mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions

WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf

Browse source 
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
This commit is contained in:
Jonas Oreland 2022-03-29 11:04:48 +02:00 committed by WebRTC LUCI CQ
parent bddfa1d9de
commit e62c2f2c77
303 changed files with 785 additions and 831 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

12
api/BUILD.gn
View file

@ -171,6 +171,7 @@ rtc_library("libjingle_peerconnection_api") {
":audio_options_api",
":callfactory_api",
":fec_controller_api",
":field_trials_view",
":frame_transformer_interface",
":libjingle_logging_api",
":media_stream_interface",
@ -200,7 +201,6 @@ rtc_library("libjingle_peerconnection_api") {
"transport:enums",
"transport:network_control",
"transport:sctp_transport_factory_interface",
"transport:webrtc_key_value_config",
"transport/rtp:rtp_source",
"units:data_rate",
"units:timestamp",
@ -1224,9 +1224,15 @@ if (rtc_include_tests) {
}
}
rtc_source_set("webrtc_key_value_config") {
rtc_source_set("field_trials_view") {
visibility = [ "*" ]
sources = [ "webrtc_key_value_config.h" ]
sources = [ "field_trials_view.h" ]
deps = [ "../rtc_base/system:rtc_export" ]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
rtc_source_set("webrtc_key_value_config") {
visibility = [ "*" ]
sources = [ "webrtc_key_value_config.h" ]
deps = [ ":field_trials_view" ]
}

2
api/audio_codecs/BUILD.gn
View file

@ -32,7 +32,7 @@ rtc_library("audio_codecs_api") {
"..:array_view",
"..:bitrate_allocation",
"..:scoped_refptr",
"../../api:webrtc_key_value_config",
"../../api:field_trials_view",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:sanitizer",

4
api/audio_codecs/L16/BUILD.gn
View file

@ -21,7 +21,7 @@ rtc_library("audio_encoder_L16") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
@ -42,7 +42,7 @@ rtc_library("audio_decoder_L16") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:pcm16b",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",

2
api/audio_codecs/L16/audio_decoder_L16.cc
View file

@ -38,7 +38,7 @@ void AudioDecoderL16::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
const Config& config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
return nullptr;
}

4
api/audio_codecs/L16/audio_decoder_L16.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -41,7 +41,7 @@ struct RTC_EXPORT AudioDecoderL16 {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/L16/audio_encoder_L16.cc
View file

@ -60,7 +60,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
const AudioEncoderL16::Config& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
AudioEncoderPcm16B::Config c;
c.sample_rate_hz = config.sample_rate_hz;
c.num_channels = config.num_channels;

4
api/audio_codecs/L16/audio_encoder_L16.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -46,7 +46,7 @@ struct RTC_EXPORT AudioEncoderL16 {
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

12
api/audio_codecs/audio_decoder_factory_template.h
View file

@ -15,8 +15,8 @@
#include <vector>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
#include "api/webrtc_key_value_config.h"
#include "rtc_base/ref_counted_object.h"
namespace webrtc {
@ -34,7 +34,7 @@ struct Helper<> {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
return nullptr;
}
};
@ -58,7 +58,7 @@ struct Helper<T, Ts...> {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
auto opt_config = T::SdpToConfig(format);
return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
: Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
@ -69,7 +69,7 @@ struct Helper<T, Ts...> {
template <typename... Ts>
class AudioDecoderFactoryT : public AudioDecoderFactory {
public:
explicit AudioDecoderFactoryT(const WebRtcKeyValueConfig* field_trials) {
explicit AudioDecoderFactoryT(const FieldTrialsView* field_trials) {
field_trials_ = field_trials;
}
@ -90,7 +90,7 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
field_trials_);
}
const WebRtcKeyValueConfig* field_trials_;
const FieldTrialsView* field_trials_;
};
} // namespace audio_decoder_factory_template_impl
@ -127,7 +127,7 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
// how it is used.
template <typename... Ts>
rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory(
const WebRtcKeyValueConfig* field_trials = nullptr) {
const FieldTrialsView* field_trials = nullptr) {
// There's no technical reason we couldn't allow zero template parameters,
// but such a factory couldn't create any decoders, and callers can do this
// by mistake by simply forgetting the <> altogether. So we forbid it in

12
api/audio_codecs/audio_encoder_factory_template.h
View file

@ -15,8 +15,8 @@
#include <vector>
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
#include "api/webrtc_key_value_config.h"
#include "rtc_base/ref_counted_object.h"
namespace webrtc {
@ -38,7 +38,7 @@ struct Helper<> {
int payload_type,
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
return nullptr;
}
};
@ -66,7 +66,7 @@ struct Helper<T, Ts...> {
int payload_type,
const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
auto opt_config = T::SdpToConfig(format);
if (opt_config) {
return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id);
@ -80,7 +80,7 @@ struct Helper<T, Ts...> {
template <typename... Ts>
class AudioEncoderFactoryT : public AudioEncoderFactory {
public:
explicit AudioEncoderFactoryT(const WebRtcKeyValueConfig* field_trials) {
explicit AudioEncoderFactoryT(const FieldTrialsView* field_trials) {
field_trials_ = field_trials;
}
@ -103,7 +103,7 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
field_trials_);
}
const WebRtcKeyValueConfig* field_trials_;
const FieldTrialsView* field_trials_;
};
} // namespace audio_encoder_factory_template_impl
@ -145,7 +145,7 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
// how it is used.
template <typename... Ts>
rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory(
const WebRtcKeyValueConfig* field_trials = nullptr) {
const FieldTrialsView* field_trials = nullptr) {
// There's no technical reason we couldn't allow zero template parameters,
// but such a factory couldn't create any encoders, and callers can do this
// by mistake by simply forgetting the <> altogether. So we forbid it in

2
api/audio_codecs/builtin_audio_encoder_factory.cc
View file

@ -48,7 +48,7 @@ struct NotAdvertised {
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr) {
const FieldTrialsView* field_trials = nullptr) {
return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
field_trials);
}

4
api/audio_codecs/g711/BUILD.gn
View file

@ -21,7 +21,7 @@ rtc_library("audio_encoder_g711") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
@ -42,7 +42,7 @@ rtc_library("audio_decoder_g711") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:g711",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",

2
api/audio_codecs/g711/audio_decoder_g711.cc
View file

@ -48,7 +48,7 @@ void AudioDecoderG711::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
const Config& config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/g711/audio_decoder_g711.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -41,7 +41,7 @@ struct RTC_EXPORT AudioDecoderG711 {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
const Config& config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/g711/audio_encoder_g711.cc
View file

@ -65,7 +65,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/g711/audio_encoder_g711.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -46,7 +46,7 @@ struct RTC_EXPORT AudioEncoderG711 {
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

4
api/audio_codecs/g722/BUILD.gn
View file

@ -27,7 +27,7 @@ rtc_library("audio_encoder_g722") {
deps = [
":audio_encoder_g722_config",
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:g722",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
@ -48,7 +48,7 @@ rtc_library("audio_decoder_g722") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:g722",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",

2
api/audio_codecs/g722/audio_decoder_g722.cc
View file

@ -37,7 +37,7 @@ void AudioDecoderG722::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/g722/audio_decoder_g722.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -35,7 +35,7 @@ struct RTC_EXPORT AudioDecoderG722 {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/g722/audio_encoder_g722.cc
View file

@ -63,7 +63,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
const AudioEncoderG722Config& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/g722/audio_encoder_g722.h
View file

@ -19,7 +19,7 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -36,7 +36,7 @@ struct RTC_EXPORT AudioEncoderG722 {
const AudioEncoderG722Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

4
api/audio_codecs/ilbc/BUILD.gn
View file

@ -27,7 +27,7 @@ rtc_library("audio_encoder_ilbc") {
deps = [
":audio_encoder_ilbc_config",
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:ilbc",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
@ -47,7 +47,7 @@ rtc_library("audio_decoder_ilbc") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:ilbc",
"../../../rtc_base:rtc_base_approved",
]

2
api/audio_codecs/ilbc/audio_decoder_ilbc.cc
View file

@ -35,7 +35,7 @@ void AudioDecoderIlbc::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
return std::make_unique<AudioDecoderIlbcImpl>();
}

