreateCallFactory();

1
sdk/BUILD.gn
View File

@ -973,7 +973,6 @@ if (is_ios || is_mac) {
"../api/crypto:frame_encryptor_interface",
"../api/rtc_event_log:rtc_event_log_factory",
"../api/task_queue:default_task_queue_factory",
"../api/transport:field_trial_based_config",
"../api/video:video_frame",
"../api/video:video_rtp_headers",
"../api/video_codecs:video_codecs_api",

3
sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
View File

@ -37,7 +37,6 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h" // nogncheck
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "modules/audio_device/include/audio_device.h" // nogncheck
#include "modules/audio_processing/include/audio_processing.h" // nogncheck
@ -191,7 +190,6 @@
}
#ifndef HAVE_NO_MEDIA
dependencies.task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_deps;
media_deps.adm = std::move(audioDeviceModule);
media_deps.task_queue_factory = dependencies.task_queue_factory.get();
@ -204,7 +202,6 @@
} else {
media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
}
media_deps.trials = dependencies.trials.get();
dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
dependencies.call_factory = webrtc::CreateCallFactory();
dependencies.event_log_factory =

1
test/network/BUILD.gn
View File

@ -93,7 +93,6 @@ rtc_library("network_emulation_pc_unittest") {
"../../api:simulated_network_api",
"../../api/rtc_event_log:rtc_event_log_factory",
"../../api/task_queue:default_task_queue_factory",
"../../api/transport:field_trial_based_config",
"../../call:simulated_network",
"../../media:rtc_audio_video",
"../../media:rtc_media_engine_defaults",

3
test/network/network_emulation_pc_unittest.cc
View File

@ -16,7 +16,6 @@
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/scoped_refptr.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "call/simulated_network.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/engine/webrtc_media_engine_defaults.h"
@ -60,7 +59,6 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
std::make_unique<RtcEventLogFactory>(pcf_deps.task_queue_factory.get());
pcf_deps.network_thread = network_thread;
pcf_deps.signaling_thread = signaling_thread;
pcf_deps.trials = std::make_unique<FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
media_deps.adm = TestAudioDeviceModule::Create(
@ -69,7 +67,6 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
kSamplingFrequency),
TestAudioDeviceModule::CreateDiscardRenderer(kSamplingFrequency),
/*speed=*/1.f);
media_deps.trials = pcf_deps.trials.get();
SetMediaEngineDefaults(&media_deps);
pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
return CreateModularPeerConnectionFactory(std::move(pcf_deps));

2
test/pc/e2e/BUILD.gn
View File

@ -58,7 +58,6 @@ if (!build_with_chromium) {
"../../../api/rtc_event_log",
"../../../api/task_queue",
"../../../api/transport:network_control",
"../../../api/transport:webrtc_key_value_config",
"../../../api/video_codecs:video_codecs_api",
"../../../rtc_base",
]
@ -270,7 +269,6 @@ if (!build_with_chromium) {
"../../../api:time_controller",
"../../../api/rtc_event_log:rtc_event_log_factory",
"../../../api/task_queue:default_task_queue_factory",
"../../../api/transport:field_trial_based_config",
"../../../api/video_codecs:builtin_video_decoder_factory",
"../../../api/video_codecs:builtin_video_encoder_factory",
"../../../media:rtc_audio_video",

3
test/pc/e2e/peer_connection_quality_test_params.h
View File

@ -21,7 +21,6 @@
#include "api/task_queue/task_queue_factory.h"
#include "api/test/peerconnection_quality_test_fixture.h"
#include "api/transport/network_control.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "rtc_base/network.h"
@ -53,8 +52,6 @@ struct PeerConnectionFactoryComponents {
// PeerConnectionFactory.
std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
std::unique_ptr<WebRtcKeyValueConfig> trials;
};
// Contains most parts from PeerConnectionDependencies. Also all fields are

11
test/pc/e2e/test_peer_factory.cc
View File

@ -16,7 +16,6 @@
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/create_time_controller.h"
#include "api/test/time_controller.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "media/engine/webrtc_media_engine.h"
@ -65,10 +64,6 @@ void SetMandatoryEntities(InjectableComponents* components,
std::make_unique<RtcEventLogFactory>(
components->pcf_dependencies->task_queue_factory.get());
}
if (!components->pcf_dependencies->trials) {
components->pcf_dependencies->trials =
std::make_unique<FieldTrialBasedConfig>();
}
}
// Returns mapping from stream label to optional spatial index.
@ -178,9 +173,6 @@ std::unique_ptr<cricket::MediaEngineInterface> CreateMediaEngine(
media_deps.video_decoder_factory =
std::move(pcf_dependencies->video_decoder_factory);
webrtc::SetMediaEngineDefaults(&media_deps);
RTC_DCHECK(pcf_dependencies->trials);
media_deps.trials = pcf_dependencies->trials.get();
return cricket::CreateMediaEngine(std::move(media_deps));
}
@ -246,9 +238,6 @@ PeerConnectionFactoryDependencies CreatePCFDependencies(
if (pcf_dependencies->neteq_factory != nullptr) {
pcf_deps.neteq_factory = std::move(pcf_dependencies->neteq_factory);
}
if (pcf_dependencies->trials != nullptr) {
pcf_deps.trials = std::move(pcf_dependencies->trials);
}
return pcf_deps;
}

1
test/peer_scenario/BUILD.gn
View File

@ -37,7 +37,6 @@ if (rtc_include_tests) {
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../api/rtc_event_log:rtc_event_log_factory",
"../../api/task_queue:default_task_queue_factory",
"../../api/transport:field_trial_based_config",
"../../api/video_codecs:builtin_video_decoder_factory",
"../../api/video_codecs:builtin_video_encoder_factory",
"../../media:rtc_audio_video",

3
test/peer_scenario/peer_scenario_client.cc
View File

@ -18,7 +18,6 @@
#include "api/rtc_event_log/rtc_event_log_factory.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/test/create_time_controller.h"
#include "api/transport/field_trial_based_config.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "media/engine/webrtc_media_engine.h"
@ -198,7 +197,6 @@ PeerScenarioClient::PeerScenarioClient(
net->time_controller()->CreateTaskQueueFactory();
pcf_deps.event_log_factory =
std::make_unique<RtcEventLogFactory>(task_queue_factory_);
pcf_deps.trials = std::make_unique<FieldTrialBasedConfig>();
cricket::MediaEngineDependencies media_deps;
media_deps.task_queue_factory = task_queue_factory_;
@ -223,7 +221,6 @@ PeerScenarioClient::PeerScenarioClient(
}
media_deps.audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
media_deps.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
media_deps.trials = pcf_deps.trials.get();
pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
pcf_deps.fec_controller_factory = nullptr;