diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn index cbf03c229a..2c0dbefc7f 100644 --- a/api/audio_codecs/opus/BUILD.gn +++ b/api/audio_codecs/opus/BUILD.gn @@ -46,12 +46,10 @@ rtc_library("audio_encoder_opus") { deps = [ ":audio_encoder_opus_config", "..:audio_codecs_api", - "../../../api:field_trials_view", "../../../modules/audio_coding:webrtc_opus", "../../../rtc_base:checks", "../../../rtc_base/system:rtc_export", "../../environment", - "//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/types:optional", ] } diff --git a/api/audio_codecs/opus/audio_encoder_opus.cc b/api/audio_codecs/opus/audio_encoder_opus.cc index 6a6d5664b6..f099aeb4be 100644 --- a/api/audio_codecs/opus/audio_encoder_opus.cc +++ b/api/audio_codecs/opus/audio_encoder_opus.cc @@ -13,14 +13,12 @@ #include #include -#include "absl/memory/memory.h" #include "absl/types/optional.h" #include "api/audio_codecs/audio_codec_pair_id.h" #include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/opus/audio_encoder_opus_config.h" -#include "api/field_trials_view.h" #include "modules/audio_coding/codecs/opus/audio_encoder_opus.h" #include "rtc_base/checks.h" @@ -41,22 +39,6 @@ AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder( return AudioEncoderOpusImpl::QueryAudioEncoder(config); } -std::unique_ptr AudioEncoderOpus::MakeAudioEncoder( - const AudioEncoderOpusConfig& config, - int payload_type, - absl::optional /*codec_pair_id*/, - const FieldTrialsView* field_trials) { - if (!config.IsOk()) { - RTC_DCHECK_NOTREACHED(); - return nullptr; - } -#pragma clang diagnostic push -#pragma clang diagnostic ignored "-Wdeprecated-declarations" - // Use WrapUnique to call deprecated constructor. - return absl::WrapUnique(new AudioEncoderOpusImpl(config, payload_type)); -#pragma clang diagnostic pop -} - std::unique_ptr AudioEncoderOpus::MakeAudioEncoder( const Environment& env, const AudioEncoderOpusConfig& config, diff --git a/api/audio_codecs/opus/audio_encoder_opus.h b/api/audio_codecs/opus/audio_encoder_opus.h index 414fe45012..d3e769eada 100644 --- a/api/audio_codecs/opus/audio_encoder_opus.h +++ b/api/audio_codecs/opus/audio_encoder_opus.h @@ -15,13 +15,11 @@ #include #include "absl/types/optional.h" -#include "api/audio_codecs/audio_codec_pair_id.h" #include "api/audio_codecs/audio_encoder.h" #include "api/audio_codecs/audio_encoder_factory.h" #include "api/audio_codecs/audio_format.h" #include "api/audio_codecs/opus/audio_encoder_opus_config.h" #include "api/environment/environment.h" -#include "api/field_trials_view.h" #include "rtc_base/system/rtc_export.h" namespace webrtc { @@ -38,11 +36,6 @@ struct RTC_EXPORT AudioEncoderOpus { const Environment& env, const AudioEncoderOpusConfig& config, const AudioEncoderFactory::Options& options); - static std::unique_ptr MakeAudioEncoder( - const AudioEncoderOpusConfig& config, - int payload_type, - absl::optional codec_pair_id = absl::nullopt, - const FieldTrialsView* field_trials = nullptr); }; } // namespace webrtc diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 0f6fbeb960..cb7163f25e 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -473,7 +473,6 @@ rtc_library("webrtc_opus") { "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs/opus:audio_encoder_opus_config", "../../api/environment", - "../../api/transport:field_trial_based_config", "../../common_audio", "../../rtc_base:buffer", "../../rtc_base:checks", diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc index 8437706122..7eb8bbd163 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc @@ -20,7 +20,6 @@ #include "absl/strings/match.h" #include "absl/strings/string_view.h" #include "api/field_trials_view.h" -#include "api/transport/field_trial_based_config.h" #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h" #include "modules/audio_coding/audio_network_adaptor/controller_manager.h" #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h" @@ -345,7 +344,7 @@ std::unique_ptr AudioEncoderOpusImpl::CreateForTesting( std::unique_ptr bitrate_smoother) { // Using `new` to access a non-public constructor. return absl::WrapUnique(new AudioEncoderOpusImpl( - env.field_trials(), config, payload_type, audio_network_adaptor_creator, + env, config, payload_type, audio_network_adaptor_creator, std::move(bitrate_smoother))); } @@ -353,7 +352,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(const Environment& env, const AudioEncoderOpusConfig& config, int payload_type) : AudioEncoderOpusImpl( - env.field_trials(), + env, config, payload_type, [this](absl::string_view config_string, RtcEventLog* event_log) { @@ -362,30 +361,19 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(const Environment& env, // We choose 5sec as initial time constant due to empirical data. std::make_unique(5'000)) {} -AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, - int payload_type) - : AudioEncoderOpusImpl( - FieldTrialBasedConfig(), - config, - payload_type, - [this](absl::string_view config_string, RtcEventLog* event_log) { - return DefaultAudioNetworkAdaptorCreator(config_string, event_log); - }, - // We choose 5sec as initial time constant due to empirical data. - std::make_unique(5000)) {} - AudioEncoderOpusImpl::AudioEncoderOpusImpl( - const FieldTrialsView& field_trials, + const Environment& env, const AudioEncoderOpusConfig& config, int payload_type, const AudioNetworkAdaptorCreator& audio_network_adaptor_creator, std::unique_ptr bitrate_smoother) : payload_type_(payload_type), - use_stable_target_for_adaptation_( - !field_trials.IsDisabled("WebRTC-Audio-StableTargetAdaptation")), - adjust_bandwidth_(field_trials.IsEnabled("WebRTC-AdjustOpusBandwidth")), + use_stable_target_for_adaptation_(!env.field_trials().IsDisabled( + "WebRTC-Audio-StableTargetAdaptation")), + adjust_bandwidth_( + env.field_trials().IsEnabled("WebRTC-AdjustOpusBandwidth")), bitrate_changed_(true), - bitrate_multipliers_(GetBitrateMultipliers(field_trials)), + bitrate_multipliers_(GetBitrateMultipliers(env.field_trials())), packet_loss_rate_(0.0), inst_(nullptr), packet_loss_fraction_smoother_(new PacketLossFractionSmoother()), diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h index 7568631b4e..b09fbf1f19 100644 --- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h +++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h @@ -61,10 +61,6 @@ class AudioEncoderOpusImpl final : public AudioEncoder { const AudioEncoderOpusConfig& config, int payload_type); - [[deprecated("bugs.webrtc.org/343086059")]] AudioEncoderOpusImpl( - const AudioEncoderOpusConfig& config, - int payload_type); - ~AudioEncoderOpusImpl() override; AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete; @@ -126,10 +122,8 @@ class AudioEncoderOpusImpl final : public AudioEncoder { private: class PacketLossFractionSmoother; - // TODO: bugs.webrtc.org/343086059 - Replace field_trials with Environment - // when public constructors that do not provide the Environment are removed. AudioEncoderOpusImpl( - const FieldTrialsView& field_trials, + const Environment& env, const AudioEncoderOpusConfig& config, int payload_type, const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,