4
api/audio_codecs/ilbc/audio_decoder_ilbc.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
namespace webrtc {
@ -31,7 +31,7 @@ struct AudioDecoderIlbc {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/ilbc/audio_encoder_ilbc.cc
View file

@ -77,7 +77,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
const AudioEncoderIlbcConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/ilbc/audio_encoder_ilbc.h
View file

@ -19,7 +19,7 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
namespace webrtc {
@ -35,7 +35,7 @@ struct AudioEncoderIlbc {
const AudioEncoderIlbcConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

8
api/audio_codecs/isac/BUILD.gn
View file

@ -65,7 +65,7 @@ rtc_library("audio_encoder_isac_fix") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:isac_fix",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",
@ -85,7 +85,7 @@ rtc_library("audio_decoder_isac_fix") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:isac_fix",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",
@ -105,7 +105,7 @@ rtc_library("audio_encoder_isac_float") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:isac",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",
@ -125,7 +125,7 @@ rtc_library("audio_decoder_isac_float") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:isac",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",

2
api/audio_codecs/isac/audio_decoder_isac_fix.cc
View file

@ -34,7 +34,7 @@ void AudioDecoderIsacFix::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderIsacFix::MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
AudioDecoderIsacFixImpl::Config c;
c.sample_rate_hz = 16000;
return std::make_unique<AudioDecoderIsacFixImpl>(c);

4
api/audio_codecs/isac/audio_decoder_isac_fix.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -32,7 +32,7 @@ struct RTC_EXPORT AudioDecoderIsacFix {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/isac/audio_decoder_isac_float.cc
View file

@ -43,7 +43,7 @@ void AudioDecoderIsacFloat::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderIsacFloat::MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
AudioDecoderIsacFloatImpl::Config c;
c.sample_rate_hz = config.sample_rate_hz;
if (!config.IsOk()) {

4
api/audio_codecs/isac/audio_decoder_isac_float.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -37,7 +37,7 @@ struct RTC_EXPORT AudioDecoderIsacFloat {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/isac/audio_encoder_isac_fix.cc
View file

@ -57,7 +57,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
AudioEncoderIsacFix::Config config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
AudioEncoderIsacFixImpl::Config c;
c.frame_size_ms = config.frame_size_ms;
c.bit_rate = config.bit_rate;

4
api/audio_codecs/isac/audio_encoder_isac_fix.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -46,7 +46,7 @@ struct RTC_EXPORT AudioEncoderIsacFix {
Config config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/isac/audio_encoder_isac_float.cc
View file

@ -69,7 +69,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
const AudioEncoderIsacFloat::Config& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
AudioEncoderIsacFloatImpl::Config c;
c.payload_type = payload_type;
c.sample_rate_hz = config.sample_rate_hz;

4
api/audio_codecs/isac/audio_encoder_isac_float.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -60,7 +60,7 @@ struct RTC_EXPORT AudioEncoderIsacFloat {
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

8
api/audio_codecs/opus/BUILD.gn
View file

@ -46,7 +46,7 @@ rtc_library("audio_encoder_opus") {
deps = [
":audio_encoder_opus_config",
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:webrtc_opus",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",
@ -66,7 +66,7 @@ rtc_library("audio_decoder_opus") {
]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:webrtc_opus",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",
@ -84,7 +84,7 @@ rtc_library("audio_encoder_multiopus") {
sources = [ "audio_encoder_multi_channel_opus.cc" ]
deps = [
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:webrtc_multiopus",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",
@ -103,7 +103,7 @@ rtc_library("audio_decoder_multiopus") {
deps = [
":audio_decoder_opus_config",
"..:audio_codecs_api",
"../../../api:webrtc_key_value_config",
"../../../api:field_trials_view",
"../../../modules/audio_coding:webrtc_multiopus",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/system:rtc_export",

2
api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
View file

@ -65,7 +65,7 @@ void AudioDecoderMultiChannelOpus::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
AudioDecoderMultiChannelOpusConfig config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
}
} // namespace webrtc

4
api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
View file

@ -19,7 +19,7 @@
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -34,7 +34,7 @@ struct RTC_EXPORT AudioDecoderMultiChannelOpus {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
AudioDecoderMultiChannelOpusConfig config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/opus/audio_decoder_opus.cc
View file

@ -74,7 +74,7 @@ void AudioDecoderOpus::AppendSupportedDecoders(
std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/opus/audio_decoder_opus.h
View file

@ -18,7 +18,7 @@
#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_codecs/audio_decoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -36,7 +36,7 @@ struct RTC_EXPORT AudioDecoderOpus {
static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
Config config,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc
View file

@ -67,7 +67,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
const AudioEncoderMultiChannelOpusConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
payload_type);
}

4
api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
View file

@ -19,7 +19,7 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -35,7 +35,7 @@ struct RTC_EXPORT AudioEncoderMultiChannelOpus {
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/opus/audio_encoder_opus.cc
View file

@ -33,7 +33,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
const AudioEncoderOpusConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> /*codec_pair_id*/,
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
if (!config.IsOk()) {
RTC_DCHECK_NOTREACHED();
return nullptr;

4
api/audio_codecs/opus/audio_encoder_opus.h
View file

@ -19,7 +19,7 @@
#include "api/audio_codecs/audio_encoder.h"
#include "api/audio_codecs/audio_format.h"
#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@ -36,7 +36,7 @@ struct RTC_EXPORT AudioEncoderOpus {
const AudioEncoderOpusConfig& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
};
} // namespace webrtc

2
api/audio_codecs/opus_audio_encoder_factory.cc
View file

@ -38,7 +38,7 @@ struct NotAdvertised {
const Config& config,
int payload_type,
absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
const WebRtcKeyValueConfig* field_trials = nullptr) {
const FieldTrialsView* field_trials = nullptr) {
return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
field_trials);
}

46
api/field_trials_view.h Normal file
View file

@ -0,0 +1,46 @@
/*
* Copyright 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_FIELD_TRIALS_VIEW_H_
#define API_FIELD_TRIALS_VIEW_H_
#include <string>
#include "absl/strings/string_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// An interface that provides a key-value mapping for configuring internal
// details of WebRTC. Note that there's no guarantess that the meaning of a
// particular key value mapping will be preserved over time and no announcements
// will be made if they are changed. It's up to the library user to ensure that
// the behavior does not break.
class RTC_EXPORT FieldTrialsView {
public:
virtual ~FieldTrialsView() = default;
// The configured value for the given key. Defaults to an empty string.
virtual std::string Lookup(absl::string_view key) const = 0;
bool IsEnabled(absl::string_view key) const {
return Lookup(key).find("Enabled") == 0;
}
bool IsDisabled(absl::string_view key) const {
return Lookup(key).find("Disabled") == 0;
}
};
// TODO(bugs.webrtc.org/10335): Remove once all migrated to
// api/field_trials_view.h
typedef FieldTrialsView WebRtcKeyValueConfig;
} // namespace webrtc
#endif // API_FIELD_TRIALS_VIEW_H_

8
api/ice_transport_interface.h
View file

@ -27,7 +27,7 @@ class IceControllerFactoryInterface;
} // namespace cricket
namespace webrtc {
class WebRtcKeyValueConfig;
class FieldTrialsView;
// An ICE transport, as represented to the outside world.
// This object is refcounted, and is therefore alive until the
@ -84,8 +84,8 @@ struct IceTransportInit final {
return ice_controller_factory_;
}
const WebRtcKeyValueConfig* field_trials() { return field_trials_; }
void set_field_trials(const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials() { return field_trials_; }
void set_field_trials(const FieldTrialsView* field_trials) {
field_trials_ = field_trials;
}
@ -96,7 +96,7 @@ struct IceTransportInit final {
AsyncResolverFactory* async_resolver_factory_ = nullptr;
RtcEventLog* event_log_ = nullptr;
cricket::IceControllerFactoryInterface* ice_controller_factory_ = nullptr;
const WebRtcKeyValueConfig* field_trials_ = nullptr;
const FieldTrialsView* field_trials_ = nullptr;
// TODO(https://crbug.com/webrtc/12657): Redesign to have const members.
};

4
api/peer_connection_interface.h
View file

@ -91,6 +91,7 @@
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
@ -117,7 +118,6 @@
#include "api/transport/enums.h"
#include "api/transport/network_control.h"
#include "api/transport/sctp_transport_factory_interface.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/turn_customizer.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/rtp_transport_controller_send_factory_interface.h"
@ -1435,7 +1435,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
std::unique_ptr<rtc::NetworkMonitorFactory> network_monitor_factory;
std::unique_ptr<NetEqFactory> neteq_factory;
std::unique_ptr<SctpTransportFactoryInterface> sctp_factory;
std::unique_ptr<WebRtcKeyValueConfig> trials;
std::unique_ptr<FieldTrialsView> trials;
std::unique_ptr<RtpTransportControllerSendFactoryInterface>
transport_controller_send_factory;
std::unique_ptr<Metronome> metronome;

16
api/transport/BUILD.gn
View file

@ -32,7 +32,7 @@ rtc_library("network_control") {
]
deps = [
":webrtc_key_value_config",
"../../api:field_trials_view",
"../rtc_event_log",
"../units:data_rate",
"../units:data_size",
@ -45,16 +45,6 @@ rtc_library("network_control") {
]
}
rtc_source_set("webrtc_key_value_config") {
visibility = [ "*" ]
sources = [ "webrtc_key_value_config.h" ]
deps = [
"../../api:webrtc_key_value_config",
"../../rtc_base/system:rtc_export",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
rtc_library("field_trial_based_config") {
visibility = [ "*" ]
sources = [
@ -62,7 +52,7 @@ rtc_library("field_trial_based_config") {
"field_trial_based_config.h",
]
deps = [
":webrtc_key_value_config",
"../../api:field_trials_view",
"../../system_wrappers:field_trial",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
@ -88,8 +78,8 @@ rtc_library("goog_cc") {
]
deps = [
":network_control",
":webrtc_key_value_config",
"..:network_state_predictor_api",
"../../api:field_trials_view",
"../../modules/congestion_controller/goog_cc",
]
absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]

4
api/transport/field_trial_based_config.h
View file

@ -13,11 +13,11 @@
#include <string>
#include "absl/strings/string_view.h"
#include "api/transport/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
namespace webrtc {
// Implementation using the field trial API fo the key value lookup.
class FieldTrialBasedConfig : public WebRtcKeyValueConfig {
class FieldTrialBasedConfig : public FieldTrialsView {
public:
std::string Lookup(absl::string_view key) const override;
};

6
api/transport/network_control.h
View file

@ -15,9 +15,9 @@
#include <memory>
#include "absl/base/attributes.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/network_types.h"
#include "api/transport/webrtc_key_value_config.h"
namespace webrtc {
@ -46,7 +46,7 @@ struct NetworkControllerConfig {
// Optional override of configuration of WebRTC internals. Using nullptr here
// indicates that the field trial API will be used.
const WebRtcKeyValueConfig* key_value_config = nullptr;
const FieldTrialsView* key_value_config = nullptr;
// Optional override of event log.
RtcEventLog* event_log = nullptr;
};
@ -132,7 +132,7 @@ class NetworkStateEstimator {
class NetworkStateEstimatorFactory {
public:
virtual std::unique_ptr<NetworkStateEstimator> Create(
const WebRtcKeyValueConfig* key_value_config) = 0;
const FieldTrialsView* key_value_config) = 0;
virtual ~NetworkStateEstimatorFactory() = default;
};
} // namespace webrtc

17
api/transport/webrtc_key_value_config.h
View file

@ -1,17 +0,0 @@
/*
* Copyright 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_
#define API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_
// TODO(bugs.webrtc.org/10335): Remove once all migrated to
// "api/webrtc_key_value_config.h".
#include "api/webrtc_key_value_config.h"
#endif // API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_

2
api/video/BUILD.gn
View file

@ -269,7 +269,7 @@ rtc_library("video_stream_decoder_create") {
deps = [
":video_stream_decoder",
"../../api:webrtc_key_value_config",
"../../api:field_trials_view",
"../../rtc_base:rtc_base_approved",
"../../video:video_stream_decoder_impl",
"../task_queue",

2
api/video/video_stream_decoder_create.cc
View file

@ -23,7 +23,7 @@ std::unique_ptr<VideoStreamDecoderInterface> CreateVideoStreamDecoder(
std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings,
// TODO(jonaso, webrtc:10335): Consider what to do with factories
// vs. field trials.
const WebRtcKeyValueConfig* field_trials) {
const FieldTrialsView* field_trials) {
return std::make_unique<VideoStreamDecoderImpl>(
callbacks, decoder_factory, task_queue_factory,
std::move(decoder_settings), field_trials);

4
api/video/video_stream_decoder_create.h
View file

@ -15,10 +15,10 @@
#include <memory>
#include <utility>
#include "api/field_trials_view.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/video/video_stream_decoder.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/webrtc_key_value_config.h"
namespace webrtc {
// The `decoder_settings` parameter is a map between:
@ -30,7 +30,7 @@ std::unique_ptr<VideoStreamDecoderInterface> CreateVideoStreamDecoder(
VideoDecoderFactory* decoder_factory,
TaskQueueFactory* task_queue_factory,
std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings,
const WebRtcKeyValueConfig* field_trials = nullptr);
const FieldTrialsView* field_trials = nullptr);
} // namespace webrtc

30
api/webrtc_key_value_config.h
View file

@ -10,32 +10,8 @@
#ifndef API_WEBRTC_KEY_VALUE_CONFIG_H_
#define API_WEBRTC_KEY_VALUE_CONFIG_H_
#include <string>
#include "absl/strings/string_view.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// An interface that provides a key-value mapping for configuring internal
// details of WebRTC. Note that there's no guarantess that the meaning of a
// particular key value mapping will be preserved over time and no announcements
// will be made if they are changed. It's up to the library user to ensure that
// the behavior does not break.
class RTC_EXPORT WebRtcKeyValueConfig {
public:
virtual ~WebRtcKeyValueConfig() = default;
// The configured value for the given key. Defaults to an empty string.
virtual std::string Lookup(absl::string_view key) const = 0;
bool IsEnabled(absl::string_view key) const {
return Lookup(key).find("Enabled") == 0;
}
bool IsDisabled(absl::string_view key) const {
return Lookup(key).find("Disabled") == 0;
}
};
} // namespace webrtc
// TODO(bugs.webrtc.org/10335): Remove once all migrated to
// api/field_trials_view.h
#include "api/field_trials_view.h"
#endif // API_WEBRTC_KEY_VALUE_CONFIG_H_

2
audio/BUILD.gn
View file

@ -42,6 +42,7 @@ rtc_library("audio") {
deps = [
"../api:array_view",
"../api:call_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:function_view",
"../api:rtp_headers",
@ -49,7 +50,6 @@ rtc_library("audio") {
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
"../api:webrtc_key_value_config",
"../api/audio:aec3_factory",
"../api/audio:audio_frame_api",
"../api/audio:audio_frame_processor",

6
audio/audio_send_stream.cc
View file

@ -88,7 +88,7 @@ std::unique_ptr<StructParametersParser> AudioAllocationConfig::Parser() {
}
AudioAllocationConfig::AudioAllocationConfig(
const WebRtcKeyValueConfig& field_trials) {
const FieldTrialsView& field_trials) {
Parser()->Parse(field_trials.Lookup(kKey));
if (priority_bitrate_raw && !priority_bitrate.IsZero()) {
RTC_LOG(LS_WARNING) << "'priority_bitrate' and '_raw' are mutually "
@ -107,7 +107,7 @@ AudioSendStream::AudioSendStream(
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: AudioSendStream(
clock,
config,
@ -142,7 +142,7 @@ AudioSendStream::AudioSendStream(
RtcEventLog* event_log,
const absl::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send,
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: clock_(clock),
field_trials_(field_trials),
rtp_transport_queue_(rtp_transport->GetWorkerQueue()),

10
audio/audio_send_stream.h
View file

@ -15,8 +15,8 @@
#include <utility>
#include <vector>
#include "api/field_trials_view.h"
#include "api/sequence_checker.h"
#include "api/webrtc_key_value_config.h"
#include "audio/audio_level.h"
#include "audio/channel_send.h"
#include "call/audio_send_stream.h"
@ -47,7 +47,7 @@ struct AudioAllocationConfig {
absl::optional<double> bitrate_priority;
std::unique_ptr<StructParametersParser> Parser();
explicit AudioAllocationConfig(const WebRtcKeyValueConfig& field_trials);
explicit AudioAllocationConfig(const FieldTrialsView& field_trials);
};
namespace internal {
class AudioState;
@ -64,7 +64,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats,
const absl::optional<RtpState>& suspended_rtp_state,
const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
// For unit tests, which need to supply a mock ChannelSend.
AudioSendStream(Clock* clock,
const webrtc::AudioSendStream::Config& config,
@ -75,7 +75,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
RtcEventLog* event_log,
const absl::optional<RtpState>& suspended_rtp_state,
std::unique_ptr<voe::ChannelSendInterface> channel_send,
const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
AudioSendStream() = delete;
AudioSendStream(const AudioSendStream&) = delete;
@ -163,7 +163,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
RTC_RUN_ON(worker_thread_checker_);
Clock* clock_;
const WebRtcKeyValueConfig& field_trials_;
const FieldTrialsView& field_trials_;
SequenceChecker worker_thread_checker_;
SequenceChecker pacer_thread_checker_;

6
audio/channel_send.cc
View file

@ -78,7 +78,7 @@ class ChannelSend : public ChannelSendInterface,
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
TransportFeedbackObserver* feedback_observer,
const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
~ChannelSend() override;
@ -459,7 +459,7 @@ ChannelSend::ChannelSend(
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
TransportFeedbackObserver* feedback_observer,
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: ssrc_(ssrc),
event_log_(rtc_event_log),
_timeStamp(0), // This is just an offset, RTP module will add it's own
@ -950,7 +950,7 @@ std::unique_ptr<ChannelSendInterface> CreateChannelSend(
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
TransportFeedbackObserver* feedback_observer,
const WebRtcKeyValueConfig& field_trials) {
const FieldTrialsView& field_trials) {
return std::make_unique<ChannelSend>(
clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log,
frame_encryptor, crypto_options, extmap_allow_mixed,

4
audio/channel_send.h
View file

@ -18,10 +18,10 @@
#include "api/audio/audio_frame.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/crypto/crypto_options.h"
#include "api/field_trials_view.h"
#include "api/frame_transformer_interface.h"
#include "api/function_view.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/webrtc_key_value_config.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
@ -137,7 +137,7 @@ std::unique_ptr<ChannelSendInterface> CreateChannelSend(
uint32_t ssrc,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
TransportFeedbackObserver* feedback_observer,
const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
} // namespace voe
} // namespace webrtc

8
call/BUILD.gn
View file

@ -42,6 +42,7 @@ rtc_library("call_interfaces") {
":rtp_interfaces",
":video_stream_api",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:network_state_predictor_api",
"../api:rtc_error",
@ -60,7 +61,6 @@ rtc_library("call_interfaces") {
"../api/task_queue",
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../api/transport:webrtc_key_value_config",
"../modules/async_audio_processing",
"../modules/audio_device",
"../modules/audio_processing",
@ -106,6 +106,7 @@ rtc_library("rtp_interfaces") {
deps = [
"../api:array_view",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:network_state_predictor_api",
"../api:rtp_headers",
@ -114,7 +115,6 @@ rtc_library("rtp_interfaces") {
"../api/rtc_event_log",
"../api/transport:bitrate_settings",
"../api/transport:network_control",
"../api/transport:webrtc_key_value_config",
"../api/units:timestamp",
"../common_video:frame_counts",
"../modules/rtp_rtcp:rtp_rtcp_format",
@ -174,11 +174,11 @@ rtc_library("rtp_sender") {
"../api:array_view",
"../api:bitrate_allocation",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:network_state_predictor_api",
"../api:rtp_parameters",
"../api:sequence_checker",
"../api:transport_api",
"../api:webrtc_key_value_config",
"../api/rtc_event_log",
"../api/transport:field_trial_based_config",
"../api/transport:goog_cc",
@ -286,12 +286,12 @@ rtc_library("call") {
"../api:array_view",
"../api:callfactory_api",
"../api:fec_controller_api",
"../api:field_trials_view",
"../api:rtp_headers",
"../api:rtp_parameters",
"../api:sequence_checker",
"../api:simulated_network_api",
"../api:transport_api",
"../api:webrtc_key_value_config",
"../api/rtc_event_log",
"../api/transport:network_control",
"../api/units:time_delta",

2
call/adaptation/BUILD.gn
View file

@ -32,10 +32,10 @@ rtc_library("resource_adaptation") {
"video_stream_input_state_provider.h",
]
deps = [
"../../api:field_trials_view",
"../../api:rtp_parameters",
"../../api:scoped_refptr",
"../../api:sequence_checker",
"../../api:webrtc_key_value_config",
"../../api/adaptation:resource_adaptation_api",
"../../api/task_queue:task_queue",
"../../api/video:video_adaptation",

2
call/adaptation/video_stream_adapter.cc
View file

@ -204,7 +204,7 @@ const VideoAdaptationCounters& Adaptation::counters() const {
VideoStreamAdapter::VideoStreamAdapter(
VideoStreamInputStateProvider* input_state_provider,
VideoStreamEncoderObserver* encoder_stats_observer,
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: input_state_provider_(input_state_provider),
encoder_stats_observer_(encoder_stats_observer),
balanced_settings_(field_trials),

4
call/adaptation/video_stream_adapter.h
View file

@ -18,10 +18,10 @@
#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/adaptation/resource.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/video/video_adaptation_counters.h"
#include "api/video/video_stream_encoder_observer.h"
#include "api/webrtc_key_value_config.h"
#include "call/adaptation/adaptation_constraint.h"
#include "call/adaptation/degradation_preference_provider.h"
#include "call/adaptation/video_source_restrictions.h"
@ -125,7 +125,7 @@ class VideoStreamAdapter {
public:
VideoStreamAdapter(VideoStreamInputStateProvider* input_state_provider,
VideoStreamEncoderObserver* encoder_stats_observer,
const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
~VideoStreamAdapter();
VideoSourceRestrictions source_restrictions() const;

6
call/call.cc
View file

@ -247,7 +247,7 @@ class Call final : public webrtc::Call,
Stats GetStats() const override;
const WebRtcKeyValueConfig& trials() const override;
const FieldTrialsView& trials() const override;
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;
@ -379,7 +379,7 @@ class Call final : public webrtc::Call,
const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
const Call::Config config_ RTC_GUARDED_BY(worker_thread_);
// Maps to config_.trials, can be used from any thread via `trials()`.
const WebRtcKeyValueConfig& trials_;
const FieldTrialsView& trials_;
NetworkState audio_network_state_ RTC_GUARDED_BY(worker_thread_);
NetworkState video_network_state_ RTC_GUARDED_BY(worker_thread_);
@ -1295,7 +1295,7 @@ Call::Stats Call::GetStats() const {
return stats;
}
const WebRtcKeyValueConfig& Call::trials() const {
const FieldTrialsView& Call::trials() const {
return trials_;
}

2
call/call.h
View file

@ -174,7 +174,7 @@ class Call {
virtual void SetClientBitratePreferences(
const BitrateSettings& preferences) = 0;
virtual const WebRtcKeyValueConfig& trials() const = 0;
virtual const FieldTrialsView& trials() const = 0;
virtual TaskQueueBase* network_thread() const = 0;
virtual TaskQueueBase* worker_thread() const = 0;

4
call/call_config.h
View file

@ -11,6 +11,7 @@
#define CALL_CALL_CONFIG_H_
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/metronome/metronome.h"
#include "api/neteq/neteq_factory.h"
#include "api/network_state_predictor.h"
@ -18,7 +19,6 @@
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/transport/webrtc_key_value_config.h"
#include "call/audio_state.h"
#include "call/rtp_transport_config.h"
#include "call/rtp_transport_controller_send_factory_interface.h"
@ -70,7 +70,7 @@ struct CallConfig {
// Key-value mapping of internal configurations to apply,
// e.g. field trials.
const WebRtcKeyValueConfig* trials = nullptr;
const FieldTrialsView* trials = nullptr;
TaskQueueBase* const network_task_queue_ = nullptr;
// RtpTransportControllerSend to use for this call.

6
call/call_factory.cc
View file

@ -31,7 +31,7 @@ namespace webrtc {
namespace {
using TimeScopedNetworkConfig = DegradedCall::TimeScopedNetworkConfig;
bool ParseConfigParam(const WebRtcKeyValueConfig& trials,
bool ParseConfigParam(const FieldTrialsView& trials,
absl::string_view exp_name,
int* field) {
std::string group = trials.Lookup(exp_name);
@ -42,7 +42,7 @@ bool ParseConfigParam(const WebRtcKeyValueConfig& trials,
}
absl::optional<TimeScopedNetworkConfig> ParseDegradationConfig(
const WebRtcKeyValueConfig& trials,
const FieldTrialsView& trials,
bool send) {
std::string exp_prefix = "WebRTCFakeNetwork";
if (send) {
@ -80,7 +80,7 @@ absl::optional<TimeScopedNetworkConfig> ParseDegradationConfig(
}
std::vector<TimeScopedNetworkConfig> GetNetworkConfigs(
const WebRtcKeyValueConfig& trials,
const FieldTrialsView& trials,
bool send) {
FieldTrialStructList<TimeScopedNetworkConfig> trials_list(
{FieldTrialStructMember("queue_length_packets",

2
call/degraded_call.cc
View file

@ -277,7 +277,7 @@ Call::Stats DegradedCall::GetStats() const {
return call_->GetStats();
}
const WebRtcKeyValueConfig& DegradedCall::trials() const {
const FieldTrialsView& DegradedCall::trials() const {
return call_->trials();
}

2
call/degraded_call.h
View file

@ -90,7 +90,7 @@ class DegradedCall : public Call, private PacketReceiver {
Stats GetStats() const override;
const WebRtcKeyValueConfig& trials() const override;
const FieldTrialsView& trials() const override;
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;

6
call/receive_time_calculator.cc
View file

@ -24,7 +24,7 @@ const char kBweReceiveTimeCorrection[] = "WebRTC-Bwe-ReceiveTimeFix";
} // namespace
ReceiveTimeCalculatorConfig::ReceiveTimeCalculatorConfig(
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: max_packet_time_repair("maxrep", TimeDelta::Millis(2000)),
stall_threshold("stall", TimeDelta::Millis(5)),
tolerance("tol", TimeDelta::Millis(1)),
@ -39,12 +39,12 @@ ReceiveTimeCalculatorConfig::ReceiveTimeCalculatorConfig(
ReceiveTimeCalculatorConfig::~ReceiveTimeCalculatorConfig() = default;
ReceiveTimeCalculator::ReceiveTimeCalculator(
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: config_(field_trials) {}
std::unique_ptr<ReceiveTimeCalculator>
ReceiveTimeCalculator::CreateFromFieldTrial(
const WebRtcKeyValueConfig& field_trials) {
const FieldTrialsView& field_trials) {
if (!field_trials.IsEnabled(kBweReceiveTimeCorrection))
return nullptr;
return std::make_unique<ReceiveTimeCalculator>(field_trials);

9
call/receive_time_calculator.h
View file

@ -14,15 +14,14 @@
#include <memory>
#include "api/field_trials_view.h"
#include "api/units/time_delta.h"
#include "api/webrtc_key_value_config.h"
#include "rtc_base/experiments/field_trial_parser.h"
namespace webrtc {
struct ReceiveTimeCalculatorConfig {
explicit ReceiveTimeCalculatorConfig(
const WebRtcKeyValueConfig& field_trials);
explicit ReceiveTimeCalculatorConfig(const FieldTrialsView& field_trials);
ReceiveTimeCalculatorConfig(const ReceiveTimeCalculatorConfig&);
ReceiveTimeCalculatorConfig& operator=(const ReceiveTimeCalculatorConfig&) =
default;
@ -44,8 +43,8 @@ struct ReceiveTimeCalculatorConfig {
class ReceiveTimeCalculator {
public:
static std::unique_ptr<ReceiveTimeCalculator> CreateFromFieldTrial(
const WebRtcKeyValueConfig& field_trials);
explicit ReceiveTimeCalculator(const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
explicit ReceiveTimeCalculator(const FieldTrialsView& field_trials);
int64_t ReconcileReceiveTimes(int64_t packet_time_us_,
int64_t system_time_us_,
int64_t safe_time_us_);

2
call/rtp_payload_params.cc
View file

@ -127,7 +127,7 @@ void SetVideoTiming(const EncodedImage& image, VideoSendTiming* timing) {
RtpPayloadParams::RtpPayloadParams(const uint32_t ssrc,
const RtpPayloadState* state,
const WebRtcKeyValueConfig& trials)
const FieldTrialsView& trials)
: ssrc_(ssrc),
generic_picture_id_experiment_(
absl::StartsWith(trials.Lookup("WebRTC-GenericPictureId"),

4
call/rtp_payload_params.h
View file

@ -15,8 +15,8 @@
#include <vector>
#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/video_codecs/video_encoder.h"
#include "api/webrtc_key_value_config.h"
#include "call/rtp_config.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
@ -34,7 +34,7 @@ class RtpPayloadParams final {
public:
RtpPayloadParams(uint32_t ssrc,
const RtpPayloadState* state,
const WebRtcKeyValueConfig& trials);
const FieldTrialsView& trials);
RtpPayloadParams(const RtpPayloadParams& other);
~RtpPayloadParams();

4
call/rtp_transport_config.h
View file

@ -13,11 +13,11 @@
#include <memory>
#include "api/field_trials_view.h"
#include "api/network_state_predictor.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/transport/webrtc_key_value_config.h"
#include "rtc_base/task_queue.h"
namespace webrtc {
@ -43,7 +43,7 @@ struct RtpTransportConfig {
// Key-value mapping of internal configurations to apply,
// e.g. field trials.
const WebRtcKeyValueConfig* trials = nullptr;
const FieldTrialsView* trials = nullptr;
};
} // namespace webrtc

8
call/rtp_transport_controller_send.cc
View file

@ -59,11 +59,11 @@ TargetRateConstraints ConvertConstraints(const BitrateConstraints& contraints,
contraints.start_bitrate_bps, clock);
}
bool IsEnabled(const WebRtcKeyValueConfig& trials, absl::string_view key) {
bool IsEnabled(const FieldTrialsView& trials, absl::string_view key) {
return absl::StartsWith(trials.Lookup(key), "Enabled");
}
bool IsDisabled(const WebRtcKeyValueConfig& trials, absl::string_view key) {
bool IsDisabled(const FieldTrialsView& trials, absl::string_view key) {
return absl::StartsWith(trials.Lookup(key), "Disabled");
}
@ -74,7 +74,7 @@ bool IsRelayed(const rtc::NetworkRoute& route) {
} // namespace
RtpTransportControllerSend::PacerSettings::PacerSettings(
const WebRtcKeyValueConfig& trials)
const FieldTrialsView& trials)
: tq_disabled("Disabled"),
holdback_window("holdback_window", PacingController::kMinSleepTime),
holdback_packets("holdback_packets",
@ -91,7 +91,7 @@ RtpTransportControllerSend::RtpTransportControllerSend(
const BitrateConstraints& bitrate_config,
std::unique_ptr<ProcessThread> process_thread,
TaskQueueFactory* task_queue_factory,
const WebRtcKeyValueConfig& trials)
const FieldTrialsView& trials)
: clock_(clock),
event_log_(event_log),
bitrate_configurator_(bitrate_config),

6
call/rtp_transport_controller_send.h
View file

@ -59,7 +59,7 @@ class RtpTransportControllerSend final
const BitrateConstraints& bitrate_config,
std::unique_ptr<ProcessThread> process_thread,
TaskQueueFactory* task_queue_factory,
const WebRtcKeyValueConfig& trials);
const FieldTrialsView& trials);
~RtpTransportControllerSend() override;
RtpTransportControllerSend(const RtpTransportControllerSend&) = delete;
@ -132,7 +132,7 @@ class RtpTransportControllerSend final
private:
struct PacerSettings {
explicit PacerSettings(const WebRtcKeyValueConfig& trials);
explicit PacerSettings(const FieldTrialsView& trials);
bool use_task_queue_pacer() const { return !tq_disabled.Get(); }
@ -223,7 +223,7 @@ class RtpTransportControllerSend final
// and deleted before any other members.
rtc::TaskQueue task_queue_;
const WebRtcKeyValueConfig& field_trials_;
const FieldTrialsView& field_trials_;
};
} // namespace webrtc

10
call/rtp_video_sender.cc
View file

@ -56,7 +56,7 @@ static const size_t kPathMTU = 1500;
using webrtc_internal_rtp_video_sender::RtpStreamSender;
bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name,
const WebRtcKeyValueConfig& trials) {
const FieldTrialsView& trials) {
const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
return true;
@ -70,7 +70,7 @@ bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name,
bool ShouldDisableRedAndUlpfec(bool flexfec_enabled,
const RtpConfig& rtp_config,
const WebRtcKeyValueConfig& trials) {
const FieldTrialsView& trials) {
// Consistency of NACK and RED+ULPFEC parameters is checked in this function.
const bool nack_enabled = rtp_config.nack.rtp_history_ms > 0;
@ -126,7 +126,7 @@ std::unique_ptr<VideoFecGenerator> MaybeCreateFecGenerator(
const RtpConfig& rtp,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
int simulcast_index,
const WebRtcKeyValueConfig& trials) {
const FieldTrialsView& trials) {
// If flexfec is configured that takes priority.
if (rtp.flexfec.payload_type >= 0) {
RTC_DCHECK_GE(rtp.flexfec.payload_type, 0);
@ -197,7 +197,7 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const WebRtcKeyValueConfig& trials) {
const FieldTrialsView& trials) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RtpRtcpInterface::Configuration configuration;
@ -359,7 +359,7 @@ RtpVideoSender::RtpVideoSender(
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const WebRtcKeyValueConfig& field_trials)
const FieldTrialsView& field_trials)
: field_trials_(field_trials),
send_side_bwe_with_overhead_(!absl::StartsWith(
field_trials_.Lookup("WebRTC-SendSideBwe-WithOverhead"),

6
call/rtp_video_sender.h
View file

@ -21,10 +21,10 @@
#include "api/call/transport.h"
#include "api/fec_controller.h"
#include "api/fec_controller_override.h"
#include "api/field_trials_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/sequence_checker.h"
#include "api/video_codecs/video_encoder.h"
#include "api/webrtc_key_value_config.h"
#include "call/rtp_config.h"
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
@ -86,7 +86,7 @@ class RtpVideoSender : public RtpVideoSenderInterface,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options, // move inside RtpTransport
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const WebRtcKeyValueConfig& field_trials);
const FieldTrialsView& field_trials);
~RtpVideoSender() override;
RtpVideoSender(const RtpVideoSender&) = delete;
@ -167,7 +167,7 @@ class RtpVideoSender : public RtpVideoSenderInterface,
DataSize overhead_per_packet,
Frequency framerate) const;
const WebRtcKeyValueConfig& field_trials_;
const FieldTrialsView& field_trials_;
const bool send_side_bwe_with_overhead_;
const bool use_frame_rate_for_overhead_;
const bool has_packet_feedback_;

6
call/rtp_video_sender_unittest.cc
View file

@ -119,7 +119,7 @@ class RtpVideoSenderTestFixture {
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
FrameCountObserver* frame_count_observer,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const WebRtcKeyValueConfig* field_trials = nullptr)
const FieldTrialsView* field_trials = nullptr)
: time_controller_(Timestamp::Millis(1000000)),
config_(CreateVideoSendStreamConfig(&transport_,
ssrcs,
@ -162,7 +162,7 @@ class RtpVideoSenderTestFixture {
int payload_type,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
FrameCountObserver* frame_count_observer,
const WebRtcKeyValueConfig* field_trials = nullptr)
const FieldTrialsView* field_trials = nullptr)
: RtpVideoSenderTestFixture(ssrcs,
rtx_ssrcs,
payload_type,
@ -176,7 +176,7 @@ class RtpVideoSenderTestFixture {
const std::vector<uint32_t>& rtx_ssrcs,
int payload_type,
const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
const WebRtcKeyValueConfig* field_trials = nullptr)
const FieldTrialsView* field_trials = nullptr)
: RtpVideoSenderTestFixture(ssrcs,
rtx_ssrcs,
payload_type,

6
media/BUILD.gn
View file

@ -51,13 +51,13 @@ rtc_library("rtc_media_base") {
":rtc_media_config",
"../api:array_view",
"../api:audio_options_api",
"../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:webrtc_key_value_config",
"../api/audio:audio_frame_processor",
"../api/audio_codecs:audio_codecs_api",
"../api/crypto:frame_decryptor_interface",
@ -249,6 +249,7 @@ rtc_library("rtc_audio_video") {
deps = [
":rtc_media_base",
"../api:call_api",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
"../api:media_stream_interface",
"../api:rtp_parameters",
@ -261,7 +262,6 @@ rtc_library("rtc_audio_video") {
"../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",
@ -453,7 +453,7 @@ rtc_library("rtc_data_sctp_transport_factory") {
]
deps = [
":rtc_data_sctp_transport_internal",
"../api:webrtc_key_value_config",
"../api:field_trials_view",
"../api/transport:sctp_transport_factory_interface",
"../rtc_base:threading",
"../rtc_base/experiments:field_trial_parser",

12
media/base/codec.cc
View file

@ -129,7 +129,7 @@ bool Codec::operator==(const Codec& c) const {
}
bool Codec::Matches(const Codec& codec,
const webrtc::WebRtcKeyValueConfig* field_trials) const {
const webrtc::FieldTrialsView* field_trials) const {
// Match the codec id/name based on the typical static/dynamic name rules.
// Matching is case-insensitive.
@ -238,9 +238,8 @@ bool AudioCodec::operator==(const AudioCodec& c) const {
return bitrate == c.bitrate && channels == c.channels && Codec::operator==(c);
}
bool AudioCodec::Matches(
const AudioCodec& codec,
const webrtc::WebRtcKeyValueConfig* field_trials) const {
bool AudioCodec::Matches(const AudioCodec& codec,
const webrtc::FieldTrialsView* field_trials) const {
// If a nonzero clockrate is specified, it must match the actual clockrate.
// If a nonzero bitrate is specified, it must match the actual bitrate,
// unless the codec is VBR (0), where we just force the supplied value.
@ -326,9 +325,8 @@ bool VideoCodec::operator==(const VideoCodec& c) const {
return Codec::operator==(c) && packetization == c.packetization;
}
bool VideoCodec::Matches(
const VideoCodec& other,
const webrtc::WebRtcKeyValueConfig* field_trials) const {
bool VideoCodec::Matches(const VideoCodec& other,
const webrtc::FieldTrialsView* field_trials) const {
return Codec::Matches(other, field_trials) &&
IsSameCodecSpecific(name, params, other.name, other.params);
}

17
media/base/codec.h
View file

@ -17,9 +17,9 @@
#include <vector>
#include "absl/types/optional.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/webrtc_key_value_config.h"
#include "media/base/media_constants.h"
#include "rtc_base/system/rtc_export.h"
@ -76,9 +76,8 @@ struct RTC_EXPORT Codec {
virtual ~Codec();
// Indicates if this codec is compatible with the specified codec.
bool Matches(
const Codec& codec,
const webrtc::WebRtcKeyValueConfig* field_trials = nullptr) const;
bool Matches(const Codec& codec,
const webrtc::FieldTrialsView* field_trials = nullptr) const;
bool MatchesCapability(const webrtc::RtpCodecCapability& capability) const;
// Find the parameter for `name` and write the value to `out`.
@ -135,9 +134,8 @@ struct AudioCodec : public Codec {
~AudioCodec() override = default;
// Indicates if this codec is compatible with the specified codec.
bool Matches(
const AudioCodec& codec,
const webrtc::WebRtcKeyValueConfig* field_trials = nullptr) const;
bool Matches(const AudioCodec& codec,
const webrtc::FieldTrialsView* field_trials = nullptr) const;
std::string ToString() const;
@ -168,9 +166,8 @@ struct RTC_EXPORT VideoCodec : public Codec {
// Indicates if this video codec is the same as the other video codec, e.g. if
// they are both VP8 or VP9, or if they are both H264 with the same H264
// profile. H264 levels however are not compared.
bool Matches(
const VideoCodec& codec,
const webrtc::WebRtcKeyValueConfig* field_trials = nullptr) const;
bool Matches(const VideoCodec& codec,
const webrtc::FieldTrialsView* field_trials = nullptr) const;
std::string ToString() const;

2
media/base/media_engine.cc
View file

@ -152,7 +152,7 @@ webrtc::RTCError CheckRtpParametersInvalidModificationAndValues(
}
CompositeMediaEngine::CompositeMediaEngine(
std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials,
std::unique_ptr<webrtc::FieldTrialsView> trials,
std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine)
: trials_(std::move(trials)),

8
media/base/media_engine.h
View file

@ -18,9 +18,9 @@
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/crypto/crypto_options.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/webrtc_key_value_config.h"
#include "call/audio_state.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
@ -137,10 +137,10 @@ class MediaEngineInterface {
// CompositeMediaEngine constructs a MediaEngine from separate
// voice and video engine classes.
// Optionally owns a WebRtcKeyValueConfig trials map.
// Optionally owns a FieldTrialsView trials map.
class CompositeMediaEngine : public MediaEngineInterface {
public:
CompositeMediaEngine(std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials,
CompositeMediaEngine(std::unique_ptr<webrtc::FieldTrialsView> trials,
std::unique_ptr<VoiceEngineInterface> audio_engine,
std::unique_ptr<VideoEngineInterface> video_engine);
CompositeMediaEngine(std::unique_ptr<VoiceEngineInterface> audio_engine,
@ -156,7 +156,7 @@ class CompositeMediaEngine : public MediaEngineInterface {
const VideoEngineInterface& video() const override;
private:
const std::unique_ptr<webrtc::WebRtcKeyValueConfig> trials_;
const std::unique_ptr<webrtc::FieldTrialsView> trials_;
const std::unique_ptr<VoiceEngineInterface> voice_engine_;
const std::unique_ptr<VideoEngineInterface> video_engine_;
};

2
media/base/test_utils.h
View file

@ -39,7 +39,7 @@ inline std::vector<T> MakeVector(const T a[], size_t s) {
template <class C>
bool ContainsMatchingCodec(const std::vector<C>& codecs,
const C& codec,
const webrtc::WebRtcKeyValueConfig* field_trials) {
const webrtc::FieldTrialsView* field_trials) {
typename std::vector<C>::const_iterator it;
for (it = codecs.begin(); it != codecs.end(); ++it) {
if (it->Matches(codec, field_trials)) {

4
media/engine/fake_webrtc_call.h
View file

@ -360,9 +360,7 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
*trials_, field_trial_string);
}
const webrtc::WebRtcKeyValueConfig& trials() const override {
return *trials_;
}
const webrtc::FieldTrialsView& trials() const override { return *trials_; }
private:
webrtc::AudioSendStream* CreateAudioSendStream(

13
media/engine/simulcast.cc
View file

@ -45,8 +45,7 @@ constexpr char kUseLegacySimulcastLayerLimitFieldTrial[] =
constexpr double kDefaultMaxRoundupRate = 0.1;
// TODO(webrtc:12415): Flip this to a kill switch when this feature launches.
bool EnableLowresBitrateInterpolation(
const webrtc::WebRtcKeyValueConfig& trials) {
bool EnableLowresBitrateInterpolation(const webrtc::FieldTrialsView& trials) {
return absl::StartsWith(
trials.Lookup("WebRTC-LowresSimulcastBitrateInterpolation"), "Enabled");
}
@ -134,7 +133,7 @@ const int kMaxScreenshareSimulcastLayers = 2;
// Multiway: Number of temporal layers for each simulcast stream.
int DefaultNumberOfTemporalLayers(int simulcast_id,
bool screenshare,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
RTC_CHECK_GE(simulcast_id, 0);
RTC_CHECK_LT(simulcast_id, webrtc::kMaxSimulcastStreams);
@ -291,7 +290,7 @@ size_t LimitSimulcastLayerCount(int width,
int height,
size_t need_layers,
size_t layer_count,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
if (!absl::StartsWith(trials.Lookup(kUseLegacySimulcastLayerLimitFieldTrial),
"Disabled")) {
// Max layers from one higher resolution in kSimulcastFormats will be used
@ -327,7 +326,7 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(
int max_qp,
bool is_screenshare_with_conference_mode,
bool temporal_layers_supported,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
RTC_DCHECK_LE(min_layers, max_layers);
RTC_DCHECK(max_layers > 1 || is_screenshare_with_conference_mode);
@ -359,7 +358,7 @@ std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
std::vector<webrtc::VideoStream> layers(layer_count);
const bool enable_lowres_bitrate_interpolation =
@ -451,7 +450,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
auto max_screenshare_layers = kMaxScreenshareSimulcastLayers;
size_t num_simulcast_layers =
std::min<int>(max_layers, max_screenshare_layers);

8
media/engine/simulcast.h
View file

@ -15,7 +15,7 @@
#include <vector>
#include "api/transport/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "api/units/data_rate.h"
#include "api/video_codecs/video_encoder_config.h"
@ -43,7 +43,7 @@ std::vector<webrtc::VideoStream> GetSimulcastConfig(
int max_qp,
bool is_screenshare_with_conference_mode,
bool temporal_layers_supported,
const webrtc::WebRtcKeyValueConfig& trials);
const webrtc::FieldTrialsView& trials);
// Gets the simulcast config layers for a non-screensharing case.
std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
@ -54,7 +54,7 @@ std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials);
const webrtc::FieldTrialsView& trials);
// Gets simulcast config layers for screenshare settings.
std::vector<webrtc::VideoStream> GetScreenshareLayers(
@ -65,7 +65,7 @@ std::vector<webrtc::VideoStream> GetScreenshareLayers(
int max_qp,
bool temporal_layers_supported,
bool base_heavy_tl3_rate_alloc,
const webrtc::WebRtcKeyValueConfig& trials);
const webrtc::FieldTrialsView& trials);
} // namespace cricket

6
media/engine/webrtc_media_engine.cc
View file

@ -30,9 +30,9 @@ 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(
std::unique_ptr<webrtc::FieldTrialsView> fallback_trials(
dependencies.trials ? nullptr : new webrtc::FieldTrialBasedConfig());
const webrtc::WebRtcKeyValueConfig& trials =
const webrtc::FieldTrialsView& trials =
dependencies.trials ? *dependencies.trials : *fallback_trials;
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
dependencies.task_queue_factory, std::move(dependencies.adm),
@ -137,7 +137,7 @@ std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions,
bool (*supported)(absl::string_view),
bool filter_redundant_extensions,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
// Don't check against old parameters; this should have been done earlier.
RTC_DCHECK(ValidateRtpExtensions(extensions, {}));
RTC_DCHECK(supported);

6
media/engine/webrtc_media_engine.h
View file

@ -19,11 +19,11 @@
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/field_trials_view.h"
#include "api/rtp_parameters.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/field_trial_based_config.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"
@ -53,7 +53,7 @@ struct MediaEngineDependencies {
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory;
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory;
const webrtc::WebRtcKeyValueConfig* trials = nullptr;
const webrtc::FieldTrialsView* trials = nullptr;
};
// CreateMediaEngine may be called on any thread, though the engine is
@ -76,7 +76,7 @@ std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions,
bool (*supported)(absl::string_view),
bool filter_redundant_extensions,
const webrtc::WebRtcKeyValueConfig& trials);
const webrtc::FieldTrialsView& trials);
webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec);

16
media/engine/webrtc_video_engine.cc
View file

@ -82,13 +82,11 @@ const char* StreamTypeToString(
return nullptr;
}
bool IsEnabled(const webrtc::WebRtcKeyValueConfig& trials,
absl::string_view name) {
bool IsEnabled(const webrtc::FieldTrialsView& trials, absl::string_view name) {
return absl::StartsWith(trials.Lookup(name), "Enabled");
}
bool IsDisabled(const webrtc::WebRtcKeyValueConfig& trials,
absl::string_view name) {
bool IsDisabled(const webrtc::FieldTrialsView& trials, absl::string_view name) {
return absl::StartsWith(trials.Lookup(name), "Disabled");
}
@ -107,7 +105,7 @@ bool IsScaleFactorsPowerOfTwo(const webrtc::VideoEncoderConfig& config) {
}
void AddDefaultFeedbackParams(VideoCodec* codec,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
// Don't add any feedback params for RED and ULPFEC.
if (codec->name == kRedCodecName || codec->name == kUlpfecCodecName)
return;
@ -164,7 +162,7 @@ template <class T>
std::vector<VideoCodec> GetPayloadTypesAndDefaultCodecs(
const T* factory,
bool is_decoder_factory,
const webrtc::WebRtcKeyValueConfig& trials) {
const webrtc::FieldTrialsView& trials) {
if (!factory) {
return {};
}
@ -331,7 +329,7 @@ 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) {
const webrtc::FieldTrialsView& trials) {
if (absl::EqualsIgnoreCase(codec_name, kVp9CodecName) ||
absl::EqualsIgnoreCase(codec_name, kAv1CodecName)) {
return true;
@ -612,7 +610,7 @@ 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)
const webrtc::FieldTrialsView& trials)
: decoder_factory_(std::move(video_decoder_factory)),
encoder_factory_(std::move(video_encoder_factory)),
trials_(trials) {
@ -3498,7 +3496,7 @@ EncoderStreamFactory::EncoderStreamFactory(
int max_qp,
bool is_screenshare,
bool conference_mode,
const webrtc::WebRtcKeyValueConfig* trials)
const webrtc::FieldTrialsView* trials)
: codec_name_(codec_name),
max_qp_(max_qp),

8
media/engine/webrtc_video_engine.h
View file

@ -94,7 +94,7 @@ class WebRtcVideoEngine : public VideoEngineInterface {
WebRtcVideoEngine(
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
const webrtc::WebRtcKeyValueConfig& trials);
const webrtc::FieldTrialsView& trials);
~WebRtcVideoEngine() override;
@ -116,7 +116,7 @@ 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_;
const webrtc::FieldTrialsView& trials_;
};
class WebRtcVideoChannel : public VideoMediaChannel,
@ -658,7 +658,7 @@ class EncoderStreamFactory
int max_qp,
bool is_screenshare,
bool conference_mode,
const webrtc::WebRtcKeyValueConfig* trials);
const webrtc::FieldTrialsView* trials);
private:
std::vector<webrtc::VideoStream> CreateEncoderStreams(
@ -686,7 +686,7 @@ class EncoderStreamFactory
// layering and various settings.
const bool conference_mode_;
const webrtc::FieldTrialBasedConfig fallback_trials_;
const webrtc::WebRtcKeyValueConfig& trials_;
const webrtc::FieldTrialsView& trials_;
};
} // namespace cricket

11
media/engine/webrtc_voice_engine.cc
View file

@ -23,7 +23,7 @@
#include "api/audio/audio_frame_processor.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 "api/field_trials_view.h"
#include "media/base/audio_source.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
@ -126,7 +126,7 @@ bool IsCodec(const AudioCodec& codec, const char* ref_name) {
bool FindCodec(const std::vector<AudioCodec>& codecs,
const AudioCodec& codec,
AudioCodec* found_codec,
const webrtc::WebRtcKeyValueConfig* field_trials) {
const webrtc::FieldTrialsView* field_trials) {
for (const AudioCodec& c : codecs) {
if (c.Matches(codec, field_trials)) {
if (found_codec != NULL) {
@ -206,8 +206,7 @@ absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
}
}
bool IsEnabled(const webrtc::WebRtcKeyValueConfig& config,
absl::string_view trial) {
bool IsEnabled(const webrtc::FieldTrialsView& config, absl::string_view trial) {
return absl::StartsWith(config.Lookup(trial), "Enabled");
}
@ -229,7 +228,7 @@ struct AdaptivePtimeConfig {
"use_slow_adaptation", &use_slow_adaptation);
}
explicit AdaptivePtimeConfig(const webrtc::WebRtcKeyValueConfig& trials) {
explicit AdaptivePtimeConfig(const webrtc::FieldTrialsView& trials) {
Parser()->Parse(trials.Lookup("WebRTC-Audio-AdaptivePtime"));
#if WEBRTC_ENABLE_PROTOBUF
webrtc::audio_network_adaptor::config::ControllerManager config;
@ -299,7 +298,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
webrtc::AudioFrameProcessor* audio_frame_processor,
const webrtc::WebRtcKeyValueConfig& trials)
const webrtc::FieldTrialsView& trials)
: task_queue_factory_(task_queue_factory),
adm_(adm),
encoder_factory_(encoder_factory),

4
media/engine/webrtc_voice_engine.h
View file

@ -17,11 +17,11 @@
#include <vector>
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/field_trials_view.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.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"
@ -55,7 +55,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
webrtc::AudioFrameProcessor* audio_frame_processor,
const webrtc::WebRtcKeyValueConfig& trials);
const webrtc::FieldTrialsView& trials);
WebRtcVoiceEngine() = delete;
WebRtcVoiceEngine(const WebRtcVoiceEngine&) = delete;

4
media/sctp/sctp_transport_factory.cc
View file

@ -10,7 +10,7 @@
#include "media/sctp/sctp_transport_factory.h"
#include "api/webrtc_key_value_config.h"
#include "api/field_trials_view.h"
#include "rtc_base/system/unused.h"
#ifdef WEBRTC_HAVE_DCSCTP
@ -26,7 +26,7 @@ namespace cricket {
SctpTransportFactory::SctpTransportFactory(
rtc::Thread* network_thread,
const webrtc::WebRtcKeyValueConfig& field_trials)
const webrtc::FieldTrialsView& field_trials)
: network_thread_(network_thread), use_usrsctp_("Disabled", false) {
RTC_UNUSED(network_thread_);
#ifdef WEBRTC_HAVE_DCSCTP

7
media/sctp/sctp_transport_factory.h
View file

@ -13,8 +13,8 @@
#include <memory>
#include "api/field_trials_view.h"
#include "api/transport/sctp_transport_factory_interface.h"
#include "api/webrtc_key_value_config.h"
#include "media/sctp/sctp_transport_internal.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/thread.h"
@ -23,9 +23,8 @@ namespace cricket {
class SctpTransportFactory : public webrtc::SctpTransportFactoryInterface {
public:
explicit SctpTransportFactory(
rtc::Thread* network_thread,
const webrtc::WebRtcKeyValueConfig& field_trials);
explicit SctpTransportFactory(rtc::Thread* network_thread,
const webrtc::FieldTrialsView& field_trials);
std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
rtc::PacketTransportInternal* transport) override;

2
modules/audio_coding/BUILD.gn
View file

@ -118,7 +118,7 @@ rtc_library("red") {
deps = [
"../../api:array_view",
"../../api:webrtc_key_value_config",
"../../api:field_trials_view",
"../../api/audio_codecs:audio_codecs_api",
"../../api/units:time_delta",
"../../common_audio",

8
modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
View file

@ -39,8 +39,7 @@ AudioEncoderCopyRed::Config::Config() = default;
AudioEncoderCopyRed::Config::Config(Config&&) = default;
AudioEncoderCopyRed::Config::~Config() = default;
size_t GetMaxRedundancyFromFieldTrial(
const WebRtcKeyValueConfig& field_trials) {
size_t GetMaxRedundancyFromFieldTrial(const FieldTrialsView& field_trials) {
const std::string red_trial =
field_trials.Lookup("WebRTC-Audio-Red-For-Opus");
size_t redundancy = 0;
@ -51,9 +50,8 @@ size_t GetMaxRedundancyFromFieldTrial(
return redundancy;
}
AudioEncoderCopyRed::AudioEncoderCopyRed(
Config&& config,
const WebRtcKeyValueConfig& field_trials)
AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config,
const FieldTrialsView& field_trials)
: speech_encoder_(std::move(config.speech_encoder)),
primary_encoded_(0, kAudioMaxRtpPacketLen),
max_packet_length_(kAudioMaxRtpPacketLen),

5
modules/audio_coding/codecs/red/audio_encoder_copy_red.h
View file

@ -21,8 +21,8 @@
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/field_trials_view.h"
#include "api/units/time_delta.h"
#include "api/webrtc_key_value_config.h"
#include "rtc_base/buffer.h"
namespace webrtc {
@ -43,8 +43,7 @@ class AudioEncoderCopyRed final : public AudioEncoder {
std::unique_ptr<AudioEncoder> speech_encoder;
};
AudioEncoderCopyRed(Config&& config,
const WebRtcKeyValueConfig& field_trials);
AudioEncoderCopyRed(Config&& config, const FieldTrialsView& field_trials);
~AudioEncoderCopyRed() override;

Some files were not shown because too many files have changed in this diff Show more